From sipping-bounces@ietf.org Wed Aug 02 16:09:45 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8N1s-0000Wq-50; Wed, 02 Aug 2006 16:09:20 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8N1q-0000U5-Sw for sipping@ietf.org; Wed, 02 Aug 2006 16:09:18 -0400 Received: from mailgate.siemenscomms.co.uk ([195.171.110.225] helo=bemg01.siemenscomms.co.uk) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G8N1p-0001dC-I1 for sipping@ietf.org; Wed, 02 Aug 2006 16:09:18 -0400 Received: from CONVERSION-DAEMON.siemenscomms.co.uk by siemenscomms.co.uk (PMDF V6.0-24 #40642) id <0J3E00E01000RF@siemenscomms.co.uk> for sipping@ietf.org; Wed, 02 Aug 2006 21:09:37 +0100 (BST) Received: from ntht207e.uksgcs.siemenscomms.co.uk ([137.223.247.82]) by siemenscomms.co.uk (PMDF V6.0-24 #40642) with ESMTP id <0J3E00D6100037@siemenscomms.co.uk> for sipping@ietf.org; Wed, 02 Aug 2006 21:09:36 +0100 (BST) Received: by ntht207e.uksgcs.siemenscomms.co.uk with Internet Mail Service (5.5.2657.72) id ; Wed, 02 Aug 2006 21:09:16 +0100 Content-return: allowed Date: Wed, 02 Aug 2006 21:09:15 +0100 From: "Elwell, John" To: sipping@ietf.org Message-id: <50B1CBA96870A34799A506B2313F266709897F3A@ntht201e.siemenscomms.co.uk> MIME-version: 1.0 X-Mailer: Internet Mail Service (5.5.2657.72) X-Spam-Score: 0.1 (/) X-Scan-Signature: 21c69d3cfc2dd19218717dbe1d974352 Subject: [Sipping] Question on do not disturb indication X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Content-Type: multipart/mixed; boundary="===============2082254971==" Errors-To: sipping-bounces@ietf.org This message is in MIME format. Since your mail reader does not understand this format, some or all of this message may not be legible. --===============2082254971== Content-return: allowed Content-type: multipart/alternative; boundary="Boundary_(ID_LCnWMiBdxWx+W6uusuwuGg)" This message is in MIME format. Since your mail reader does not understand this format, some or all of this message may not be legible. --Boundary_(ID_LCnWMiBdxWx+W6uusuwuGg) Content-type: text/plain Content-transfer-encoding: 7BIT RFC 3261 suggests the use of response code 480 Temporarily Unavailable for indicating do not disturb (DND), but it is not used exclusively for this purpose. There are other documents (e.g., RFC 4504) that suggest the use of 486 Busy Here, but again this is not helpful if the receiver wants to distinguish between DND and ordinary busy. Presence can be used to indicate this condition through the RPID note element, but that does not say why a particular session establishment attempt has failed. What are the best practices for indicating the DND condition? Does anyone see a need for an explicit indicator for this purpose, e.g., a new SIP response code? John --Boundary_(ID_LCnWMiBdxWx+W6uusuwuGg) Content-type: text/html Content-transfer-encoding: 7BIT Question on do not disturb indication

RFC 3261 suggests the use of response code 480 Temporarily Unavailable for indicating do not disturb (DND), but it is not used exclusively for this purpose. There are other documents (e.g., RFC 4504) that suggest the use of 486 Busy Here, but again this is not helpful if the receiver wants to distinguish between DND and ordinary busy. Presence can be used to indicate this condition through the RPID note element, but that does not say why a particular session establishment attempt has failed. What are the best practices for indicating the DND condition? Does anyone see a need for an explicit indicator for this purpose, e.g., a new SIP response code?

John


--Boundary_(ID_LCnWMiBdxWx+W6uusuwuGg)-- --===============2082254971== Content-Type: text/plain; charset="us-ascii" MIME-Version: 1.0 Content-Transfer-Encoding: 7bit Content-Disposition: inline _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP --===============2082254971==-- From sipping-bounces@ietf.org Wed Aug 02 17:15:49 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8O3x-0006Rl-Rl; Wed, 02 Aug 2006 17:15:33 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8O3v-0006M1-JW for sipping@ietf.org; Wed, 02 Aug 2006 17:15:31 -0400 Received: from [65.220.123.2] (helo=mail.pingtel.com) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G8O3p-0005mX-4p for sipping@ietf.org; Wed, 02 Aug 2006 17:15:31 -0400 Received: from localhost.localdomain (pi.pingtel.com [10.1.1.12]) by mail.pingtel.com (Postfix) with ESMTP id C44116C020; Wed, 2 Aug 2006 16:14:53 -0400 (EDT) Subject: Re: [Sipping] Question on do not disturb indication From: Scott Lawrence To: "Elwell, John" In-Reply-To: <50B1CBA96870A34799A506B2313F266709897F3A@ntht201e.siemenscomms.co.uk> References: <50B1CBA96870A34799A506B2313F266709897F3A@ntht201e.siemenscomms.co.uk> Content-Type: text/plain Organization: Pingtel Corp. http://www.pingtel.com/ Date: Wed, 02 Aug 2006 17:14:30 -0400 Message-Id: <1154553270.2866.86.camel@sukothai.pingtel.com> Mime-Version: 1.0 X-Mailer: Evolution 2.6.2 (2.6.2-1.fc5.5) Content-Transfer-Encoding: 7bit X-Spam-Score: 0.1 (/) X-Scan-Signature: ea4ac80f790299f943f0a53be7e1a21a Cc: sipping@ietf.org X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org On Wed, 2006-08-02 at 21:09 +0100, Elwell, John wrote: > RFC 3261 suggests the use of response code 480 Temporarily Unavailable > for indicating do not disturb (DND), but it is not used exclusively > for this purpose. There are other documents (e.g., RFC 4504) that > suggest the use of 486 Busy Here, but again this is not helpful if the > receiver wants to distinguish between DND and ordinary busy. Presence > can be used to indicate this condition through the RPID note element, > but that does not say why a particular session establishment attempt > has failed. What are the best practices for indicating the DND > condition? Does anyone see a need for an explicit indicator for this > purpose, e.g., a new SIP response code? Personally, I like 480 for this. If the phone wanted to be sneaky, it could just mute the ringing and allow the transaction to time out without explicitly rejecting it. That prevents the caller from being able to tell they are being disregarded, but some users might prefer that forwarding take effect sooner and not like that. The one thing that I _don't_ like to see is a 603 being used for this; that messes up any upstream forking. -- Scott Lawrence tel:+1-781-938-5306;ext=162 or sip:slawrence@pingtel.com sipXpbx project coordinator - SIPfoundry http://www.sipfoundry.org/sipX Chief Architect - Pingtel Corp. http://www.pingtel.com/ _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Wed Aug 02 18:31:11 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8PF4-00077C-H3; Wed, 02 Aug 2006 18:31:06 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8PF3-000777-CU for sipping@ietf.org; Wed, 02 Aug 2006 18:31:05 -0400 Received: from sj-iport-4.cisco.com ([171.68.10.86]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G8PF1-0003UD-1P for sipping@ietf.org; Wed, 02 Aug 2006 18:31:05 -0400 Received: from sj-dkim-7.cisco.com ([171.68.10.88]) by sj-iport-4.cisco.com with ESMTP; 02 Aug 2006 15:31:03 -0700 X-IronPort-AV: i="4.07,206,1151910000"; d="scan'208"; a="1843885096:sNHT26685348" Received: from sj-core-3.cisco.com (sj-core-3.cisco.com [171.68.223.137]) by sj-dkim-7.cisco.com (8.12.11.20060308/8.12.11) with ESMTP id k72MV2Y2012592; Wed, 2 Aug 2006 15:31:02 -0700 Received: from xbh-rtp-201.amer.cisco.com (xbh-rtp-201.cisco.com [64.102.31.12]) by sj-core-3.cisco.com (8.12.10/8.12.6) with ESMTP id k72MV1Bc019294; Wed, 2 Aug 2006 15:31:02 -0700 (PDT) Received: from xfe-rtp-202.amer.cisco.com ([64.102.31.21]) by xbh-rtp-201.amer.cisco.com with Microsoft SMTPSVC(6.0.3790.1830); Wed, 2 Aug 2006 18:30:46 -0400 Received: from [161.44.79.104] ([161.44.79.104]) by xfe-rtp-202.amer.cisco.com with Microsoft SMTPSVC(6.0.3790.1830); Wed, 2 Aug 2006 18:30:45 -0400 Message-ID: <44D12795.4080901@cisco.com> Date: Wed, 02 Aug 2006 18:30:45 -0400 From: Paul Kyzivat User-Agent: Thunderbird 1.5.0.5 (Windows/20060719) MIME-Version: 1.0 To: Scott Lawrence Subject: Re: [Sipping] Question on do not disturb indication References: <50B1CBA96870A34799A506B2313F266709897F3A@ntht201e.siemenscomms.co.uk> <1154553270.2866.86.camel@sukothai.pingtel.com> In-Reply-To: <1154553270.2866.86.camel@sukothai.pingtel.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Content-Transfer-Encoding: 7bit X-OriginalArrivalTime: 02 Aug 2006 22:30:45.0747 (UTC) FILETIME=[4BED9430:01C6B683] DKIM-Signature: a=rsa-sha1; q=dns; l=1661; t=1154557862; x=1155421862; c=relaxed/simple; s=sjdkim7002; h=Content-Type:From:Subject:Content-Transfer-Encoding:MIME-Version; d=cisco.com; i=pkyzivat@cisco.com; z=From:Paul=20Kyzivat=20 |Subject:Re=3A=20[Sipping]=20Question=20on=20do=20not=20disturb=20indication; X=v=3Dcisco.com=3B=20h=3Dyu9ohsv1kFqX2uC2CMjguD3sTnY=3D; b=nV4cRI9sYIegDfsW1qqnkw5h+ZU+yrE73tc20jtoJInZJEpuw2yABHkwgUF54wiaYOW0ijd1 oIntziJVFN0RJkk/8B/j9dohWS10oJn7PyGffegp7wfNpqJLmSaUHH8e; Authentication-Results: sj-dkim-7.cisco.com; header.From=pkyzivat@cisco.com; dkim=pass ( sig from cisco.com verified; ); X-Spam-Score: 0.0 (/) X-Scan-Signature: bb8f917bb6b8da28fc948aeffb74aa17 Cc: sipping@ietf.org, "Elwell, John" X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org I pretty much agree with Scott. In many/most cases I won't want the caller to know why his call is going unanswered. Both 480 and 486 are in effect "white lies". Which one to use may depend on which impression you want to give to the caller. Another one - if you want to be more rude - is 403 Forbidden. I don't see a need for a *special* code that is unique to this service. Paul Scott Lawrence wrote: > On Wed, 2006-08-02 at 21:09 +0100, Elwell, John wrote: >> RFC 3261 suggests the use of response code 480 Temporarily Unavailable >> for indicating do not disturb (DND), but it is not used exclusively >> for this purpose. There are other documents (e.g., RFC 4504) that >> suggest the use of 486 Busy Here, but again this is not helpful if the >> receiver wants to distinguish between DND and ordinary busy. Presence >> can be used to indicate this condition through the RPID note element, >> but that does not say why a particular session establishment attempt >> has failed. What are the best practices for indicating the DND >> condition? Does anyone see a need for an explicit indicator for this >> purpose, e.g., a new SIP response code? > > Personally, I like 480 for this. > > If the phone wanted to be sneaky, it could just mute the ringing and > allow the transaction to time out without explicitly rejecting it. That > prevents the caller from being able to tell they are being disregarded, > but some users might prefer that forwarding take effect sooner and not > like that. > > The one thing that I _don't_ like to see is a 603 being used for this; > that messes up any upstream forking. > > _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Wed Aug 02 18:52:03 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8PZJ-0000rr-A4; Wed, 02 Aug 2006 18:52:01 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8PZH-0000rk-4m for sipping@ietf.org; Wed, 02 Aug 2006 18:51:59 -0400 Received: from sccrmhc14.comcast.net ([63.240.77.84]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G8PZF-0005Mk-V6 for sipping@ietf.org; Wed, 02 Aug 2006 18:51:59 -0400 Received: from dragon.ariadne.com (dworley.hsd1.ma.comcast.net[24.34.79.42]) by comcast.net (sccrmhc14) with ESMTP id <2006080222515701400ch0s6e>; Wed, 2 Aug 2006 22:51:57 +0000 Received: from dragon.ariadne.com (dragon.ariadne.com [127.0.0.1]) by dragon.ariadne.com (8.12.8/8.12.8) with ESMTP id k72MpvZW020064 for ; Wed, 2 Aug 2006 18:51:57 -0400 Received: (from worley@localhost) by dragon.ariadne.com (8.12.8/8.12.8/Submit) id k72Mpvkp020060; Wed, 2 Aug 2006 18:51:57 -0400 Date: Wed, 2 Aug 2006 18:51:57 -0400 Message-Id: <200608022251.k72Mpvkp020060@dragon.ariadne.com> To: sipping@ietf.org From: Dale.Worley@comcast.net In-reply-to: <1154553270.2866.86.camel@sukothai.pingtel.com> (slawrence@pingtel.com) Subject: Re: [Sipping] Question on do not disturb indication References: <50B1CBA96870A34799A506B2313F266709897F3A@ntht201e.siemenscomms.co.uk> <1154553270.2866.86.camel@sukothai.pingtel.com> X-Spam-Score: 0.2 (/) X-Scan-Signature: 7bac9cb154eb5790ae3b2913587a40de X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org From: Scott Lawrence The one thing that I _don't_ like to see is a 603 being used for this; that messes up any upstream forking. Personally, I think we should file a bug against the *existence* of 6xx codes. They just can't be made to work correctly in the light of multiple routings and forwardings of a URI. Dale _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Wed Aug 02 18:58:43 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8Pfm-0006Vg-H0; Wed, 02 Aug 2006 18:58:42 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8Pfl-0006Tb-Lh for sipping@ietf.org; Wed, 02 Aug 2006 18:58:41 -0400 Received: from mail120.messagelabs.com ([216.82.250.83]) by ietf-mx.ietf.org with smtp (Exim 4.43) id 1G8Pfk-00068a-9j for sipping@ietf.org; Wed, 02 Aug 2006 18:58:41 -0400 X-VirusChecked: Checked X-Env-Sender: mdolly@att.com X-Msg-Ref: server-2.tower-120.messagelabs.com!1154559291!12313940!31 X-StarScan-Version: 5.5.10.7; banners=-,-,- X-Originating-IP: [134.24.146.4] Received: (qmail 1143 invoked from network); 2 Aug 2006 22:58:39 -0000 Received: from unknown (HELO attrh3i.attrh.att.com) (134.24.146.4) by server-2.tower-120.messagelabs.com with SMTP; 2 Aug 2006 22:58:39 -0000 Received: from OCCLUST04EVS1.ugd.att.com (135.38.164.12) by attrh3i.attrh.att.com (7.2.052) id 44C39E850016E329; Wed, 2 Aug 2006 18:58:37 -0400 X-MimeOLE: Produced By Microsoft Exchange V6.5 Content-class: urn:content-classes:message MIME-Version: 1.0 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: quoted-printable Subject: RE: [Sipping] Question on do not disturb indication Date: Wed, 2 Aug 2006 17:58:36 -0500 Message-ID: <28F05913385EAC43AF019413F674A017101B6C81@OCCLUST04EVS1.ugd.att.com> In-Reply-To: <44D12795.4080901@cisco.com> X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: [Sipping] Question on do not disturb indication Thread-Index: Aca2g2r/28Oh2xC2R+O8begJmi2HoAAA1QLA From: "Dolly, Martin C, ALABS" To: "Paul Kyzivat" , "Scott Lawrence" X-Spam-Score: 0.0 (/) X-Scan-Signature: 5a9a1bd6c2d06a21d748b7d0070ddcb8 Cc: "Elwell, John" , sipping@ietf.org X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Is not this a question for the market to decide?=20 Do any of you think we can speak for J.Q. Public? Or should. =20 -----Original Message----- From: Paul Kyzivat [mailto:pkyzivat@cisco.com]=20 Sent: Wednesday, August 02, 2006 6:31 PM To: Scott Lawrence Cc: sipping@ietf.org; Elwell, John Subject: Re: [Sipping] Question on do not disturb indication I pretty much agree with Scott. In many/most cases I won't want the=20 caller to know why his call is going unanswered. Both 480 and 486 are in effect "white lies". Which one to use may depend on which impression you want to give to the caller. Another one - if you want to be more rude -=20 is 403 Forbidden. I don't see a need for a *special* code that is unique to this service. Paul Scott Lawrence wrote: > On Wed, 2006-08-02 at 21:09 +0100, Elwell, John wrote: >> RFC 3261 suggests the use of response code 480 Temporarily Unavailable >> for indicating do not disturb (DND), but it is not used exclusively >> for this purpose. There are other documents (e.g., RFC 4504) that >> suggest the use of 486 Busy Here, but again this is not helpful if the >> receiver wants to distinguish between DND and ordinary busy. Presence >> can be used to indicate this condition through the RPID note element, >> but that does not say why a particular session establishment attempt >> has failed. What are the best practices for indicating the DND >> condition? Does anyone see a need for an explicit indicator for this >> purpose, e.g., a new SIP response code?=20 >=20 > Personally, I like 480 for this. =20 >=20 > If the phone wanted to be sneaky, it could just mute the ringing and > allow the transaction to time out without explicitly rejecting it. That > prevents the caller from being able to tell they are being disregarded, > but some users might prefer that forwarding take effect sooner and not > like that. >=20 > The one thing that I _don't_ like to see is a 603 being used for this; > that messes up any upstream forking. >=20 >=20 _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Wed Aug 02 19:24:50 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8Q4w-0003eo-QO; Wed, 02 Aug 2006 19:24:42 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8Q4v-0003bL-6V for sipping@ietf.org; Wed, 02 Aug 2006 19:24:41 -0400 Received: from zcars04f.nortel.com ([47.129.242.57]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G8Q4t-0008LW-RU for sipping@ietf.org; Wed, 02 Aug 2006 19:24:41 -0400 Received: from zrc2hxm0.corp.nortel.com (zrc2hxm0.corp.nortel.com [47.103.123.71]) by zcars04f.nortel.com (Switch-2.2.6/Switch-2.2.0) with ESMTP id k72NOXG03522; Wed, 2 Aug 2006 19:24:34 -0400 (EDT) X-MimeOLE: Produced By Microsoft Exchange V6.5.7226.0 Content-class: urn:content-classes:message MIME-Version: 1.0 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: quoted-printable Subject: RE: [Sipping] Question on do not disturb indication Date: Wed, 2 Aug 2006 18:24:09 -0500 Message-ID: <1ECE0EB50388174790F9694F77522CCF0BF08368@zrc2hxm0.corp.nortel.com> In-Reply-To: <44D12795.4080901@cisco.com> X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: [Sipping] Question on do not disturb indication Thread-Index: Aca2g3JKRNWXN+TVRtuAF1P9b2YPcAABfjDQ From: "Francois Audet" To: "Paul Kyzivat" , "Scott Lawrence" X-Spam-Score: 0.0 (/) X-Scan-Signature: c3a18ef96977fc9bcc21a621cbf1174b Cc: "Elwell, John" , sipping@ietf.org X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org I agree: I don't see a need for a special code for this, as it seems to make pretty much exactly to existing practice. In fact, even in the "old days" you had different features corresponding to the two codes. - Do-not-Disturb is pretty much like 480 - Make-busy is pretty much like 486 Some vendors have "make busy", others have "do not disturb", some may have both. On the 603 issue, I'd like to disagree with everybody else... Sometimes being able to decline the call everywhere is actually what you want. As an example, I have multiple forked device. If I send a 603, it triggers the proxy to use whatever alternate routing is sees fit (e.g., voicemail, attendant, etc.). That's the desired behavior. Problem with a 480/486 with that scenario is that it won't trigger the alternate routing and will keep on ringing the other fork. That may not be the desired behavior. My 2 cents.=20 > -----Original Message----- > From: Paul Kyzivat [mailto:pkyzivat@cisco.com]=20 > Sent: Wednesday, August 02, 2006 3:31 PM > To: Scott Lawrence > Cc: sipping@ietf.org; Elwell, John > Subject: Re: [Sipping] Question on do not disturb indication >=20 > I pretty much agree with Scott. In many/most cases I won't=20 > want the caller to know why his call is going unanswered.=20 > Both 480 and 486 are in effect "white lies". Which one to use=20 > may depend on which impression you want to give to the=20 > caller. Another one - if you want to be more rude - is 403 Forbidden. >=20 > I don't see a need for a *special* code that is unique to=20 > this service. >=20 > Paul >=20 > Scott Lawrence wrote: > > On Wed, 2006-08-02 at 21:09 +0100, Elwell, John wrote: > >> RFC 3261 suggests the use of response code 480 Temporarily=20 > >> Unavailable for indicating do not disturb (DND), but it is=20 > not used=20 > >> exclusively for this purpose. There are other documents (e.g., RFC=20 > >> 4504) that suggest the use of 486 Busy Here, but again this is not=20 > >> helpful if the receiver wants to distinguish between DND=20 > and ordinary=20 > >> busy. Presence can be used to indicate this condition through the=20 > >> RPID note element, but that does not say why a particular session=20 > >> establishment attempt has failed. What are the best practices for=20 > >> indicating the DND condition? Does anyone see a need for=20 > an explicit=20 > >> indicator for this purpose, e.g., a new SIP response code? > >=20 > > Personally, I like 480 for this. =20 > >=20 > > If the phone wanted to be sneaky, it could just mute the=20 > ringing and=20 > > allow the transaction to time out without explicitly rejecting it. =20 > > That prevents the caller from being able to tell they are being=20 > > disregarded, but some users might prefer that forwarding=20 > take effect=20 > > sooner and not like that. > >=20 > > The one thing that I _don't_ like to see is a 603 being=20 > used for this;=20 > > that messes up any upstream forking. > >=20 > >=20 >=20 > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP=20 > Use sip-implementors@cs.columbia.edu for questions on current=20 > sip Use sip@ietf.org for new developments of core SIP >=20 _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Wed Aug 02 19:58:16 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8QbF-0006DL-TX; Wed, 02 Aug 2006 19:58:05 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8QbE-0006DG-N8 for sipping@ietf.org; Wed, 02 Aug 2006 19:58:04 -0400 Received: from [65.220.123.2] (helo=mail.pingtel.com) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G8Qb9-0004hJ-7T for sipping@ietf.org; Wed, 02 Aug 2006 19:58:04 -0400 Received: from localhost.localdomain (pi.pingtel.com [10.1.1.12]) by mail.pingtel.com (Postfix) with ESMTP id 516316C01D; Wed, 2 Aug 2006 18:57:25 -0400 (EDT) Subject: RE: [Sipping] Question on do not disturb indication From: Scott Lawrence To: Francois Audet In-Reply-To: <1ECE0EB50388174790F9694F77522CCF0BF08368@zrc2hxm0.corp.nortel.com> References: <1ECE0EB50388174790F9694F77522CCF0BF08368@zrc2hxm0.corp.nortel.com> Content-Type: text/plain Organization: Pingtel Corp. http://www.pingtel.com/ Date: Wed, 02 Aug 2006 19:57:01 -0400 Message-Id: <1154563021.3841.13.camel@localhost.localdomain> Mime-Version: 1.0 X-Mailer: Evolution 2.6.2 (2.6.2-1.fc5.5) Content-Transfer-Encoding: 7bit X-Spam-Score: 0.1 (/) X-Scan-Signature: 8b431ad66d60be2d47c7bfeb879db82c Cc: "Elwell, John" , Paul Kyzivat , sipping@ietf.org X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org On Wed, 2006-08-02 at 18:24 -0500, Francois Audet wrote: > I agree: I don't see a need for a special code for this, as it seems to > make > pretty much exactly to existing practice. > On the 603 issue, I'd like to disagree with everybody else... > Sometimes being able to decline the call everywhere is actually what you > want. > As an example, I have multiple forked device. If I send a 603, it > triggers the proxy to > use whatever alternate routing is sees fit (e.g., voicemail, attendant, > etc.). That's > the desired behavior. Well, that depends on what you think 603 means. 3261 says: 6xx responses indicate that a server has definitive information about a particular user, not just the particular instance indicated in the Request-URI. and for 603 specifically it says: This status response is returned only if the client knows that no other end point will answer the request. I read that to mean that a forking proxy should abandon all forks for the call and terminate the request - the response asserts that the voicemail (another end point) will not answer. That might be an appropriate response for a proxy to send if it is authoritative for the domain and knows that a particular user does not exist, but I can't see how a user agent can ever have enough global knowledge to justify a 6xx response. > Problem with a 480/486 with that scenario is that it won't trigger the > alternate routing > and will keep on ringing the other fork. That may not be the desired > behavior. This is interesting - you seem to have exactly the opposite belief about these. No 4xx response will affect other forks in our proxy. -- Scott Lawrence tel:+1-781-938-5306;ext=162 or sip:slawrence@pingtel.com sipXpbx project coordinator - SIPfoundry http://www.sipfoundry.org/sipX Chief Architect - Pingtel Corp. http://www.pingtel.com/ _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Wed Aug 02 23:17:52 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8TiM-0003x8-D0; Wed, 02 Aug 2006 23:17:38 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8TiJ-0003x2-Lk for sipping@ietf.org; Wed, 02 Aug 2006 23:17:35 -0400 Received: from nylon.softarmor.com ([66.135.38.164]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G8TiH-0001pX-B2 for sipping@ietf.org; Wed, 02 Aug 2006 23:17:35 -0400 Received: from [192.168.2.104] (sj-natpool-220.cisco.com [128.107.248.220]) (authenticated bits=0) by nylon.softarmor.com (8.13.1/8.13.1) with ESMTP id k733JYIj003808 (version=TLSv1/SSLv3 cipher=RC4-SHA bits=128 verify=NO); Wed, 2 Aug 2006 22:19:35 -0500 In-Reply-To: <28F05913385EAC43AF019413F674A017101B6C81@OCCLUST04EVS1.ugd.att.com> References: <28F05913385EAC43AF019413F674A017101B6C81@OCCLUST04EVS1.ugd.att.com> Mime-Version: 1.0 (Apple Message framework v752.2) Content-Type: text/plain; charset=US-ASCII; delsp=yes; format=flowed Message-Id: <091AED7C-4DF3-4045-B40B-A88A5A814A67@softarmor.com> Content-Transfer-Encoding: 7bit From: Dean Willis Subject: Re: [Sipping] Question on do not disturb indication Date: Wed, 2 Aug 2006 22:17:24 -0500 To: "Dolly, Martin C, ALABS" X-Mailer: Apple Mail (2.752.2) X-Spam-Score: 0.0 (/) X-Scan-Signature: 79899194edc4f33a41f49410777972f8 Cc: sipping@ietf.org, Paul Kyzivat , "Elwell, John" , Scott Lawrence X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org On Aug 2, 2006, at 5:58 PM, Dolly, Martin C, ALABS wrote: > Is not this a question for the market to decide? > Do any of you think we can speak for J.Q. Public? Or should. Well, I personally speak for at least a portion of the service- consuming market. Myself. After all, I AM an AT&T customer for some of my services. And a Comcast customer, Verizon customer, Skype customer, etc. Or maybe we should have a national poll on which code to use? Heck, people on the street can't seven seem to be able to pick a president, much less tell a 404 from an ACM. So, as a service-consuming member of the market, I'd rather have the IETF recommend something that mostly works, thanks. -- Dean _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Wed Aug 02 23:27:59 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8TsM-0004Z9-IX; Wed, 02 Aug 2006 23:27:58 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8TsJ-0004Yu-Ji for sipping@ietf.org; Wed, 02 Aug 2006 23:27:55 -0400 Received: from wip-ec-wd.wipro.com ([203.91.193.32]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G8TsH-0002Vz-7B for sipping@ietf.org; Wed, 02 Aug 2006 23:27:55 -0400 Received: from wip-ec-wd.wipro.com (localhost.wipro.com [127.0.0.1]) by localhost (Postfix) with ESMTP id 71068205F4 for ; Thu, 3 Aug 2006 08:54:37 +0530 (IST) Received: from blr-ec-bh01.wipro.com (blr-ec-bh01.wipro.com [10.201.50.91]) by wip-ec-wd.wipro.com (Postfix) with ESMTP id 5E8A3205DA for ; Thu, 3 Aug 2006 08:54:37 +0530 (IST) Received: from blr-m2-msg.wipro.com ([10.116.50.99]) by blr-ec-bh01.wipro.com with Microsoft SMTPSVC(6.0.3790.1830); Thu, 3 Aug 2006 08:57:51 +0530 X-MimeOLE: Produced By Microsoft Exchange V6.5 Content-class: urn:content-classes:message MIME-Version: 1.0 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: quoted-printable Subject: RE: [Sipping] Some thoughts on fixing early media Date: Thu, 3 Aug 2006 08:55:31 +0530 Message-ID: <532B18E13CF9E64380EF5FDDE265E071384C58@blr-m2-msg.wipro.com> In-Reply-To: <897DA809E857CD40A8419CF517948E2101F6FC5B@blr-k1-msg.wipro.com> X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: [Sipping] Some thoughts on fixing early media Thread-Index: Acav6si/KX4K3NwnQFGu7NssoL4r7gAFBZXwAAT8rwABjPJ3AA== From: To: , , X-OriginalArrivalTime: 03 Aug 2006 03:27:51.0206 (UTC) FILETIME=[CCBAC460:01C6B6AC] X-Spam-Score: 0.2 (/) X-Scan-Signature: cdb443e3957ca9b4c5b55e78cfcf4b26 Cc: dean.willis@softarmor.com, sipping@ietf.org, christer.holmberg@ericsson.com X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Hi Sayan, Would it not be the UAS's responsibility to decide that the user should be alerted only after the Early Media is played? I think that the UAS which has early media to play, can decide to postpone the alerting till it has been sent to the UAC. UAC need not have to control this - I think this should satisfy all the use cases needed. (for example "please hang-up in ten seconds from now unless you wish to be billed a huge amount" or something like that should be played before UAS alerts the user.) Regards, Ravi. -- =20 Ravishankar. G. Shiroor Wipro Technologies, Bangalore. =20 ravishankar.shiroor@wipro.com -- =20 > -----Original Message----- > From: sayan.chowdhury@wipro.com [mailto:sayan.chowdhury@wipro.com]=20 > Sent: Tuesday, July 25, 2006 11:21 PM > To: bstucker@nortel.com; pkyzivat@cisco.com > Cc: christer.holmberg@ericsson.com; sipping@ietf.org;=20 > dean.willis@softarmor.com > Subject: RE: [Sipping] Some thoughts on fixing early media >=20 > Hello Brian , > I am trying to put a call flow here , (I will not try to draw=20 > as my mail client is messing it up)=20 >=20 > 1>Callee sends an offer with emflow=3Dnone > 2>gets back say two 183 responses (establishing early dialog)=20 > both with > early media class as two-way. > 3>it selects the 183 with the higher q-value and more=20 > critical em class > and adds new offer in the PRACK with emflow as sendrecv. > 4>PRACKs the other 183 as well but with emflow=3Dnone and gets=20 > back 200 OK > for that. > 4>Gets back the answer in the 200 OK to PRACK for the 183 with higher > q-value. > 5>Does early media.=20 > 6>Sends an UPDATE to the next 183 with lower q-value with emflow as > sendrcv > ... continues >=20 > If my understanding of the flow is correct , then in the=20 > above scenario how does the UAC indicate to the UAS(s) not to=20 > alert the user until early media is over ?=20 >=20 > Do we need a confirm-status mechanism as well wherein the UAC=20 > can tell when to alert the UAS ? > There may be use cases when the UAC wants to defer the=20 > alerting until the UAC has heard all the early media "flows". > For eg , when the UAC chooses which UAS it wants to have a=20 > regular media session with after hearing to all the early=20 > media sessions. >=20 > Regards , > Sayan =20 >=20 > -----Original Message----- > From: Brian Stucker [mailto:bstucker@nortel.com] > Sent: Tuesday, July 25, 2006 9:02 PM > To: Paul Kyzivat > Cc: Dean Willis; sipping; Christer Holmberg (JO/LMF) > Subject: RE: [Sipping] Some thoughts on fixing early media >=20 > Yes, this is pretty much what's going to happen. I'll add=20 > that to the text, actually, since I think it deserves being=20 > called out. >=20 > In cases where network interworking is not present (ie. no=20 > SIP-to-PSTN type gateways are present in the routing tree)=20 > things will basically behave as they always have been because=20 > most SIP endpoints do not generate early media. It's services=20 > like CRBT and PSTN gateways that get us into trouble. For the=20 > other endpoints we simply don't wish to clip the media if=20 > possible. For the ones that do generate early media, we can=20 > fork to them, but don't necessarily want to hear what they=20 > all have to say in as a chorus. >=20 > Regards, > Brian >=20 > > -----Original Message----- > > From: Paul Kyzivat [mailto:pkyzivat@cisco.com] > > Sent: Tuesday, July 25, 2006 8:03 AM > > To: Stucker, Brian (RICH1:B620) > > Cc: Christer Holmberg (JO/LMF); Dean Willis; Stephen Sprunk; sipping > > Subject: Re: [Sipping] Some thoughts on fixing early media > >=20 > > If the UAC only gives permission to one tine of the fork to send=20 > > media, and preconditions are used to prevent alerting until=20 > permission >=20 > > to send is granted, then parallel forking has been turned=20 > into serial=20 > > forking under the control of the UAC. > >=20 > > While it seems a little odd, it is perhaps the best that=20 > can be done=20 > > under the circumstances. It works well in that those=20 > destinations that >=20 > > don't require early media can all alert in parallel, and=20 > only the ones >=20 > > requiring early media are serialized. That is a huge=20 > improvement over=20 > > the proxy making the choice, because it doesn't know which=20 > ones will=20 > > require early media. > >=20 > > Paul > >=20 > > Brian Stucker wrote: > > > Yes, I think you understand what I am suggesting. > > >=20 > > > Regards, > > > Brian > > >=20 > > > -----Original Message----- > > > From: Christer Holmberg (JO/LMF) > > > [mailto:christer.holmberg@ericsson.com] > > >=20 > > > Sent: Sunday, July 23, 2006 5:20 PM > > > To: Stucker, Brian (RICH1:B620); Dean Willis; Stephen Sprunk > > > Cc: sipping > > > Subject: VS: [Sipping] Some thoughts on fixing early media > > >=20 > > > Hi, > > >=20 > > >> I'm proposing that the far end would not alert until=20 > permission to=20 > > >> send > > >=20 > > >> media had been granted. > > >=20 > > > That's called pre-conditions, isn't it? :) > > > =20 > > > Or, you could also send an INVITE WITH SDP, but the UAS would not=20 > > > altert the user until some other type of indicator has been > > sent to it. > > > =20 > > > Regards, > > > =20 > > > Christer > > > =20 > > >=20 > > >=20 > > > -----Original Message----- > > > From: Christer Holmberg [mailto:christer.holmberg@ericsson.com] > > > Sent: Sunday, July 23, 2006 12:14 AM > > > To: Stucker, Brian (RICH1:B620); Dean Willis; Stephen Sprunk > > > Cc: sipping > > > Subject: RE: [Sipping] Some thoughts on fixing early media > > >=20 > > > Hi Brian, > > >=20 > > > What if an UAS wants to send 200 OK (and media) before=20 > the UAC has=20 > > > told him it's ok to send media? Someone would have to send > > an UPDATE, > > > to switch the media to sendrecv. > > >=20 > > > Or, are you proposing that the UAS, until it's given=20 > permission to=20 > > > send media, wouldn't even alert the user behind it? > > >=20 > > > Regards, > > >=20 > > > Christer > > >=20 > > >=20 > > >=20 > > >=20 > > > _______________________________________________ > > > Sipping mailing list =20 > https://www1.ietf.org/mailman/listinfo/sipping > > > This list is for NEW development of the application of SIP Use=20 > > > sip-implementors@cs.columbia.edu for questions on current sip Use=20 > > > sip@ietf.org for new developments of core SIP > > >=20 > >=20 >=20 > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP=20 > Use sip-implementors@cs.columbia.edu for questions on current=20 > sip Use sip@ietf.org for new developments of core SIP >=20 > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP=20 > Use sip-implementors@cs.columbia.edu for questions on current=20 > sip Use sip@ietf.org for new developments of core SIP >=20 _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Thu Aug 03 00:01:08 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8UON-0008NV-Rn; Thu, 03 Aug 2006 00:01:03 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8UOM-0008F2-6D for sipping@ietf.org; Thu, 03 Aug 2006 00:01:02 -0400 Received: from zrtps0kp.nortel.com ([47.140.192.56]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G8UOL-0004Zv-Md for sipping@ietf.org; Thu, 03 Aug 2006 00:01:02 -0400 Received: from zrc2hxm2.corp.nortel.com (zrc2hxm2.corp.nortel.com [47.103.123.73]) by zrtps0kp.nortel.com (Switch-2.2.6/Switch-2.2.0) with ESMTP id k733pQA02684; Wed, 2 Aug 2006 23:51:26 -0400 (EDT) X-MimeOLE: Produced By Microsoft Exchange V6.5.7226.0 Content-class: urn:content-classes:message MIME-Version: 1.0 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: quoted-printable Subject: RE: [Sipping] Some thoughts on fixing early media Date: Wed, 2 Aug 2006 22:51:27 -0500 Message-ID: <6FC4416DDE56C44DA0AEE67BC7CA43710FBDA0FC@zrc2hxm2.corp.nortel.com> In-Reply-To: <532B18E13CF9E64380EF5FDDE265E071384C58@blr-m2-msg.wipro.com> X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: [Sipping] Some thoughts on fixing early media thread-index: Acav6si/KX4K3NwnQFGu7NssoL4r7gAFBZXwAAT8rwABjPJ3AAAaSNXw From: "Brian Stucker" To: , , X-Spam-Score: 0.0 (/) X-Scan-Signature: 2b2ad76aced9b1d558e34a970a85c027 Cc: dean.willis@softarmor.com, sipping@ietf.org, christer.holmberg@ericsson.com X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Ok, so... I have a call that forks. The first guy starts sending me colorful ringback, the second guy sends me "please hang up or I'm about to bill you a huge amount in 10 seconds". According to common practice with UAC design, it plays neither early media and instead generates local ringback. You never hear the warning, the colorful ringback service is broken and the originator gets billed a huge amount, but the terminator didn't alert until after the warning was played. :) If the UAC has no control, then the UAS has no guarantee that what they're wanting to send will get presented to the originator. Regards, Brian=20 > -----Original Message----- > From: ravishankar.shiroor@wipro.com=20 > [mailto:ravishankar.shiroor@wipro.com]=20 > Sent: Wednesday, August 02, 2006 11:26 PM > To: sayan.chowdhury@wipro.com; Stucker, Brian (RICH1:B620);=20 > pkyzivat@cisco.com > Cc: christer.holmberg@ericsson.com; sipping@ietf.org;=20 > dean.willis@softarmor.com > Subject: RE: [Sipping] Some thoughts on fixing early media >=20 > Hi Sayan, >=20 > Would it not be the UAS's responsibility to decide that the=20 > user should be alerted only after the Early Media is played? > I think that the UAS which has early media to play, can=20 > decide to postpone the alerting till it has been sent to the=20 > UAC. UAC need not have to control this - I think this should=20 > satisfy all the use cases needed. (for example "please=20 > hang-up in ten seconds from now unless you wish to be billed=20 > a huge amount" or something like that should be played before=20 > UAS alerts the user.) >=20 > Regards, > Ravi. >=20 >=20 > -- > =20 > Ravishankar. G. Shiroor > Wipro Technologies, Bangalore. > =20 > ravishankar.shiroor@wipro.com > -- > =20 >=20 > > -----Original Message----- > > From: sayan.chowdhury@wipro.com [mailto:sayan.chowdhury@wipro.com] > > Sent: Tuesday, July 25, 2006 11:21 PM > > To: bstucker@nortel.com; pkyzivat@cisco.com > > Cc: christer.holmberg@ericsson.com; sipping@ietf.org;=20 > > dean.willis@softarmor.com > > Subject: RE: [Sipping] Some thoughts on fixing early media > >=20 > > Hello Brian , > > I am trying to put a call flow here , (I will not try to draw as my=20 > > mail client is messing it up) > >=20 > > 1>Callee sends an offer with emflow=3Dnone > > 2>gets back say two 183 responses (establishing early dialog) > > both with > > early media class as two-way. > > 3>it selects the 183 with the higher q-value and more > > critical em class > > and adds new offer in the PRACK with emflow as sendrecv. > > 4>PRACKs the other 183 as well but with emflow=3Dnone and gets > > back 200 OK > > for that. > > 4>Gets back the answer in the 200 OK to PRACK for the 183=20 > with higher > > q-value. > > 5>Does early media.=20 > > 6>Sends an UPDATE to the next 183 with lower q-value with emflow as > > sendrcv > > ... continues > >=20 > > If my understanding of the flow is correct , then in the above=20 > > scenario how does the UAC indicate to the UAS(s) not to=20 > alert the user=20 > > until early media is over ? > >=20 > > Do we need a confirm-status mechanism as well wherein the=20 > UAC can tell=20 > > when to alert the UAS ? > > There may be use cases when the UAC wants to defer the=20 > alerting until=20 > > the UAC has heard all the early media "flows". > > For eg , when the UAC chooses which UAS it wants to have a regular=20 > > media session with after hearing to all the early media sessions. > >=20 > > Regards , > > Sayan > >=20 > > -----Original Message----- > > From: Brian Stucker [mailto:bstucker@nortel.com] > > Sent: Tuesday, July 25, 2006 9:02 PM > > To: Paul Kyzivat > > Cc: Dean Willis; sipping; Christer Holmberg (JO/LMF) > > Subject: RE: [Sipping] Some thoughts on fixing early media > >=20 > > Yes, this is pretty much what's going to happen. I'll add=20 > that to the=20 > > text, actually, since I think it deserves being called out. > >=20 > > In cases where network interworking is not present (ie. no=20 > SIP-to-PSTN=20 > > type gateways are present in the routing tree) things will=20 > basically=20 > > behave as they always have been because most SIP endpoints do not=20 > > generate early media. It's services like CRBT and PSTN=20 > gateways that=20 > > get us into trouble. For the other endpoints we simply=20 > don't wish to=20 > > clip the media if possible. For the ones that do generate=20 > early media,=20 > > we can fork to them, but don't necessarily want to hear=20 > what they all=20 > > have to say in as a chorus. > >=20 > > Regards, > > Brian > >=20 > > > -----Original Message----- > > > From: Paul Kyzivat [mailto:pkyzivat@cisco.com] > > > Sent: Tuesday, July 25, 2006 8:03 AM > > > To: Stucker, Brian (RICH1:B620) > > > Cc: Christer Holmberg (JO/LMF); Dean Willis; Stephen=20 > Sprunk; sipping > > > Subject: Re: [Sipping] Some thoughts on fixing early media > > >=20 > > > If the UAC only gives permission to one tine of the fork to send=20 > > > media, and preconditions are used to prevent alerting until > > permission > >=20 > > > to send is granted, then parallel forking has been turned > > into serial > > > forking under the control of the UAC. > > >=20 > > > While it seems a little odd, it is perhaps the best that > > can be done > > > under the circumstances. It works well in that those > > destinations that > >=20 > > > don't require early media can all alert in parallel, and > > only the ones > >=20 > > > requiring early media are serialized. That is a huge > > improvement over > > > the proxy making the choice, because it doesn't know which > > ones will > > > require early media. > > >=20 > > > Paul > > >=20 > > > Brian Stucker wrote: > > > > Yes, I think you understand what I am suggesting. > > > >=20 > > > > Regards, > > > > Brian > > > >=20 > > > > -----Original Message----- > > > > From: Christer Holmberg (JO/LMF) > > > > [mailto:christer.holmberg@ericsson.com] > > > >=20 > > > > Sent: Sunday, July 23, 2006 5:20 PM > > > > To: Stucker, Brian (RICH1:B620); Dean Willis; Stephen Sprunk > > > > Cc: sipping > > > > Subject: VS: [Sipping] Some thoughts on fixing early media > > > >=20 > > > > Hi, > > > >=20 > > > >> I'm proposing that the far end would not alert until > > permission to > > > >> send > > > >=20 > > > >> media had been granted. > > > >=20 > > > > That's called pre-conditions, isn't it? :) > > > > =20 > > > > Or, you could also send an INVITE WITH SDP, but the UAS=20 > would not=20 > > > > altert the user until some other type of indicator has been > > > sent to it. > > > > =20 > > > > Regards, > > > > =20 > > > > Christer > > > > =20 > > > >=20 > > > >=20 > > > > -----Original Message----- > > > > From: Christer Holmberg [mailto:christer.holmberg@ericsson.com] > > > > Sent: Sunday, July 23, 2006 12:14 AM > > > > To: Stucker, Brian (RICH1:B620); Dean Willis; Stephen Sprunk > > > > Cc: sipping > > > > Subject: RE: [Sipping] Some thoughts on fixing early media > > > >=20 > > > > Hi Brian, > > > >=20 > > > > What if an UAS wants to send 200 OK (and media) before > > the UAC has > > > > told him it's ok to send media? Someone would have to send > > > an UPDATE, > > > > to switch the media to sendrecv. > > > >=20 > > > > Or, are you proposing that the UAS, until it's given > > permission to > > > > send media, wouldn't even alert the user behind it? > > > >=20 > > > > Regards, > > > >=20 > > > > Christer > > > >=20 > > > >=20 > > > >=20 > > > >=20 > > > > _______________________________________________ > > > > Sipping mailing list > > https://www1.ietf.org/mailman/listinfo/sipping > > > > This list is for NEW development of the application of SIP Use=20 > > > > sip-implementors@cs.columbia.edu for questions on=20 > current sip Use=20 > > > > sip@ietf.org for new developments of core SIP > > > >=20 > > >=20 > >=20 > > _______________________________________________ > > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > > This list is for NEW development of the application of SIP Use=20 > > sip-implementors@cs.columbia.edu for questions on current sip Use=20 > > sip@ietf.org for new developments of core SIP > >=20 > > _______________________________________________ > > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > > This list is for NEW development of the application of SIP Use=20 > > sip-implementors@cs.columbia.edu for questions on current sip Use=20 > > sip@ietf.org for new developments of core SIP > >=20 >=20 _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Thu Aug 03 00:33:52 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8Utz-0002G4-1y; Thu, 03 Aug 2006 00:33:43 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8Utx-0002Ft-LD for sipping@ietf.org; Thu, 03 Aug 2006 00:33:41 -0400 Received: from wip-ec-wd.wipro.com ([203.91.193.32]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G8Utu-0006gw-Sh for sipping@ietf.org; Thu, 03 Aug 2006 00:33:41 -0400 Received: from wip-ec-wd.wipro.com (localhost.wipro.com [127.0.0.1]) by localhost (Postfix) with ESMTP id 70FA5205F4 for ; Thu, 3 Aug 2006 10:00:20 +0530 (IST) Received: from blr-ec-bh01.wipro.com (blr-ec-bh01.wipro.com [10.201.50.91]) by wip-ec-wd.wipro.com (Postfix) with ESMTP id 5C6A3205DA for ; Thu, 3 Aug 2006 10:00:20 +0530 (IST) Received: from blr-m2-msg.wipro.com ([10.116.50.99]) by blr-ec-bh01.wipro.com with Microsoft SMTPSVC(6.0.3790.1830); Thu, 3 Aug 2006 10:03:34 +0530 X-MimeOLE: Produced By Microsoft Exchange V6.5 Content-class: urn:content-classes:message MIME-Version: 1.0 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: quoted-printable Subject: RE: [Sipping] Some thoughts on fixing early media Date: Thu, 3 Aug 2006 10:05:14 +0530 Message-ID: <532B18E13CF9E64380EF5FDDE265E071384CC8@blr-m2-msg.wipro.com> In-Reply-To: <6FC4416DDE56C44DA0AEE67BC7CA43710FBDA0FC@zrc2hxm2.corp.nortel.com> X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: [Sipping] Some thoughts on fixing early media Thread-Index: Acav6si/KX4K3NwnQFGu7NssoL4r7gAFBZXwAAT8rwABjPJ3AAAaSNXwAAFv6hA= From: To: , , X-OriginalArrivalTime: 03 Aug 2006 04:33:34.0333 (UTC) FILETIME=[FB0432D0:01C6B6B5] X-Spam-Score: 0.2 (/) X-Scan-Signature: 7f3fa64b9851a63d7f3174ef64114da7 Cc: dean.willis@softarmor.com, sipping@ietf.org, christer.holmberg@ericsson.com X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org I am assuming that we will go with the option that the early medias will be played sequentially as per your suggestion. The thing I am suggesting is that the alerting-the-user need not be controlled by the UAC, but it is can be controlled by UAS, as it has enough information to make that decision. According to the q value set, either the colorful ringtone or "please hangup.." will be played first. Whichever UAS finishes playing the early media first (according to the q value set), will start alerting the user. -- =20 Ravishankar. G. Shiroor Wipro Technologies, Bangalore. =20 ravishankar.shiroor@wipro.com -- =20 > -----Original Message----- > From: Stucker, Brian (RICH1:B620) [mailto:BSTUCKER@nortel.com]=20 > Sent: Thursday, August 03, 2006 9:21 AM > To: RaviShankar Shiroor (WT01 - Product Engineering=20 > Solutions); Sayan Chowdhury (WT01 - IP-Multimedia Carrier &=20 > Ent Networks); pkyzivat@cisco.com > Cc: christer.holmberg@ericsson.com; sipping@ietf.org;=20 > dean.willis@softarmor.com > Subject: RE: [Sipping] Some thoughts on fixing early media >=20 > Ok, so... >=20 > I have a call that forks. The first guy starts sending me=20 > colorful ringback, the second guy sends me "please hang up or=20 > I'm about to bill you a huge amount in 10 seconds". According=20 > to common practice with UAC design, it plays neither early=20 > media and instead generates local ringback. >=20 > You never hear the warning, the colorful ringback service is=20 > broken and the originator gets billed a huge amount, but the=20 > terminator didn't alert until after the warning was played. :) >=20 > If the UAC has no control, then the UAS has no guarantee that=20 > what they're wanting to send will get presented to the originator. >=20 > Regards, > Brian=20 >=20 > > -----Original Message----- > > From: ravishankar.shiroor@wipro.com > > [mailto:ravishankar.shiroor@wipro.com] > > Sent: Wednesday, August 02, 2006 11:26 PM > > To: sayan.chowdhury@wipro.com; Stucker, Brian (RICH1:B620);=20 > > pkyzivat@cisco.com > > Cc: christer.holmberg@ericsson.com; sipping@ietf.org;=20 > > dean.willis@softarmor.com > > Subject: RE: [Sipping] Some thoughts on fixing early media > >=20 > > Hi Sayan, > >=20 > > Would it not be the UAS's responsibility to decide that the user=20 > > should be alerted only after the Early Media is played? > > I think that the UAS which has early media to play, can decide to=20 > > postpone the alerting till it has been sent to the UAC. UAC=20 > need not=20 > > have to control this - I think this should satisfy all the=20 > use cases=20 > > needed. (for example "please hang-up in ten seconds from now unless=20 > > you wish to be billed a huge amount" or something like that=20 > should be=20 > > played before UAS alerts the user.) > >=20 > > Regards, > > Ravi. > >=20 > >=20 > > -- > > =20 > > Ravishankar. G. Shiroor > > Wipro Technologies, Bangalore. > > =20 > > ravishankar.shiroor@wipro.com > > -- > > =20 > >=20 > > > -----Original Message----- > > > From: sayan.chowdhury@wipro.com [mailto:sayan.chowdhury@wipro.com] > > > Sent: Tuesday, July 25, 2006 11:21 PM > > > To: bstucker@nortel.com; pkyzivat@cisco.com > > > Cc: christer.holmberg@ericsson.com; sipping@ietf.org;=20 > > > dean.willis@softarmor.com > > > Subject: RE: [Sipping] Some thoughts on fixing early media > > >=20 > > > Hello Brian , > > > I am trying to put a call flow here , (I will not try to=20 > draw as my=20 > > > mail client is messing it up) > > >=20 > > > 1>Callee sends an offer with emflow=3Dnone > > > 2>gets back say two 183 responses (establishing early dialog) > > > both with > > > early media class as two-way. > > > 3>it selects the 183 with the higher q-value and more > > > critical em class > > > and adds new offer in the PRACK with emflow as sendrecv. > > > 4>PRACKs the other 183 as well but with emflow=3Dnone and gets > > > back 200 OK > > > for that. > > > 4>Gets back the answer in the 200 OK to PRACK for the 183 > > with higher > > > q-value. > > > 5>Does early media.=20 > > > 6>Sends an UPDATE to the next 183 with lower q-value with=20 > emflow as > > > sendrcv > > > ... continues > > >=20 > > > If my understanding of the flow is correct , then in the above=20 > > > scenario how does the UAC indicate to the UAS(s) not to > > alert the user > > > until early media is over ? > > >=20 > > > Do we need a confirm-status mechanism as well wherein the > > UAC can tell > > > when to alert the UAS ? > > > There may be use cases when the UAC wants to defer the > > alerting until > > > the UAC has heard all the early media "flows". > > > For eg , when the UAC chooses which UAS it wants to have=20 > a regular=20 > > > media session with after hearing to all the early media sessions. > > >=20 > > > Regards , > > > Sayan > > >=20 > > > -----Original Message----- > > > From: Brian Stucker [mailto:bstucker@nortel.com] > > > Sent: Tuesday, July 25, 2006 9:02 PM > > > To: Paul Kyzivat > > > Cc: Dean Willis; sipping; Christer Holmberg (JO/LMF) > > > Subject: RE: [Sipping] Some thoughts on fixing early media > > >=20 > > > Yes, this is pretty much what's going to happen. I'll add > > that to the > > > text, actually, since I think it deserves being called out. > > >=20 > > > In cases where network interworking is not present (ie. no > > SIP-to-PSTN > > > type gateways are present in the routing tree) things will > > basically > > > behave as they always have been because most SIP endpoints do not=20 > > > generate early media. It's services like CRBT and PSTN > > gateways that > > > get us into trouble. For the other endpoints we simply > > don't wish to > > > clip the media if possible. For the ones that do generate > > early media, > > > we can fork to them, but don't necessarily want to hear > > what they all > > > have to say in as a chorus. > > >=20 > > > Regards, > > > Brian > > >=20 > > > > -----Original Message----- > > > > From: Paul Kyzivat [mailto:pkyzivat@cisco.com] > > > > Sent: Tuesday, July 25, 2006 8:03 AM > > > > To: Stucker, Brian (RICH1:B620) > > > > Cc: Christer Holmberg (JO/LMF); Dean Willis; Stephen > > Sprunk; sipping > > > > Subject: Re: [Sipping] Some thoughts on fixing early media > > > >=20 > > > > If the UAC only gives permission to one tine of the=20 > fork to send=20 > > > > media, and preconditions are used to prevent alerting until > > > permission > > >=20 > > > > to send is granted, then parallel forking has been turned > > > into serial > > > > forking under the control of the UAC. > > > >=20 > > > > While it seems a little odd, it is perhaps the best that > > > can be done > > > > under the circumstances. It works well in that those > > > destinations that > > >=20 > > > > don't require early media can all alert in parallel, and > > > only the ones > > >=20 > > > > requiring early media are serialized. That is a huge > > > improvement over > > > > the proxy making the choice, because it doesn't know which > > > ones will > > > > require early media. > > > >=20 > > > > Paul > > > >=20 > > > > Brian Stucker wrote: > > > > > Yes, I think you understand what I am suggesting. > > > > >=20 > > > > > Regards, > > > > > Brian > > > > >=20 > > > > > -----Original Message----- > > > > > From: Christer Holmberg (JO/LMF)=20 > > > > > [mailto:christer.holmberg@ericsson.com] > > > > >=20 > > > > > Sent: Sunday, July 23, 2006 5:20 PM > > > > > To: Stucker, Brian (RICH1:B620); Dean Willis; Stephen Sprunk > > > > > Cc: sipping > > > > > Subject: VS: [Sipping] Some thoughts on fixing early media > > > > >=20 > > > > > Hi, > > > > >=20 > > > > >> I'm proposing that the far end would not alert until > > > permission to > > > > >> send > > > > >=20 > > > > >> media had been granted. > > > > >=20 > > > > > That's called pre-conditions, isn't it? :) > > > > > =20 > > > > > Or, you could also send an INVITE WITH SDP, but the UAS > > would not > > > > > altert the user until some other type of indicator has been > > > > sent to it. > > > > > =20 > > > > > Regards, > > > > > =20 > > > > > Christer > > > > > =20 > > > > >=20 > > > > >=20 > > > > > -----Original Message----- > > > > > From: Christer Holmberg=20 > [mailto:christer.holmberg@ericsson.com] > > > > > Sent: Sunday, July 23, 2006 12:14 AM > > > > > To: Stucker, Brian (RICH1:B620); Dean Willis; Stephen Sprunk > > > > > Cc: sipping > > > > > Subject: RE: [Sipping] Some thoughts on fixing early media > > > > >=20 > > > > > Hi Brian, > > > > >=20 > > > > > What if an UAS wants to send 200 OK (and media) before > > > the UAC has > > > > > told him it's ok to send media? Someone would have to send > > > > an UPDATE, > > > > > to switch the media to sendrecv. > > > > >=20 > > > > > Or, are you proposing that the UAS, until it's given > > > permission to > > > > > send media, wouldn't even alert the user behind it? > > > > >=20 > > > > > Regards, > > > > >=20 > > > > > Christer > > > > >=20 > > > > >=20 > > > > >=20 > > > > >=20 > > > > > _______________________________________________ > > > > > Sipping mailing list > > > https://www1.ietf.org/mailman/listinfo/sipping > > > > > This list is for NEW development of the application=20 > of SIP Use=20 > > > > > sip-implementors@cs.columbia.edu for questions on > > current sip Use > > > > > sip@ietf.org for new developments of core SIP > > > > >=20 > > > >=20 > > >=20 > > > _______________________________________________ > > > Sipping mailing list =20 > https://www1.ietf.org/mailman/listinfo/sipping > > > This list is for NEW development of the application of SIP Use=20 > > > sip-implementors@cs.columbia.edu for questions on current sip Use=20 > > > sip@ietf.org for new developments of core SIP > > >=20 > > > _______________________________________________ > > > Sipping mailing list =20 > https://www1.ietf.org/mailman/listinfo/sipping > > > This list is for NEW development of the application of SIP Use=20 > > > sip-implementors@cs.columbia.edu for questions on current sip Use=20 > > > sip@ietf.org for new developments of core SIP > > >=20 > >=20 >=20 _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Thu Aug 03 01:47:05 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8W2o-0000Ex-Hk; Thu, 03 Aug 2006 01:46:54 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8W2n-0000Cr-5q for sipping@ietf.org; Thu, 03 Aug 2006 01:46:53 -0400 Received: from mailgw4.ericsson.se ([193.180.251.62]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G8W2m-0005FB-02 for sipping@ietf.org; Thu, 03 Aug 2006 01:46:53 -0400 Received: from esealmw129.eemea.ericsson.se (unknown [153.88.254.120]) by mailgw4.ericsson.se (Symantec Mail Security) with ESMTP id B6E3B6E0001; Thu, 3 Aug 2006 07:46:02 +0200 (CEST) Received: from esealmw127.eemea.ericsson.se ([153.88.254.171]) by esealmw129.eemea.ericsson.se with Microsoft SMTPSVC(6.0.3790.1830); Thu, 3 Aug 2006 07:46:02 +0200 Received: from esealmw113.eemea.ericsson.se ([153.88.200.4]) by esealmw127.eemea.ericsson.se with Microsoft SMTPSVC(6.0.3790.1830); Thu, 3 Aug 2006 07:46:01 +0200 X-MimeOLE: Produced By Microsoft Exchange V6.5 Content-class: urn:content-classes:message MIME-Version: 1.0 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: quoted-printable Subject: RE: [Sipping] Some thoughts on fixing early media Date: Thu, 3 Aug 2006 07:46:01 +0200 Message-ID: <5EB80D22825EEE42872083AD5BFFB594017F12A3@esealmw113.eemea.ericsson.se> In-Reply-To: <6FC4416DDE56C44DA0AEE67BC7CA43710FBDA0FC@zrc2hxm2.corp.nortel.com> X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: [Sipping] Some thoughts on fixing early media Thread-Index: Acav6si/KX4K3NwnQFGu7NssoL4r7gAFBZXwAAT8rwABjPJ3AAAaSNXwAAPgO6A= From: "Christer Holmberg \(JO/LMF\)" To: "Brian Stucker" , , , X-OriginalArrivalTime: 03 Aug 2006 05:46:01.0853 (UTC) FILETIME=[1A571AD0:01C6B6C0] X-Brightmail-Tracker: AAAAAA== X-Spam-Score: 0.0 (/) X-Scan-Signature: 40161b1d86420e0807d771943d981d25 Cc: sipping@ietf.org, dean.willis@softarmor.com X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Hi Brian, I think there are two issues. One is that the UAC gives permission to the UAS to play early media. But, once that permission is given I am not sure the UAC needs to give the UAS permission to alert the user. I think, as Ravinhankar said, it's an UAS implementation issue when it alterts the user. Also, even if the UAS is NOT given permisson to play early media (or, the UAC indicates it won't listen to it), I think it's an UAS implementation issue whether it "waits" for permission, or alterts the UAS anyway, depending on the importance of the early media that the UAS would have wanted to play.=20 So, if a UAS would like to send a "please hang up or I'll bill you a huge amount in 10 seconds" early media message, but is not given permission to do so, I guess it should not wait for permission more than 10 seconds, but instead send an error response. Regards, Christer -----Original Message----- From: Brian Stucker [mailto:bstucker@nortel.com]=20 Sent: 3. elokuuta 2006 6:51 To: ravishankar.shiroor@wipro.com; sayan.chowdhury@wipro.com; pkyzivat@cisco.com Cc: Christer Holmberg (JO/LMF); sipping@ietf.org; dean.willis@softarmor.com Subject: RE: [Sipping] Some thoughts on fixing early media Ok, so... I have a call that forks. The first guy starts sending me colorful ringback, the second guy sends me "please hang up or I'm about to bill you a huge amount in 10 seconds". According to common practice with UAC design, it plays neither early media and instead generates local ringback. You never hear the warning, the colorful ringback service is broken and the originator gets billed a huge amount, but the terminator didn't alert until after the warning was played. :) If the UAC has no control, then the UAS has no guarantee that what they're wanting to send will get presented to the originator. Regards, Brian=20 > -----Original Message----- > From: ravishankar.shiroor@wipro.com > [mailto:ravishankar.shiroor@wipro.com] > Sent: Wednesday, August 02, 2006 11:26 PM > To: sayan.chowdhury@wipro.com; Stucker, Brian (RICH1:B620);=20 > pkyzivat@cisco.com > Cc: christer.holmberg@ericsson.com; sipping@ietf.org;=20 > dean.willis@softarmor.com > Subject: RE: [Sipping] Some thoughts on fixing early media >=20 > Hi Sayan, >=20 > Would it not be the UAS's responsibility to decide that the user=20 > should be alerted only after the Early Media is played? > I think that the UAS which has early media to play, can decide to=20 > postpone the alerting till it has been sent to the UAC. UAC need not=20 > have to control this - I think this should satisfy all the use cases=20 > needed. (for example "please hang-up in ten seconds from now unless=20 > you wish to be billed a huge amount" or something like that should be=20 > played before UAS alerts the user.) >=20 > Regards, > Ravi. >=20 >=20 > -- > =20 > Ravishankar. G. Shiroor > Wipro Technologies, Bangalore. > =20 > ravishankar.shiroor@wipro.com > -- > =20 >=20 > > -----Original Message----- > > From: sayan.chowdhury@wipro.com [mailto:sayan.chowdhury@wipro.com] > > Sent: Tuesday, July 25, 2006 11:21 PM > > To: bstucker@nortel.com; pkyzivat@cisco.com > > Cc: christer.holmberg@ericsson.com; sipping@ietf.org;=20 > > dean.willis@softarmor.com > > Subject: RE: [Sipping] Some thoughts on fixing early media > >=20 > > Hello Brian , > > I am trying to put a call flow here , (I will not try to draw as my=20 > > mail client is messing it up) > >=20 > > 1>Callee sends an offer with emflow=3Dnone > > 2>gets back say two 183 responses (establishing early dialog) > > both with > > early media class as two-way. > > 3>it selects the 183 with the higher q-value and more > > critical em class > > and adds new offer in the PRACK with emflow as sendrecv. > > 4>PRACKs the other 183 as well but with emflow=3Dnone and gets > > back 200 OK > > for that. > > 4>Gets back the answer in the 200 OK to PRACK for the 183 > with higher > > q-value. > > 5>Does early media.=20 > > 6>Sends an UPDATE to the next 183 with lower q-value with emflow as > > sendrcv > > ... continues > >=20 > > If my understanding of the flow is correct , then in the above=20 > > scenario how does the UAC indicate to the UAS(s) not to > alert the user > > until early media is over ? > >=20 > > Do we need a confirm-status mechanism as well wherein the > UAC can tell > > when to alert the UAS ? > > There may be use cases when the UAC wants to defer the > alerting until > > the UAC has heard all the early media "flows". > > For eg , when the UAC chooses which UAS it wants to have a regular=20 > > media session with after hearing to all the early media sessions. > >=20 > > Regards , > > Sayan > >=20 > > -----Original Message----- > > From: Brian Stucker [mailto:bstucker@nortel.com] > > Sent: Tuesday, July 25, 2006 9:02 PM > > To: Paul Kyzivat > > Cc: Dean Willis; sipping; Christer Holmberg (JO/LMF) > > Subject: RE: [Sipping] Some thoughts on fixing early media > >=20 > > Yes, this is pretty much what's going to happen. I'll add > that to the > > text, actually, since I think it deserves being called out. > >=20 > > In cases where network interworking is not present (ie. no > SIP-to-PSTN > > type gateways are present in the routing tree) things will > basically > > behave as they always have been because most SIP endpoints do not=20 > > generate early media. It's services like CRBT and PSTN > gateways that > > get us into trouble. For the other endpoints we simply > don't wish to > > clip the media if possible. For the ones that do generate > early media, > > we can fork to them, but don't necessarily want to hear > what they all > > have to say in as a chorus. > >=20 > > Regards, > > Brian > >=20 > > > -----Original Message----- > > > From: Paul Kyzivat [mailto:pkyzivat@cisco.com] > > > Sent: Tuesday, July 25, 2006 8:03 AM > > > To: Stucker, Brian (RICH1:B620) > > > Cc: Christer Holmberg (JO/LMF); Dean Willis; Stephen > Sprunk; sipping > > > Subject: Re: [Sipping] Some thoughts on fixing early media > > >=20 > > > If the UAC only gives permission to one tine of the fork to send=20 > > > media, and preconditions are used to prevent alerting until > > permission > >=20 > > > to send is granted, then parallel forking has been turned > > into serial > > > forking under the control of the UAC. > > >=20 > > > While it seems a little odd, it is perhaps the best that > > can be done > > > under the circumstances. It works well in that those > > destinations that > >=20 > > > don't require early media can all alert in parallel, and > > only the ones > >=20 > > > requiring early media are serialized. That is a huge > > improvement over > > > the proxy making the choice, because it doesn't know which > > ones will > > > require early media. > > >=20 > > > Paul > > >=20 > > > Brian Stucker wrote: > > > > Yes, I think you understand what I am suggesting. > > > >=20 > > > > Regards, > > > > Brian > > > >=20 > > > > -----Original Message----- > > > > From: Christer Holmberg (JO/LMF)=20 > > > > [mailto:christer.holmberg@ericsson.com] > > > >=20 > > > > Sent: Sunday, July 23, 2006 5:20 PM > > > > To: Stucker, Brian (RICH1:B620); Dean Willis; Stephen Sprunk > > > > Cc: sipping > > > > Subject: VS: [Sipping] Some thoughts on fixing early media > > > >=20 > > > > Hi, > > > >=20 > > > >> I'm proposing that the far end would not alert until > > permission to > > > >> send > > > >=20 > > > >> media had been granted. > > > >=20 > > > > That's called pre-conditions, isn't it? :) > > > > =20 > > > > Or, you could also send an INVITE WITH SDP, but the UAS > would not > > > > altert the user until some other type of indicator has been > > > sent to it. > > > > =20 > > > > Regards, > > > > =20 > > > > Christer > > > > =20 > > > >=20 > > > >=20 > > > > -----Original Message----- > > > > From: Christer Holmberg [mailto:christer.holmberg@ericsson.com] > > > > Sent: Sunday, July 23, 2006 12:14 AM > > > > To: Stucker, Brian (RICH1:B620); Dean Willis; Stephen Sprunk > > > > Cc: sipping > > > > Subject: RE: [Sipping] Some thoughts on fixing early media > > > >=20 > > > > Hi Brian, > > > >=20 > > > > What if an UAS wants to send 200 OK (and media) before > > the UAC has > > > > told him it's ok to send media? Someone would have to send > > > an UPDATE, > > > > to switch the media to sendrecv. > > > >=20 > > > > Or, are you proposing that the UAS, until it's given > > permission to > > > > send media, wouldn't even alert the user behind it? > > > >=20 > > > > Regards, > > > >=20 > > > > Christer > > > >=20 > > > >=20 > > > >=20 > > > >=20 > > > > _______________________________________________ > > > > Sipping mailing list > > https://www1.ietf.org/mailman/listinfo/sipping > > > > This list is for NEW development of the application of SIP Use=20 > > > > sip-implementors@cs.columbia.edu for questions on > current sip Use > > > > sip@ietf.org for new developments of core SIP > > > >=20 > > >=20 > >=20 > > _______________________________________________ > > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > > This list is for NEW development of the application of SIP Use=20 > > sip-implementors@cs.columbia.edu for questions on current sip Use=20 > > sip@ietf.org for new developments of core SIP > >=20 > > _______________________________________________ > > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > > This list is for NEW development of the application of SIP Use=20 > > sip-implementors@cs.columbia.edu for questions on current sip Use=20 > > sip@ietf.org for new developments of core SIP > >=20 >=20 _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Thu Aug 03 02:18:24 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8WXE-0004WC-E2; Thu, 03 Aug 2006 02:18:20 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8WXC-0004W7-Ku for sipping@ietf.org; Thu, 03 Aug 2006 02:18:18 -0400 Received: from mailgate.siemenscomms.co.uk ([195.171.110.225] helo=bemg01.siemenscomms.co.uk) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G8WX7-0007ws-9J for sipping@ietf.org; Thu, 03 Aug 2006 02:18:18 -0400 Received: from CONVERSION-DAEMON.siemenscomms.co.uk by siemenscomms.co.uk (PMDF V6.0-24 #40642) id <0J3E00101S6WCL@siemenscomms.co.uk> for sipping@ietf.org; Thu, 03 Aug 2006 07:18:32 +0100 (BST) Received: from ntht207e.uksgcs.siemenscomms.co.uk ([137.223.247.82]) by siemenscomms.co.uk (PMDF V6.0-24 #40642) with ESMTP id <0J3E00M50S6WQB@siemenscomms.co.uk>; Thu, 03 Aug 2006 07:18:32 +0100 (BST) Received: by ntht207e.uksgcs.siemenscomms.co.uk with Internet Mail Service (5.5.2657.72) id ; Thu, 03 Aug 2006 07:18:11 +0100 Content-return: allowed Date: Thu, 03 Aug 2006 07:18:10 +0100 From: "Elwell, John" Subject: RE: [Sipping] Question on do not disturb indication To: Scott Lawrence Message-id: <50B1CBA96870A34799A506B2313F266709897F65@ntht201e.siemenscomms.co.uk> MIME-version: 1.0 X-Mailer: Internet Mail Service (5.5.2657.72) Content-type: text/plain Content-transfer-encoding: 7BIT X-Spam-Score: 0.0 (/) X-Scan-Signature: 5a9a1bd6c2d06a21d748b7d0070ddcb8 Cc: sipping@ietf.org X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Scott, Thanks for your response. I too like 480, but some people are telling me that it has too many other uses, so if you want to convey a definite indication that you are in DND they felt something else might be needed. The chief cause of concern seems to be that RFC 3398 recommends mapping a couple of Q.850 causes to 480 (19 no answer from user and 20 subscriber absent), and although these both have the general meaning of user temporarily unavailable, neither has quite the semantics of DND. John > -----Original Message----- > From: Scott Lawrence [mailto:slawrence@pingtel.com] > Sent: 02 August 2006 22:15 > To: Elwell, John > Cc: sipping@ietf.org > Subject: Re: [Sipping] Question on do not disturb indication > > On Wed, 2006-08-02 at 21:09 +0100, Elwell, John wrote: > > RFC 3261 suggests the use of response code 480 Temporarily > Unavailable > > for indicating do not disturb (DND), but it is not used exclusively > > for this purpose. There are other documents (e.g., RFC 4504) that > > suggest the use of 486 Busy Here, but again this is not > helpful if the > > receiver wants to distinguish between DND and ordinary > busy. Presence > > can be used to indicate this condition through the RPID > note element, > > but that does not say why a particular session establishment attempt > > has failed. What are the best practices for indicating the DND > > condition? Does anyone see a need for an explicit indicator for this > > purpose, e.g., a new SIP response code? > > Personally, I like 480 for this. > > If the phone wanted to be sneaky, it could just mute the ringing and > allow the transaction to time out without explicitly > rejecting it. That > prevents the caller from being able to tell they are being > disregarded, > but some users might prefer that forwarding take effect sooner and not > like that. > > The one thing that I _don't_ like to see is a 603 being used for this; > that messes up any upstream forking. > > > -- > Scott Lawrence tel:+1-781-938-5306;ext=162 or > sip:slawrence@pingtel.com > sipXpbx project coordinator - SIPfoundry > http://www.sipfoundry.org/sipX > Chief Architect - Pingtel Corp. http://www.pingtel.com/ > > _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Thu Aug 03 02:27:38 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8WgD-0004kx-Hr; Thu, 03 Aug 2006 02:27:37 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8WgC-0004eg-2x for sipping@ietf.org; Thu, 03 Aug 2006 02:27:36 -0400 Received: from mailgate.siemenscomms.co.uk ([195.171.110.225] helo=bemg01.siemenscomms.co.uk) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G8WgB-0000b9-Ju for sipping@ietf.org; Thu, 03 Aug 2006 02:27:36 -0400 Received: from CONVERSION-DAEMON.siemenscomms.co.uk by siemenscomms.co.uk (PMDF V6.0-24 #40642) id <0J3E00201SL9YZ@siemenscomms.co.uk> for sipping@ietf.org; Thu, 03 Aug 2006 07:27:39 +0100 (BST) Received: from ntht207e.uksgcs.siemenscomms.co.uk ([137.223.247.82]) by siemenscomms.co.uk (PMDF V6.0-24 #40642) with ESMTP id <0J3E002E5SKZ8V@siemenscomms.co.uk>; Thu, 03 Aug 2006 07:26:59 +0100 (BST) Received: by ntht207e.uksgcs.siemenscomms.co.uk with Internet Mail Service (5.5.2657.72) id ; Thu, 03 Aug 2006 07:26:38 +0100 Content-return: allowed Date: Thu, 03 Aug 2006 07:26:38 +0100 From: "Elwell, John" Subject: RE: [Sipping] Question on do not disturb indication To: Francois Audet , Paul Kyzivat , Scott Lawrence Message-id: <50B1CBA96870A34799A506B2313F266709897F66@ntht201e.siemenscomms.co.uk> MIME-version: 1.0 X-Mailer: Internet Mail Service (5.5.2657.72) Content-type: text/plain Content-transfer-encoding: 7BIT X-Spam-Score: 0.0 (/) X-Scan-Signature: 5d7a7e767f20255fce80fa0b77fb2433 Cc: sipping@ietf.org X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Francois, See below. John > -----Original Message----- > From: Francois Audet [mailto:audet@nortel.com] > Sent: 03 August 2006 00:24 > To: Paul Kyzivat; Scott Lawrence > Cc: sipping@ietf.org; Elwell, John > Subject: RE: [Sipping] Question on do not disturb indication > > I agree: I don't see a need for a special code for this, as > it seems to > make > pretty much exactly to existing practice. > > In fact, even in the "old days" you had different features > corresponding > to > the two codes. > - Do-not-Disturb is pretty much like 480 > - Make-busy is pretty much like 486 [JRE] I agree that "make busy" is a separate feature and is best dealt with by 486. But where I explicitly want the caller to know that I am in DND (possibly with a retry-after time), do you think that 480 is sufficient, bearing in mind that it gets used for other purposes too, e.g., in RFC3398? > > Some vendors have "make busy", others have "do not disturb", some may > have > both. > > On the 603 issue, I'd like to disagree with everybody else... > Sometimes being able to decline the call everywhere is > actually what you > want. > As an example, I have multiple forked device. If I send a 603, it > triggers the proxy to > use whatever alternate routing is sees fit (e.g., voicemail, > attendant, > etc.). That's > the desired behavior. [JRE] I tend to agree that 603 is useful, but I find the other 6xx codes less useful. 603 is saying I reject this call, period. That is quite different from DND, where I don't want to be disturbed on my present device but might be quite happy for my call to another device, where an assistant perhaps could answer, or voicemail. > > Problem with a 480/486 with that scenario is that it won't trigger the > alternate routing > and will keep on ringing the other fork. That may not be the desired > behavior. > > My 2 cents. > > > -----Original Message----- > > From: Paul Kyzivat [mailto:pkyzivat@cisco.com] > > Sent: Wednesday, August 02, 2006 3:31 PM > > To: Scott Lawrence > > Cc: sipping@ietf.org; Elwell, John > > Subject: Re: [Sipping] Question on do not disturb indication > > > > I pretty much agree with Scott. In many/most cases I won't > > want the caller to know why his call is going unanswered. > > Both 480 and 486 are in effect "white lies". Which one to use > > may depend on which impression you want to give to the > > caller. Another one - if you want to be more rude - is 403 > Forbidden. > > > > I don't see a need for a *special* code that is unique to > > this service. > > > > Paul > > > > Scott Lawrence wrote: > > > On Wed, 2006-08-02 at 21:09 +0100, Elwell, John wrote: > > >> RFC 3261 suggests the use of response code 480 Temporarily > > >> Unavailable for indicating do not disturb (DND), but it is > > not used > > >> exclusively for this purpose. There are other documents > (e.g., RFC > > >> 4504) that suggest the use of 486 Busy Here, but again > this is not > > >> helpful if the receiver wants to distinguish between DND > > and ordinary > > >> busy. Presence can be used to indicate this condition > through the > > >> RPID note element, but that does not say why a > particular session > > >> establishment attempt has failed. What are the best > practices for > > >> indicating the DND condition? Does anyone see a need for > > an explicit > > >> indicator for this purpose, e.g., a new SIP response code? > > > > > > Personally, I like 480 for this. > > > > > > If the phone wanted to be sneaky, it could just mute the > > ringing and > > > allow the transaction to time out without explicitly > rejecting it. > > > That prevents the caller from being able to tell they are being > > > disregarded, but some users might prefer that forwarding > > take effect > > > sooner and not like that. > > > > > > The one thing that I _don't_ like to see is a 603 being > > used for this; > > > that messes up any upstream forking. > > > > > > > > > > _______________________________________________ > > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > > This list is for NEW development of the application of SIP > > Use sip-implementors@cs.columbia.edu for questions on current > > sip Use sip@ietf.org for new developments of core SIP > > > _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Thu Aug 03 02:28:58 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8WhW-00063f-EW; Thu, 03 Aug 2006 02:28:58 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8WhU-00063T-G7 for sipping@ietf.org; Thu, 03 Aug 2006 02:28:56 -0400 Received: from mailgate.siemenscomms.co.uk ([195.171.110.225] helo=bemg01.siemenscomms.co.uk) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G8WhT-0000mq-7F for sipping@ietf.org; Thu, 03 Aug 2006 02:28:56 -0400 Received: from CONVERSION-DAEMON.siemenscomms.co.uk by siemenscomms.co.uk (PMDF V6.0-24 #40642) id <0J3E00401SOH6K@siemenscomms.co.uk> for sipping@ietf.org; Thu, 03 Aug 2006 07:29:06 +0100 (BST) Received: from ntht207e.uksgcs.siemenscomms.co.uk ([137.223.247.82]) by siemenscomms.co.uk (PMDF V6.0-24 #40642) with ESMTP id <0J3E002FRSO38V@siemenscomms.co.uk>; Thu, 03 Aug 2006 07:28:51 +0100 (BST) Received: by ntht207e.uksgcs.siemenscomms.co.uk with Internet Mail Service (5.5.2657.72) id ; Thu, 03 Aug 2006 07:28:30 +0100 Content-return: allowed Date: Thu, 03 Aug 2006 07:28:28 +0100 From: "Elwell, John" Subject: RE: [Sipping] Question on do not disturb indication To: Dean Willis , "Dolly, Martin C, ALABS" Message-id: <50B1CBA96870A34799A506B2313F266709897F68@ntht201e.siemenscomms.co.uk> MIME-version: 1.0 X-Mailer: Internet Mail Service (5.5.2657.72) Content-type: text/plain Content-transfer-encoding: 7BIT X-Spam-Score: 0.0 (/) X-Scan-Signature: 52e1467c2184c31006318542db5614d5 Cc: Paul Kyzivat , sipping@ietf.org, Scott Lawrence X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Dean, So what would you like the IETF to recommend? Do you believe 480 mostly works, or do you think something new is needed? John > -----Original Message----- > From: Dean Willis [mailto:dean.willis@softarmor.com] > Sent: 03 August 2006 04:17 > To: Dolly, Martin C, ALABS > Cc: Paul Kyzivat; Scott Lawrence; Elwell, John; sipping@ietf.org > Subject: Re: [Sipping] Question on do not disturb indication > > > On Aug 2, 2006, at 5:58 PM, Dolly, Martin C, ALABS wrote: > > > Is not this a question for the market to decide? > > Do any of you think we can speak for J.Q. Public? Or should. > > Well, I personally speak for at least a portion of the service- > consuming market. Myself. After all, I AM an AT&T customer for some > of my services. And a Comcast customer, Verizon customer, Skype > customer, etc. > > Or maybe we should have a national poll on which code to use? Heck, > people on the street can't seven seem to be able to pick a > president, > much less tell a 404 from an ACM. > > So, as a service-consuming member of the market, I'd rather have the > IETF recommend something that mostly works, thanks. > > -- > Dean > _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Thu Aug 03 03:37:54 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8Xm2-0002Xw-S6; Thu, 03 Aug 2006 03:37:42 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8Xm1-0002XY-3d for sipping@ietf.org; Thu, 03 Aug 2006 03:37:41 -0400 Received: from host10.216.41.24.conversent.net ([216.41.24.10] helo=acmepacket.com) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G8Xlz-00058l-Re for sipping@ietf.org; Thu, 03 Aug 2006 03:37:41 -0400 Received: from hkaplan [24.54.31.12] by acmepacket.com with ESMTP (SMTPD-9.03) id A7C90668; Thu, 03 Aug 2006 03:37:45 -0400 From: "Hadriel Kaplan" To: "'Scott Lawrence'" Subject: RE: [Sipping] Question on do not disturb indication Date: Thu, 3 Aug 2006 03:37:35 -0400 Message-ID: <03bb01c6b6cf$b0757110$6601a8c0@acmepacket.com> MIME-Version: 1.0 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: 7bit X-Mailer: Microsoft Office Outlook 11 X-MimeOLE: Produced By Microsoft MimeOLE V6.00.2900.2869 Thread-Index: Aca2j490qNOTlo1PTKel8s0H3KM0hQAOptTg In-Reply-To: <1154563021.3841.13.camel@localhost.localdomain> X-Spam-Score: 0.1 (/) X-Scan-Signature: 9466e0365fc95844abaf7c3f15a05c7d Cc: sipping@ietf.org X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org > -----Original Message----- > From: Scott Lawrence [mailto:slawrence@pingtel.com] > [about a 603 response...] > I read that to mean that a forking proxy should abandon all forks for > the call and terminate the request - the response asserts that the > voicemail (another end point) will not answer. > > That might be an appropriate response for a proxy to send if it is > authoritative for the domain and knows that a particular user does not > exist, but I can't see how a user agent can ever have enough global > knowledge to justify a 6xx response. That would be a 604 (Does not exist anywhere), no? A 603 could be useful someday for personal black-lists; for example where the user agent may decide who to give 603's to based on the From, and not send them to voicemail. The problem is just because I pressed DND doesn't mean I don't want it to go to voicemail - I do want it to go to voicemail. So 6xx is wrong. But sending a 480 typically only makes it go to voicemail if there are no other contacts to try, or they all send 480. So I have to press DND on all my phones; but I'm not sure that's avoidable. -hadriel _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Thu Aug 03 07:18:24 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8bDL-000141-Px; Thu, 03 Aug 2006 07:18:07 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8bDK-00013w-Kl for sipping@ietf.org; Thu, 03 Aug 2006 07:18:06 -0400 Received: from nf-out-0910.google.com ([64.233.182.186]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G8bDI-00019b-4X for sipping@ietf.org; Thu, 03 Aug 2006 07:18:06 -0400 Received: by nf-out-0910.google.com with SMTP id n15so1037878nfc for ; Thu, 03 Aug 2006 04:18:03 -0700 (PDT) DomainKey-Signature: a=rsa-sha1; q=dns; c=nofws; s=beta; d=gmail.com; h=received:message-id:date:from:sender:to:subject:cc:in-reply-to:mime-version:content-type:content-transfer-encoding:content-disposition:references:x-google-sender-auth; b=ByAAmud3t3S7hQSFEVH3HzzfLaTt+6k46QFGHTiCPFOh7XD1eEfq4zaElK4wc11FlYrLjqoUzBldEbyR5sczxJW2Gkytdw+G8mtEHihlZj/C3F5Nzgez0FCvi3UQkwJM1aOW0mAoaryLNc24SCROQpL7eTavxBP7OFvdn4md51I= Received: by 10.48.254.1 with SMTP id b1mr3609756nfi; Thu, 03 Aug 2006 04:18:02 -0700 (PDT) Received: by 10.48.164.7 with HTTP; Thu, 3 Aug 2006 04:18:02 -0700 (PDT) Message-ID: Date: Thu, 3 Aug 2006 07:18:02 -0400 From: "Xiaotao Wu" To: "Francois Audet" Subject: Re: [Sipping] Question on do not disturb indication In-Reply-To: <1ECE0EB50388174790F9694F77522CCF0BF08368@zrc2hxm0.corp.nortel.com> MIME-Version: 1.0 Content-Type: text/plain; charset=ISO-8859-1; format=flowed Content-Transfer-Encoding: 7bit Content-Disposition: inline References: <44D12795.4080901@cisco.com> <1ECE0EB50388174790F9694F77522CCF0BF08368@zrc2hxm0.corp.nortel.com> X-Google-Sender-Auth: ec7a41b6703578ea X-Spam-Score: 0.0 (/) X-Scan-Signature: 9a2be21919e71dc6faef12b370c4ecf5 Cc: sipping@ietf.org, Paul Kyzivat , "Elwell, John" , Scott Lawrence X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org On 8/2/06, Francois Audet wrote: > On the 603 issue, I'd like to disagree with everybody else... > Sometimes being able to decline the call everywhere is actually what you > want. > As an example, I have multiple forked device. If I send a 603, it > triggers the proxy to > use whatever alternate routing is sees fit (e.g., voicemail, attendant, > etc.). That's > the desired behavior. > > Problem with a 480/486 with that scenario is that it won't trigger the > alternate routing > and will keep on ringing the other fork. That may not be the desired > behavior. I would expect the response should be sent by the proxy, instead by endpoints. If it's send by an endpoint, things may get complicated. For example, Bob has a desk phone, a cell phone, a voicemail, and a secretary. It's hard to compose a response from his desk phone to indicate that he does not want to ring his desk phone, and cell phone, but ring his secretary phone and voicemail. So, I think the problem is more on what's the expected code for the caller, not on the routing behavior of the request/response. -Xiaotao > > My 2 cents. > > > -----Original Message----- > > From: Paul Kyzivat [mailto:pkyzivat@cisco.com] > > Sent: Wednesday, August 02, 2006 3:31 PM > > To: Scott Lawrence > > Cc: sipping@ietf.org; Elwell, John > > Subject: Re: [Sipping] Question on do not disturb indication > > > > I pretty much agree with Scott. In many/most cases I won't > > want the caller to know why his call is going unanswered. > > Both 480 and 486 are in effect "white lies". Which one to use > > may depend on which impression you want to give to the > > caller. Another one - if you want to be more rude - is 403 Forbidden. > > > > I don't see a need for a *special* code that is unique to > > this service. > > > > Paul > > > > Scott Lawrence wrote: > > > On Wed, 2006-08-02 at 21:09 +0100, Elwell, John wrote: > > >> RFC 3261 suggests the use of response code 480 Temporarily > > >> Unavailable for indicating do not disturb (DND), but it is > > not used > > >> exclusively for this purpose. There are other documents (e.g., RFC > > >> 4504) that suggest the use of 486 Busy Here, but again this is not > > >> helpful if the receiver wants to distinguish between DND > > and ordinary > > >> busy. Presence can be used to indicate this condition through the > > >> RPID note element, but that does not say why a particular session > > >> establishment attempt has failed. What are the best practices for > > >> indicating the DND condition? Does anyone see a need for > > an explicit > > >> indicator for this purpose, e.g., a new SIP response code? > > > > > > Personally, I like 480 for this. > > > > > > If the phone wanted to be sneaky, it could just mute the > > ringing and > > > allow the transaction to time out without explicitly rejecting it. > > > That prevents the caller from being able to tell they are being > > > disregarded, but some users might prefer that forwarding > > take effect > > > sooner and not like that. > > > > > > The one thing that I _don't_ like to see is a 603 being > > used for this; > > > that messes up any upstream forking. > > > > > > > > > > _______________________________________________ > > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > > This list is for NEW development of the application of SIP > > Use sip-implementors@cs.columbia.edu for questions on current > > sip Use sip@ietf.org for new developments of core SIP > > > > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP > Use sip-implementors@cs.columbia.edu for questions on current sip > Use sip@ietf.org for new developments of core SIP > -- -Xiaotao =========================================================== Name : Xiaotao Wu Email : xwu@avaya.com, xiaotaow@gmail.com Web : http://www.cs.columbia.edu/~xiaotaow Phone : (732) 852-2133 SIP : sip:xiaotaow@cs.columbia.edu Address : 1M-240, 307 Middletown Lincroft Road Lincroft, NJ 07738-1526, U.S.A. _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Thu Aug 03 08:06:00 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8bxb-0006Gl-DS; Thu, 03 Aug 2006 08:05:55 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8bxZ-0006Gg-Pe for sipping@ietf.org; Thu, 03 Aug 2006 08:05:53 -0400 Received: from jes-fe2.zx.nl ([194.187.76.133]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G8bxY-0005WK-CO for sipping@ietf.org; Thu, 03 Aug 2006 08:05:53 -0400 Received: from Inbox ([89.205.129.198]) by jes-fe2.zx.nl (Sun Java System Messaging Server 6.2-2.05 (built Apr 28 2005)) with ESMTP id <0J3F00I48B8RG700@jes-fe2.zx.nl> for sipping@ietf.org; Thu, 03 Aug 2006 14:10:20 +0100 (WEST) Date: Thu, 03 Aug 2006 14:05:44 +0200 From: Jeroen Van Bemmel Subject: RE: [Sipping] Question on do not disturb indication To: Xiaotao Wu , Francois Audet Message-id: <0J3F00I49B8SG700@jes-fe2.zx.nl> MIME-version: 1.0 Content-type: text/plain; charset=iso-8859-1 Content-transfer-encoding: quoted-printable Importance: normal X-Priority: 3 X-Spam-Score: 1.2 (+) X-Scan-Signature: 7a6398bf8aaeabc7a7bb696b6b0a2aad Cc: "Elwell, John" , Paul Kyzivat , sipping@ietf.org, Scott Lawrence X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org So, I think the problem is more on what's the expected code for the caller, not on the routing behavior of the request/response. (JvB) The reason phrase is (and was always) intended for this. "480 Do not = disturb (use IM, call my secretary or leave a voiemail)" would seem appropr= iate. The semantics are hard to convey to a machine, therefore ideally the = choice for any follow ups should be made by the caller IMHO. A 6xx would gu= arantee that outcome Regards, Jeroen=20 _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Thu Aug 03 08:12:00 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8c3T-0003Wj-AN; Thu, 03 Aug 2006 08:11:59 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8c3R-0003QF-5F for sipping@ietf.org; Thu, 03 Aug 2006 08:11:57 -0400 Received: from zcars04e.nortel.com ([47.129.242.56]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G8c3Q-0006fW-Kd for sipping@ietf.org; Thu, 03 Aug 2006 08:11:57 -0400 Received: from zrc2hxm2.corp.nortel.com (zrc2hxm2.corp.nortel.com [47.103.123.73]) by zcars04e.nortel.com (Switch-2.2.0/Switch-2.2.0) with ESMTP id k73C60e08093; Thu, 3 Aug 2006 08:06:01 -0400 (EDT) X-MimeOLE: Produced By Microsoft Exchange V6.5.7226.0 Content-class: urn:content-classes:message MIME-Version: 1.0 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: quoted-printable Subject: RE: [Sipping] Some thoughts on fixing early media Date: Thu, 3 Aug 2006 07:11:49 -0500 Message-ID: <6FC4416DDE56C44DA0AEE67BC7CA43710FBDA1B2@zrc2hxm2.corp.nortel.com> In-Reply-To: <532B18E13CF9E64380EF5FDDE265E071384CC8@blr-m2-msg.wipro.com> X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: [Sipping] Some thoughts on fixing early media thread-index: Acav6si/KX4K3NwnQFGu7NssoL4r7gAFBZXwAAT8rwABjPJ3AAAaSNXwAAFv6hAAEBABkA== From: "Brian Stucker" To: , , X-Spam-Score: 0.0 (/) X-Scan-Signature: 9f79b8e383fd3af2b1b5b1d0910f6094 Cc: dean.willis@softarmor.com, sipping@ietf.org, christer.holmberg@ericsson.com X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Ah, sorry... misunderstood you. Yes, I think existing mechanisms (ie. preconditions) should be acceptable to control any alerting behavior.=20 Regards, Brian > -----Original Message----- > From: ravishankar.shiroor@wipro.com=20 > [mailto:ravishankar.shiroor@wipro.com]=20 > Sent: Thursday, August 03, 2006 12:35 AM > To: Stucker, Brian (RICH1:B620); sayan.chowdhury@wipro.com;=20 > pkyzivat@cisco.com > Cc: christer.holmberg@ericsson.com; sipping@ietf.org;=20 > dean.willis@softarmor.com > Subject: RE: [Sipping] Some thoughts on fixing early media >=20 > I am assuming that we will go with the option that the early=20 > medias will be played sequentially as per your suggestion. > The thing I am suggesting is that the alerting-the-user need=20 > not be controlled by the UAC, but it is can be controlled by=20 > UAS, as it has enough information to make that decision. > According to the q value set, either the colorful ringtone or=20 > "please hangup.." will be played first. > Whichever UAS finishes playing the early media first=20 > (according to the q value set), will start alerting the user. >=20 > -- > =20 > Ravishankar. G. Shiroor > Wipro Technologies, Bangalore. > =20 > ravishankar.shiroor@wipro.com > -- > =20 >=20 > > -----Original Message----- > > From: Stucker, Brian (RICH1:B620) [mailto:BSTUCKER@nortel.com] > > Sent: Thursday, August 03, 2006 9:21 AM > > To: RaviShankar Shiroor (WT01 - Product Engineering=20 > Solutions); Sayan=20 > > Chowdhury (WT01 - IP-Multimedia Carrier & Ent Networks);=20 > > pkyzivat@cisco.com > > Cc: christer.holmberg@ericsson.com; sipping@ietf.org;=20 > > dean.willis@softarmor.com > > Subject: RE: [Sipping] Some thoughts on fixing early media > >=20 > > Ok, so... > >=20 > > I have a call that forks. The first guy starts sending me colorful=20 > > ringback, the second guy sends me "please hang up or I'm=20 > about to bill=20 > > you a huge amount in 10 seconds". According to common practice with=20 > > UAC design, it plays neither early media and instead=20 > generates local=20 > > ringback. > >=20 > > You never hear the warning, the colorful ringback service is broken=20 > > and the originator gets billed a huge amount, but the terminator=20 > > didn't alert until after the warning was played. :) > >=20 > > If the UAC has no control, then the UAS has no guarantee that what=20 > > they're wanting to send will get presented to the originator. > >=20 > > Regards, > > Brian > >=20 > > > -----Original Message----- > > > From: ravishankar.shiroor@wipro.com > > > [mailto:ravishankar.shiroor@wipro.com] > > > Sent: Wednesday, August 02, 2006 11:26 PM > > > To: sayan.chowdhury@wipro.com; Stucker, Brian (RICH1:B620);=20 > > > pkyzivat@cisco.com > > > Cc: christer.holmberg@ericsson.com; sipping@ietf.org;=20 > > > dean.willis@softarmor.com > > > Subject: RE: [Sipping] Some thoughts on fixing early media > > >=20 > > > Hi Sayan, > > >=20 > > > Would it not be the UAS's responsibility to decide that the user=20 > > > should be alerted only after the Early Media is played? > > > I think that the UAS which has early media to play, can decide to=20 > > > postpone the alerting till it has been sent to the UAC. UAC > > need not > > > have to control this - I think this should satisfy all the > > use cases > > > needed. (for example "please hang-up in ten seconds from=20 > now unless=20 > > > you wish to be billed a huge amount" or something like that > > should be > > > played before UAS alerts the user.) > > >=20 > > > Regards, > > > Ravi. > > >=20 > > >=20 > > > -- > > > =20 > > > Ravishankar. G. Shiroor > > > Wipro Technologies, Bangalore. > > > =20 > > > ravishankar.shiroor@wipro.com > > > -- > > > =20 > > >=20 > > > > -----Original Message----- > > > > From: sayan.chowdhury@wipro.com=20 > [mailto:sayan.chowdhury@wipro.com] > > > > Sent: Tuesday, July 25, 2006 11:21 PM > > > > To: bstucker@nortel.com; pkyzivat@cisco.com > > > > Cc: christer.holmberg@ericsson.com; sipping@ietf.org;=20 > > > > dean.willis@softarmor.com > > > > Subject: RE: [Sipping] Some thoughts on fixing early media > > > >=20 > > > > Hello Brian , > > > > I am trying to put a call flow here , (I will not try to > > draw as my > > > > mail client is messing it up) > > > >=20 > > > > 1>Callee sends an offer with emflow=3Dnone > > > > 2>gets back say two 183 responses (establishing early dialog) > > > > both with > > > > early media class as two-way. > > > > 3>it selects the 183 with the higher q-value and more > > > > critical em class > > > > and adds new offer in the PRACK with emflow as sendrecv. > > > > 4>PRACKs the other 183 as well but with emflow=3Dnone and gets > > > > back 200 OK > > > > for that. > > > > 4>Gets back the answer in the 200 OK to PRACK for the 183 > > > with higher > > > > q-value. > > > > 5>Does early media.=20 > > > > 6>Sends an UPDATE to the next 183 with lower q-value with > > emflow as > > > > sendrcv > > > > ... continues > > > >=20 > > > > If my understanding of the flow is correct , then in the above=20 > > > > scenario how does the UAC indicate to the UAS(s) not to > > > alert the user > > > > until early media is over ? > > > >=20 > > > > Do we need a confirm-status mechanism as well wherein the > > > UAC can tell > > > > when to alert the UAS ? > > > > There may be use cases when the UAC wants to defer the > > > alerting until > > > > the UAC has heard all the early media "flows". > > > > For eg , when the UAC chooses which UAS it wants to have > > a regular > > > > media session with after hearing to all the early media=20 > sessions. > > > >=20 > > > > Regards , > > > > Sayan > > > >=20 > > > > -----Original Message----- > > > > From: Brian Stucker [mailto:bstucker@nortel.com] > > > > Sent: Tuesday, July 25, 2006 9:02 PM > > > > To: Paul Kyzivat > > > > Cc: Dean Willis; sipping; Christer Holmberg (JO/LMF) > > > > Subject: RE: [Sipping] Some thoughts on fixing early media > > > >=20 > > > > Yes, this is pretty much what's going to happen. I'll add > > > that to the > > > > text, actually, since I think it deserves being called out. > > > >=20 > > > > In cases where network interworking is not present (ie. no > > > SIP-to-PSTN > > > > type gateways are present in the routing tree) things will > > > basically > > > > behave as they always have been because most SIP=20 > endpoints do not=20 > > > > generate early media. It's services like CRBT and PSTN > > > gateways that > > > > get us into trouble. For the other endpoints we simply > > > don't wish to > > > > clip the media if possible. For the ones that do generate > > > early media, > > > > we can fork to them, but don't necessarily want to hear > > > what they all > > > > have to say in as a chorus. > > > >=20 > > > > Regards, > > > > Brian > > > >=20 > > > > > -----Original Message----- > > > > > From: Paul Kyzivat [mailto:pkyzivat@cisco.com] > > > > > Sent: Tuesday, July 25, 2006 8:03 AM > > > > > To: Stucker, Brian (RICH1:B620) > > > > > Cc: Christer Holmberg (JO/LMF); Dean Willis; Stephen > > > Sprunk; sipping > > > > > Subject: Re: [Sipping] Some thoughts on fixing early media > > > > >=20 > > > > > If the UAC only gives permission to one tine of the > > fork to send > > > > > media, and preconditions are used to prevent alerting until > > > > permission > > > >=20 > > > > > to send is granted, then parallel forking has been turned > > > > into serial > > > > > forking under the control of the UAC. > > > > >=20 > > > > > While it seems a little odd, it is perhaps the best that > > > > can be done > > > > > under the circumstances. It works well in that those > > > > destinations that > > > >=20 > > > > > don't require early media can all alert in parallel, and > > > > only the ones > > > >=20 > > > > > requiring early media are serialized. That is a huge > > > > improvement over > > > > > the proxy making the choice, because it doesn't know which > > > > ones will > > > > > require early media. > > > > >=20 > > > > > Paul > > > > >=20 > > > > > Brian Stucker wrote: > > > > > > Yes, I think you understand what I am suggesting. > > > > > >=20 > > > > > > Regards, > > > > > > Brian > > > > > >=20 > > > > > > -----Original Message----- > > > > > > From: Christer Holmberg (JO/LMF)=20 > > > > > > [mailto:christer.holmberg@ericsson.com] > > > > > >=20 > > > > > > Sent: Sunday, July 23, 2006 5:20 PM > > > > > > To: Stucker, Brian (RICH1:B620); Dean Willis; Stephen Sprunk > > > > > > Cc: sipping > > > > > > Subject: VS: [Sipping] Some thoughts on fixing early media > > > > > >=20 > > > > > > Hi, > > > > > >=20 > > > > > >> I'm proposing that the far end would not alert until > > > > permission to > > > > > >> send > > > > > >=20 > > > > > >> media had been granted. > > > > > >=20 > > > > > > That's called pre-conditions, isn't it? :) > > > > > > =20 > > > > > > Or, you could also send an INVITE WITH SDP, but the UAS > > > would not > > > > > > altert the user until some other type of indicator has been > > > > > sent to it. > > > > > > =20 > > > > > > Regards, > > > > > > =20 > > > > > > Christer > > > > > > =20 > > > > > >=20 > > > > > >=20 > > > > > > -----Original Message----- > > > > > > From: Christer Holmberg > > [mailto:christer.holmberg@ericsson.com] > > > > > > Sent: Sunday, July 23, 2006 12:14 AM > > > > > > To: Stucker, Brian (RICH1:B620); Dean Willis; Stephen Sprunk > > > > > > Cc: sipping > > > > > > Subject: RE: [Sipping] Some thoughts on fixing early media > > > > > >=20 > > > > > > Hi Brian, > > > > > >=20 > > > > > > What if an UAS wants to send 200 OK (and media) before > > > > the UAC has > > > > > > told him it's ok to send media? Someone would have to send > > > > > an UPDATE, > > > > > > to switch the media to sendrecv. > > > > > >=20 > > > > > > Or, are you proposing that the UAS, until it's given > > > > permission to > > > > > > send media, wouldn't even alert the user behind it? > > > > > >=20 > > > > > > Regards, > > > > > >=20 > > > > > > Christer > > > > > >=20 > > > > > >=20 > > > > > >=20 > > > > > >=20 > > > > > > _______________________________________________ > > > > > > Sipping mailing list > > > > https://www1.ietf.org/mailman/listinfo/sipping > > > > > > This list is for NEW development of the application > > of SIP Use > > > > > > sip-implementors@cs.columbia.edu for questions on > > > current sip Use > > > > > > sip@ietf.org for new developments of core SIP > > > > > >=20 > > > > >=20 > > > >=20 > > > > _______________________________________________ > > > > Sipping mailing list > > https://www1.ietf.org/mailman/listinfo/sipping > > > > This list is for NEW development of the application of SIP Use=20 > > > > sip-implementors@cs.columbia.edu for questions on=20 > current sip Use=20 > > > > sip@ietf.org for new developments of core SIP > > > >=20 > > > > _______________________________________________ > > > > Sipping mailing list > > https://www1.ietf.org/mailman/listinfo/sipping > > > > This list is for NEW development of the application of SIP Use=20 > > > > sip-implementors@cs.columbia.edu for questions on=20 > current sip Use=20 > > > > sip@ietf.org for new developments of core SIP > > > >=20 > > >=20 > >=20 >=20 _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Thu Aug 03 08:18:45 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8c9n-0008G0-Kn; Thu, 03 Aug 2006 08:18:31 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8c9l-0008CG-Ns for sipping@ietf.org; Thu, 03 Aug 2006 08:18:29 -0400 Received: from zrtps0kn.nortel.com ([47.140.192.55]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G8c9k-0007WQ-6Z for sipping@ietf.org; Thu, 03 Aug 2006 08:18:29 -0400 Received: from zrc2hxm2.corp.nortel.com (zrc2hxm2.corp.nortel.com [47.103.123.73]) by zrtps0kn.nortel.com (Switch-2.2.6/Switch-2.2.0) with ESMTP id k73CIOh00902; Thu, 3 Aug 2006 08:18:24 -0400 (EDT) X-MimeOLE: Produced By Microsoft Exchange V6.5.7226.0 Content-class: urn:content-classes:message MIME-Version: 1.0 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: quoted-printable Subject: RE: [Sipping] Some thoughts on fixing early media Date: Thu, 3 Aug 2006 07:18:22 -0500 Message-ID: <6FC4416DDE56C44DA0AEE67BC7CA43710FBDA1BC@zrc2hxm2.corp.nortel.com> In-Reply-To: <5EB80D22825EEE42872083AD5BFFB594017F12A3@esealmw113.eemea.ericsson.se> X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: [Sipping] Some thoughts on fixing early media thread-index: Acav6si/KX4K3NwnQFGu7NssoL4r7gAFBZXwAAT8rwABjPJ3AAAaSNXwAAPgO6AADbB7sA== From: "Brian Stucker" To: "Christer Holmberg \(JO/LMF\)" , , , X-Spam-Score: 0.0 (/) X-Scan-Signature: 16a2b98d831858659c646b3dec9ed22b Cc: sipping@ietf.org, dean.willis@softarmor.com X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Yes, I misunderstood his comment to mean that existing mechanisms would work for early media. I agree, unless someone can show me a use case where the UAC must control the alerting of the UAS in the context of early media, I'll include it. Otherwise I'll update the draft in the next revision to spell out the interaction and specify that it's up to the UAS to decide how long to queue for. Regards, Brian=20 > -----Original Message----- > From: Christer Holmberg (JO/LMF)=20 > [mailto:christer.holmberg@ericsson.com]=20 > Sent: Thursday, August 03, 2006 1:46 AM > To: Stucker, Brian (RICH1:B620);=20 > ravishankar.shiroor@wipro.com; sayan.chowdhury@wipro.com;=20 > pkyzivat@cisco.com > Cc: sipping@ietf.org; dean.willis@softarmor.com > Subject: RE: [Sipping] Some thoughts on fixing early media >=20 >=20 > Hi Brian, >=20 > I think there are two issues. >=20 > One is that the UAC gives permission to the UAS to play early media. > But, once that permission is given I am not sure the UAC=20 > needs to give the UAS permission to alert the user. I think,=20 > as Ravinhankar said, it's an UAS implementation issue when it=20 > alterts the user. >=20 > Also, even if the UAS is NOT given permisson to play early=20 > media (or, the UAC indicates it won't listen to it), I think=20 > it's an UAS implementation issue whether it "waits" for=20 > permission, or alterts the UAS anyway, depending on the=20 > importance of the early media that the UAS would have wanted to play.=20 >=20 > So, if a UAS would like to send a "please hang up or I'll=20 > bill you a huge amount in 10 seconds" early media message,=20 > but is not given permission to do so, I guess it should not=20 > wait for permission more than 10 seconds, but instead send an=20 > error response. >=20 > Regards, >=20 > Christer >=20 >=20 > -----Original Message----- > From: Brian Stucker [mailto:bstucker@nortel.com] > Sent: 3. elokuuta 2006 6:51 > To: ravishankar.shiroor@wipro.com; sayan.chowdhury@wipro.com;=20 > pkyzivat@cisco.com > Cc: Christer Holmberg (JO/LMF); sipping@ietf.org;=20 > dean.willis@softarmor.com > Subject: RE: [Sipping] Some thoughts on fixing early media >=20 > Ok, so... >=20 > I have a call that forks. The first guy starts sending me=20 > colorful ringback, the second guy sends me "please hang up or=20 > I'm about to bill you a huge amount in 10 seconds". According=20 > to common practice with UAC design, it plays neither early=20 > media and instead generates local ringback. >=20 > You never hear the warning, the colorful ringback service is=20 > broken and the originator gets billed a huge amount, but the=20 > terminator didn't alert until after the warning was played. :) >=20 > If the UAC has no control, then the UAS has no guarantee that=20 > what they're wanting to send will get presented to the originator. >=20 > Regards, > Brian=20 >=20 > > -----Original Message----- > > From: ravishankar.shiroor@wipro.com > > [mailto:ravishankar.shiroor@wipro.com] > > Sent: Wednesday, August 02, 2006 11:26 PM > > To: sayan.chowdhury@wipro.com; Stucker, Brian (RICH1:B620);=20 > > pkyzivat@cisco.com > > Cc: christer.holmberg@ericsson.com; sipping@ietf.org;=20 > > dean.willis@softarmor.com > > Subject: RE: [Sipping] Some thoughts on fixing early media > >=20 > > Hi Sayan, > >=20 > > Would it not be the UAS's responsibility to decide that the user=20 > > should be alerted only after the Early Media is played? > > I think that the UAS which has early media to play, can decide to=20 > > postpone the alerting till it has been sent to the UAC. UAC=20 > need not=20 > > have to control this - I think this should satisfy all the=20 > use cases=20 > > needed. (for example "please hang-up in ten seconds from now unless=20 > > you wish to be billed a huge amount" or something like that=20 > should be=20 > > played before UAS alerts the user.) > >=20 > > Regards, > > Ravi. > >=20 > >=20 > > -- > > =20 > > Ravishankar. G. Shiroor > > Wipro Technologies, Bangalore. > > =20 > > ravishankar.shiroor@wipro.com > > -- > > =20 > >=20 > > > -----Original Message----- > > > From: sayan.chowdhury@wipro.com [mailto:sayan.chowdhury@wipro.com] > > > Sent: Tuesday, July 25, 2006 11:21 PM > > > To: bstucker@nortel.com; pkyzivat@cisco.com > > > Cc: christer.holmberg@ericsson.com; sipping@ietf.org;=20 > > > dean.willis@softarmor.com > > > Subject: RE: [Sipping] Some thoughts on fixing early media > > >=20 > > > Hello Brian , > > > I am trying to put a call flow here , (I will not try to=20 > draw as my=20 > > > mail client is messing it up) > > >=20 > > > 1>Callee sends an offer with emflow=3Dnone > > > 2>gets back say two 183 responses (establishing early dialog) > > > both with > > > early media class as two-way. > > > 3>it selects the 183 with the higher q-value and more > > > critical em class > > > and adds new offer in the PRACK with emflow as sendrecv. > > > 4>PRACKs the other 183 as well but with emflow=3Dnone and gets > > > back 200 OK > > > for that. > > > 4>Gets back the answer in the 200 OK to PRACK for the 183 > > with higher > > > q-value. > > > 5>Does early media.=20 > > > 6>Sends an UPDATE to the next 183 with lower q-value with=20 > emflow as > > > sendrcv > > > ... continues > > >=20 > > > If my understanding of the flow is correct , then in the above=20 > > > scenario how does the UAC indicate to the UAS(s) not to > > alert the user > > > until early media is over ? > > >=20 > > > Do we need a confirm-status mechanism as well wherein the > > UAC can tell > > > when to alert the UAS ? > > > There may be use cases when the UAC wants to defer the > > alerting until > > > the UAC has heard all the early media "flows". > > > For eg , when the UAC chooses which UAS it wants to have=20 > a regular=20 > > > media session with after hearing to all the early media sessions. > > >=20 > > > Regards , > > > Sayan > > >=20 > > > -----Original Message----- > > > From: Brian Stucker [mailto:bstucker@nortel.com] > > > Sent: Tuesday, July 25, 2006 9:02 PM > > > To: Paul Kyzivat > > > Cc: Dean Willis; sipping; Christer Holmberg (JO/LMF) > > > Subject: RE: [Sipping] Some thoughts on fixing early media > > >=20 > > > Yes, this is pretty much what's going to happen. I'll add > > that to the > > > text, actually, since I think it deserves being called out. > > >=20 > > > In cases where network interworking is not present (ie. no > > SIP-to-PSTN > > > type gateways are present in the routing tree) things will > > basically > > > behave as they always have been because most SIP endpoints do not=20 > > > generate early media. It's services like CRBT and PSTN > > gateways that > > > get us into trouble. For the other endpoints we simply > > don't wish to > > > clip the media if possible. For the ones that do generate > > early media, > > > we can fork to them, but don't necessarily want to hear > > what they all > > > have to say in as a chorus. > > >=20 > > > Regards, > > > Brian > > >=20 > > > > -----Original Message----- > > > > From: Paul Kyzivat [mailto:pkyzivat@cisco.com] > > > > Sent: Tuesday, July 25, 2006 8:03 AM > > > > To: Stucker, Brian (RICH1:B620) > > > > Cc: Christer Holmberg (JO/LMF); Dean Willis; Stephen > > Sprunk; sipping > > > > Subject: Re: [Sipping] Some thoughts on fixing early media > > > >=20 > > > > If the UAC only gives permission to one tine of the=20 > fork to send=20 > > > > media, and preconditions are used to prevent alerting until > > > permission > > >=20 > > > > to send is granted, then parallel forking has been turned > > > into serial > > > > forking under the control of the UAC. > > > >=20 > > > > While it seems a little odd, it is perhaps the best that > > > can be done > > > > under the circumstances. It works well in that those > > > destinations that > > >=20 > > > > don't require early media can all alert in parallel, and > > > only the ones > > >=20 > > > > requiring early media are serialized. That is a huge > > > improvement over > > > > the proxy making the choice, because it doesn't know which > > > ones will > > > > require early media. > > > >=20 > > > > Paul > > > >=20 > > > > Brian Stucker wrote: > > > > > Yes, I think you understand what I am suggesting. > > > > >=20 > > > > > Regards, > > > > > Brian > > > > >=20 > > > > > -----Original Message----- > > > > > From: Christer Holmberg (JO/LMF)=20 > > > > > [mailto:christer.holmberg@ericsson.com] > > > > >=20 > > > > > Sent: Sunday, July 23, 2006 5:20 PM > > > > > To: Stucker, Brian (RICH1:B620); Dean Willis; Stephen Sprunk > > > > > Cc: sipping > > > > > Subject: VS: [Sipping] Some thoughts on fixing early media > > > > >=20 > > > > > Hi, > > > > >=20 > > > > >> I'm proposing that the far end would not alert until > > > permission to > > > > >> send > > > > >=20 > > > > >> media had been granted. > > > > >=20 > > > > > That's called pre-conditions, isn't it? :) > > > > > =20 > > > > > Or, you could also send an INVITE WITH SDP, but the UAS > > would not > > > > > altert the user until some other type of indicator has been > > > > sent to it. > > > > > =20 > > > > > Regards, > > > > > =20 > > > > > Christer > > > > > =20 > > > > >=20 > > > > >=20 > > > > > -----Original Message----- > > > > > From: Christer Holmberg=20 > [mailto:christer.holmberg@ericsson.com] > > > > > Sent: Sunday, July 23, 2006 12:14 AM > > > > > To: Stucker, Brian (RICH1:B620); Dean Willis; Stephen Sprunk > > > > > Cc: sipping > > > > > Subject: RE: [Sipping] Some thoughts on fixing early media > > > > >=20 > > > > > Hi Brian, > > > > >=20 > > > > > What if an UAS wants to send 200 OK (and media) before > > > the UAC has > > > > > told him it's ok to send media? Someone would have to send > > > > an UPDATE, > > > > > to switch the media to sendrecv. > > > > >=20 > > > > > Or, are you proposing that the UAS, until it's given > > > permission to > > > > > send media, wouldn't even alert the user behind it? > > > > >=20 > > > > > Regards, > > > > >=20 > > > > > Christer > > > > >=20 > > > > >=20 > > > > >=20 > > > > >=20 > > > > > _______________________________________________ > > > > > Sipping mailing list > > > https://www1.ietf.org/mailman/listinfo/sipping > > > > > This list is for NEW development of the application=20 > of SIP Use=20 > > > > > sip-implementors@cs.columbia.edu for questions on > > current sip Use > > > > > sip@ietf.org for new developments of core SIP > > > > >=20 > > > >=20 > > >=20 > > > _______________________________________________ > > > Sipping mailing list =20 > https://www1.ietf.org/mailman/listinfo/sipping > > > This list is for NEW development of the application of SIP Use=20 > > > sip-implementors@cs.columbia.edu for questions on current sip Use=20 > > > sip@ietf.org for new developments of core SIP > > >=20 > > > _______________________________________________ > > > Sipping mailing list =20 > https://www1.ietf.org/mailman/listinfo/sipping > > > This list is for NEW development of the application of SIP Use=20 > > > sip-implementors@cs.columbia.edu for questions on current sip Use=20 > > > sip@ietf.org for new developments of core SIP > > >=20 > >=20 >=20 _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Thu Aug 03 08:36:45 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8cRP-0007K8-QJ; Thu, 03 Aug 2006 08:36:43 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8cRN-0007FH-NI for sipping@ietf.org; Thu, 03 Aug 2006 08:36:41 -0400 Received: from [65.220.123.2] (helo=mail.pingtel.com) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G8cRI-0000eZ-EB for sipping@ietf.org; Thu, 03 Aug 2006 08:36:41 -0400 Received: from localhost.localdomain (pi.pingtel.com [10.1.1.12]) by mail.pingtel.com (Postfix) with ESMTP id 4913B6C01D; Thu, 3 Aug 2006 07:36:06 -0400 (EDT) Subject: RE: [Sipping] Question on do not disturb indication From: Scott Lawrence To: Hadriel Kaplan In-Reply-To: <03bb01c6b6cf$b0757110$6601a8c0@acmepacket.com> References: <03bb01c6b6cf$b0757110$6601a8c0@acmepacket.com> Content-Type: text/plain Organization: Pingtel Corp. http://www.pingtel.com/ Date: Thu, 03 Aug 2006 08:35:43 -0400 Message-Id: <1154608543.2859.30.camel@sukothai.pingtel.com> Mime-Version: 1.0 X-Mailer: Evolution 2.6.2 (2.6.2-1.fc5.5) Content-Transfer-Encoding: 7bit X-Spam-Score: 0.1 (/) X-Scan-Signature: 4adaf050708fb13be3316a9eee889caa Cc: sipping@ietf.org X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org On Thu, 2006-08-03 at 03:37 -0400, Hadriel Kaplan wrote: > > > -----Original Message----- > > From: Scott Lawrence [mailto:slawrence@pingtel.com] > > > [about a 603 response...] > > I read that to mean that a forking proxy should abandon all forks for > > the call and terminate the request - the response asserts that the > > voicemail (another end point) will not answer. > > > > That might be an appropriate response for a proxy to send if it is > > authoritative for the domain and knows that a particular user does not > > exist, but I can't see how a user agent can ever have enough global > > knowledge to justify a 6xx response. > > That would be a 604 (Does not exist anywhere), no? > A 603 could be useful someday for personal black-lists; for example where > the user agent may decide who to give 603's to based on the From, and not > send them to voicemail. > > The problem is just because I pressed DND doesn't mean I don't want it to go > to voicemail - I do want it to go to voicemail. So 6xx is wrong. But > sending a 480 typically only makes it go to voicemail if there are no other > contacts to try, or they all send 480. So I have to press DND on all my > phones; but I'm not sure that's avoidable. Yes - I call that the 'ring twice problem'; if I ignore or reject a call on my desk phone, then my personal forwarding rules send it to my cell and I have to ignore or reject it there before it will go to my voicemail. I control whether or not I have that problem by deciding whether or not to have that forwarding in place. The only solution I can think of for avoiding ring-twice would be a response that indicates that all forks that in the class 'my personal phones' should be abandoned, but that phones in other classes, such as 'my assistant', or 'my voicemail' may proceed. Perhaps such a thing could be built with the caller prefs mechanisms, but I'm not optimistic that it would see wide implementation (I probably would not try to implement it in our proxy, for example). -- Scott Lawrence tel:+1-781-938-5306;ext=162 or sip:slawrence@pingtel.com sipXpbx project coordinator - SIPfoundry http://www.sipfoundry.org/sipX Chief Architect - Pingtel Corp. http://www.pingtel.com/ _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Thu Aug 03 08:37:15 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8cRv-0007iT-NT; Thu, 03 Aug 2006 08:37:15 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8cRu-0007iO-7R for sipping@ietf.org; Thu, 03 Aug 2006 08:37:14 -0400 Received: from zcars04f.nortel.com ([47.129.242.57]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G8cRs-0000jY-PE for sipping@ietf.org; Thu, 03 Aug 2006 08:37:14 -0400 Received: from zrc2hxm2.corp.nortel.com (zrc2hxm2.corp.nortel.com [47.103.123.73]) by zcars04f.nortel.com (Switch-2.2.6/Switch-2.2.0) with ESMTP id k73Cb9926797; Thu, 3 Aug 2006 08:37:09 -0400 (EDT) X-MimeOLE: Produced By Microsoft Exchange V6.5.7226.0 Content-class: urn:content-classes:message MIME-Version: 1.0 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: quoted-printable Subject: RE: [Sipping] Question on do not disturb indication Date: Thu, 3 Aug 2006 07:36:36 -0500 Message-ID: <6FC4416DDE56C44DA0AEE67BC7CA43710FBDA1D8@zrc2hxm2.corp.nortel.com> In-Reply-To: X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: [Sipping] Question on do not disturb indication thread-index: Aca27pxEB1DxeyrlQ42aB5K1T5SMoQACfWNA From: "Brian Stucker" To: "Xiaotao Wu" , "Francois Audet" X-Spam-Score: 0.0 (/) X-Scan-Signature: c54bc2f42d02429833c0ca4b8725abd7 Cc: "Elwell, John" , Paul Kyzivat , sipping@ietf.org, Scott Lawrence X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org I agree with Francois. A 603 isn't a horrible thing. Essentially with SIP you have two choices here, kill one forked leg of the call (4xx response) or kill all of the forked legs (603). What the proxy does with it from there for alternate routing is up to the proxy. I don't know that the example Xiaotao gives is a reasonable use case. If the user does not wish for their desk and cell phone to ring, then you 486 those legs individually. The only other mechanism 3261 gives you to do that is with a 603 response. To be any more refined than that would require extensions so that endpoints can coordinate their actions more closely. If we go with the notion that 603 is bad, then if Xiaotao's user's cellphone is out in their car, and they don't want the call to go to voicemail on the cellphone but go to the office secretary/voicemail , all they can do is send a 603 from their desk phone. Regards, Brian > -----Original Message----- > From: Xiaotao Wu [mailto:xwu@avaya.com]=20 > Sent: Thursday, August 03, 2006 7:18 AM > To: Audet, Francois (SC100:3055) > Cc: sipping@ietf.org; Paul Kyzivat; Elwell, John; Scott Lawrence > Subject: Re: [Sipping] Question on do not disturb indication >=20 > On 8/2/06, Francois Audet wrote: > > On the 603 issue, I'd like to disagree with everybody else... > > Sometimes being able to decline the call everywhere is=20 > actually what=20 > > you want. > > As an example, I have multiple forked device. If I send a 603, it=20 > > triggers the proxy to use whatever alternate routing is sees fit=20 > > (e.g., voicemail, attendant, etc.). That's the desired behavior. > > > > Problem with a 480/486 with that scenario is that it won't=20 > trigger the=20 > > alternate routing and will keep on ringing the other fork. That may=20 > > not be the desired behavior. >=20 > I would expect the response should be sent by the proxy,=20 > instead by endpoints. If it's send by an endpoint, things may=20 > get complicated. > For example, Bob has a desk phone, a cell phone, a voicemail,=20 > and a secretary. It's hard to compose a response from his=20 > desk phone to indicate that he does not want to ring his desk=20 > phone, and cell phone, but ring his secretary phone and voicemail. >=20 > So, I think the problem is more on what's the expected code=20 > for the caller, not on the routing behavior of the request/response. >=20 > -Xiaotao >=20 > > > > My 2 cents. > > > > > -----Original Message----- > > > From: Paul Kyzivat [mailto:pkyzivat@cisco.com] > > > Sent: Wednesday, August 02, 2006 3:31 PM > > > To: Scott Lawrence > > > Cc: sipping@ietf.org; Elwell, John > > > Subject: Re: [Sipping] Question on do not disturb indication > > > > > > I pretty much agree with Scott. In many/most cases I=20 > won't want the=20 > > > caller to know why his call is going unanswered. > > > Both 480 and 486 are in effect "white lies". Which one to use may=20 > > > depend on which impression you want to give to the=20 > caller. Another=20 > > > one - if you want to be more rude - is 403 Forbidden. > > > > > > I don't see a need for a *special* code that is unique to this=20 > > > service. > > > > > > Paul > > > > > > Scott Lawrence wrote: > > > > On Wed, 2006-08-02 at 21:09 +0100, Elwell, John wrote: > > > >> RFC 3261 suggests the use of response code 480 Temporarily=20 > > > >> Unavailable for indicating do not disturb (DND), but it is > > > not used > > > >> exclusively for this purpose. There are other documents (e.g.,=20 > > > >> RFC > > > >> 4504) that suggest the use of 486 Busy Here, but again this is=20 > > > >> not helpful if the receiver wants to distinguish between DND > > > and ordinary > > > >> busy. Presence can be used to indicate this condition=20 > through the=20 > > > >> RPID note element, but that does not say why a=20 > particular session=20 > > > >> establishment attempt has failed. What are the best=20 > practices for=20 > > > >> indicating the DND condition? Does anyone see a need for > > > an explicit > > > >> indicator for this purpose, e.g., a new SIP response code? > > > > > > > > Personally, I like 480 for this. > > > > > > > > If the phone wanted to be sneaky, it could just mute the > > > ringing and > > > > allow the transaction to time out without explicitly=20 > rejecting it. > > > > That prevents the caller from being able to tell they are being=20 > > > > disregarded, but some users might prefer that forwarding > > > take effect > > > > sooner and not like that. > > > > > > > > The one thing that I _don't_ like to see is a 603 being > > > used for this; > > > > that messes up any upstream forking. > > > > > > > > > > > > > > _______________________________________________ > > > Sipping mailing list =20 > https://www1.ietf.org/mailman/listinfo/sipping > > > This list is for NEW development of the application of SIP Use=20 > > > sip-implementors@cs.columbia.edu for questions on current sip Use=20 > > > sip@ietf.org for new developments of core SIP > > > > > > > _______________________________________________ > > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > > This list is for NEW development of the application of SIP Use=20 > > sip-implementors@cs.columbia.edu for questions on current sip Use=20 > > sip@ietf.org for new developments of core SIP > > >=20 >=20 > -- > -Xiaotao >=20 > = =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D= =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D= =3D=3D=3D=3D=3D=3D=3D=3D=3D > Name : Xiaotao Wu > Email : xwu@avaya.com, xiaotaow@gmail.com > Web : http://www.cs.columbia.edu/~xiaotaow > Phone : (732) 852-2133 > SIP : sip:xiaotaow@cs.columbia.edu > Address : 1M-240, 307 Middletown Lincroft Road > Lincroft, NJ 07738-1526, U.S.A. >=20 > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP=20 > Use sip-implementors@cs.columbia.edu for questions on current=20 > sip Use sip@ietf.org for new developments of core SIP >=20 _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Thu Aug 03 08:40:21 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8cUt-0001Ak-Ce; Thu, 03 Aug 2006 08:40:19 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8cUr-0001Aa-Rk for sipping@ietf.org; Thu, 03 Aug 2006 08:40:17 -0400 Received: from [65.220.123.2] (helo=mail.pingtel.com) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G8cUm-0000pD-Ke for sipping@ietf.org; Thu, 03 Aug 2006 08:40:17 -0400 Received: from localhost.localdomain (pi.pingtel.com [10.1.1.12]) by mail.pingtel.com (Postfix) with ESMTP id C4C926C01D; Thu, 3 Aug 2006 07:39:41 -0400 (EDT) Subject: RE: [Sipping] Question on do not disturb indication From: Scott Lawrence To: Jeroen Van Bemmel In-Reply-To: <0J3F00I49B8SG700@jes-fe2.zx.nl> References: <0J3F00I49B8SG700@jes-fe2.zx.nl> Content-Type: text/plain Organization: Pingtel Corp. http://www.pingtel.com/ Date: Thu, 03 Aug 2006 08:39:19 -0400 Message-Id: <1154608759.2859.32.camel@sukothai.pingtel.com> Mime-Version: 1.0 X-Mailer: Evolution 2.6.2 (2.6.2-1.fc5.5) Content-Transfer-Encoding: 7bit X-Spam-Score: 0.1 (/) X-Scan-Signature: 856eb5f76e7a34990d1d457d8e8e5b7f Cc: sipping@ietf.org, Francois Audet , Xiaotao Wu , Paul Kyzivat , "Elwell, John" X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org On Thu, 2006-08-03 at 14:05 +0200, Jeroen Van Bemmel wrote: > So, I think the problem is more on what's the expected code for the > caller, not on the routing behavior of the request/response. > > (JvB) The reason phrase is (and was always) intended for this. "480 Do > not disturb (use IM, call my secretary or leave a voiemail)" would > seem appropriate. The semantics are hard to convey to a machine, > therefore ideally the choice for any follow ups should be made by the > caller IMHO. A 6xx would guarantee that outcome I don't understand which response you are arguing the UA should send. -- Scott Lawrence tel:+1-781-938-5306;ext=162 or sip:slawrence@pingtel.com sipXpbx project coordinator - SIPfoundry http://www.sipfoundry.org/sipX Chief Architect - Pingtel Corp. http://www.pingtel.com/ _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Thu Aug 03 08:42:51 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8cXL-0002sW-EP; Thu, 03 Aug 2006 08:42:51 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8cXK-0002oZ-Cu for sipping@ietf.org; Thu, 03 Aug 2006 08:42:50 -0400 Received: from stsc1260-eth-s1-s1p1-vip.va.neustar.com ([156.154.16.129] helo=chiedprmail1.ietf.org) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G8ayI-0008Ha-A0 for sipping@ietf.org; Thu, 03 Aug 2006 07:02:34 -0400 Received: from szxga01-in.huawei.com ([61.144.161.53]) by chiedprmail1.ietf.org with esmtp (Exim 4.43) id 1G8aJG-0001SN-3T for sipping@ietf.org; Thu, 03 Aug 2006 06:20:14 -0400 Received: from huawei.com (szxga01-in [172.24.2.3]) by szxga01-in.huawei.com (iPlanet Messaging Server 5.2 HotFix 1.25 (built Mar 3 2004)) with ESMTP id <0J3F00JD03G5PR@szxga01-in.huawei.com> for sipping@ietf.org; Thu, 03 Aug 2006 18:21:42 +0800 (CST) Received: from huawei.com ([172.24.1.18]) by szxga01-in.huawei.com (iPlanet Messaging Server 5.2 HotFix 1.25 (built Mar 3 2004)) with ESMTP id <0J3F008G13G5EY@szxga01-in.huawei.com> for sipping@ietf.org; Thu, 03 Aug 2006 18:21:41 +0800 (CST) Received: from x18916 ([10.76.176.230]) by szxml03-in.huawei.com (iPlanet Messaging Server 5.2 HotFix 1.25 (built Mar 3 2004)) with ESMTPA id <0J3F00E7N3GWXW@szxml03-in.huawei.com> for sipping@ietf.org; Thu, 03 Aug 2006 18:22:11 +0800 (CST) Date: Thu, 03 Aug 2006 18:18:57 +0800 From: Peili Xu Subject: RE: [Sipping] Question on do not disturb indication In-reply-to: <50B1CBA96870A34799A506B2313F266709897F65@ntht201e.siemenscomms.co.uk> To: "'Elwell, John'" , 'Scott Lawrence' Message-id: <001901c6b6e6$3b226140$e6b04c0a@china.huawei.com> MIME-version: 1.0 X-MIMEOLE: Produced By Microsoft MimeOLE V6.00.2800.1106 X-Mailer: Microsoft Office Outlook 11 Content-type: text/plain; charset=us-ascii Content-transfer-encoding: 7BIT Thread-index: Aca2xKvevQ1SLkKUQheK9KKx+7stnwAIO8Ow X-Spam-Score: -2.6 (--) X-Scan-Signature: 6cca30437e2d04f45110f2ff8dc1b1d5 Cc: sipping@ietf.org X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Hi John, As my understanding, in PSTN where DND has been used for a long time, there is no specific cause value for it. I do not see strong requirments in SIP network . Peili -----Original Message----- From: Elwell, John [mailto:john.elwell@siemens.com] Sent: Thursday, August 03, 2006 2:18 PM To: Scott Lawrence Cc: sipping@ietf.org Subject: RE: [Sipping] Question on do not disturb indication Scott, Thanks for your response. I too like 480, but some people are telling me that it has too many other uses, so if you want to convey a definite indication that you are in DND they felt something else might be needed. The chief cause of concern seems to be that RFC 3398 recommends mapping a couple of Q.850 causes to 480 (19 no answer from user and 20 subscriber absent), and although these both have the general meaning of user temporarily unavailable, neither has quite the semantics of DND. John > -----Original Message----- > From: Scott Lawrence [mailto:slawrence@pingtel.com] > Sent: 02 August 2006 22:15 > To: Elwell, John > Cc: sipping@ietf.org > Subject: Re: [Sipping] Question on do not disturb indication > > On Wed, 2006-08-02 at 21:09 +0100, Elwell, John wrote: > > RFC 3261 suggests the use of response code 480 Temporarily > Unavailable > > for indicating do not disturb (DND), but it is not used exclusively > > for this purpose. There are other documents (e.g., RFC 4504) that > > suggest the use of 486 Busy Here, but again this is not > helpful if the > > receiver wants to distinguish between DND and ordinary > busy. Presence > > can be used to indicate this condition through the RPID > note element, > > but that does not say why a particular session establishment attempt > > has failed. What are the best practices for indicating the DND > > condition? Does anyone see a need for an explicit indicator for this > > purpose, e.g., a new SIP response code? > > Personally, I like 480 for this. > > If the phone wanted to be sneaky, it could just mute the ringing and > allow the transaction to time out without explicitly rejecting it. > That prevents the caller from being able to tell they are being > disregarded, but some users might prefer that forwarding take effect > sooner and not like that. > > The one thing that I _don't_ like to see is a 603 being used for this; > that messes up any upstream forking. > > > -- > Scott Lawrence tel:+1-781-938-5306;ext=162 or > sip:slawrence@pingtel.com > sipXpbx project coordinator - SIPfoundry > http://www.sipfoundry.org/sipX > Chief Architect - Pingtel Corp. http://www.pingtel.com/ > > _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Thu Aug 03 09:16:20 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8d3j-0007K7-Kl; Thu, 03 Aug 2006 09:16:19 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8d3i-0007Ha-RO for sipping@ietf.org; Thu, 03 Aug 2006 09:16:18 -0400 Received: from [65.220.123.2] (helo=mail.pingtel.com) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G8d3d-0005du-JF for sipping@ietf.org; Thu, 03 Aug 2006 09:16:18 -0400 Received: from localhost.localdomain (pi.pingtel.com [10.1.1.12]) by mail.pingtel.com (Postfix) with ESMTP id 456046C01D; Thu, 3 Aug 2006 08:15:43 -0400 (EDT) Subject: RE: [Sipping] Question on do not disturb indication From: Scott Lawrence To: Brian Stucker In-Reply-To: <6FC4416DDE56C44DA0AEE67BC7CA43710FBDA1D8@zrc2hxm2.corp.nortel.com> References: <6FC4416DDE56C44DA0AEE67BC7CA43710FBDA1D8@zrc2hxm2.corp.nortel.com> Content-Type: text/plain Organization: Pingtel Corp. http://www.pingtel.com/ Date: Thu, 03 Aug 2006 09:15:20 -0400 Message-Id: <1154610920.2859.38.camel@sukothai.pingtel.com> Mime-Version: 1.0 X-Mailer: Evolution 2.6.2 (2.6.2-1.fc5.5) Content-Transfer-Encoding: 7bit X-Spam-Score: 0.1 (/) X-Scan-Signature: b19722fc8d3865b147c75ae2495625f2 Cc: sipping@ietf.org, Francois Audet , Xiaotao Wu , Paul Kyzivat , "Elwell, John" X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org On Thu, 2006-08-03 at 07:36 -0500, Brian Stucker wrote: > I agree with Francois. A 603 isn't a horrible thing. Essentially with > SIP you have two choices here, kill one forked leg of the call (4xx > response) or kill all of the forked legs (603). What the proxy does with > it from there for alternate routing is up to the proxy. I don't think that it is - I think that 603 means that the proxy should stop trying to route the call and reject it back to the caller. If that isn't what it means, then we need a bug against 3261 to fix it. > I don't know that the example Xiaotao gives is a reasonable use case. If > the user does not wish for their desk and cell phone to ring, then you > 486 those legs individually. The only other mechanism 3261 gives you to > do that is with a 603 response. To be any more refined than that would > require extensions so that endpoints can coordinate their actions more > closely. > > If we go with the notion that 603 is bad, then if Xiaotao's user's > cellphone is out in their car, and they don't want the call to go to > voicemail on the cellphone but go to the office secretary/voicemail , > all they can do is send a 603 from their desk phone. But voicemail is, from the point of view of the proxy, just another fork for the same user. That's my point to begin with - either 603 means 'kill all forks', or it doesn't, and if it does then it kills voicemail too. -- Scott Lawrence tel:+1-781-938-5306;ext=162 or sip:slawrence@pingtel.com sipXpbx project coordinator - SIPfoundry http://www.sipfoundry.org/sipX Chief Architect - Pingtel Corp. http://www.pingtel.com/ _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Thu Aug 03 10:00:13 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8dk8-0003oO-3k; Thu, 03 Aug 2006 10:00:08 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8dk6-0003oJ-Fx for sipping@ietf.org; Thu, 03 Aug 2006 10:00:06 -0400 Received: from rtp-iport-1.cisco.com ([64.102.122.148]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G8dk4-0008RV-2z for sipping@ietf.org; Thu, 03 Aug 2006 10:00:06 -0400 Received: from rtp-dkim-2.cisco.com ([64.102.121.159]) by rtp-iport-1.cisco.com with ESMTP; 03 Aug 2006 07:00:03 -0700 X-BrightmailFiltered: true X-Brightmail-Tracker: AAAAAA== X-IronPort-AV: i="4.07,208,1151910000"; d="scan'208"; a="34404205:sNHT26520012" Received: from rtp-core-2.cisco.com (rtp-core-2.cisco.com [64.102.124.13]) by rtp-dkim-2.cisco.com (8.12.11.20060308/8.12.11) with ESMTP id k73E020u006284; Thu, 3 Aug 2006 10:00:02 -0400 Received: from xbh-rtp-201.amer.cisco.com (xbh-rtp-201.cisco.com [64.102.31.12]) by rtp-core-2.cisco.com (8.12.10/8.12.6) with ESMTP id k73E02dU003490; Thu, 3 Aug 2006 10:00:02 -0400 (EDT) Received: from xfe-rtp-201.amer.cisco.com ([64.102.31.38]) by xbh-rtp-201.amer.cisco.com with Microsoft SMTPSVC(6.0.3790.1830); Thu, 3 Aug 2006 10:00:02 -0400 Received: from [161.44.79.104] ([161.44.79.104]) by xfe-rtp-201.amer.cisco.com with Microsoft SMTPSVC(6.0.3790.1830); Thu, 3 Aug 2006 10:00:02 -0400 Message-ID: <44D20161.9050506@cisco.com> Date: Thu, 03 Aug 2006 10:00:01 -0400 From: Paul Kyzivat User-Agent: Thunderbird 1.5.0.5 (Windows/20060719) MIME-Version: 1.0 To: Dale.Worley@comcast.net Subject: Re: [Sipping] Question on do not disturb indication References: <50B1CBA96870A34799A506B2313F266709897F3A@ntht201e.siemenscomms.co.uk> <1154553270.2866.86.camel@sukothai.pingtel.com> <200608022251.k72Mpvkp020060@dragon.ariadne.com> In-Reply-To: <200608022251.k72Mpvkp020060@dragon.ariadne.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Content-Transfer-Encoding: 7bit X-OriginalArrivalTime: 03 Aug 2006 14:00:02.0596 (UTC) FILETIME=[1D998E40:01C6B705] DKIM-Signature: a=rsa-sha1; q=dns; l=3671; t=1154613602; x=1155477602; c=relaxed/simple; s=rtpdkim2001; h=Content-Type:From:Subject:Content-Transfer-Encoding:MIME-Version; d=cisco.com; i=pkyzivat@cisco.com; z=From:Paul=20Kyzivat=20 |Subject:Re=3A=20[Sipping]=20Question=20on=20do=20not=20disturb=20indication |To:Dale.Worley@comcast.net; X=v=3Dcisco.com=3B=20h=3Dyu9ohsv1kFqX2uC2CMjguD3sTnY=3D; b=KCoqz8VxnXcK1YB3TbS04xKsjAn1a2FJXfkw7W6wbCWDymWixcoC/dSbZ8B3rvg9epxUFuo6 EwvtO8ZWHd4ZKVio1Aix/Hy5CqqtndRgV71DbPA2yFPr/qdv8Y8mCRR9; Authentication-Results: rtp-dkim-2.cisco.com; header.From=pkyzivat@cisco.com; dkim=pass ( sig from cisco.com verified; ); X-Spam-Score: 0.0 (/) X-Scan-Signature: c3a18ef96977fc9bcc21a621cbf1174b Cc: sipping@ietf.org X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org I went at reviewed 3261 on this and found some interesting things: - Global Failures 6xx 6xx responses indicate that a server has definitive information about a particular user, not just the particular instance indicated in the Request-URI. (This refers to a "particular user", but how that relates to a call in progress isn't clear. Do two extensions on one AOR both belong to the same "user", or not? Should the phone and a VM server backing up the phone be viewed as the same user? If a proxy is configured to offer calls first to Alice's phone, and if no answer then to Alice's assistant Bob, does a 6xx response by Alice's phone say anything about Bob?) - 603 Decline: This status response is returned only if the client knows that no other end point will answer the request. (This is in conflict with the language in 6xx because it talks about end point rather than user. It doesn't appear that the difference is because of anything special about 603 vs the general 6xx.) - 604 Does Not Exist Anywhere: The server has authoritative information that the user indicated in the *** Request-URI *** does not exist anywhere. (This again differs from the 6xx description in a way that doesn't seem related to the specifics of 604 vs 6xx. This one does talk about *user*, and is more specific about what it means by user. But not in a very helpful way. It explicitly calls out the R-URI, and not say the To-URI. The R-URI typically only identifies one UA, which means this response may not technically mean anything different than 404.} This to me indicates a general lack of specificity in 3261 about the scope of applicability of the 6xx codes. As we are finding with many aspects of 3261, as we progress we find a need to tighten things up - clarify things that ambiguous or just wrong. But before that can be done, we need to figure out how we thing it ought to work. IMO there has to be some discretion in how 6xx codes are handled upstream - this interacts with what the upstream node is trying to accomplish. For instance: - if a proxy intends to try one or more contacts of a single AOR, and it receives a 6xx from one of them, it normally shouldn't try the others. - if a proxy intends to try one or more contacts of AOR-1 and then try one or more contacts of AOR-2, and it receives a 6xx from one of the AOR-1 contacts, then it should have the discretion to still try AOR-2. Whether it does this or not may depend on what it is trying to achieve by trying both. There clearly is a fine line here. The nodes receiving the 6xx responses need some discretion. But the nodes sending the 6xx need to have enough definition about what the codes mean to choose which one to return, so that the right thing might happen. Paul Dale.Worley@comcast.net wrote: > From: Scott Lawrence > > The one thing that I _don't_ like to see is a 603 being used for this; > that messes up any upstream forking. > > Personally, I think we should file a bug against the *existence* of > 6xx codes. They just can't be made to work correctly in the light of > multiple routings and forwardings of a URI. > > Dale > > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP > Use sip-implementors@cs.columbia.edu for questions on current sip > Use sip@ietf.org for new developments of core SIP > _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Thu Aug 03 10:16:26 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8dzs-00014I-Nw; Thu, 03 Aug 2006 10:16:24 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8dzq-000146-Q3 for sipping@ietf.org; Thu, 03 Aug 2006 10:16:22 -0400 Received: from zcars04e.nortel.com ([47.129.242.56]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G8dzp-0000mw-Et for sipping@ietf.org; Thu, 03 Aug 2006 10:16:22 -0400 Received: from zrc2hxm2.corp.nortel.com (zrc2hxm2.corp.nortel.com [47.103.123.73]) by zcars04e.nortel.com (Switch-2.2.0/Switch-2.2.0) with ESMTP id k73EAPe05109; Thu, 3 Aug 2006 10:10:26 -0400 (EDT) X-MimeOLE: Produced By Microsoft Exchange V6.5.7226.0 Content-class: urn:content-classes:message MIME-Version: 1.0 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: quoted-printable Subject: RE: [Sipping] Question on do not disturb indication Date: Thu, 3 Aug 2006 09:16:15 -0500 Message-ID: <6FC4416DDE56C44DA0AEE67BC7CA43710FBDA37E@zrc2hxm2.corp.nortel.com> In-Reply-To: <1154610920.2859.38.camel@sukothai.pingtel.com> X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: [Sipping] Question on do not disturb indication thread-index: Aca2/vYQuAyo/mqiQ5apaMvemL4QGgAAjLdQ From: "Brian Stucker" To: "Scott Lawrence" X-Spam-Score: 0.0 (/) X-Scan-Signature: 3002fc2e661cd7f114cb6bae92fe88f1 Cc: sipping@ietf.org, Francois Audet , Xiaotao Wu , Paul Kyzivat , "Elwell, John" X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org I think we're missing an important distinction: nested target sets. Let's say I have a proxy serving user X has two target sets: A and B. Target set A consists of two routes that are forked sequentially: target set B for user X, and go to voicemail for user X. Target set B consists of several routes that are forked in parallel: user X at device 1, user X at device 2, etc... Now, take a look at RFC-3261, section 16.6 around response contexts (third paragraph) and step 5 of section 16.7. Ok, so let's run our thought experiment... When a call comes in for User X, the proxy executes target set A for user X (sequentially fork to user X's target set B). This in turn executes target set B for user X (the user's registered contacts and other endpoints). Device 2 for User X sends back a 603.=20 This, according to RFC-3261, kills off all the other legs to target set B. At that point, there is no normative language in section 16.7 that requires the proxy to take any action towards target set A. It only MUST NOT create any new branches out of the context of target set B. And so the call then sequentially forks to the next destination in target set A, i.e. voicemail. Regards, Brian > -----Original Message----- > From: Scott Lawrence [mailto:slawrence@pingtel.com]=20 > Sent: Thursday, August 03, 2006 9:15 AM > To: Stucker, Brian (RICH1:B620) > Cc: Xiaotao Wu; Audet, Francois (SC100:3055);=20 > sipping@ietf.org; Paul Kyzivat; Elwell, John > Subject: RE: [Sipping] Question on do not disturb indication >=20 > On Thu, 2006-08-03 at 07:36 -0500, Brian Stucker wrote: > > I agree with Francois. A 603 isn't a horrible thing.=20 > Essentially with=20 > > SIP you have two choices here, kill one forked leg of the call (4xx > > response) or kill all of the forked legs (603). What the proxy does=20 > > with it from there for alternate routing is up to the proxy. >=20 > I don't think that it is - I think that 603 means that the=20 > proxy should stop trying to route the call and reject it back=20 > to the caller. If that isn't what it means, then we need a=20 > bug against 3261 to fix it. >=20 > > I don't know that the example Xiaotao gives is a reasonable=20 > use case.=20 > > If the user does not wish for their desk and cell phone to=20 > ring, then=20 > > you > > 486 those legs individually. The only other mechanism 3261=20 > gives you=20 > > to do that is with a 603 response. To be any more refined than that=20 > > would require extensions so that endpoints can coordinate their=20 > > actions more closely. > >=20 > > If we go with the notion that 603 is bad, then if Xiaotao's user's=20 > > cellphone is out in their car, and they don't want the call=20 > to go to=20 > > voicemail on the cellphone but go to the office=20 > secretary/voicemail ,=20 > > all they can do is send a 603 from their desk phone. >=20 > But voicemail is, from the point of view of the proxy, just=20 > another fork for the same user. That's my point to begin=20 > with - either 603 means 'kill all forks', or it doesn't, and=20 > if it does then it kills voicemail too. >=20 > -- > Scott Lawrence tel:+1-781-938-5306;ext=3D162 or=20 > sip:slawrence@pingtel.com > sipXpbx project coordinator - SIPfoundry =20 > http://www.sipfoundry.org/sipX > Chief Architect - Pingtel Corp. http://www.pingtel.com/ >=20 >=20 >=20 _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Thu Aug 03 10:30:20 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8eDH-0003F2-9e; Thu, 03 Aug 2006 10:30:15 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8eDF-0003Ex-T6 for sipping@ietf.org; Thu, 03 Aug 2006 10:30:13 -0400 Received: from [65.220.123.2] (helo=mail.pingtel.com) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G8eDA-0001Q5-L2 for sipping@ietf.org; Thu, 03 Aug 2006 10:30:13 -0400 Received: from localhost.localdomain (pi.pingtel.com [10.1.1.12]) by mail.pingtel.com (Postfix) with ESMTP id 3A1D76C020; Thu, 3 Aug 2006 09:29:38 -0400 (EDT) Subject: RE: [Sipping] Question on do not disturb indication From: Scott Lawrence To: "Elwell, John" In-Reply-To: <50B1CBA96870A34799A506B2313F2667098D7B24@ntht201e.siemenscomms.co.uk> References: <50B1CBA96870A34799A506B2313F2667098D7B24@ntht201e.siemenscomms.co.uk> Content-Type: text/plain Organization: Pingtel Corp. http://www.pingtel.com/ Date: Thu, 03 Aug 2006 10:29:15 -0400 Message-Id: <1154615356.2859.57.camel@sukothai.pingtel.com> Mime-Version: 1.0 X-Mailer: Evolution 2.6.2 (2.6.2-1.fc5.5) Content-Transfer-Encoding: 7bit X-Spam-Score: 0.1 (/) X-Scan-Signature: b19722fc8d3865b147c75ae2495625f2 Cc: sipping@ietf.org X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org On Thu, 2006-08-03 at 15:18 +0100, Elwell, John wrote: > Scott, > > To solve your 'ring twice' problem, we could, as an alternative to your > suggestion, have an explicit indication of DND, and so at least this would > allow a proxy that is scripted to take special action on DND (e.g., go to > voicemail) to know when to take that action, rather than assuming 480 is DND > or relying on a reason phrase in 480. At this point, the difficulty in the proxy is that it doesn't know or care which forks are phones and which are voicemail and which are rude recorded messages. It just forks calls to the configured/registered contacts. Other than a couple of special cases called out by 3261 like authentication challenges, it treats all 4xx responses exactly the same. I don't actually think that there is a 'problem' to be solved other than user agents asserting global knowledge by sending 6xx responses. To get back to your original question - whether or not there is an explicit response code needed for 'I am in DND mode, so your call has been ignored'... I don't think so. We have a framework for providing presence information that is rich enough to provide that information, and to mediate who gets which view of it. I think that the response to invite should be as simple as it is now (480 is fine for this), and if user agents want to provide a richer experience then they should support presence. -- Scott Lawrence tel:+1-781-938-5306;ext=162 or sip:slawrence@pingtel.com sipXpbx project coordinator - SIPfoundry http://www.sipfoundry.org/sipX Chief Architect - Pingtel Corp. http://www.pingtel.com/ _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Thu Aug 03 11:21:39 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8f0k-0006tk-Pc; Thu, 03 Aug 2006 11:21:22 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8f0j-0006tf-9C for sipping@ietf.org; Thu, 03 Aug 2006 11:21:21 -0400 Received: from zcars04f.nortel.com ([47.129.242.57]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G8f0g-0004BU-Ti for sipping@ietf.org; Thu, 03 Aug 2006 11:21:21 -0400 Received: from zrc2hxm2.corp.nortel.com (zrc2hxm2.corp.nortel.com [47.103.123.73]) by zcars04f.nortel.com (Switch-2.2.6/Switch-2.2.0) with ESMTP id k73FLF923561; Thu, 3 Aug 2006 11:21:15 -0400 (EDT) X-MimeOLE: Produced By Microsoft Exchange V6.5.7226.0 Content-class: urn:content-classes:message MIME-Version: 1.0 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: quoted-printable Subject: RE: [Sipping] Question on do not disturb indication Date: Thu, 3 Aug 2006 10:20:42 -0500 Message-ID: <6FC4416DDE56C44DA0AEE67BC7CA43710FBDA56A@zrc2hxm2.corp.nortel.com> In-Reply-To: <1154615356.2859.57.camel@sukothai.pingtel.com> X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: [Sipping] Question on do not disturb indication thread-index: Aca3CYhrQLUgCsHySIilSxSTBe9cwAABqRYg From: "Brian Stucker" To: "Scott Lawrence" , "Elwell, John" X-Spam-Score: 0.0 (/) X-Scan-Signature: 082a9cbf4d599f360ac7f815372a6a15 Cc: sipping@ietf.org X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org IMHO- 480 seems too generic a response from the UAS (although I have no problem with sending this to the UAC). What's wrong with 486 from the UAS? Regards, Brian =20 > -----Original Message----- > From: Scott Lawrence [mailto:slawrence@pingtel.com]=20 > Sent: Thursday, August 03, 2006 10:29 AM > To: Elwell, John > Cc: sipping@ietf.org > Subject: RE: [Sipping] Question on do not disturb indication >=20 > On Thu, 2006-08-03 at 15:18 +0100, Elwell, John wrote: > > Scott, > >=20 > > To solve your 'ring twice' problem, we could, as an alternative to=20 > > your suggestion, have an explicit indication of DND, and so=20 > at least=20 > > this would allow a proxy that is scripted to take special action on=20 > > DND (e.g., go to > > voicemail) to know when to take that action, rather than=20 > assuming 480=20 > > is DND or relying on a reason phrase in 480. >=20 > At this point, the difficulty in the proxy is that it doesn't=20 > know or care which forks are phones and which are voicemail=20 > and which are rude recorded messages. It just forks calls to=20 > the configured/registered contacts. Other than a couple of=20 > special cases called out by 3261 like authentication=20 > challenges, it treats all 4xx responses exactly the same. >=20 > I don't actually think that there is a 'problem' to be solved=20 > other than user agents asserting global knowledge by sending=20 > 6xx responses. >=20 > To get back to your original question - whether or not there=20 > is an explicit response code needed for 'I am in DND mode, so=20 > your call has been ignored'... I don't think so. We have a=20 > framework for providing presence information that is rich=20 > enough to provide that information, and to mediate who gets=20 > which view of it. I think that the response to invite should=20 > be as simple as it is now (480 is fine for this), and if user=20 > agents want to provide a richer experience then they should=20 > support presence. >=20 > -- > Scott Lawrence tel:+1-781-938-5306;ext=3D162 or=20 > sip:slawrence@pingtel.com > sipXpbx project coordinator - SIPfoundry =20 > http://www.sipfoundry.org/sipX > Chief Architect - Pingtel Corp. http://www.pingtel.com/ >=20 >=20 >=20 > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP > Use sip-implementors@cs.columbia.edu for questions on current sip > Use sip@ietf.org for new developments of core SIP >=20 _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Thu Aug 03 11:44:36 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8fND-000112-2q; Thu, 03 Aug 2006 11:44:35 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8fNC-00010e-Ak for sipping@ietf.org; Thu, 03 Aug 2006 11:44:34 -0400 Received: from [65.220.123.2] (helo=mail.pingtel.com) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G8fN7-0005MO-2L for sipping@ietf.org; Thu, 03 Aug 2006 11:44:34 -0400 Received: from localhost.localdomain (pi.pingtel.com [10.1.1.12]) by mail.pingtel.com (Postfix) with ESMTP id 9C11A6C01D; Thu, 3 Aug 2006 10:43:58 -0400 (EDT) Subject: RE: [Sipping] Question on do not disturb indication From: Scott Lawrence To: Brian Stucker In-Reply-To: <6FC4416DDE56C44DA0AEE67BC7CA43710FBDA37E@zrc2hxm2.corp.nortel.com> References: <6FC4416DDE56C44DA0AEE67BC7CA43710FBDA37E@zrc2hxm2.corp.nortel.com> Content-Type: text/plain Organization: Pingtel Corp. http://www.pingtel.com/ Date: Thu, 03 Aug 2006 11:43:36 -0400 Message-Id: <1154619816.2859.68.camel@sukothai.pingtel.com> Mime-Version: 1.0 X-Mailer: Evolution 2.6.2 (2.6.2-1.fc5.5) Content-Transfer-Encoding: 7bit X-Spam-Score: 0.1 (/) X-Scan-Signature: 69a74e02bbee44ab4f8eafdbcedd94a1 Cc: sipping@ietf.org, Francois Audet , Xiaotao Wu , Paul Kyzivat , "Elwell, John" X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org On Thu, 2006-08-03 at 09:16 -0500, Brian Stucker wrote: > I think we're missing an important distinction: nested target sets. > > Let's say I have a proxy serving user X has two target sets: A and B. > > Target set A consists of two routes that are forked sequentially: target > set B for user X, and go to voicemail for user X. > > Target set B consists of several routes that are forked in parallel: > user X at device 1, user X at device 2, etc... > > Now, take a look at RFC-3261, section 16.6 around response contexts > (third paragraph) and step 5 of section 16.7. > > Ok, so let's run our thought experiment... > > When a call comes in for User X, the proxy executes target set A for > user X (sequentially fork to user X's target set B). This in turn > executes target set B for user X (the user's registered contacts and > other endpoints). Device 2 for User X sends back a 603. > > This, according to RFC-3261, kills off all the other legs to target set > B. At that point, there is no normative language in section 16.7 that > requires the proxy to take any action towards target set A. It only MUST > NOT create any new branches out of the context of target set B. > > And so the call then sequentially forks to the next destination in > target set A, i.e. voicemail. I recognize the utility of the 'target set' concept, but I don't know of any standard way to express it in the existing protocol. Do you? -- Scott Lawrence tel:+1-781-938-5306;ext=162 or sip:slawrence@pingtel.com sipXpbx project coordinator - SIPfoundry http://www.sipfoundry.org/sipX Chief Architect - Pingtel Corp. http://www.pingtel.com/ _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Thu Aug 03 12:00:25 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8fcT-0008Ge-Hs; Thu, 03 Aug 2006 12:00:21 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8fcL-00081V-Tu for sipping@ietf.org; Thu, 03 Aug 2006 12:00:14 -0400 Received: from zcars04e.nortel.com ([47.129.242.56]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G8fcK-0006Tx-JO for sipping@ietf.org; Thu, 03 Aug 2006 12:00:13 -0400 Received: from zrc2hxm2.corp.nortel.com (zrc2hxm2.corp.nortel.com [47.103.123.73]) by zcars04e.nortel.com (Switch-2.2.0/Switch-2.2.0) with ESMTP id k73FsIe26293; Thu, 3 Aug 2006 11:54:19 -0400 (EDT) X-MimeOLE: Produced By Microsoft Exchange V6.5.7226.0 Content-class: urn:content-classes:message MIME-Version: 1.0 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: quoted-printable Subject: RE: [Sipping] Question on do not disturb indication Date: Thu, 3 Aug 2006 11:00:03 -0500 Message-ID: <6FC4416DDE56C44DA0AEE67BC7CA43710FBDA669@zrc2hxm2.corp.nortel.com> In-Reply-To: <1154619816.2859.68.camel@sukothai.pingtel.com> X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: [Sipping] Question on do not disturb indication thread-index: Aca3E975zNq5NwoHTTeBXbvJCi5RJgAAbnhQ From: "Brian Stucker" To: "Scott Lawrence" X-Spam-Score: 0.0 (/) X-Scan-Signature: 244a2fd369eaf00ce6820a760a3de2e8 Cc: sipping@ietf.org, Francois Audet , Xiaotao Wu , Paul Kyzivat , "Elwell, John" X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org I think CPL captures it pretty well.=20 Since it's local policy of the proxy as to how it expects the relevant services to work, short of standardizing a DND service (which is likely to prove very difficult) what do you see as missing? Regards, Brian =20 > -----Original Message----- > From: Scott Lawrence [mailto:slawrence@pingtel.com]=20 > Sent: Thursday, August 03, 2006 11:44 AM > To: Stucker, Brian (RICH1:B620) > Cc: Xiaotao Wu; Audet, Francois (SC100:3055);=20 > sipping@ietf.org; Paul Kyzivat; Elwell, John > Subject: RE: [Sipping] Question on do not disturb indication >=20 > On Thu, 2006-08-03 at 09:16 -0500, Brian Stucker wrote: > > I think we're missing an important distinction: nested target sets. > >=20 > > Let's say I have a proxy serving user X has two target=20 > sets: A and B. > >=20 > > Target set A consists of two routes that are forked sequentially:=20 > > target set B for user X, and go to voicemail for user X. > >=20 > > Target set B consists of several routes that are forked in parallel: > > user X at device 1, user X at device 2, etc... > >=20 > > Now, take a look at RFC-3261, section 16.6 around response contexts=20 > > (third paragraph) and step 5 of section 16.7. > >=20 > > Ok, so let's run our thought experiment... > >=20 > > When a call comes in for User X, the proxy executes target=20 > set A for=20 > > user X (sequentially fork to user X's target set B). This in turn=20 > > executes target set B for user X (the user's registered=20 > contacts and=20 > > other endpoints). Device 2 for User X sends back a 603. > >=20 > > This, according to RFC-3261, kills off all the other legs to target=20 > > set B. At that point, there is no normative language in=20 > section 16.7=20 > > that requires the proxy to take any action towards target set A. It=20 > > only MUST NOT create any new branches out of the context of=20 > target set B. > >=20 > > And so the call then sequentially forks to the next destination in=20 > > target set A, i.e. voicemail. >=20 > I recognize the utility of the 'target set' concept, but I=20 > don't know of any standard way to express it in the existing=20 > protocol. Do you? >=20 > -- > Scott Lawrence tel:+1-781-938-5306;ext=3D162 or=20 > sip:slawrence@pingtel.com > sipXpbx project coordinator - SIPfoundry =20 > http://www.sipfoundry.org/sipX > Chief Architect - Pingtel Corp. http://www.pingtel.com/ >=20 >=20 >=20 _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Thu Aug 03 12:03:29 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8ffU-0002uA-QX; Thu, 03 Aug 2006 12:03:28 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8ffS-0002u5-LZ for sipping@ietf.org; Thu, 03 Aug 2006 12:03:26 -0400 Received: from [65.220.123.2] (helo=mail.pingtel.com) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G8ffN-0006mF-Em for sipping@ietf.org; Thu, 03 Aug 2006 12:03:26 -0400 Received: from localhost.localdomain (pi.pingtel.com [10.1.1.12]) by mail.pingtel.com (Postfix) with ESMTP id 437E46C01D; Thu, 3 Aug 2006 11:02:51 -0400 (EDT) Subject: RE: [Sipping] Question on do not disturb indication From: Scott Lawrence To: Brian Stucker In-Reply-To: <6FC4416DDE56C44DA0AEE67BC7CA43710FBDA669@zrc2hxm2.corp.nortel.com> References: <6FC4416DDE56C44DA0AEE67BC7CA43710FBDA669@zrc2hxm2.corp.nortel.com> Content-Type: text/plain Organization: Pingtel Corp. http://www.pingtel.com/ Date: Thu, 03 Aug 2006 12:02:28 -0400 Message-Id: <1154620948.2859.76.camel@sukothai.pingtel.com> Mime-Version: 1.0 X-Mailer: Evolution 2.6.2 (2.6.2-1.fc5.5) Content-Transfer-Encoding: 7bit X-Spam-Score: 0.1 (/) X-Scan-Signature: cf4fa59384e76e63313391b70cd0dd25 Cc: sipping@ietf.org, Francois Audet , Xiaotao Wu , Paul Kyzivat , "Elwell, John" X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org On Thu, 2006-08-03 at 11:00 -0500, Brian Stucker wrote: > I think CPL captures it pretty well. > > Since it's local policy of the proxy as to how it expects the relevant > services to work, short of standardizing a DND service (which is likely > to prove very difficult) what do you see as missing? Something that tells me that the scope of a 6xx response is the class or all classes. -- Scott Lawrence tel:+1-781-938-5306;ext=162 or sip:slawrence@pingtel.com sipXpbx project coordinator - SIPfoundry http://www.sipfoundry.org/sipX Chief Architect - Pingtel Corp. http://www.pingtel.com/ _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Thu Aug 03 13:23:35 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8guj-0008Ll-DP; Thu, 03 Aug 2006 13:23:17 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8gui-0008Ld-5h for sipping@ietf.org; Thu, 03 Aug 2006 13:23:16 -0400 Received: from [216.148.227.155] (helo=rwcrmhc15.comcast.net) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G8guf-00079H-T8 for sipping@ietf.org; Thu, 03 Aug 2006 13:23:16 -0400 Received: from dragon.ariadne.com (dworley.hsd1.ma.comcast.net[24.34.79.42]) by comcast.net (rwcrmhc15) with ESMTP id <20060803172308m1500r44vfe>; Thu, 3 Aug 2006 17:23:08 +0000 Received: from dragon.ariadne.com (dragon.ariadne.com [127.0.0.1]) by dragon.ariadne.com (8.12.8/8.12.8) with ESMTP id k73HN7ZW013656 for ; Thu, 3 Aug 2006 13:23:07 -0400 Received: (from worley@localhost) by dragon.ariadne.com (8.12.8/8.12.8/Submit) id k73HN7pQ013652; Thu, 3 Aug 2006 13:23:07 -0400 Date: Thu, 3 Aug 2006 13:23:07 -0400 Message-Id: <200608031723.k73HN7pQ013652@dragon.ariadne.com> To: sipping@ietf.org From: Dale.Worley@comcast.net In-reply-to: <44D20161.9050506@cisco.com> (pkyzivat@cisco.com) Subject: Re: [Sipping] Question on do not disturb indication References: <50B1CBA96870A34799A506B2313F266709897F3A@ntht201e.siemenscomms.co.uk> <1154553270.2866.86.camel@sukothai.pingtel.com> <200608022251.k72Mpvkp020060@dragon.ariadne.com> <44D20161.9050506@cisco.com> X-Spam-Score: 0.2 (/) X-Scan-Signature: 97adf591118a232206bdb5a27b217034 X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org The difficulty with the 6xx codes is that their effect is prescribed in RFC 3261, section 16.7, item 9 -- immediately terminate *all* forks of this INVITE, all the way back to the UAC. And the UAS that generates the 6xx cannot know (in general) what the original request-URI was, and what its semantics are. Consider when Able is a member of a support group. If Able puts his phone DND, an incoming call to the support group should fail over to Baker. But if Able's phone responds with a 6xx, the caller receives the 6xx. I suppose we could make an exception when the UAS recognizes the To-URI and possesses authoritative information about it. From: Paul Kyzivat IMO there has to be some discretion in how 6xx codes are handled upstream - this interacts with what the upstream node is trying to accomplish. For instance: - if a proxy intends to try one or more contacts of a single AOR, and it receives a 6xx from one of them, it normally shouldn't try the others. - if a proxy intends to try one or more contacts of AOR-1 and then try one or more contacts of AOR-2, and it receives a 6xx from one of the AOR-1 contacts, then it should have the discretion to still try AOR-2. Whether it does this or not may depend on what it is trying to achieve by trying both. This sounds reasonable to me, but we need to amend RFC 3261 to make it possible. Dale _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Thu Aug 03 13:32:33 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8h3M-0005dM-9C; Thu, 03 Aug 2006 13:32:12 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8h3L-0005cv-DU for sipping@ietf.org; Thu, 03 Aug 2006 13:32:11 -0400 Received: from alnrmhc12.comcast.net ([204.127.225.92]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G8h3K-0000Ox-6k for sipping@ietf.org; Thu, 03 Aug 2006 13:32:11 -0400 Received: from dragon.ariadne.com (dworley.hsd1.ma.comcast.net[24.34.79.42]) by comcast.net (alnrmhc12) with ESMTP id <20060803173209b1200i5t3be>; Thu, 3 Aug 2006 17:32:09 +0000 Received: from dragon.ariadne.com (dragon.ariadne.com [127.0.0.1]) by dragon.ariadne.com (8.12.8/8.12.8) with ESMTP id k73HW8ZW013886 for ; Thu, 3 Aug 2006 13:32:08 -0400 Received: (from worley@localhost) by dragon.ariadne.com (8.12.8/8.12.8/Submit) id k73HW8DN013882; Thu, 3 Aug 2006 13:32:08 -0400 Date: Thu, 3 Aug 2006 13:32:08 -0400 Message-Id: <200608031732.k73HW8DN013882@dragon.ariadne.com> To: sipping@ietf.org From: Dale.Worley@comcast.net In-reply-to: <6FC4416DDE56C44DA0AEE67BC7CA43710FBDA37E@zrc2hxm2.corp.nortel.com> (bstucker@nortel.com) Subject: Re: [Sipping] Question on do not disturb indication References: <6FC4416DDE56C44DA0AEE67BC7CA43710FBDA37E@zrc2hxm2.corp.nortel.com> X-Spam-Score: 0.2 (/) X-Scan-Signature: 798b2e660f1819ae38035ac1d8d5e3ab X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org From: "Brian Stucker" When a call comes in for User X, the proxy executes target set A for user X (sequentially fork to user X's target set B). This in turn executes target set B for user X (the user's registered contacts and other endpoints). Device 2 for User X sends back a 603. This, according to RFC-3261, kills off all the other legs to target set B. At that point, there is no normative language in section 16.7 that requires the proxy to take any action towards target set A. It only MUST NOT create any new branches out of the context of target set B. And so the call then sequentially forks to the next destination in target set A, i.e. voicemail. Not quite -- the forking process for target set B cancels all remaining branches, and then *forwards the 603 response to its parent forking process*, the one for target set A. Then *that* forking process cancels all the remaining branches for target set A. And so on, recursively back to the caller. Dale _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Thu Aug 03 13:54:52 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8hP4-0003pN-LV; Thu, 03 Aug 2006 13:54:38 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8hP3-0003jB-Ab for sipping@ietf.org; Thu, 03 Aug 2006 13:54:37 -0400 Received: from stsc1260-eth-s1-s1p1-vip.va.neustar.com ([156.154.16.129] helo=chiedprmail1.ietf.org) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G8eNx-0001qU-OY for sipping@ietf.org; Thu, 03 Aug 2006 10:41:17 -0400 Received: from mailgate.siemenscomms.co.uk ([195.171.110.225] helo=bemg01.siemenscomms.co.uk) by chiedprmail1.ietf.org with esmtp (Exim 4.43) id 1G8e2m-0006Oe-Rt for sipping@ietf.org; Thu, 03 Aug 2006 10:19:27 -0400 Received: from CONVERSION-DAEMON.siemenscomms.co.uk by siemenscomms.co.uk (PMDF V6.0-24 #40642) id <0J3F00801EGCHH@siemenscomms.co.uk> for sipping@ietf.org; Thu, 03 Aug 2006 15:19:25 +0100 (BST) Received: from ntht207e.uksgcs.siemenscomms.co.uk ([137.223.247.82]) by siemenscomms.co.uk (PMDF V6.0-24 #40642) with ESMTP id <0J3F00807EEUB2@siemenscomms.co.uk>; Thu, 03 Aug 2006 15:18:30 +0100 (BST) Received: by ntht207e.uksgcs.siemenscomms.co.uk with Internet Mail Service (5.5.2657.72) id ; Thu, 03 Aug 2006 15:18:09 +0100 Content-return: allowed Date: Thu, 03 Aug 2006 15:18:08 +0100 From: "Elwell, John" Subject: RE: [Sipping] Question on do not disturb indication To: Scott Lawrence , Hadriel Kaplan Message-id: <50B1CBA96870A34799A506B2313F2667098D7B24@ntht201e.siemenscomms.co.uk> MIME-version: 1.0 X-Mailer: Internet Mail Service (5.5.2657.72) Content-type: text/plain Content-transfer-encoding: 7BIT X-Spam-Score: -2.6 (--) X-Scan-Signature: 31247fb3be228bb596db9127becad0bc Cc: sipping@ietf.org X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Scott, To solve your 'ring twice' problem, we could, as an alternative to your suggestion, have an explicit indication of DND, and so at least this would allow a proxy that is scripted to take special action on DND (e.g., go to voicemail) to know when to take that action, rather than assuming 480 is DND or relying on a reason phrase in 480. John > -----Original Message----- > From: Scott Lawrence [mailto:slawrence@pingtel.com] > Sent: 03 August 2006 13:36 > To: Hadriel Kaplan > Cc: sipping@ietf.org > Subject: RE: [Sipping] Question on do not disturb indication > > On Thu, 2006-08-03 at 03:37 -0400, Hadriel Kaplan wrote: > > > > > -----Original Message----- > > > From: Scott Lawrence [mailto:slawrence@pingtel.com] > > > > > [about a 603 response...] > > > I read that to mean that a forking proxy should abandon > all forks for > > > the call and terminate the request - the response asserts that the > > > voicemail (another end point) will not answer. > > > > > > That might be an appropriate response for a proxy to send if it is > > > authoritative for the domain and knows that a particular > user does not > > > exist, but I can't see how a user agent can ever have > enough global > > > knowledge to justify a 6xx response. > > > > That would be a 604 (Does not exist anywhere), no? > > A 603 could be useful someday for personal black-lists; for > example where > > the user agent may decide who to give 603's to based on the > From, and not > > send them to voicemail. > > > > The problem is just because I pressed DND doesn't mean I > don't want it to go > > to voicemail - I do want it to go to voicemail. So 6xx is > wrong. But > > sending a 480 typically only makes it go to voicemail if > there are no other > > contacts to try, or they all send 480. So I have to press > DND on all my > > phones; but I'm not sure that's avoidable. > > Yes - I call that the 'ring twice problem'; if I ignore or > reject a call > on my desk phone, then my personal forwarding rules send it to my cell > and I have to ignore or reject it there before it will go to my > voicemail. I control whether or not I have that problem by deciding > whether or not to have that forwarding in place. > > The only solution I can think of for avoiding ring-twice would be a > response that indicates that all forks that in the class 'my personal > phones' should be abandoned, but that phones in other classes, such as > 'my assistant', or 'my voicemail' may proceed. Perhaps such a thing > could be built with the caller prefs mechanisms, but I'm not > optimistic > that it would see wide implementation (I probably would not try to > implement it in our proxy, for example). > > -- > Scott Lawrence tel:+1-781-938-5306;ext=162 or > sip:slawrence@pingtel.com > sipXpbx project coordinator - SIPfoundry > http://www.sipfoundry.org/sipX > Chief Architect - Pingtel Corp. http://www.pingtel.com/ > > > > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP > Use sip-implementors@cs.columbia.edu for questions on current sip > Use sip@ietf.org for new developments of core SIP > _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Thu Aug 03 14:07:51 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8hbE-0000A0-8N; Thu, 03 Aug 2006 14:07:12 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8hbC-00008W-6V for sipping@ietf.org; Thu, 03 Aug 2006 14:07:10 -0400 Received: from zcars04e.nortel.com ([47.129.242.56]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G8hbA-0005AY-Rv for sipping@ietf.org; Thu, 03 Aug 2006 14:07:10 -0400 Received: from zrc2hxm0.corp.nortel.com (zrc2hxm0.corp.nortel.com [47.103.123.71]) by zcars04e.nortel.com (Switch-2.2.0/Switch-2.2.0) with ESMTP id k73I0Pe25876; Thu, 3 Aug 2006 14:00:25 -0400 (EDT) X-MimeOLE: Produced By Microsoft Exchange V6.5.7226.0 Content-class: urn:content-classes:message MIME-Version: 1.0 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: quoted-printable Subject: RE: [Sipping] Question on do not disturb indication Date: Thu, 3 Aug 2006 13:06:14 -0500 Message-ID: <1ECE0EB50388174790F9694F77522CCF0BF08B79@zrc2hxm0.corp.nortel.com> In-Reply-To: <1154563021.3841.13.camel@localhost.localdomain> X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: [Sipping] Question on do not disturb indication Thread-Index: Aca2j3wXrnCHa+llT/2qvioXUkT0fgAl0JGA From: "Francois Audet" To: "Scott Lawrence" X-Spam-Score: 0.0 (/) X-Scan-Signature: b280b4db656c3ca28dd62e5e0b03daa8 Cc: "Elwell, John" , Paul Kyzivat , sipping@ietf.org X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org What I am saying is this: 603 means that the request to sip:dude@example.com will positively not be=20 answered by sip:dude@example.com If the proxy's logic is that the session is retargeted to an attendant, or a voicemail service, a hunt group, or just dropped completely, that's fine. It's the desired behavior. As for your last claim, we are saying exactly the same thing: No 4xx response will affect=20 forks in our proxy (i think you misread what I said). What I am trying to say is that IF the desired behavior is to affect ALL forks, a 480/488 won't work. > -----Original Message----- > From: Scott Lawrence [mailto:slawrence@pingtel.com]=20 > Sent: Wednesday, August 02, 2006 4:57 PM > To: Audet, Francois (SC100:3055) > Cc: Paul Kyzivat; sipping@ietf.org; Elwell, John > Subject: RE: [Sipping] Question on do not disturb indication >=20 > On Wed, 2006-08-02 at 18:24 -0500, Francois Audet wrote: > > I agree: I don't see a need for a special code for this, as=20 > it seems=20 > > to make pretty much exactly to existing practice. >=20 > > On the 603 issue, I'd like to disagree with everybody else... > > Sometimes being able to decline the call everywhere is=20 > actually what=20 > > you want. > > As an example, I have multiple forked device. If I send a 603, it=20 > > triggers the proxy to use whatever alternate routing is sees fit=20 > > (e.g., voicemail, attendant, etc.). That's the desired behavior. >=20 > Well, that depends on what you think 603 means. =20 >=20 > 3261 says: >=20 > 6xx responses indicate that a server has definitive=20 > information about > a particular user, not just the particular instance=20 > indicated in the > Request-URI. >=20 > and for 603 specifically it says: >=20 > This > status response is returned only if the client knows that no other > end point will answer the request. >=20 > I read that to mean that a forking proxy should abandon all=20 > forks for the call and terminate the request - the response=20 > asserts that the voicemail (another end point) will not answer. >=20 > That might be an appropriate response for a proxy to send if=20 > it is authoritative for the domain and knows that a=20 > particular user does not exist, but I can't see how a user=20 > agent can ever have enough global knowledge to justify a 6xx response. >=20 > > Problem with a 480/486 with that scenario is that it won't=20 > trigger the=20 > > alternate routing and will keep on ringing the other fork. That may=20 > > not be the desired behavior. >=20 > This is interesting - you seem to have exactly the opposite=20 > belief about these. No 4xx response will affect other forks=20 > in our proxy. >=20 > -- > Scott Lawrence tel:+1-781-938-5306;ext=3D162 or=20 > sip:slawrence@pingtel.com > sipXpbx project coordinator - SIPfoundry =20 > http://www.sipfoundry.org/sipX > Chief Architect - Pingtel Corp. http://www.pingtel.com/ >=20 >=20 >=20 _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Thu Aug 03 14:10:22 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8heG-0001rS-D8; Thu, 03 Aug 2006 14:10:20 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8heF-0001rI-7t for sipping@ietf.org; Thu, 03 Aug 2006 14:10:19 -0400 Received: from zrtps0kp.nortel.com ([47.140.192.56]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G8heD-0005lP-Vp for sipping@ietf.org; Thu, 03 Aug 2006 14:10:19 -0400 Received: from zrc2hxm0.corp.nortel.com (zrc2hxm0.corp.nortel.com [47.103.123.71]) by zrtps0kp.nortel.com (Switch-2.2.6/Switch-2.2.0) with ESMTP id k73IABO13893; Thu, 3 Aug 2006 14:10:12 -0400 (EDT) X-MimeOLE: Produced By Microsoft Exchange V6.5.7226.0 Content-class: urn:content-classes:message MIME-Version: 1.0 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: quoted-printable Subject: RE: [Sipping] Question on do not disturb indication Date: Thu, 3 Aug 2006 13:10:08 -0500 Message-ID: <1ECE0EB50388174790F9694F77522CCF0BF08B85@zrc2hxm0.corp.nortel.com> In-Reply-To: <50B1CBA96870A34799A506B2313F266709897F66@ntht201e.siemenscomms.co.uk> X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: [Sipping] Question on do not disturb indication Thread-Index: Aca2xkopbeIjijuISAKlTavCd20UEwAYUA/w From: "Francois Audet" To: "Elwell, John" , "Paul Kyzivat" , "Scott Lawrence" X-Spam-Score: 0.0 (/) X-Scan-Signature: 7a6398bf8aaeabc7a7bb696b6b0a2aad Cc: sipping@ietf.org X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org =20 > [JRE] I agree that "make busy" is a separate feature and is=20 > best dealt with by 486. But where I explicitly want the=20 > caller to know that I am in DND (possibly with a retry-after=20 > time), do you think that 480 is sufficient, bearing in mind=20 > that it gets used for other purposes too, e.g., in RFC3398? Personally, I think it is sufficient, and I also think that=20 mapping DND to 480 is pretty much established by now. My perspective is Enterprise-centric however: there might be some Service Provider services that need a more precise indication. I could be convinced that it is needed... _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Thu Aug 03 14:12:10 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8hfy-0002YG-JR; Thu, 03 Aug 2006 14:12:06 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8hfw-0002Wr-EQ for sipping@ietf.org; Thu, 03 Aug 2006 14:12:04 -0400 Received: from zcars04f.nortel.com ([47.129.242.57]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G8hfv-0005tj-1J for sipping@ietf.org; Thu, 03 Aug 2006 14:12:04 -0400 Received: from zrc2hxm0.corp.nortel.com (zrc2hxm0.corp.nortel.com [47.103.123.71]) by zcars04f.nortel.com (Switch-2.2.6/Switch-2.2.0) with ESMTP id k73IBN906803; Thu, 3 Aug 2006 14:11:23 -0400 (EDT) X-MimeOLE: Produced By Microsoft Exchange V6.5.7226.0 Content-class: urn:content-classes:message MIME-Version: 1.0 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: quoted-printable Subject: RE: [Sipping] Question on do not disturb indication Date: Thu, 3 Aug 2006 13:11:19 -0500 Message-ID: <1ECE0EB50388174790F9694F77522CCF0BF08B8F@zrc2hxm0.corp.nortel.com> In-Reply-To: X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: [Sipping] Question on do not disturb indication Thread-Index: Aca27oKy/+xqgSl5Q3a22itbRZlzGwAOZOLg From: "Francois Audet" To: "Xiaotao Wu" X-Spam-Score: 0.0 (/) X-Scan-Signature: 68ba2b07ef271dba6ee42a93832cfa4c Cc: sipping@ietf.org, Paul Kyzivat , "Elwell, John" , Scott Lawrence X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Sure you can do it on phones. You an have a button on the phone to "Decline" an incoming call dynamically. You can have a DND feature (either permanent or dynamic). This is widely done today.=20 > -----Original Message----- > From: xiaotaow@gmail.com [mailto:xiaotaow@gmail.com] On=20 > Behalf Of Xiaotao Wu > Sent: Thursday, August 03, 2006 4:18 AM > To: Audet, Francois (SC100:3055) > Cc: Paul Kyzivat; Scott Lawrence; Elwell, John; sipping@ietf.org > Subject: Re: [Sipping] Question on do not disturb indication >=20 > On 8/2/06, Francois Audet wrote: > > On the 603 issue, I'd like to disagree with everybody else... > > Sometimes being able to decline the call everywhere is=20 > actually what=20 > > you want. > > As an example, I have multiple forked device. If I send a 603, it=20 > > triggers the proxy to use whatever alternate routing is sees fit=20 > > (e.g., voicemail, attendant, etc.). That's the desired behavior. > > > > Problem with a 480/486 with that scenario is that it won't=20 > trigger the=20 > > alternate routing and will keep on ringing the other fork. That may=20 > > not be the desired behavior. >=20 > I would expect the response should be sent by the proxy,=20 > instead by endpoints. If it's send by an endpoint, things may=20 > get complicated. > For example, Bob has a desk phone, a cell phone, a voicemail,=20 > and a secretary. It's hard to compose a response from his=20 > desk phone to indicate that he does not want to ring his desk=20 > phone, and cell phone, but ring his secretary phone and voicemail. >=20 > So, I think the problem is more on what's the expected code=20 > for the caller, not on the routing behavior of the request/response. >=20 > -Xiaotao >=20 > > > > My 2 cents. > > > > > -----Original Message----- > > > From: Paul Kyzivat [mailto:pkyzivat@cisco.com] > > > Sent: Wednesday, August 02, 2006 3:31 PM > > > To: Scott Lawrence > > > Cc: sipping@ietf.org; Elwell, John > > > Subject: Re: [Sipping] Question on do not disturb indication > > > > > > I pretty much agree with Scott. In many/most cases I=20 > won't want the=20 > > > caller to know why his call is going unanswered. > > > Both 480 and 486 are in effect "white lies". Which one to use may=20 > > > depend on which impression you want to give to the=20 > caller. Another=20 > > > one - if you want to be more rude - is 403 Forbidden. > > > > > > I don't see a need for a *special* code that is unique to this=20 > > > service. > > > > > > Paul > > > > > > Scott Lawrence wrote: > > > > On Wed, 2006-08-02 at 21:09 +0100, Elwell, John wrote: > > > >> RFC 3261 suggests the use of response code 480 Temporarily=20 > > > >> Unavailable for indicating do not disturb (DND), but it is > > > not used > > > >> exclusively for this purpose. There are other documents (e.g.,=20 > > > >> RFC > > > >> 4504) that suggest the use of 486 Busy Here, but again this is=20 > > > >> not helpful if the receiver wants to distinguish between DND > > > and ordinary > > > >> busy. Presence can be used to indicate this condition=20 > through the=20 > > > >> RPID note element, but that does not say why a=20 > particular session=20 > > > >> establishment attempt has failed. What are the best=20 > practices for=20 > > > >> indicating the DND condition? Does anyone see a need for > > > an explicit > > > >> indicator for this purpose, e.g., a new SIP response code? > > > > > > > > Personally, I like 480 for this. > > > > > > > > If the phone wanted to be sneaky, it could just mute the > > > ringing and > > > > allow the transaction to time out without explicitly=20 > rejecting it. > > > > That prevents the caller from being able to tell they are being=20 > > > > disregarded, but some users might prefer that forwarding > > > take effect > > > > sooner and not like that. > > > > > > > > The one thing that I _don't_ like to see is a 603 being > > > used for this; > > > > that messes up any upstream forking. > > > > > > > > > > > > > > _______________________________________________ > > > Sipping mailing list =20 > https://www1.ietf.org/mailman/listinfo/sipping > > > This list is for NEW development of the application of SIP Use=20 > > > sip-implementors@cs.columbia.edu for questions on current sip Use=20 > > > sip@ietf.org for new developments of core SIP > > > > > > > _______________________________________________ > > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > > This list is for NEW development of the application of SIP Use=20 > > sip-implementors@cs.columbia.edu for questions on current sip Use=20 > > sip@ietf.org for new developments of core SIP > > >=20 >=20 > -- > -Xiaotao >=20 > = =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D= =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D= =3D=3D=3D=3D=3D=3D=3D=3D=3D > Name : Xiaotao Wu > Email : xwu@avaya.com, xiaotaow@gmail.com > Web : http://www.cs.columbia.edu/~xiaotaow > Phone : (732) 852-2133 > SIP : sip:xiaotaow@cs.columbia.edu > Address : 1M-240, 307 Middletown Lincroft Road > Lincroft, NJ 07738-1526, U.S.A. >=20 _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Thu Aug 03 14:13:06 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8hgw-0003DY-7Y; Thu, 03 Aug 2006 14:13:06 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8hgu-0003DI-3o for sipping@ietf.org; Thu, 03 Aug 2006 14:13:04 -0400 Received: from zrtps0kn.nortel.com ([47.140.192.55]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G8hgs-00062g-Rf for sipping@ietf.org; Thu, 03 Aug 2006 14:13:04 -0400 Received: from zrc2hxm0.corp.nortel.com (zrc2hxm0.corp.nortel.com [47.103.123.71]) by zrtps0kn.nortel.com (Switch-2.2.6/Switch-2.2.0) with ESMTP id k73ICuh21496; Thu, 3 Aug 2006 14:12:56 -0400 (EDT) X-MimeOLE: Produced By Microsoft Exchange V6.5.7226.0 Content-class: urn:content-classes:message MIME-Version: 1.0 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: quoted-printable Subject: RE: [Sipping] Question on do not disturb indication Date: Thu, 3 Aug 2006 13:12:14 -0500 Message-ID: <1ECE0EB50388174790F9694F77522CCF0BF08B97@zrc2hxm0.corp.nortel.com> In-Reply-To: <0J3F00I49B8SG700@jes-fe2.zx.nl> X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: [Sipping] Question on do not disturb indication Thread-Index: Aca29SxXKMRh9slqT1aPGHmR0PXjiAAMx5YQ From: "Francois Audet" To: "Jeroen Van Bemmel" , "Xiaotao Wu" X-Spam-Score: 0.0 (/) X-Scan-Signature: ffa9dfbbe7cc58b3fa6b8ae3e57b0aa3 Cc: "Elwell, John" , Paul Kyzivat , sipping@ietf.org, Scott Lawrence X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org If you provide an alternate address, then it should be a 3XX response. =20 > -----Original Message----- > From: Jeroen Van Bemmel [mailto:jbemmel@zonnet.nl]=20 > Sent: Thursday, August 03, 2006 5:06 AM > To: Xiaotao Wu; Audet, Francois (SC100:3055) > Cc: sipping@ietf.org; Paul Kyzivat; Elwell, John; Scott Lawrence > Subject: RE: [Sipping] Question on do not disturb indication >=20 > So, I think the problem is more on what's the expected code=20 > for the caller, not on the routing behavior of the request/response. >=20 > (JvB) The reason phrase is (and was always) intended for=20 > this. "480 Do not disturb (use IM, call my secretary or leave=20 > a voiemail)" would seem appropriate. The semantics are hard=20 > to convey to a machine, therefore ideally the choice for any=20 > follow ups should be made by the caller IMHO. A 6xx would=20 > guarantee that outcome >=20 > Regards, > Jeroen=20 >=20 >=20 >=20 _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Thu Aug 03 14:15:44 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8hjT-0004pw-Gp; Thu, 03 Aug 2006 14:15:43 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8hjR-0004oM-T8 for sipping@ietf.org; Thu, 03 Aug 2006 14:15:41 -0400 Received: from zcars04e.nortel.com ([47.129.242.56]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G8hjQ-0006Ty-J0 for sipping@ietf.org; Thu, 03 Aug 2006 14:15:41 -0400 Received: from zrc2hxm0.corp.nortel.com (zrc2hxm0.corp.nortel.com [47.103.123.71]) by zcars04e.nortel.com (Switch-2.2.0/Switch-2.2.0) with ESMTP id k73I9Me26751; Thu, 3 Aug 2006 14:09:22 -0400 (EDT) X-MimeOLE: Produced By Microsoft Exchange V6.5.7226.0 Content-class: urn:content-classes:message MIME-Version: 1.0 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: quoted-printable Subject: RE: [Sipping] Question on do not disturb indication Date: Thu, 3 Aug 2006 13:15:11 -0500 Message-ID: <1ECE0EB50388174790F9694F77522CCF0BF08BA3@zrc2hxm0.corp.nortel.com> In-Reply-To: <1154610920.2859.38.camel@sukothai.pingtel.com> X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: [Sipping] Question on do not disturb indication Thread-Index: Aca2/veCjlQ7TaYkR+Kq2DcriUM7OQAKYsuQ From: "Francois Audet" To: "Scott Lawrence" , "Brian Stucker" X-Spam-Score: 0.0 (/) X-Scan-Signature: 52f7a77164458f8c7b36b66787c853da Cc: sipping@ietf.org, Paul Kyzivat , Xiaotao Wu , "Elwell, John" X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Scott, They key point is 603 kills all the forks, yes. Going somewhere else (like a voicemail or attendant) is a retargeting. It's not the same target. It's just an alternate route that the proxy can have. Nothing particularly new about this. Cutting a cable will achieve the same effect. > -----Original Message----- > From: Scott Lawrence [mailto:slawrence@pingtel.com]=20 > Sent: Thursday, August 03, 2006 6:15 AM > To: Stucker, Brian (RICH1:B620) > Cc: Xiaotao Wu; Audet, Francois (SC100:3055);=20 > sipping@ietf.org; Paul Kyzivat; Elwell, John > Subject: RE: [Sipping] Question on do not disturb indication >=20 > On Thu, 2006-08-03 at 07:36 -0500, Brian Stucker wrote: > > I agree with Francois. A 603 isn't a horrible thing.=20 > Essentially with=20 > > SIP you have two choices here, kill one forked leg of the call (4xx > > response) or kill all of the forked legs (603). What the proxy does=20 > > with it from there for alternate routing is up to the proxy. >=20 > I don't think that it is - I think that 603 means that the=20 > proxy should stop trying to route the call and reject it back=20 > to the caller. If that isn't what it means, then we need a=20 > bug against 3261 to fix it. >=20 > > I don't know that the example Xiaotao gives is a reasonable=20 > use case.=20 > > If the user does not wish for their desk and cell phone to=20 > ring, then=20 > > you > > 486 those legs individually. The only other mechanism 3261=20 > gives you=20 > > to do that is with a 603 response. To be any more refined than that=20 > > would require extensions so that endpoints can coordinate their=20 > > actions more closely. > >=20 > > If we go with the notion that 603 is bad, then if Xiaotao's user's=20 > > cellphone is out in their car, and they don't want the call=20 > to go to=20 > > voicemail on the cellphone but go to the office=20 > secretary/voicemail ,=20 > > all they can do is send a 603 from their desk phone. >=20 > But voicemail is, from the point of view of the proxy, just=20 > another fork for the same user. That's my point to begin=20 > with - either 603 means 'kill all forks', or it doesn't, and=20 > if it does then it kills voicemail too. >=20 > -- > Scott Lawrence tel:+1-781-938-5306;ext=3D162 or=20 > sip:slawrence@pingtel.com > sipXpbx project coordinator - SIPfoundry =20 > http://www.sipfoundry.org/sipX > Chief Architect - Pingtel Corp. http://www.pingtel.com/ >=20 >=20 >=20 _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Thu Aug 03 14:21:03 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8hoc-0003FK-Ie; Thu, 03 Aug 2006 14:21:02 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8hob-0003FF-DX for sipping@ietf.org; Thu, 03 Aug 2006 14:21:01 -0400 Received: from zcars04f.nortel.com ([47.129.242.57]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G8hoa-0007Qq-2i for sipping@ietf.org; Thu, 03 Aug 2006 14:21:01 -0400 Received: from zrc2hxm0.corp.nortel.com (zrc2hxm0.corp.nortel.com [47.103.123.71]) by zcars04f.nortel.com (Switch-2.2.6/Switch-2.2.0) with ESMTP id k73IKZ909313; Thu, 3 Aug 2006 14:20:35 -0400 (EDT) X-MimeOLE: Produced By Microsoft Exchange V6.5.7226.0 Content-class: urn:content-classes:message MIME-Version: 1.0 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: quoted-printable Subject: RE: [Sipping] Question on do not disturb indication Date: Thu, 3 Aug 2006 13:20:04 -0500 Message-ID: <1ECE0EB50388174790F9694F77522CCF0BF08BBC@zrc2hxm0.corp.nortel.com> In-Reply-To: <200608031723.k73HN7pQ013652@dragon.ariadne.com> X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: [Sipping] Question on do not disturb indication Thread-Index: Aca3Iez1L5d4cmyxRFCWo9ixyeVEywABx69w From: "Francois Audet" To: , X-Spam-Score: 0.0 (/) X-Scan-Signature: 0a7aa2e6e558383d84476dc338324fab Cc: X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org I agree with Paul as well. But I don't think RFC 3261 necessarily needs to be amended. It's always been legal for a proxy to retarget however it pleases. What I do see however is that we have never been very clear in our documentation about normal forking (i.e., same targer AOR) versus=20 forking with retargeting (when a fork goes to a different AOR).=20 (In both cases, it applies to both sequential and parallel forking). > -----Original Message----- > From: Dale.Worley@comcast.net [mailto:Dale.Worley@comcast.net]=20 > Sent: Thursday, August 03, 2006 10:23 AM > To: sipping@ietf.org > Subject: Re: [Sipping] Question on do not disturb indication >=20 > The difficulty with the 6xx codes is that their effect is=20 > prescribed in RFC 3261, section 16.7, item 9 -- immediately=20 > terminate *all* forks of this INVITE, all the way back to the=20 > UAC. And the UAS that generates the 6xx cannot know (in=20 > general) what the original request-URI was, and what its=20 > semantics are. >=20 > Consider when Able is a member of a support group. If Able=20 > puts his phone DND, an incoming call to the support group=20 > should fail over to Baker. But if Able's phone responds with=20 > a 6xx, the caller receives the 6xx. >=20 > I suppose we could make an exception when the UAS recognizes=20 > the To-URI and possesses authoritative information about it. >=20 > From: Paul Kyzivat >=20 > IMO there has to be some discretion in how 6xx codes are handled=20 > upstream - this interacts with what the upstream node is trying to=20 > accomplish. For instance: >=20 > - if a proxy intends to try one or more contacts of a single AOR, > and it receives a 6xx from one of them, it normally shouldn't > try the others. >=20 > - if a proxy intends to try one or more contacts of AOR-1 and then > try one or more contacts of AOR-2, and it receives a 6xx from > one of the AOR-1 contacts, then it should have the discretion > to still try AOR-2. Whether it does this or not may depend on > what it is trying to achieve by trying both. >=20 > This sounds reasonable to me, but we need to amend RFC 3261=20 > to make it possible. >=20 > Dale >=20 > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP=20 > Use sip-implementors@cs.columbia.edu for questions on current=20 > sip Use sip@ietf.org for new developments of core SIP >=20 _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Thu Aug 03 14:27:55 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8hvA-0000b1-Sl; Thu, 03 Aug 2006 14:27:48 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8hv9-0000aZ-NE for sipping@ietf.org; Thu, 03 Aug 2006 14:27:47 -0400 Received: from sj-iport-6.cisco.com ([171.71.176.117]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G8hv8-0000yC-Cc for sipping@ietf.org; Thu, 03 Aug 2006 14:27:47 -0400 Received: from sj-dkim-1.cisco.com ([171.71.179.21]) by sj-iport-6.cisco.com with ESMTP; 03 Aug 2006 11:27:45 -0700 Received: from sj-core-4.cisco.com (sj-core-4.cisco.com [171.68.223.138]) by sj-dkim-1.cisco.com (8.12.11.20060308/8.12.11) with ESMTP id k73IRjai022405; Thu, 3 Aug 2006 11:27:45 -0700 Received: from xbh-rtp-211.amer.cisco.com (xbh-rtp-211.cisco.com [64.102.31.102]) by sj-core-4.cisco.com (8.12.10/8.12.6) with ESMTP id k73IRimS015077; Thu, 3 Aug 2006 11:27:45 -0700 (PDT) Received: from xfe-rtp-201.amer.cisco.com ([64.102.31.38]) by xbh-rtp-211.amer.cisco.com with Microsoft SMTPSVC(6.0.3790.1830); Thu, 3 Aug 2006 14:27:45 -0400 Received: from [161.44.79.104] ([161.44.79.104]) by xfe-rtp-201.amer.cisco.com with Microsoft SMTPSVC(6.0.3790.1830); Thu, 3 Aug 2006 14:27:44 -0400 Message-ID: <44D24020.2020809@cisco.com> Date: Thu, 03 Aug 2006 14:27:44 -0400 From: Paul Kyzivat User-Agent: Thunderbird 1.5.0.5 (Windows/20060719) MIME-Version: 1.0 To: Dale.Worley@comcast.net Subject: Re: [Sipping] Question on do not disturb indication References: <50B1CBA96870A34799A506B2313F266709897F3A@ntht201e.siemenscomms.co.uk> <1154553270.2866.86.camel@sukothai.pingtel.com> <200608022251.k72Mpvkp020060@dragon.ariadne.com> <44D20161.9050506@cisco.com> <200608031723.k73HN7pQ013652@dragon.ariadne.com> In-Reply-To: <200608031723.k73HN7pQ013652@dragon.ariadne.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Content-Transfer-Encoding: 7bit X-OriginalArrivalTime: 03 Aug 2006 18:27:44.0784 (UTC) FILETIME=[8368F100:01C6B72A] DKIM-Signature: a=rsa-sha1; q=dns; l=1649; t=1154629665; x=1155493665; c=relaxed/simple; s=sjdkim1002; h=Content-Type:From:Subject:Content-Transfer-Encoding:MIME-Version; d=cisco.com; i=pkyzivat@cisco.com; z=From:Paul=20Kyzivat=20 |Subject:Re=3A=20[Sipping]=20Question=20on=20do=20not=20disturb=20indication; X=v=3Dcisco.com=3B=20h=3Dyu9ohsv1kFqX2uC2CMjguD3sTnY=3D; b=XDUeLB+w+XxU4wk7P2Djhz/misg/17TVryISuaLtjQMZIw8isiGuisDeokBDluJwrAfwLJTC oDIUoCxfgAWrvCTPSpkQnaxkF9N2LZOyhKENytQtt5S2H0xr6EfjECeF; Authentication-Results: sj-dkim-1.cisco.com; header.From=pkyzivat@cisco.com; dkim=pass ( sig from cisco.com verified; ); X-Spam-Score: 0.0 (/) X-Scan-Signature: cab78e1e39c4b328567edb48482b6a69 Cc: sipping@ietf.org X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Dale.Worley@comcast.net wrote: > The difficulty with the 6xx codes is that their effect is prescribed > in RFC 3261, section 16.7, item 9 -- immediately terminate *all* forks > of this INVITE, all the way back to the UAC. Reviewing 16.7, I agree with you. > And the UAS that > generates the 6xx cannot know (in general) what the original > request-URI was, and what its semantics are. Yup. > Consider when Able is a member of a support group. If Able puts his > phone DND, an incoming call to the support group should fail over to > Baker. But if Able's phone responds with a 6xx, the caller receives > the 6xx. > > I suppose we could make an exception when the UAS recognizes the > To-URI and possesses authoritative information about it. > > From: Paul Kyzivat > > IMO there has to be some discretion in how 6xx codes are handled > upstream - this interacts with what the upstream node is trying to > accomplish. For instance: > > - if a proxy intends to try one or more contacts of a single AOR, > and it receives a 6xx from one of them, it normally shouldn't > try the others. > > - if a proxy intends to try one or more contacts of AOR-1 and then > try one or more contacts of AOR-2, and it receives a 6xx from > one of the AOR-1 contacts, then it should have the discretion > to still try AOR-2. Whether it does this or not may depend on > what it is trying to achieve by trying both. > > This sounds reasonable to me, but we need to amend RFC 3261 to make it > possible. Yeah, does seem that way. Paul _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Thu Aug 03 14:45:27 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8iCC-0000mw-9U; Thu, 03 Aug 2006 14:45:24 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8iCA-0000lP-C9 for sipping@ietf.org; Thu, 03 Aug 2006 14:45:22 -0400 Received: from mailgate.siemenscomms.co.uk ([195.171.110.225] helo=bemg01.siemenscomms.co.uk) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G8hzv-0002pd-Iw for sipping@ietf.org; Thu, 03 Aug 2006 14:32:44 -0400 Received: from CONVERSION-DAEMON.siemenscomms.co.uk by siemenscomms.co.uk (PMDF V6.0-24 #40642) id <0J3F00101Q72RL@siemenscomms.co.uk> for sipping@ietf.org; Thu, 03 Aug 2006 19:33:03 +0100 (BST) Received: from ntht207e.uksgcs.siemenscomms.co.uk ([137.223.247.82]) by siemenscomms.co.uk (PMDF V6.0-24 #40642) with ESMTP id <0J3F00LB1Q72QU@siemenscomms.co.uk>; Thu, 03 Aug 2006 19:33:02 +0100 (BST) Received: by ntht207e.uksgcs.siemenscomms.co.uk with Internet Mail Service (5.5.2657.72) id ; Thu, 03 Aug 2006 19:32:42 +0100 Content-return: allowed Date: Thu, 03 Aug 2006 19:32:41 +0100 From: "Elwell, John" Subject: RE: [Sipping] Question on do not disturb indication To: Francois Audet , Paul Kyzivat , Scott Lawrence Message-id: <50B1CBA96870A34799A506B2313F2667098D7D21@ntht201e.siemenscomms.co.uk> MIME-version: 1.0 X-Mailer: Internet Mail Service (5.5.2657.72) Content-type: text/plain Content-transfer-encoding: 7BIT X-Spam-Score: 0.0 (/) X-Scan-Signature: ea4ac80f790299f943f0a53be7e1a21a Cc: sipping@ietf.org X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Francois, It is not only that I want the caller to know that I am in DND - also my proxy, if it is scripted to perform special behaviour on DND (e.g., retarget), then I would like it to know. I am not sure that 480 is sufficiently precise. John > -----Original Message----- > From: Francois Audet [mailto:audet@nortel.com] > Sent: 03 August 2006 19:10 > To: Elwell, John; Paul Kyzivat; Scott Lawrence > Cc: sipping@ietf.org > Subject: RE: [Sipping] Question on do not disturb indication > > > > [JRE] I agree that "make busy" is a separate feature and is > > best dealt with by 486. But where I explicitly want the > > caller to know that I am in DND (possibly with a retry-after > > time), do you think that 480 is sufficient, bearing in mind > > that it gets used for other purposes too, e.g., in RFC3398? > > Personally, I think it is sufficient, and I also think that > mapping DND to 480 is pretty much established by now. > > My perspective is Enterprise-centric however: there might be > some Service Provider services that need a more precise indication. > I could be convinced that it is needed... > _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Thu Aug 03 14:45:39 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8iCR-0001I8-QV; Thu, 03 Aug 2006 14:45:39 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8iCQ-0001Gk-3X for sipping@ietf.org; Thu, 03 Aug 2006 14:45:38 -0400 Received: from jes-fe1.zx.nl ([194.187.76.132]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G8hww-0001ky-Nx for sipping@ietf.org; Thu, 03 Aug 2006 14:29:43 -0400 Received: from Inbox ([84.241.193.195]) by jes-fe1.zx.nl (Sun Java System Messaging Server 6.2-2.05 (built Apr 28 2005)) with ESMTP id <0J3F00HIISL0AR30@jes-fe1.zx.nl> for sipping@ietf.org; Thu, 03 Aug 2006 20:25:03 +0100 (WEST) Date: Thu, 03 Aug 2006 20:29:35 +0200 From: Jeroen Van Bemmel Subject: RE: [Sipping] Question on do not disturb indication To: Francois Audet , Xiaotao Wu Message-id: <0J3F00HIJSL1AR30@jes-fe1.zx.nl> MIME-version: 1.0 Content-type: text/plain; charset=iso-8859-1 Content-transfer-encoding: quoted-printable Importance: normal X-Priority: 3 X-Spam-Score: 1.2 (+) X-Scan-Signature: 0ddefe323dd869ab027dbfff7eff0465 Cc: "Elwell, John" , Paul Kyzivat , sipping@ietf.org, Scott Lawrence X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org ...but 3xx wouldn't stop other devices from ringing To clarify: I was argueing for a 6xx response. In addition, the body could = have some HTML with links for alternative means of contact. Could be person= alized, etc IMHO there should not be too much complicated automatic handling in the pro= xies. Better to present the case to the caller, and let him decide Regards, Jeroen PS but perhaps a "3xx code with 6xx effect" (or vice versa) could help -----Oorspronkelijk bericht ----- Van: "Francois Audet" Aan: "Jeroen Van Bemmel" ; "Xiaotao Wu" CC: sipping@ietf.org; "Paul Kyzivat" ; "Elwell, John" <= john.elwell@siemens.com>; "Scott Lawrence" Verzonden: 3-8-06 20:12 Onderwerp: RE: [Sipping] Question on do not disturb indication If you provide an alternate address, then it should be a 3XX response. =20 > -----Original Message----- > From: Jeroen Van Bemmel [mailto:jbemmel@zonnet.nl]=20 > Sent: Thursday, August 03, 2006 5:06 AM > To: Xiaotao Wu; Audet, Francois (SC100:3055) > Cc: sipping@ietf.org; Paul Kyzivat; Elwell, John; Scott Lawrence > Subject: RE: [Sipping] Question on do not disturb indication >=20 > So, I think the problem is more on what's the expected code=20 > for the caller, not on the routing behavior of the request/response. >=20 > (JvB) The reason phrase is (and was always) intended for=20 > this. "480 Do not disturb (use IM, call my secretary or leave=20 > a voiemail)" would seem appropriate. The semantics are hard=20 > to convey to a machine, therefore ideally the choice for any=20 > follow ups should be made by the caller IMHO. A 6xx would=20 > guarantee that outcome >=20 > Regards, > Jeroen=20 >=20 >=20 >=20 _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Thu Aug 03 15:08:53 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8iYq-0007Og-I0; Thu, 03 Aug 2006 15:08:48 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8iYp-0007KQ-0q for sipping@ietf.org; Thu, 03 Aug 2006 15:08:47 -0400 Received: from web31504.mail.mud.yahoo.com ([68.142.198.133]) by ietf-mx.ietf.org with smtp (Exim 4.43) id 1G8iYn-0000IQ-K7 for sipping@ietf.org; Thu, 03 Aug 2006 15:08:47 -0400 Received: (qmail 83016 invoked by uid 60001); 3 Aug 2006 19:08:44 -0000 DomainKey-Signature: a=rsa-sha1; q=dns; c=nofws; s=s1024; d=yahoo.com; h=Message-ID:Received:Date:From:Subject:To:MIME-Version:Content-Type:Content-Transfer-Encoding; b=XVJayK6a9ip0YWZVik6s/WwyTfQLdaT39Y7XM4+Xp5lfgZMDHyKq8OxYvuVQKwIe3Xw/dzL+eQDD6cmW0uibXEiNlr0c0HHWBgqNwxrhV4Y/obLaJAe/B/xXfjL+YM/GNMkdleUZj0wBZK5/yZfxAc9JSGS/w4KDIP1aCTXTgK8= ; Message-ID: <20060803190844.83014.qmail@web31504.mail.mud.yahoo.com> Received: from [198.152.12.67] by web31504.mail.mud.yahoo.com via HTTP; Thu, 03 Aug 2006 20:08:44 BST Date: Thu, 3 Aug 2006 20:08:44 +0100 (BST) From: Kapil Nayar To: sipping@ietf.org MIME-Version: 1.0 Content-Type: text/plain; charset=iso-8859-1 Content-Transfer-Encoding: 8bit X-Spam-Score: 0.0 (/) X-Scan-Signature: 79899194edc4f33a41f49410777972f8 Subject: [Sipping] Clarification on RFC 3959 - early media X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Hi The RFC mentions the 183-session progress to have both the early-offer and the answer which is replied with the early-answer in PRACK. The RFC seems to imply that we DONOT need an answer (SDP) in the 200-OK. Is it correct? thanks, -kapil ___________________________________________________________ All new Yahoo! Mail "The new Interface is stunning in its simplicity and ease of use." - PC Magazine http://uk.docs.yahoo.com/nowyoucan.html _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Thu Aug 03 17:06:45 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8kOj-0003Cb-Ni; Thu, 03 Aug 2006 17:06:29 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8kOj-0003CW-56 for sipping@ietf.org; Thu, 03 Aug 2006 17:06:29 -0400 Received: from nylon.softarmor.com ([66.135.38.164]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G8kOh-0008Bo-Rl for sipping@ietf.org; Thu, 03 Aug 2006 17:06:29 -0400 Received: from [192.168.2.104] (sj-natpool-220.cisco.com [128.107.248.220]) (authenticated bits=0) by nylon.softarmor.com (8.13.1/8.13.1) with ESMTP id k73L84pN008107 (version=TLSv1/SSLv3 cipher=RC4-SHA bits=128 verify=NO); Thu, 3 Aug 2006 16:08:05 -0500 In-Reply-To: <50B1CBA96870A34799A506B2313F266709897F68@ntht201e.siemenscomms.co.uk> References: <50B1CBA96870A34799A506B2313F266709897F68@ntht201e.siemenscomms.co.uk> Mime-Version: 1.0 (Apple Message framework v752.2) Content-Type: text/plain; charset=US-ASCII; delsp=yes; format=flowed Message-Id: <1B5A1FDA-AA89-4B8E-807E-2F170054D828@softarmor.com> Content-Transfer-Encoding: 7bit From: Dean Willis Subject: Re: [Sipping] Question on do not disturb indication Date: Thu, 3 Aug 2006 16:05:54 -0500 To: "Elwell, John" X-Mailer: Apple Mail (2.752.2) X-Spam-Score: 0.0 (/) X-Scan-Signature: 4adaf050708fb13be3316a9eee889caa Cc: "Dolly, Martin C, ALABS" , Paul Kyzivat , sipping@ietf.org, Scott Lawrence X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org On Aug 3, 2006, at 1:28 AM, Elwell, John wrote: > Dean, > > So what would you like the IETF to recommend? Do you believe 480 > mostly > works, or do you think something new is needed? > > John Hey, I'm just a consumer. I don't want to know the technical details. I just want it to work, mostly. If it were up to me, I'd be asking the question: What part of DND is important for a calling human to understand, vs. a calling automata? Is there a distinction needed? If we believe that we'd like to convey a distinction between "Temporarily unavailable" and "Temporarily unavailable due to subscriber choice" to a human, perhaps a reason phrase on a 480 would be adequate? Like "480 Do Not Disturb", maybe. An automata is just likely to retry the call periodically anyhow, but a human might stop and think "Oh yeah, it IS the middle of the night in Singapore . . .". Question: How beneficial would a Retry-After value be in rate- limiting the retries of automata? Now, I will say that Martin is right in that there might be many reasons to reject a call, that DND is just the tip of the iceberg, and that there's no way we can define them all. Now, here's an interesting thought. If I use a reason phrase on 480, and couple it to the right software, I can use SIP INVITE requests as a covert-channel replacement for HTTP GET operations (or file transfer, or whatever) , but peer-to-peer using the SIP infrastructure as a rendezvous service. Don't try this at home, kids. -- Dean _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Thu Aug 03 21:07:49 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8oA9-0001k7-I6; Thu, 03 Aug 2006 21:07:41 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8oA8-0001g3-0m for sipping@ietf.org; Thu, 03 Aug 2006 21:07:40 -0400 Received: from nf-out-0910.google.com ([64.233.182.189]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G8oA6-0003OK-G8 for sipping@ietf.org; Thu, 03 Aug 2006 21:07:40 -0400 Received: by nf-out-0910.google.com with SMTP id q29so1218208nfc for ; Thu, 03 Aug 2006 18:07:37 -0700 (PDT) DomainKey-Signature: a=rsa-sha1; q=dns; c=nofws; s=beta; d=gmail.com; h=received:message-id:date:from:sender:to:subject:cc:in-reply-to:mime-version:content-type:content-transfer-encoding:content-disposition:references:x-google-sender-auth; b=kldqUGfRMbgzD+01q1j92JfC0L9YcfABILSmyR/NJKW6jwIM+1XUAdXvIF7k/pSdYvxhBK8ndvmE8dbVuaac0GiLGpYFqoy+06rzWifZCXgqcGeuc1On165xs40LlEME8oH1zi8379oDziBkJBos66h61I+2MsgO+M7k/Ntm4KA= Received: by 10.49.10.3 with SMTP id n3mr4707762nfi; Thu, 03 Aug 2006 18:07:37 -0700 (PDT) Received: by 10.48.164.7 with HTTP; Thu, 3 Aug 2006 18:07:37 -0700 (PDT) Message-ID: Date: Thu, 3 Aug 2006 21:07:37 -0400 From: "Xiaotao Wu" To: "Francois Audet" Subject: Re: [Sipping] Question on do not disturb indication In-Reply-To: <1ECE0EB50388174790F9694F77522CCF0BF08B8F@zrc2hxm0.corp.nortel.com> MIME-Version: 1.0 Content-Type: text/plain; charset=ISO-8859-1; format=flowed Content-Transfer-Encoding: 7bit Content-Disposition: inline References: <1ECE0EB50388174790F9694F77522CCF0BF08B8F@zrc2hxm0.corp.nortel.com> X-Google-Sender-Auth: 54a91509ff1ee201 X-Spam-Score: 0.0 (/) X-Scan-Signature: cd3fc8e909678b38737fc606dec187f0 Cc: "Elwell, John" , Paul Kyzivat , sipping@ietf.org, Scott Lawrence X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org On 8/3/06, Francois Audet wrote: > Sure you can do it on phones. > > You an have a button on the phone to "Decline" an incoming call > dynamically. > You can have a DND feature (either permanent or dynamic). Which entity to ebable the feature is different from which entity to execute the service logic. Sure you can enable it on the phone, but the service logic can still be executed on proxy servers. For example, Microsoft Office Communicator uses CSTA to transmit the "Set DND" command to a gateway, the "Set DND" is not executed on the Communicator itself. -Xiaotao > > This is widely done today. > > > -----Original Message----- > > From: xiaotaow@gmail.com [mailto:xiaotaow@gmail.com] On > > Behalf Of Xiaotao Wu > > Sent: Thursday, August 03, 2006 4:18 AM > > To: Audet, Francois (SC100:3055) > > Cc: Paul Kyzivat; Scott Lawrence; Elwell, John; sipping@ietf.org > > Subject: Re: [Sipping] Question on do not disturb indication > > > > On 8/2/06, Francois Audet wrote: > > > On the 603 issue, I'd like to disagree with everybody else... > > > Sometimes being able to decline the call everywhere is > > actually what > > > you want. > > > As an example, I have multiple forked device. If I send a 603, it > > > triggers the proxy to use whatever alternate routing is sees fit > > > (e.g., voicemail, attendant, etc.). That's the desired behavior. > > > > > > Problem with a 480/486 with that scenario is that it won't > > trigger the > > > alternate routing and will keep on ringing the other fork. That may > > > not be the desired behavior. > > > > I would expect the response should be sent by the proxy, > > instead by endpoints. If it's send by an endpoint, things may > > get complicated. > > For example, Bob has a desk phone, a cell phone, a voicemail, > > and a secretary. It's hard to compose a response from his > > desk phone to indicate that he does not want to ring his desk > > phone, and cell phone, but ring his secretary phone and voicemail. > > > > So, I think the problem is more on what's the expected code > > for the caller, not on the routing behavior of the request/response. > > > > -Xiaotao > > > > > > > > My 2 cents. > > > > > > > -----Original Message----- > > > > From: Paul Kyzivat [mailto:pkyzivat@cisco.com] > > > > Sent: Wednesday, August 02, 2006 3:31 PM > > > > To: Scott Lawrence > > > > Cc: sipping@ietf.org; Elwell, John > > > > Subject: Re: [Sipping] Question on do not disturb indication > > > > > > > > I pretty much agree with Scott. In many/most cases I > > won't want the > > > > caller to know why his call is going unanswered. > > > > Both 480 and 486 are in effect "white lies". Which one to use may > > > > depend on which impression you want to give to the > > caller. Another > > > > one - if you want to be more rude - is 403 Forbidden. > > > > > > > > I don't see a need for a *special* code that is unique to this > > > > service. > > > > > > > > Paul > > > > > > > > Scott Lawrence wrote: > > > > > On Wed, 2006-08-02 at 21:09 +0100, Elwell, John wrote: > > > > >> RFC 3261 suggests the use of response code 480 Temporarily > > > > >> Unavailable for indicating do not disturb (DND), but it is > > > > not used > > > > >> exclusively for this purpose. There are other documents (e.g., > > > > >> RFC > > > > >> 4504) that suggest the use of 486 Busy Here, but again this is > > > > >> not helpful if the receiver wants to distinguish between DND > > > > and ordinary > > > > >> busy. Presence can be used to indicate this condition > > through the > > > > >> RPID note element, but that does not say why a > > particular session > > > > >> establishment attempt has failed. What are the best > > practices for > > > > >> indicating the DND condition? Does anyone see a need for > > > > an explicit > > > > >> indicator for this purpose, e.g., a new SIP response code? > > > > > > > > > > Personally, I like 480 for this. > > > > > > > > > > If the phone wanted to be sneaky, it could just mute the > > > > ringing and > > > > > allow the transaction to time out without explicitly > > rejecting it. > > > > > That prevents the caller from being able to tell they are being > > > > > disregarded, but some users might prefer that forwarding > > > > take effect > > > > > sooner and not like that. > > > > > > > > > > The one thing that I _don't_ like to see is a 603 being > > > > used for this; > > > > > that messes up any upstream forking. > > > > > > > > > > > > > > > > > > _______________________________________________ > > > > Sipping mailing list > > https://www1.ietf.org/mailman/listinfo/sipping > > > > This list is for NEW development of the application of SIP Use > > > > sip-implementors@cs.columbia.edu for questions on current sip Use > > > > sip@ietf.org for new developments of core SIP > > > > > > > > > > _______________________________________________ > > > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > > > This list is for NEW development of the application of SIP Use > > > sip-implementors@cs.columbia.edu for questions on current sip Use > > > sip@ietf.org for new developments of core SIP > > > > > > > > > -- > > -Xiaotao > > > > =========================================================== > > Name : Xiaotao Wu > > Email : xwu@avaya.com, xiaotaow@gmail.com > > Web : http://www.cs.columbia.edu/~xiaotaow > > Phone : (732) 852-2133 > > SIP : sip:xiaotaow@cs.columbia.edu > > Address : 1M-240, 307 Middletown Lincroft Road > > Lincroft, NJ 07738-1526, U.S.A. > > > > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP > Use sip-implementors@cs.columbia.edu for questions on current sip > Use sip@ietf.org for new developments of core SIP > -- -Xiaotao =========================================================== Name : Xiaotao Wu Email : xwu@avaya.com, xiaotaow@gmail.com Web : http://www.cs.columbia.edu/~xiaotaow Phone : (732) 852-2133 SIP : sip:xiaotaow@cs.columbia.edu Address : 1M-240, 307 Middletown Lincroft Road Lincroft, NJ 07738-1526, U.S.A. _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Thu Aug 03 21:14:42 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8oGv-0008AJ-E0; Thu, 03 Aug 2006 21:14:41 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8oGt-00087g-FE for sipping@ietf.org; Thu, 03 Aug 2006 21:14:39 -0400 Received: from zrtps0kp.nortel.com ([47.140.192.56]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G8oGq-0005x1-W9 for sipping@ietf.org; Thu, 03 Aug 2006 21:14:39 -0400 Received: from zrc2hxm0.corp.nortel.com (zrc2hxm0.corp.nortel.com [47.103.123.71]) by zrtps0kp.nortel.com (Switch-2.2.6/Switch-2.2.0) with ESMTP id k741EXa17414; Thu, 3 Aug 2006 21:14:33 -0400 (EDT) X-MimeOLE: Produced By Microsoft Exchange V6.5.7226.0 Content-class: urn:content-classes:message MIME-Version: 1.0 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: quoted-printable Subject: RE: [Sipping] Question on do not disturb indication Date: Thu, 3 Aug 2006 20:14:32 -0500 Message-ID: <1ECE0EB50388174790F9694F77522CCF0BFA2EDD@zrc2hxm0.corp.nortel.com> In-Reply-To: X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: [Sipping] Question on do not disturb indication Thread-Index: Aca3Ymuzs66kmSXiSJOOEVuUKCOiNAAAJxEw From: "Francois Audet" To: "Xiaotao Wu" X-Spam-Score: 0.0 (/) X-Scan-Signature: 311e798ce51dbeacf5cdfcc8e9fda21b Cc: "Elwell, John" , Paul Kyzivat , sipping@ietf.org, Scott Lawrence X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Yeah, sure, but this is IETF (SIP), not ECMA (CSTA). But I agree with you that if you have some other protocol, you can do DND on the proxy instead of the endpoint. You could also have a Web Page that allows you to affect routing on the proxy in a similar way.=20 > -----Original Message----- > From: xiaotaow@gmail.com [mailto:xiaotaow@gmail.com] On=20 > Behalf Of Xiaotao Wu > Sent: Thursday, August 03, 2006 6:08 PM > To: Audet, Francois (SC100:3055) > Cc: sipping@ietf.org; Paul Kyzivat; Elwell, John; Scott Lawrence > Subject: Re: [Sipping] Question on do not disturb indication >=20 > On 8/3/06, Francois Audet wrote: > > Sure you can do it on phones. > > > > You an have a button on the phone to "Decline" an incoming call=20 > > dynamically. > > You can have a DND feature (either permanent or dynamic). >=20 > Which entity to ebable the feature is different from which=20 > entity to execute the service logic. Sure you can enable it=20 > on the phone, but the service logic can still be executed on=20 > proxy servers. For example, Microsoft Office Communicator=20 > uses CSTA to transmit the "Set DND" > command to a gateway, the "Set DND" is not executed on the=20 > Communicator itself. >=20 > -Xiaotao >=20 > > > > This is widely done today. > > > > > -----Original Message----- > > > From: xiaotaow@gmail.com [mailto:xiaotaow@gmail.com] On Behalf Of=20 > > > Xiaotao Wu > > > Sent: Thursday, August 03, 2006 4:18 AM > > > To: Audet, Francois (SC100:3055) > > > Cc: Paul Kyzivat; Scott Lawrence; Elwell, John; sipping@ietf.org > > > Subject: Re: [Sipping] Question on do not disturb indication > > > > > > On 8/2/06, Francois Audet wrote: > > > > On the 603 issue, I'd like to disagree with everybody else... > > > > Sometimes being able to decline the call everywhere is > > > actually what > > > > you want. > > > > As an example, I have multiple forked device. If I send=20 > a 603, it=20 > > > > triggers the proxy to use whatever alternate routing is=20 > sees fit=20 > > > > (e.g., voicemail, attendant, etc.). That's the desired behavior. > > > > > > > > Problem with a 480/486 with that scenario is that it won't > > > trigger the > > > > alternate routing and will keep on ringing the other fork. That=20 > > > > may not be the desired behavior. > > > > > > I would expect the response should be sent by the proxy,=20 > instead by=20 > > > endpoints. If it's send by an endpoint, things may get=20 > complicated. > > > For example, Bob has a desk phone, a cell phone, a=20 > voicemail, and a=20 > > > secretary. It's hard to compose a response from his desk phone to=20 > > > indicate that he does not want to ring his desk phone, and cell=20 > > > phone, but ring his secretary phone and voicemail. > > > > > > So, I think the problem is more on what's the expected=20 > code for the=20 > > > caller, not on the routing behavior of the request/response. > > > > > > -Xiaotao > > > > > > > > > > > My 2 cents. > > > > > > > > > -----Original Message----- > > > > > From: Paul Kyzivat [mailto:pkyzivat@cisco.com] > > > > > Sent: Wednesday, August 02, 2006 3:31 PM > > > > > To: Scott Lawrence > > > > > Cc: sipping@ietf.org; Elwell, John > > > > > Subject: Re: [Sipping] Question on do not disturb indication > > > > > > > > > > I pretty much agree with Scott. In many/most cases I > > > won't want the > > > > > caller to know why his call is going unanswered. > > > > > Both 480 and 486 are in effect "white lies". Which one to use=20 > > > > > may depend on which impression you want to give to the > > > caller. Another > > > > > one - if you want to be more rude - is 403 Forbidden. > > > > > > > > > > I don't see a need for a *special* code that is=20 > unique to this=20 > > > > > service. > > > > > > > > > > Paul > > > > > > > > > > Scott Lawrence wrote: > > > > > > On Wed, 2006-08-02 at 21:09 +0100, Elwell, John wrote: > > > > > >> RFC 3261 suggests the use of response code 480 Temporarily=20 > > > > > >> Unavailable for indicating do not disturb (DND), but it is > > > > > not used > > > > > >> exclusively for this purpose. There are other documents=20 > > > > > >> (e.g., RFC > > > > > >> 4504) that suggest the use of 486 Busy Here, but=20 > again this=20 > > > > > >> is not helpful if the receiver wants to=20 > distinguish between=20 > > > > > >> DND > > > > > and ordinary > > > > > >> busy. Presence can be used to indicate this condition > > > through the > > > > > >> RPID note element, but that does not say why a > > > particular session > > > > > >> establishment attempt has failed. What are the best > > > practices for > > > > > >> indicating the DND condition? Does anyone see a need for > > > > > an explicit > > > > > >> indicator for this purpose, e.g., a new SIP response code? > > > > > > > > > > > > Personally, I like 480 for this. > > > > > > > > > > > > If the phone wanted to be sneaky, it could just mute the > > > > > ringing and > > > > > > allow the transaction to time out without explicitly > > > rejecting it. > > > > > > That prevents the caller from being able to tell they are=20 > > > > > > being disregarded, but some users might prefer that=20 > forwarding > > > > > take effect > > > > > > sooner and not like that. > > > > > > > > > > > > The one thing that I _don't_ like to see is a 603 being > > > > > used for this; > > > > > > that messes up any upstream forking. > > > > > > > > > > > > > > > > > > > > > > _______________________________________________ > > > > > Sipping mailing list > > > https://www1.ietf.org/mailman/listinfo/sipping > > > > > This list is for NEW development of the application=20 > of SIP Use=20 > > > > > sip-implementors@cs.columbia.edu for questions on current sip=20 > > > > > Use sip@ietf.org for new developments of core SIP > > > > > > > > > > > > > _______________________________________________ > > > > Sipping mailing list =20 > > > > https://www1.ietf.org/mailman/listinfo/sipping > > > > This list is for NEW development of the application of SIP Use=20 > > > > sip-implementors@cs.columbia.edu for questions on=20 > current sip Use=20 > > > > sip@ietf.org for new developments of core SIP > > > > > > > > > > > > > -- > > > -Xiaotao > > > > > > = =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D= =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D= =3D=3D=3D=3D=3D=3D=3D=3D=3D > > > Name : Xiaotao Wu > > > Email : xwu@avaya.com, xiaotaow@gmail.com > > > Web : http://www.cs.columbia.edu/~xiaotaow > > > Phone : (732) 852-2133 > > > SIP : sip:xiaotaow@cs.columbia.edu > > > Address : 1M-240, 307 Middletown Lincroft Road > > > Lincroft, NJ 07738-1526, U.S.A. > > > > > > > _______________________________________________ > > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > > This list is for NEW development of the application of SIP Use=20 > > sip-implementors@cs.columbia.edu for questions on current sip Use=20 > > sip@ietf.org for new developments of core SIP > > >=20 >=20 > -- > -Xiaotao >=20 > = =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D= =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D= =3D=3D=3D=3D=3D=3D=3D=3D=3D > Name : Xiaotao Wu > Email : xwu@avaya.com, xiaotaow@gmail.com > Web : http://www.cs.columbia.edu/~xiaotaow > Phone : (732) 852-2133 > SIP : sip:xiaotaow@cs.columbia.edu > Address : 1M-240, 307 Middletown Lincroft Road > Lincroft, NJ 07738-1526, U.S.A. >=20 _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Thu Aug 03 22:15:54 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8pE4-0004Jm-DQ; Thu, 03 Aug 2006 22:15:48 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8pE2-0004Ce-NR for sipping@ietf.org; Thu, 03 Aug 2006 22:15:46 -0400 Received: from sccrmhc13.comcast.net ([204.127.200.83]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G8pE1-0006Lo-Hb for sipping@ietf.org; Thu, 03 Aug 2006 22:15:46 -0400 Received: from dragon.ariadne.com (dworley.hsd1.ma.comcast.net[24.34.79.42]) by comcast.net (sccrmhc13) with ESMTP id <2006080402154501300a7v0le>; Fri, 4 Aug 2006 02:15:45 +0000 Received: from dragon.ariadne.com (dragon.ariadne.com [127.0.0.1]) by dragon.ariadne.com (8.12.8/8.12.8) with ESMTP id k742FiZW025966 for ; Thu, 3 Aug 2006 22:15:44 -0400 Received: (from worley@localhost) by dragon.ariadne.com (8.12.8/8.12.8/Submit) id k742FicH025962; Thu, 3 Aug 2006 22:15:44 -0400 Date: Thu, 3 Aug 2006 22:15:44 -0400 Message-Id: <200608040215.k742FicH025962@dragon.ariadne.com> To: sipping@ietf.org From: Dale.Worley@comcast.net In-reply-to: <1ECE0EB50388174790F9694F77522CCF0BF08BBC@zrc2hxm0.corp.nortel.com> (audet@nortel.com) Subject: Re: [Sipping] Question on do not disturb indication References: <1ECE0EB50388174790F9694F77522CCF0BF08BBC@zrc2hxm0.corp.nortel.com> X-Spam-Score: 0.2 (/) X-Scan-Signature: cf4fa59384e76e63313391b70cd0dd25 X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org From: "Francois Audet" But I don't think RFC 3261 necessarily needs to be amended. It's always been legal for a proxy to retarget however it pleases. I beg to differ -- RFC 3261 is unpleasantly clear on the effect of 6xx responses. Handling *other* responses leaves the proxy with a lot of flexibility in deciding the "target set", but for 6xx, the target set is effectively emptied. Though I see no reason not to change that definition -- I see the current definition of 6xx as unusable due to evil interaction with any system that uses "interesting" retargeting, so "there are no current uses of the feature". Dale _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Thu Aug 03 22:29:38 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8pRH-00073N-CK; Thu, 03 Aug 2006 22:29:27 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8pRG-00072p-57 for sipping@ietf.org; Thu, 03 Aug 2006 22:29:26 -0400 Received: from sccrmhc12.comcast.net ([204.127.200.82]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G8pRE-0001DI-VG for sipping@ietf.org; Thu, 03 Aug 2006 22:29:26 -0400 Received: from dragon.ariadne.com (dworley.hsd1.ma.comcast.net[24.34.79.42]) by comcast.net (sccrmhc12) with ESMTP id <2006080402292401200skc4oe>; Fri, 4 Aug 2006 02:29:24 +0000 Received: from dragon.ariadne.com (dragon.ariadne.com [127.0.0.1]) by dragon.ariadne.com (8.12.8/8.12.8) with ESMTP id k742TOZW026445 for ; Thu, 3 Aug 2006 22:29:24 -0400 Received: (from worley@localhost) by dragon.ariadne.com (8.12.8/8.12.8/Submit) id k742TOU7026441; Thu, 3 Aug 2006 22:29:24 -0400 Date: Thu, 3 Aug 2006 22:29:24 -0400 Message-Id: <200608040229.k742TOU7026441@dragon.ariadne.com> To: sipping@ietf.org From: Dale.Worley@comcast.net In-reply-to: <1154619816.2859.68.camel@sukothai.pingtel.com> (slawrence@pingtel.com) Subject: Re: [Sipping] Question on do not disturb indication References: <6FC4416DDE56C44DA0AEE67BC7CA43710FBDA37E@zrc2hxm2.corp.nortel.com> <1154619816.2859.68.camel@sukothai.pingtel.com> X-Spam-Score: 0.2 (/) X-Scan-Signature: 7655788c23eb79e336f5f8ba8bce7906 X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org From: Scott Lawrence I recognize the utility of the 'target set' concept, but I don't know of any standard way to express it in the existing protocol. Do you? A "nested target set" concept would work in some cases -- The 6xx would be qualified (implicitly or explicitly) by a target set identifier, and the response would be cascaded back through the forking proxies until it reached the one that was the point-of-first-forking for that target set. The limitation would be that the UA would have to know that the request had been routed to it through that particular target set before 6xx-ing it with the scope of that particular target set. In the PSTN this is more likely to be possible, as there are a much smaller number of forking proxies (switches), and a particular UA (phone) is usually fed from only one of them, which handles all the target sets that could ever route to that UA. Things can be "more general" (messier) in SIP. Dale _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Fri Aug 04 07:34:03 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8xwA-0003w2-0A; Fri, 04 Aug 2006 07:33:54 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8xw7-0003vx-Ty for sipping@ietf.org; Fri, 04 Aug 2006 07:33:51 -0400 Received: from jes-fe2.zx.nl ([194.187.76.133]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G8xw6-00017E-Jg for sipping@ietf.org; Fri, 04 Aug 2006 07:33:51 -0400 Received: from Inbox ([89.205.141.30]) by jes-fe2.zx.nl (Sun Java System Messaging Server 6.2-2.05 (built Apr 28 2005)) with ESMTP id <0J3H00II44FOG7A0@jes-fe2.zx.nl> for sipping@ietf.org; Fri, 04 Aug 2006 13:38:23 +0100 (WEST) Date: Fri, 04 Aug 2006 13:33:48 +0200 From: Jeroen Van Bemmel Subject: RE: [Sipping] Question on do not disturb indication To: Dale.Worley@comcast.net, sipping@ietf.org Message-id: <0J3H00II54FQG7A0@jes-fe2.zx.nl> MIME-version: 1.0 Content-type: text/plain; charset=iso-8859-1 Content-transfer-encoding: quoted-printable Importance: normal X-Priority: 3 X-Spam-Score: 1.2 (+) X-Scan-Signature: 39bd8f8cbb76cae18b7e23f7cf6b2b9f Cc: X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org It occurred to me that perhaps the best response would be 200, immediately = followed by BYE. That will take care of the "retargeting people" who interp= ret 6xx as "only CANCEL this set" (and then retarget to voicemail) Regards, Jeroen -----Oorspronkelijk bericht ----- Van: Dale.Worley@comcast.net Aan: sipping@ietf.org Verzonden: 4-8-06 04:15 Onderwerp: Re: [Sipping] Question on do not disturb indication From: "Francois Audet" But I don't think RFC 3261 necessarily needs to be amended. It's always been legal for a proxy to retarget however it pleases. I beg to differ -- RFC 3261 is unpleasantly clear on the effect of 6xx responses. Handling *other* responses leaves the proxy with a lot of flexibility in deciding the "target set", but for 6xx, the target set is effectively emptied. Though I see no reason not to change that definition -- I see the current definition of 6xx as unusable due to evil interaction with any system that uses "interesting" retargeting, so "there are no current uses of the feature". Dale _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Fri Aug 04 07:34:03 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8xwG-0003y0-LL; Fri, 04 Aug 2006 07:34:00 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8xwF-0003xV-6K for sipping@ietf.org; Fri, 04 Aug 2006 07:33:59 -0400 Received: from nf-out-0910.google.com ([64.233.182.191]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G8xwD-00018D-BH for sipping@ietf.org; Fri, 04 Aug 2006 07:33:59 -0400 Received: by nf-out-0910.google.com with SMTP id q29so1382743nfc for ; Fri, 04 Aug 2006 04:33:56 -0700 (PDT) DomainKey-Signature: a=rsa-sha1; q=dns; c=nofws; s=beta; d=gmail.com; h=received:message-id:date:from:sender:to:subject:cc:in-reply-to:mime-version:content-type:content-transfer-encoding:content-disposition:references:x-google-sender-auth; b=DG0XRpaJbSxQGVyrNvgZKxlZoViJW+/ZvX2VuuNBzV+ZWFf3Btclzq9tZqsr0yYbjNYK2pc/Z4/IVdjhW41XHK2d2gIlu3BoaBRmcS2nxajq7x7EujQN4zLh70UMo6iSXXgsga71oi/rYutGjc3+N3n3e6TxtgV//BcgFrRnoC0= Received: by 10.49.19.18 with SMTP id w18mr5106693nfi; Fri, 04 Aug 2006 04:33:56 -0700 (PDT) Received: by 10.48.164.7 with HTTP; Fri, 4 Aug 2006 04:33:56 -0700 (PDT) Message-ID: Date: Fri, 4 Aug 2006 07:33:56 -0400 From: "Xiaotao Wu" To: "Francois Audet" Subject: Re: [Sipping] Question on do not disturb indication In-Reply-To: <1ECE0EB50388174790F9694F77522CCF0BFA2EDD@zrc2hxm0.corp.nortel.com> MIME-Version: 1.0 Content-Type: text/plain; charset=ISO-8859-1; format=flowed Content-Transfer-Encoding: 7bit Content-Disposition: inline References: <1ECE0EB50388174790F9694F77522CCF0BFA2EDD@zrc2hxm0.corp.nortel.com> X-Google-Sender-Auth: 25586ed8bbf87963 X-Spam-Score: 0.0 (/) X-Scan-Signature: 8a85b14f27c9dcbe0719e27d46abc1f8 Cc: sipping@ietf.org, Paul Kyzivat , "Elwell, John" , Scott Lawrence X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org On 8/3/06, Francois Audet wrote: > Yeah, sure, but this is IETF (SIP), not ECMA (CSTA). CSTA is just one example in the existing application of doing this. HTTP (as you said), SOAP can all be used to do this. It's not about whether it's IETF or ECMA. It's about which place we should put the service at. Certainly you can handle it on an endpoints, but you then need to make this endpoint be able to control the routing to other endpoints. That should not be the task of endpoings (well, maybe in a untrusted P2P environment, this is useful). So, I think performing DND on a proxy server (if there is, and maybe with the help of CPL or something like) is more appropriate than performing the service on endpoints. -Xiaotao > > But I agree with you that if you have some other protocol, you can do > DND > on the proxy instead of the endpoint. > > You could also have a Web Page that allows you to affect routing on the > proxy > in a similar way. > > > -----Original Message----- > > From: xiaotaow@gmail.com [mailto:xiaotaow@gmail.com] On > > Behalf Of Xiaotao Wu > > Sent: Thursday, August 03, 2006 6:08 PM > > To: Audet, Francois (SC100:3055) > > Cc: sipping@ietf.org; Paul Kyzivat; Elwell, John; Scott Lawrence > > Subject: Re: [Sipping] Question on do not disturb indication > > > > On 8/3/06, Francois Audet wrote: > > > Sure you can do it on phones. > > > > > > You an have a button on the phone to "Decline" an incoming call > > > dynamically. > > > You can have a DND feature (either permanent or dynamic). > > > > Which entity to ebable the feature is different from which > > entity to execute the service logic. Sure you can enable it > > on the phone, but the service logic can still be executed on > > proxy servers. For example, Microsoft Office Communicator > > uses CSTA to transmit the "Set DND" > > command to a gateway, the "Set DND" is not executed on the > > Communicator itself. > > > > -Xiaotao > > > > > > > > This is widely done today. > > > > > > > -----Original Message----- > > > > From: xiaotaow@gmail.com [mailto:xiaotaow@gmail.com] On Behalf Of > > > > Xiaotao Wu > > > > Sent: Thursday, August 03, 2006 4:18 AM > > > > To: Audet, Francois (SC100:3055) > > > > Cc: Paul Kyzivat; Scott Lawrence; Elwell, John; sipping@ietf.org > > > > Subject: Re: [Sipping] Question on do not disturb indication > > > > > > > > On 8/2/06, Francois Audet wrote: > > > > > On the 603 issue, I'd like to disagree with everybody else... > > > > > Sometimes being able to decline the call everywhere is > > > > actually what > > > > > you want. > > > > > As an example, I have multiple forked device. If I send > > a 603, it > > > > > triggers the proxy to use whatever alternate routing is > > sees fit > > > > > (e.g., voicemail, attendant, etc.). That's the desired behavior. > > > > > > > > > > Problem with a 480/486 with that scenario is that it won't > > > > trigger the > > > > > alternate routing and will keep on ringing the other fork. That > > > > > may not be the desired behavior. > > > > > > > > I would expect the response should be sent by the proxy, > > instead by > > > > endpoints. If it's send by an endpoint, things may get > > complicated. > > > > For example, Bob has a desk phone, a cell phone, a > > voicemail, and a > > > > secretary. It's hard to compose a response from his desk phone to > > > > indicate that he does not want to ring his desk phone, and cell > > > > phone, but ring his secretary phone and voicemail. > > > > > > > > So, I think the problem is more on what's the expected > > code for the > > > > caller, not on the routing behavior of the request/response. > > > > > > > > -Xiaotao > > > > > > > > > > > > > > My 2 cents. > > > > > > > > > > > -----Original Message----- > > > > > > From: Paul Kyzivat [mailto:pkyzivat@cisco.com] > > > > > > Sent: Wednesday, August 02, 2006 3:31 PM > > > > > > To: Scott Lawrence > > > > > > Cc: sipping@ietf.org; Elwell, John > > > > > > Subject: Re: [Sipping] Question on do not disturb indication > > > > > > > > > > > > I pretty much agree with Scott. In many/most cases I > > > > won't want the > > > > > > caller to know why his call is going unanswered. > > > > > > Both 480 and 486 are in effect "white lies". Which one to use > > > > > > may depend on which impression you want to give to the > > > > caller. Another > > > > > > one - if you want to be more rude - is 403 Forbidden. > > > > > > > > > > > > I don't see a need for a *special* code that is > > unique to this > > > > > > service. > > > > > > > > > > > > Paul > > > > > > > > > > > > Scott Lawrence wrote: > > > > > > > On Wed, 2006-08-02 at 21:09 +0100, Elwell, John wrote: > > > > > > >> RFC 3261 suggests the use of response code 480 Temporarily > > > > > > >> Unavailable for indicating do not disturb (DND), but it is > > > > > > not used > > > > > > >> exclusively for this purpose. There are other documents > > > > > > >> (e.g., RFC > > > > > > >> 4504) that suggest the use of 486 Busy Here, but > > again this > > > > > > >> is not helpful if the receiver wants to > > distinguish between > > > > > > >> DND > > > > > > and ordinary > > > > > > >> busy. Presence can be used to indicate this condition > > > > through the > > > > > > >> RPID note element, but that does not say why a > > > > particular session > > > > > > >> establishment attempt has failed. What are the best > > > > practices for > > > > > > >> indicating the DND condition? Does anyone see a need for > > > > > > an explicit > > > > > > >> indicator for this purpose, e.g., a new SIP response code? > > > > > > > > > > > > > > Personally, I like 480 for this. > > > > > > > > > > > > > > If the phone wanted to be sneaky, it could just mute the > > > > > > ringing and > > > > > > > allow the transaction to time out without explicitly > > > > rejecting it. > > > > > > > That prevents the caller from being able to tell they are > > > > > > > being disregarded, but some users might prefer that > > forwarding > > > > > > take effect > > > > > > > sooner and not like that. > > > > > > > > > > > > > > The one thing that I _don't_ like to see is a 603 being > > > > > > used for this; > > > > > > > that messes up any upstream forking. > > > > > > > > > > > > > > > > > > > > > > > > > > _______________________________________________ > > > > > > Sipping mailing list > > > > https://www1.ietf.org/mailman/listinfo/sipping > > > > > > This list is for NEW development of the application > > of SIP Use > > > > > > sip-implementors@cs.columbia.edu for questions on current sip > > > > > > Use sip@ietf.org for new developments of core SIP > > > > > > > > > > > > > > > > _______________________________________________ > > > > > Sipping mailing list > > > > > https://www1.ietf.org/mailman/listinfo/sipping > > > > > This list is for NEW development of the application of SIP Use > > > > > sip-implementors@cs.columbia.edu for questions on > > current sip Use > > > > > sip@ietf.org for new developments of core SIP > > > > > > > > > > > > > > > > > -- > > > > -Xiaotao > > > > > > > > =========================================================== > > > > Name : Xiaotao Wu > > > > Email : xwu@avaya.com, xiaotaow@gmail.com > > > > Web : http://www.cs.columbia.edu/~xiaotaow > > > > Phone : (732) 852-2133 > > > > SIP : sip:xiaotaow@cs.columbia.edu > > > > Address : 1M-240, 307 Middletown Lincroft Road > > > > Lincroft, NJ 07738-1526, U.S.A. > > > > > > > > > > _______________________________________________ > > > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > > > This list is for NEW development of the application of SIP Use > > > sip-implementors@cs.columbia.edu for questions on current sip Use > > > sip@ietf.org for new developments of core SIP > > > > > > > > > -- > > -Xiaotao > > > > =========================================================== > > Name : Xiaotao Wu > > Email : xwu@avaya.com, xiaotaow@gmail.com > > Web : http://www.cs.columbia.edu/~xiaotaow > > Phone : (732) 852-2133 > > SIP : sip:xiaotaow@cs.columbia.edu > > Address : 1M-240, 307 Middletown Lincroft Road > > Lincroft, NJ 07738-1526, U.S.A. > > > > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP > Use sip-implementors@cs.columbia.edu for questions on current sip > Use sip@ietf.org for new developments of core SIP > -- -Xiaotao =========================================================== Name : Xiaotao Wu Email : xwu@avaya.com, xiaotaow@gmail.com Web : http://www.cs.columbia.edu/~xiaotaow Phone : (732) 852-2133 SIP : sip:xiaotaow@cs.columbia.edu Address : 1M-240, 307 Middletown Lincroft Road Lincroft, NJ 07738-1526, U.S.A. _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Fri Aug 04 07:39:10 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8y1G-000474-CN; Fri, 04 Aug 2006 07:39:10 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8y1F-00046z-7J for sipping@ietf.org; Fri, 04 Aug 2006 07:39:09 -0400 Received: from nf-out-0910.google.com ([64.233.182.189]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G8y1C-0001Ui-IX for sipping@ietf.org; Fri, 04 Aug 2006 07:39:09 -0400 Received: by nf-out-0910.google.com with SMTP id q29so1384275nfc for ; Fri, 04 Aug 2006 04:39:05 -0700 (PDT) DomainKey-Signature: a=rsa-sha1; q=dns; c=nofws; s=beta; d=gmail.com; h=received:message-id:date:from:sender:to:subject:cc:in-reply-to:mime-version:content-type:content-transfer-encoding:content-disposition:references:x-google-sender-auth; b=M2s7ZAD6Y/ApWPqTbadkvAAA+ppgVuDMpp0oJE44g8Eqw4164VoBsPoAEp2suQd2fOPQ9KFW24VpNbBsd+vGOfmvitJnGTjQurLEmWCVcW4z6WoaOwHdNxc+qCEXQ3Zw7uD5E+efadrmDpgSCRsoBUje9tFjedZ+r1Iv5kM5Re0= Received: by 10.49.93.13 with SMTP id v13mr3808118nfl; Fri, 04 Aug 2006 04:39:05 -0700 (PDT) Received: by 10.48.164.7 with HTTP; Fri, 4 Aug 2006 04:39:05 -0700 (PDT) Message-ID: Date: Fri, 4 Aug 2006 07:39:05 -0400 From: "Xiaotao Wu" To: "Francois Audet" Subject: Re: [Sipping] Question on do not disturb indication In-Reply-To: MIME-Version: 1.0 Content-Type: text/plain; charset=ISO-8859-1; format=flowed Content-Transfer-Encoding: 7bit Content-Disposition: inline References: <1ECE0EB50388174790F9694F77522CCF0BFA2EDD@zrc2hxm0.corp.nortel.com> X-Google-Sender-Auth: 57529fae5efc05f2 X-Spam-Score: 0.0 (/) X-Scan-Signature: 2a9ffb6f997442a3b543bcdaf483b990 Cc: sipping@ietf.org, Paul Kyzivat , "Elwell, John" , Scott Lawrence X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org On 8/4/06, Xiaotao Wu wrote: > On 8/3/06, Francois Audet wrote: > > Yeah, sure, but this is IETF (SIP), not ECMA (CSTA). > > CSTA is just one example in the existing application of doing this. > HTTP (as you said), SOAP can all be used to do this. It's not about > whether it's IETF or ECMA. > It's about which place we should put the service at. Certainly you > can handle it on an endpoints, but you then need to make this endpoint > be able to control the routing to other endpoints. That should not be the task > of endpoings (well, maybe in a untrusted P2P environment, this is useful). > So, I think performing DND on a proxy server (if there is, and maybe > with the help of CPL > or something like) is more appropriate than performing the service on > endpoints. I think 480 is the right code we should use. It clearly convey the DND information to the caller. If we really want to perform the service on an endpoint, we need to define a way to express the user's prefered routing, but still use 480 as the response code. -Xiaotao > > -Xiaotao > > > > > But I agree with you that if you have some other protocol, you can do > > DND > > on the proxy instead of the endpoint. > > > > You could also have a Web Page that allows you to affect routing on the > > proxy > > in a similar way. > > > > > -----Original Message----- > > > From: xiaotaow@gmail.com [mailto:xiaotaow@gmail.com] On > > > Behalf Of Xiaotao Wu > > > Sent: Thursday, August 03, 2006 6:08 PM > > > To: Audet, Francois (SC100:3055) > > > Cc: sipping@ietf.org; Paul Kyzivat; Elwell, John; Scott Lawrence > > > Subject: Re: [Sipping] Question on do not disturb indication > > > > > > On 8/3/06, Francois Audet wrote: > > > > Sure you can do it on phones. > > > > > > > > You an have a button on the phone to "Decline" an incoming call > > > > dynamically. > > > > You can have a DND feature (either permanent or dynamic). > > > > > > Which entity to ebable the feature is different from which > > > entity to execute the service logic. Sure you can enable it > > > on the phone, but the service logic can still be executed on > > > proxy servers. For example, Microsoft Office Communicator > > > uses CSTA to transmit the "Set DND" > > > command to a gateway, the "Set DND" is not executed on the > > > Communicator itself. > > > > > > -Xiaotao > > > > > > > > > > > This is widely done today. > > > > > > > > > -----Original Message----- > > > > > From: xiaotaow@gmail.com [mailto:xiaotaow@gmail.com] On Behalf Of > > > > > Xiaotao Wu > > > > > Sent: Thursday, August 03, 2006 4:18 AM > > > > > To: Audet, Francois (SC100:3055) > > > > > Cc: Paul Kyzivat; Scott Lawrence; Elwell, John; sipping@ietf.org > > > > > Subject: Re: [Sipping] Question on do not disturb indication > > > > > > > > > > On 8/2/06, Francois Audet wrote: > > > > > > On the 603 issue, I'd like to disagree with everybody else... > > > > > > Sometimes being able to decline the call everywhere is > > > > > actually what > > > > > > you want. > > > > > > As an example, I have multiple forked device. If I send > > > a 603, it > > > > > > triggers the proxy to use whatever alternate routing is > > > sees fit > > > > > > (e.g., voicemail, attendant, etc.). That's the desired behavior. > > > > > > > > > > > > Problem with a 480/486 with that scenario is that it won't > > > > > trigger the > > > > > > alternate routing and will keep on ringing the other fork. That > > > > > > may not be the desired behavior. > > > > > > > > > > I would expect the response should be sent by the proxy, > > > instead by > > > > > endpoints. If it's send by an endpoint, things may get > > > complicated. > > > > > For example, Bob has a desk phone, a cell phone, a > > > voicemail, and a > > > > > secretary. It's hard to compose a response from his desk phone to > > > > > indicate that he does not want to ring his desk phone, and cell > > > > > phone, but ring his secretary phone and voicemail. > > > > > > > > > > So, I think the problem is more on what's the expected > > > code for the > > > > > caller, not on the routing behavior of the request/response. > > > > > > > > > > -Xiaotao > > > > > > > > > > > > > > > > > My 2 cents. > > > > > > > > > > > > > -----Original Message----- > > > > > > > From: Paul Kyzivat [mailto:pkyzivat@cisco.com] > > > > > > > Sent: Wednesday, August 02, 2006 3:31 PM > > > > > > > To: Scott Lawrence > > > > > > > Cc: sipping@ietf.org; Elwell, John > > > > > > > Subject: Re: [Sipping] Question on do not disturb indication > > > > > > > > > > > > > > I pretty much agree with Scott. In many/most cases I > > > > > won't want the > > > > > > > caller to know why his call is going unanswered. > > > > > > > Both 480 and 486 are in effect "white lies". Which one to use > > > > > > > may depend on which impression you want to give to the > > > > > caller. Another > > > > > > > one - if you want to be more rude - is 403 Forbidden. > > > > > > > > > > > > > > I don't see a need for a *special* code that is > > > unique to this > > > > > > > service. > > > > > > > > > > > > > > Paul > > > > > > > > > > > > > > Scott Lawrence wrote: > > > > > > > > On Wed, 2006-08-02 at 21:09 +0100, Elwell, John wrote: > > > > > > > >> RFC 3261 suggests the use of response code 480 Temporarily > > > > > > > >> Unavailable for indicating do not disturb (DND), but it is > > > > > > > not used > > > > > > > >> exclusively for this purpose. There are other documents > > > > > > > >> (e.g., RFC > > > > > > > >> 4504) that suggest the use of 486 Busy Here, but > > > again this > > > > > > > >> is not helpful if the receiver wants to > > > distinguish between > > > > > > > >> DND > > > > > > > and ordinary > > > > > > > >> busy. Presence can be used to indicate this condition > > > > > through the > > > > > > > >> RPID note element, but that does not say why a > > > > > particular session > > > > > > > >> establishment attempt has failed. What are the best > > > > > practices for > > > > > > > >> indicating the DND condition? Does anyone see a need for > > > > > > > an explicit > > > > > > > >> indicator for this purpose, e.g., a new SIP response code? > > > > > > > > > > > > > > > > Personally, I like 480 for this. > > > > > > > > > > > > > > > > If the phone wanted to be sneaky, it could just mute the > > > > > > > ringing and > > > > > > > > allow the transaction to time out without explicitly > > > > > rejecting it. > > > > > > > > That prevents the caller from being able to tell they are > > > > > > > > being disregarded, but some users might prefer that > > > forwarding > > > > > > > take effect > > > > > > > > sooner and not like that. > > > > > > > > > > > > > > > > The one thing that I _don't_ like to see is a 603 being > > > > > > > used for this; > > > > > > > > that messes up any upstream forking. > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > _______________________________________________ > > > > > > > Sipping mailing list > > > > > https://www1.ietf.org/mailman/listinfo/sipping > > > > > > > This list is for NEW development of the application > > > of SIP Use > > > > > > > sip-implementors@cs.columbia.edu for questions on current sip > > > > > > > Use sip@ietf.org for new developments of core SIP > > > > > > > > > > > > > > > > > > > _______________________________________________ > > > > > > Sipping mailing list > > > > > > https://www1.ietf.org/mailman/listinfo/sipping > > > > > > This list is for NEW development of the application of SIP Use > > > > > > sip-implementors@cs.columbia.edu for questions on > > > current sip Use > > > > > > sip@ietf.org for new developments of core SIP > > > > > > > > > > > > > > > > > > > > > -- > > > > > -Xiaotao > > > > > > > > > > =========================================================== > > > > > Name : Xiaotao Wu > > > > > Email : xwu@avaya.com, xiaotaow@gmail.com > > > > > Web : http://www.cs.columbia.edu/~xiaotaow > > > > > Phone : (732) 852-2133 > > > > > SIP : sip:xiaotaow@cs.columbia.edu > > > > > Address : 1M-240, 307 Middletown Lincroft Road > > > > > Lincroft, NJ 07738-1526, U.S.A. > > > > > > > > > > > > > _______________________________________________ > > > > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > > > > This list is for NEW development of the application of SIP Use > > > > sip-implementors@cs.columbia.edu for questions on current sip Use > > > > sip@ietf.org for new developments of core SIP > > > > > > > > > > > > > -- > > > -Xiaotao > > > > > > =========================================================== > > > Name : Xiaotao Wu > > > Email : xwu@avaya.com, xiaotaow@gmail.com > > > Web : http://www.cs.columbia.edu/~xiaotaow > > > Phone : (732) 852-2133 > > > SIP : sip:xiaotaow@cs.columbia.edu > > > Address : 1M-240, 307 Middletown Lincroft Road > > > Lincroft, NJ 07738-1526, U.S.A. > > > > > > > _______________________________________________ > > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > > This list is for NEW development of the application of SIP > > Use sip-implementors@cs.columbia.edu for questions on current sip > > Use sip@ietf.org for new developments of core SIP > > > > > -- > -Xiaotao > > =========================================================== > Name : Xiaotao Wu > Email : xwu@avaya.com, xiaotaow@gmail.com > Web : http://www.cs.columbia.edu/~xiaotaow > Phone : (732) 852-2133 > SIP : sip:xiaotaow@cs.columbia.edu > Address : 1M-240, 307 Middletown Lincroft Road > Lincroft, NJ 07738-1526, U.S.A. > -- -Xiaotao =========================================================== Name : Xiaotao Wu Email : xwu@avaya.com, xiaotaow@gmail.com Web : http://www.cs.columbia.edu/~xiaotaow Phone : (732) 852-2133 SIP : sip:xiaotaow@cs.columbia.edu Address : 1M-240, 307 Middletown Lincroft Road Lincroft, NJ 07738-1526, U.S.A. _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Fri Aug 04 09:20:50 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8zba-0004Kz-UB; Fri, 04 Aug 2006 09:20:46 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G8zbZ-0004Gf-Fk for sipping@ietf.org; Fri, 04 Aug 2006 09:20:45 -0400 Received: from rtp-iport-2.cisco.com ([64.102.122.149]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G8zbX-00072O-3v for sipping@ietf.org; Fri, 04 Aug 2006 09:20:45 -0400 Received: from rtp-dkim-2.cisco.com ([64.102.121.159]) by rtp-iport-2.cisco.com with ESMTP; 04 Aug 2006 09:20:43 -0400 X-IronPort-AV: i="4.07,211,1151899200"; d="scan'208"; a="95115848:sNHT26859420" Received: from rtp-core-1.cisco.com (rtp-core-1.cisco.com [64.102.124.12]) by rtp-dkim-2.cisco.com (8.12.11.20060308/8.12.11) with ESMTP id k74DKgPj001441; Fri, 4 Aug 2006 09:20:42 -0400 Received: from xbh-rtp-201.amer.cisco.com (xbh-rtp-201.cisco.com [64.102.31.12]) by rtp-core-1.cisco.com (8.12.10/8.12.6) with ESMTP id k74DKgdp007432; Fri, 4 Aug 2006 09:20:42 -0400 (EDT) Received: from xfe-rtp-201.amer.cisco.com ([64.102.31.38]) by xbh-rtp-201.amer.cisco.com with Microsoft SMTPSVC(6.0.3790.1830); Fri, 4 Aug 2006 09:20:42 -0400 Received: from [161.44.79.104] ([161.44.79.104]) by xfe-rtp-201.amer.cisco.com with Microsoft SMTPSVC(6.0.3790.1830); Fri, 4 Aug 2006 09:20:42 -0400 Message-ID: <44D349A9.4010100@cisco.com> Date: Fri, 04 Aug 2006 09:20:41 -0400 From: Paul Kyzivat User-Agent: Thunderbird 1.5.0.5 (Windows/20060719) MIME-Version: 1.0 To: Kapil Nayar Subject: Re: [Sipping] Clarification on RFC 3959 - early media References: <20060803190844.83014.qmail@web31504.mail.mud.yahoo.com> In-Reply-To: <20060803190844.83014.qmail@web31504.mail.mud.yahoo.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Content-Transfer-Encoding: 7bit X-OriginalArrivalTime: 04 Aug 2006 13:20:42.0188 (UTC) FILETIME=[C919A0C0:01C6B7C8] DKIM-Signature: a=rsa-sha1; q=dns; l=696; t=1154697642; x=1155561642; c=relaxed/simple; s=rtpdkim2001; h=Content-Type:From:Subject:Content-Transfer-Encoding:MIME-Version; d=cisco.com; i=pkyzivat@cisco.com; z=From:Paul=20Kyzivat=20 |Subject:Re=3A=20[Sipping]=20Clarification=20on=20RFC=203959=20-=20early=20media |To:Kapil=20Nayar=20; X=v=3Dcisco.com=3B=20h=3D/ddYJqkrd9IxqsyFuDtUIsYBJE0=3D; b=kAV3dl897Do+r+B7nXMv0kdP1OGT9rP7wTEwQBNaBMeq6aF90XdMsYXn6G0Zhyt/6JA9skmI dEH9r6rqqC7k9gPo2r2/5Fd6yoL+Ge/+ZWr2UxD2PV2pi7H9B/4l4Qxb; Authentication-Results: rtp-dkim-2.cisco.com; header.From=pkyzivat@cisco.com; dkim=pass ( sig from cisco.com verified; ); X-Spam-Score: 0.0 (/) X-Scan-Signature: 798b2e660f1819ae38035ac1d8d5e3ab Cc: sipping@ietf.org X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Kapil Nayar wrote: > Hi > > The RFC mentions the 183-session progress to have both > the early-offer and the answer which is replied with > the early-answer in PRACK. > > The RFC seems to imply that we DONOT need an answer > (SDP) in the 200-OK. > Is it correct? *IF* the offer/answer cycle has been *completed* prior to the sending of the 200 for the INVITE, then SDP need not be conveyed in the 200. To get in this situation requires the use of reliable provisional responses. This is a complex subject. Aspects of it are spread across many RFCs, and much is underspecified. See draft-sawada-sipping-sip-offeranswer-01 for a more complete specification. Paul _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Fri Aug 04 11:12:14 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G91LQ-0000re-RR; Fri, 04 Aug 2006 11:12:12 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G91LP-0000rZ-7Q for sipping@ietf.org; Fri, 04 Aug 2006 11:12:11 -0400 Received: from rtp-iport-2.cisco.com ([64.102.122.149]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G91LN-0004y2-Pv for sipping@ietf.org; Fri, 04 Aug 2006 11:12:11 -0400 Received: from rtp-dkim-2.cisco.com ([64.102.121.159]) by rtp-iport-2.cisco.com with ESMTP; 04 Aug 2006 11:12:09 -0400 X-IronPort-AV: i="4.07,211,1151899200"; d="scan'208"; a="95143047:sNHT27295948" Received: from rtp-core-2.cisco.com (rtp-core-2.cisco.com [64.102.124.13]) by rtp-dkim-2.cisco.com (8.12.11.20060308/8.12.11) with ESMTP id k74FC987008340; Fri, 4 Aug 2006 11:12:09 -0400 Received: from xbh-rtp-211.amer.cisco.com (xbh-rtp-211.cisco.com [64.102.31.102]) by rtp-core-2.cisco.com (8.12.10/8.12.6) with ESMTP id k74FC5e2019064; Fri, 4 Aug 2006 11:12:05 -0400 (EDT) Received: from xfe-rtp-201.amer.cisco.com ([64.102.31.38]) by xbh-rtp-211.amer.cisco.com with Microsoft SMTPSVC(6.0.3790.1830); Fri, 4 Aug 2006 11:12:05 -0400 Received: from [192.168.1.100] ([10.86.243.53]) by xfe-rtp-201.amer.cisco.com with Microsoft SMTPSVC(6.0.3790.1830); Fri, 4 Aug 2006 11:12:05 -0400 Message-ID: <44D363C4.5040708@cisco.com> Date: Fri, 04 Aug 2006 11:12:04 -0400 From: Jonathan Rosenberg User-Agent: Mozilla/5.0 (Windows; U; Windows NT 5.0; en-US; rv:1.7.8) Gecko/20050511 X-Accept-Language: en-us, en MIME-Version: 1.0 To: Dean Willis Subject: Re: [Sipping] Question on do not disturb indication References: <50B1CBA96870A34799A506B2313F266709897F68@ntht201e.siemenscomms.co.uk> <1B5A1FDA-AA89-4B8E-807E-2F170054D828@softarmor.com> In-Reply-To: <1B5A1FDA-AA89-4B8E-807E-2F170054D828@softarmor.com> Content-Type: text/plain; charset=us-ascii; format=flowed Content-Transfer-Encoding: 7bit X-OriginalArrivalTime: 04 Aug 2006 15:12:05.0117 (UTC) FILETIME=[586FAAD0:01C6B7D8] DKIM-Signature: a=rsa-sha1; q=dns; l=3345; t=1154704329; x=1155568329; c=relaxed/simple; s=rtpdkim2001; h=Content-Type:From:Subject:Content-Transfer-Encoding:MIME-Version; d=cisco.com; i=jdrosen@cisco.com; z=From:Jonathan=20Rosenberg=20 |Subject:Re=3A=20[Sipping]=20Question=20on=20do=20not=20disturb=20indication |To:Dean=20Willis=20; X=v=3Dcisco.com=3B=20h=3DGFxES4RmmDu49iLt0Ijo850qQOk=3D; b=0DIQ+ETTBTx4bMaTIhrGETMIWk/dtlo3wtj++Tl9eik0IvCDCn+tBw6rgoFT8fKT3YPVbGQZ oXCfQi0NCQG9ch/SS1zF96C6jV2yHoWx+PRxksPGoAiAJwRREzrgY50+; Authentication-Results: rtp-dkim-2.cisco.com; header.From=jdrosen@cisco.com; dkim=pass ( sig from cisco.com verified; ); X-Spam-Score: 0.0 (/) X-Scan-Signature: c3a18ef96977fc9bcc21a621cbf1174b Cc: "Dolly, Martin C, ALABS" , Paul Kyzivat , "Elwell, John" , sipping@ietf.org, Scott Lawrence X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org I agree with Dean here. Why do you want to know, at the calling UA side, that the call was busy because of DND? Whether you need a new code depends on the answer: 1. if you want the human user to know that this was DND so they can do something appropriate, a reason phrase is exactly the right thing 2. if you want to have the calling UA to do something different, what would it do differently on a 'your call was rejected by DND' vs. 'your call was rejected because the user is on another call'? If the answer is nothing, then you don't need a new response code. I'm also leary about defining new response codes to do a specific terminating application that is running. Saying that this was rejected due to 'DND' requiers you to have a standard definition for DND. I don't want to do that. As others have pointed out, a reasonable implementation for DND is that the termianting network simply lets the phone ring and eventually responds with a 480. That should be allowed. Thanks, Jonathan R. Dean Willis wrote: > > On Aug 3, 2006, at 1:28 AM, Elwell, John wrote: > >> Dean, >> >> So what would you like the IETF to recommend? Do you believe 480 mostly >> works, or do you think something new is needed? >> >> John > > > > Hey, I'm just a consumer. I don't want to know the technical details. I > just want it to work, mostly. > > If it were up to me, I'd be asking the question: > > What part of DND is important for a calling human to understand, vs. a > calling automata? Is there a distinction needed? > > If we believe that we'd like to convey a distinction between > "Temporarily unavailable" and "Temporarily unavailable due to > subscriber choice" to a human, perhaps a reason phrase on a 480 would > be adequate? Like "480 Do Not Disturb", maybe. > > > An automata is just likely to retry the call periodically anyhow, but a > human might stop and think "Oh yeah, it IS the middle of the night in > Singapore . . .". > > Question: How beneficial would a Retry-After value be in rate- limiting > the retries of automata? > > Now, I will say that Martin is right in that there might be many > reasons to reject a call, that DND is just the tip of the iceberg, and > that there's no way we can define them all. > > > Now, here's an interesting thought. If I use a reason phrase on 480, > and couple it to the right software, I can use SIP INVITE requests as a > covert-channel replacement for HTTP GET operations (or file transfer, > or whatever) , but peer-to-peer using the SIP infrastructure as a > rendezvous service. Don't try this at home, kids. > > -- > Dean > > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP > Use sip-implementors@cs.columbia.edu for questions on current sip > Use sip@ietf.org for new developments of core SIP > -- Jonathan D. Rosenberg, Ph.D. 600 Lanidex Plaza Cisco Fellow Parsippany, NJ 07054-2711 Cisco Systems jdrosen@cisco.com FAX: (973) 952-5050 http://www.jdrosen.net PHONE: (973) 952-5000 http://www.cisco.com _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Fri Aug 04 11:23:16 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G91W3-0002pR-GJ; Fri, 04 Aug 2006 11:23:11 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G91W2-0002pM-Al for sipping@ietf.org; Fri, 04 Aug 2006 11:23:10 -0400 Received: from mailgate.siemenscomms.co.uk ([195.171.110.225] helo=bemg01.siemenscomms.co.uk) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G91W1-0005Y5-0A for sipping@ietf.org; Fri, 04 Aug 2006 11:23:10 -0400 Received: from CONVERSION-DAEMON.siemenscomms.co.uk by siemenscomms.co.uk (PMDF V6.0-24 #40642) id <0J3H00901C2CKE@siemenscomms.co.uk> for sipping@ietf.org; Fri, 04 Aug 2006 16:23:15 +0100 (BST) Received: from ntht207e.uksgcs.siemenscomms.co.uk ([137.223.247.82]) by siemenscomms.co.uk (PMDF V6.0-24 #40642) with ESMTP id <0J3H0083JBXRP7@siemenscomms.co.uk>; Fri, 04 Aug 2006 16:20:15 +0100 (BST) Received: by ntht207e.uksgcs.siemenscomms.co.uk with Internet Mail Service (5.5.2657.72) id ; Fri, 04 Aug 2006 16:19:55 +0100 Content-return: allowed Date: Fri, 04 Aug 2006 16:19:55 +0100 From: "Elwell, John" Subject: RE: [Sipping] Question on do not disturb indication To: Paul Kyzivat , Dean Willis Message-id: <50B1CBA96870A34799A506B2313F2667098D83CE@ntht201e.siemenscomms.co.uk> MIME-version: 1.0 X-Mailer: Internet Mail Service (5.5.2657.72) Content-type: text/plain Content-transfer-encoding: 7BIT X-Spam-Score: 0.0 (/) X-Scan-Signature: cd26b070c2577ac175cd3a6d878c6248 Cc: "Dolly, Martin C, ALABS" , sipping@ietf.org, Scott Lawrence X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Paul, If it doesn't allow the proxy or UAC to take automated action, I find it hard to agree with your opinion that Error-Info seems a reasonable approach. A new response code would allow an automated action (particularly by a proxy), and a UAC could also put up a suitable display. Admittedly this would not be extensible to other conditions (except by defining yet more response codes), but adding response codes for a small number of common conditions ought to be feasible. John > -----Original Message----- > From: Paul Kyzivat [mailto:pkyzivat@cisco.com] > Sent: 04 August 2006 14:13 > To: Dean Willis > Cc: Elwell, John; Dolly, Martin C, ALABS; Scott Lawrence; > sipping@ietf.org > Subject: Re: [Sipping] Question on do not disturb indication > > > Dean Willis wrote: > > > If we believe that we'd like to convey a distinction between > > "Temporarily unavailable" and "Temporarily unavailable due > to subscriber > > choice" to a human, perhaps a reason phrase on a 480 would > be adequate? > > Like "480 Do Not Disturb", maybe. > > I have never understood the purpose of reason phrases, or the > value of > sending different ones with the same response code. > > Doing so implies an expectation that the UAC will do > something different > as a result. Its hard to do something different > algorithmically, because > there are no useful specs on what can be there. (Some UASs > may choose to > send the reason phrases translated into the language of the UAS > developer - say Klingon.) > > About the best one could hope for would be for some UACs to *display* > the value, uninterpretted. But that only works for UACs that have the > capability to do so, and that in fact do so. Realistically, providing > that is only useful to geeks. > > > Now, I will say that Martin is right in that there might be > many reasons > > to reject a call, that DND is just the tip of the iceberg, and that > > there's no way we can define them all. > > Right. > > There *is* a mechanism defined for this: Error-Info. There is > good and > bad to it: > > - good: because the value is a URI (or several) it is arbitrarily > extensible to new reasons > > - good: it can make it possible for the UAC to obtain > something that can > be rendered to the calling user, without the UAC needing to know all > possible reasons. > > - bad: its still hard for the UAC or a proxy to take > programatic action > based on the reason because there are no standards for what > the reasons > might be. It is possible that two different UASs might choose > different > URIs to represent the same reason. > > On balance, this seems like a reasonable approach to me. > > Paul > _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Fri Aug 04 11:29:26 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G91c5-0000dj-Ot; Fri, 04 Aug 2006 11:29:25 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G91c4-0000dS-4l for sipping@ietf.org; Fri, 04 Aug 2006 11:29:24 -0400 Received: from rtp-iport-1.cisco.com ([64.102.122.148]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G91c0-0005ud-Ep for sipping@ietf.org; Fri, 04 Aug 2006 11:29:24 -0400 Received: from rtp-dkim-1.cisco.com ([64.102.121.158]) by rtp-iport-1.cisco.com with ESMTP; 04 Aug 2006 08:28:54 -0700 X-BrightmailFiltered: true X-Brightmail-Tracker: AAAAAA== X-IronPort-AV: i="4.07,212,1151910000"; d="scan'208"; a="34607373:sNHT28236283272" Received: from rtp-core-2.cisco.com (rtp-core-2.cisco.com [64.102.124.13]) by rtp-dkim-1.cisco.com (8.12.11.20060308/8.12.11) with ESMTP id k74FSs9p018165; Fri, 4 Aug 2006 11:28:54 -0400 Received: from xbh-rtp-211.amer.cisco.com (xbh-rtp-211.cisco.com [64.102.31.102]) by rtp-core-2.cisco.com (8.12.10/8.12.6) with ESMTP id k74FSse2023376; Fri, 4 Aug 2006 11:28:54 -0400 (EDT) Received: from xfe-rtp-201.amer.cisco.com ([64.102.31.38]) by xbh-rtp-211.amer.cisco.com with Microsoft SMTPSVC(6.0.3790.1830); Fri, 4 Aug 2006 11:28:54 -0400 Received: from [192.168.1.100] ([10.86.243.53]) by xfe-rtp-201.amer.cisco.com with Microsoft SMTPSVC(6.0.3790.1830); Fri, 4 Aug 2006 11:28:54 -0400 Message-ID: <44D367B6.3060903@cisco.com> Date: Fri, 04 Aug 2006 11:28:54 -0400 From: Jonathan Rosenberg User-Agent: Mozilla/5.0 (Windows; U; Windows NT 5.0; en-US; rv:1.7.8) Gecko/20050511 X-Accept-Language: en-us, en MIME-Version: 1.0 To: Paul Kyzivat Subject: Re: [Sipping] Question on do not disturb indication References: <50B1CBA96870A34799A506B2313F266709897F3A@ntht201e.siemenscomms.co.uk> <1154553270.2866.86.camel@sukothai.pingtel.com> <200608022251.k72Mpvkp020060@dragon.ariadne.com> <44D20161.9050506@cisco.com> In-Reply-To: <44D20161.9050506@cisco.com> Content-Type: text/plain; charset=us-ascii; format=flowed Content-Transfer-Encoding: 7bit X-OriginalArrivalTime: 04 Aug 2006 15:28:54.0537 (UTC) FILETIME=[B218EF90:01C6B7DA] DKIM-Signature: a=rsa-sha1; q=dns; l=7210; t=1154705334; x=1155569334; c=relaxed/simple; s=rtpdkim1001; h=Content-Type:From:Subject:Content-Transfer-Encoding:MIME-Version; d=cisco.com; i=jdrosen@cisco.com; z=From:Jonathan=20Rosenberg=20 |Subject:Re=3A=20[Sipping]=20Question=20on=20do=20not=20disturb=20indication |To:Paul=20Kyzivat=20; X=v=3Dcisco.com=3B=20h=3DGFxES4RmmDu49iLt0Ijo850qQOk=3D; b=ZYRt4nEp56RzYR82Vn6CNmWyNm3nvqMQ1tozlOFovTI5p5SQcQbHXHY1zNYygnP8KbAcF1T6 dvAJ2bBxkFeVsrIKimEhYxqM1V0Zn9bC9KQoZf/C9n6FWgE+hYtPmEh8; Authentication-Results: rtp-dkim-1.cisco.com; header.From=jdrosen@cisco.com; dkim=pass ( sig from cisco.com verified; ); X-Spam-Score: 0.0 (/) X-Scan-Signature: 20f22c03b5c66958bff5ef54fcda6e48 Cc: sipping@ietf.org X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org I agree with Paul and Dale that the current prescribed behavior for 6xx makes it hard to apply to the use case we're talking about here. I think its worth taking a step back here and discussing the broader problem, which is one I recall we've touched on a few times but has never really been resolved. The general problem is how a UA can impact routing logic in a proxy at the time of call setup. We have several use cases here that are important: 1. when a user has multiple devices, and a receives a call, they would like to press a button on their phone called 'send to voicemail' which causes the call to go to voicemail. This should terminate any other branches for other UAs for that user if any are in progress, and halt any sequential search for othe UAs for that user if any were planned, and go to voicemail. 2. Like use case 1, except that the user wants the call to be rejected outright, and not even sent to voicemail. 3. Like use case 1, except that the user wants the caller to hear ringing for a while more and then get voicemail (sort of a polite block, where the caller can't differentiate this case from one where the called party wasn't there. Dean alluded to this). 4. send-to-X: the user wants the caller to be immediately redirected to somewhere else These cases all share the common problem that, its not that a response is being returend from a particular UA instance (in which case proxy logic should try others), but rather, a response is being returned from the *user* that is meant to ask the proxy to definitiely do something else. There are sevearl ways one might solve this problem: 1. The UA uploads a CPL script when it registers; that CPL script tells the proxy what to do based on specific response codes. For example, it can tell the proxy that a 476 means forward to voicemail. This has the benefit of not requiring standardized definitions of codes, and of leaving a lot of flexibility for different services. However, it falls apart when there are multiple UA in use and requires CPL support which has not seen wide success for many reasons. 2. We define one or more new response codes which have the desired effects. I think this is going to be hard actually, given that the four use cases above each might require a new response code 3. We have a header field in the response which tells the proxy that this response shuold be treated as if it came from *all* UA instances forked to from the AOR, thus cancelling all other forks for that AOR. It feels to me like we need some requirements work and problem definition here. I'd argue that this is perfect fodder for our MIG BoF next IETF. -Jonathan R. Paul Kyzivat wrote: > I went at reviewed 3261 on this and found some interesting things: > > - Global Failures 6xx > > 6xx responses indicate that a server has definitive information about > a particular user, not just the particular instance indicated in the > Request-URI. > > (This refers to a "particular user", but how that relates to a call > in progress isn't clear. Do two extensions on one AOR both belong > to the same "user", or not? Should the phone and a VM server backing > up the phone be viewed as the same user? If a proxy is configured > to offer calls first to Alice's phone, and if no answer then to > Alice's assistant Bob, does a 6xx response by Alice's phone say > anything about Bob?) > > - 603 Decline: > > This status response is returned only if the client knows > that no other end point will answer the request. > > (This is in conflict with the language in 6xx because it talks > about end point rather than user. It doesn't appear that the > difference is because of anything special about 603 vs the > general 6xx.) > > - 604 Does Not Exist Anywhere: > > The server has authoritative information that the user indicated in > the *** Request-URI *** does not exist anywhere. > > (This again differs from the 6xx description in a way that > doesn't seem related to the specifics of 604 vs 6xx. This one > does talk about *user*, and is more specific about what it means > by user. But not in a very helpful way. It explicitly calls out > the R-URI, and not say the To-URI. The R-URI typically only > identifies one UA, which means this response may not technically > mean anything different than 404.} > > This to me indicates a general lack of specificity in 3261 about the > scope of applicability of the 6xx codes. As we are finding with many > aspects of 3261, as we progress we find a need to tighten things up - > clarify things that ambiguous or just wrong. But before that can be > done, we need to figure out how we thing it ought to work. > > IMO there has to be some discretion in how 6xx codes are handled > upstream - this interacts with what the upstream node is trying to > accomplish. For instance: > > - if a proxy intends to try one or more contacts of a single AOR, > and it receives a 6xx from one of them, it normally shouldn't > try the others. > > - if a proxy intends to try one or more contacts of AOR-1 and then > try one or more contacts of AOR-2, and it receives a 6xx from > one of the AOR-1 contacts, then it should have the discretion > to still try AOR-2. Whether it does this or not may depend on > what it is trying to achieve by trying both. > > There clearly is a fine line here. The nodes receiving the 6xx responses > need some discretion. But the nodes sending the 6xx need to have enough > definition about what the codes mean to choose which one to return, so > that the right thing might happen. > > Paul > > > > Dale.Worley@comcast.net wrote: > >> From: Scott Lawrence >> >> The one thing that I _don't_ like to see is a 603 being used for this; >> that messes up any upstream forking. >> >> Personally, I think we should file a bug against the *existence* of >> 6xx codes. They just can't be made to work correctly in the light of >> multiple routings and forwardings of a URI. >> >> Dale >> >> _______________________________________________ >> Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping >> This list is for NEW development of the application of SIP >> Use sip-implementors@cs.columbia.edu for questions on current sip >> Use sip@ietf.org for new developments of core SIP >> > > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP > Use sip-implementors@cs.columbia.edu for questions on current sip > Use sip@ietf.org for new developments of core SIP > -- Jonathan D. Rosenberg, Ph.D. 600 Lanidex Plaza Cisco Fellow Parsippany, NJ 07054-2711 Cisco Systems jdrosen@cisco.com FAX: (973) 952-5050 http://www.jdrosen.net PHONE: (973) 952-5000 http://www.cisco.com _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Fri Aug 04 12:07:42 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G92D0-00051e-6R; Fri, 04 Aug 2006 12:07:34 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G92Cy-00051Y-PH for sipping@ietf.org; Fri, 04 Aug 2006 12:07:32 -0400 Received: from nf-out-0910.google.com ([64.233.182.189]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G92Cx-0008BS-6k for sipping@ietf.org; Fri, 04 Aug 2006 12:07:32 -0400 Received: by nf-out-0910.google.com with SMTP id n29so207331nfc for ; Fri, 04 Aug 2006 09:07:30 -0700 (PDT) DomainKey-Signature: a=rsa-sha1; q=dns; c=nofws; s=beta; d=gmail.com; h=received:message-id:date:from:reply-to:sender:to:subject:cc:in-reply-to:mime-version:content-type:content-transfer-encoding:content-disposition:references:x-google-sender-auth; b=pOzMqMG5avqJLJXtBd2hTychxQsezCbW5/hNrmwrhKO5ElYXZXFV6w0R6xhXyj1qFlaTuRpHt3n4JUn3pEgGcaVo/yKxJmFkdIFBBVoOcXsuGuMJTm88vHm1YilXBFz3Sb0Ksbp4hvrjJhenFCiVv4ZXc9upUX54bwR6P1BY0eM= Received: by 10.49.90.4 with SMTP id s4mr5493185nfl; Fri, 04 Aug 2006 09:07:30 -0700 (PDT) Received: by 10.48.164.7 with HTTP; Fri, 4 Aug 2006 09:07:30 -0700 (PDT) Message-ID: Date: Fri, 4 Aug 2006 12:07:30 -0400 From: "Xiaotao Wu" To: "Jonathan Rosenberg" Subject: Re: [Sipping] Question on do not disturb indication In-Reply-To: <44D367B6.3060903@cisco.com> MIME-Version: 1.0 Content-Type: text/plain; charset=ISO-8859-1; format=flowed Content-Transfer-Encoding: 7bit Content-Disposition: inline References: <50B1CBA96870A34799A506B2313F266709897F3A@ntht201e.siemenscomms.co.uk> <1154553270.2866.86.camel@sukothai.pingtel.com> <200608022251.k72Mpvkp020060@dragon.ariadne.com> <44D20161.9050506@cisco.com> <44D367B6.3060903@cisco.com> X-Google-Sender-Auth: e1326b916d97a841 X-Spam-Score: 0.0 (/) X-Scan-Signature: 225414c974e0d6437992164e91287a51 Cc: Paul Kyzivat , sipping@ietf.org X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list Reply-To: xiaotaow@gmail.com List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org > > There are sevearl ways one might solve this problem: > > 1. The UA uploads a CPL script when it registers; that CPL script tells > the proxy what to do based on specific response codes. For example, it > can tell the proxy that a 476 means forward to voicemail. This has the > benefit of not requiring standardized definitions of codes, and of > leaving a lot of flexibility for different services. However, it falls > apart when there are multiple UA in use and requires CPL support which > has not seen wide success for many reasons. > > 2. We define one or more new response codes which have the desired > effects. I think this is going to be hard actually, given that the four > use cases above each might require a new response code > > 3. We have a header field in the response which tells the proxy that > this response shuold be treated as if it came from *all* UA instances > forked to from the AOR, thus cancelling all other forks for that AOR. 4. Putting a script (e.g., a CPL-like script) as a MIME part in the response to instruct the proxy what to do. This is different from option 1, which invokes service logic by a specific response code. For example, for DND, we can still use 480 response code, but have a script instructing the proxy to proxy the call to voicemail. -Xiaotao > > > It feels to me like we need some requirements work and problem > definition here. I'd argue that this is perfect fodder for our MIG BoF > next IETF. > > -Jonathan R. > > > > Paul Kyzivat wrote: > > > I went at reviewed 3261 on this and found some interesting things: > > > > - Global Failures 6xx > > > > 6xx responses indicate that a server has definitive information about > > a particular user, not just the particular instance indicated in the > > Request-URI. > > > > (This refers to a "particular user", but how that relates to a call > > in progress isn't clear. Do two extensions on one AOR both belong > > to the same "user", or not? Should the phone and a VM server backing > > up the phone be viewed as the same user? If a proxy is configured > > to offer calls first to Alice's phone, and if no answer then to > > Alice's assistant Bob, does a 6xx response by Alice's phone say > > anything about Bob?) > > > > - 603 Decline: > > > > This status response is returned only if the client knows > > that no other end point will answer the request. > > > > (This is in conflict with the language in 6xx because it talks > > about end point rather than user. It doesn't appear that the > > difference is because of anything special about 603 vs the > > general 6xx.) > > > > - 604 Does Not Exist Anywhere: > > > > The server has authoritative information that the user indicated in > > the *** Request-URI *** does not exist anywhere. > > > > (This again differs from the 6xx description in a way that > > doesn't seem related to the specifics of 604 vs 6xx. This one > > does talk about *user*, and is more specific about what it means > > by user. But not in a very helpful way. It explicitly calls out > > the R-URI, and not say the To-URI. The R-URI typically only > > identifies one UA, which means this response may not technically > > mean anything different than 404.} > > > > This to me indicates a general lack of specificity in 3261 about the > > scope of applicability of the 6xx codes. As we are finding with many > > aspects of 3261, as we progress we find a need to tighten things up - > > clarify things that ambiguous or just wrong. But before that can be > > done, we need to figure out how we thing it ought to work. > > > > IMO there has to be some discretion in how 6xx codes are handled > > upstream - this interacts with what the upstream node is trying to > > accomplish. For instance: > > > > - if a proxy intends to try one or more contacts of a single AOR, > > and it receives a 6xx from one of them, it normally shouldn't > > try the others. > > > > - if a proxy intends to try one or more contacts of AOR-1 and then > > try one or more contacts of AOR-2, and it receives a 6xx from > > one of the AOR-1 contacts, then it should have the discretion > > to still try AOR-2. Whether it does this or not may depend on > > what it is trying to achieve by trying both. > > > > There clearly is a fine line here. The nodes receiving the 6xx responses > > need some discretion. But the nodes sending the 6xx need to have enough > > definition about what the codes mean to choose which one to return, so > > that the right thing might happen. > > > > Paul > > > > > > > > Dale.Worley@comcast.net wrote: > > > >> From: Scott Lawrence > >> > >> The one thing that I _don't_ like to see is a 603 being used for this; > >> that messes up any upstream forking. > >> > >> Personally, I think we should file a bug against the *existence* of > >> 6xx codes. They just can't be made to work correctly in the light of > >> multiple routings and forwardings of a URI. > >> > >> Dale > >> > >> _______________________________________________ > >> Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > >> This list is for NEW development of the application of SIP > >> Use sip-implementors@cs.columbia.edu for questions on current sip > >> Use sip@ietf.org for new developments of core SIP > >> > > > > _______________________________________________ > > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > > This list is for NEW development of the application of SIP > > Use sip-implementors@cs.columbia.edu for questions on current sip > > Use sip@ietf.org for new developments of core SIP > > > > -- > Jonathan D. Rosenberg, Ph.D. 600 Lanidex Plaza > Cisco Fellow Parsippany, NJ 07054-2711 > Cisco Systems > jdrosen@cisco.com FAX: (973) 952-5050 > http://www.jdrosen.net PHONE: (973) 952-5000 > http://www.cisco.com > > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP > Use sip-implementors@cs.columbia.edu for questions on current sip > Use sip@ietf.org for new developments of core SIP > -- -Xiaotao =========================================================== Name : Xiaotao Wu Email : xwu@avaya.com, xiaotaow@gmail.com Web : http://www.cs.columbia.edu/~xiaotaow Phone : (732) 852-2133 SIP : sip:xiaotaow@cs.columbia.edu Address : 1M-240, 307 Middletown Lincroft Road Lincroft, NJ 07738-1526, U.S.A. _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Fri Aug 04 12:26:46 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G92VV-00054O-OI; Fri, 04 Aug 2006 12:26:41 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G92VS-00051f-RD for sipping@ietf.org; Fri, 04 Aug 2006 12:26:38 -0400 Received: from zcars04f.nortel.com ([47.129.242.57]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G92V3-0001XE-BA for sipping@ietf.org; Fri, 04 Aug 2006 12:26:15 -0400 Received: from zrc2hxm0.corp.nortel.com (zrc2hxm0.corp.nortel.com [47.103.123.71]) by zcars04f.nortel.com (Switch-2.2.6/Switch-2.2.0) with ESMTP id k74GQ7Q09919; Fri, 4 Aug 2006 12:26:07 -0400 (EDT) X-MimeOLE: Produced By Microsoft Exchange V6.5.7226.0 Content-class: urn:content-classes:message MIME-Version: 1.0 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: quoted-printable Subject: RE: [Sipping] Question on do not disturb indication Date: Fri, 4 Aug 2006 11:26:05 -0500 Message-ID: <1ECE0EB50388174790F9694F77522CCF0BFA33E6@zrc2hxm0.corp.nortel.com> In-Reply-To: X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: [Sipping] Question on do not disturb indication Thread-Index: Aca3ukBWJG3aDoOqT7qQJkWQzmlVMQAJ7jdw From: "Francois Audet" To: "Xiaotao Wu" X-Spam-Score: 0.0 (/) X-Scan-Signature: 025f8c5000216988bfe31585db759250 Cc: sipping@ietf.org, Paul Kyzivat , "Elwell, John" , Scott Lawrence X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org That's because you are thinking of a network with very centralized control... If there is not that much coordination between the operator of the proxy, and the manufacturer of the terminal, then you don't really have a choice. For example, if I install a proxy from Open Source, or some other manufacturer, and terminals are not tighly controlled, there will be no SOAP or other interface to do this dynamic DND. An operator of a service, e.g., in an Enterprise, may choose to deploy the simplest most basic SIP proxy possible. We can speculate all we want on config protocols between terminals and proxy, but the=20 fact is all terminals that I'm aware of have the capability to do this type of feature=20 on the terminal itself.=20 Some of them have the ability to choose if these features are done "on the device" or on a centralized proxy. > -----Original Message----- > From: xiaotaow@gmail.com [mailto:xiaotaow@gmail.com] On=20 > Behalf Of Xiaotao Wu > Sent: Friday, August 04, 2006 4:34 AM > To: Audet, Francois (SC100:3055) > Cc: Elwell, John; Paul Kyzivat; sipping@ietf.org; Scott Lawrence > Subject: Re: [Sipping] Question on do not disturb indication >=20 > On 8/3/06, Francois Audet wrote: > > Yeah, sure, but this is IETF (SIP), not ECMA (CSTA). >=20 > CSTA is just one example in the existing application of doing this. > HTTP (as you said), SOAP can all be used to do this. It's not=20 > about whether it's IETF or ECMA. > It's about which place we should put the service at.=20 > Certainly you can handle it on an endpoints, but you then=20 > need to make this endpoint be able to control the routing to=20 > other endpoints. That should not be the task of endpoings=20 > (well, maybe in a untrusted P2P environment, this is useful). > So, I think performing DND on a proxy server (if there is,=20 > and maybe with the help of CPL or something like) is more=20 > appropriate than performing the service on endpoints. >=20 > -Xiaotao >=20 > > > > But I agree with you that if you have some other protocol,=20 > you can do=20 > > DND on the proxy instead of the endpoint. > > > > You could also have a Web Page that allows you to affect routing on=20 > > the proxy in a similar way. > > > > > -----Original Message----- > > > From: xiaotaow@gmail.com [mailto:xiaotaow@gmail.com] On Behalf Of=20 > > > Xiaotao Wu > > > Sent: Thursday, August 03, 2006 6:08 PM > > > To: Audet, Francois (SC100:3055) > > > Cc: sipping@ietf.org; Paul Kyzivat; Elwell, John; Scott Lawrence > > > Subject: Re: [Sipping] Question on do not disturb indication > > > > > > On 8/3/06, Francois Audet wrote: > > > > Sure you can do it on phones. > > > > > > > > You an have a button on the phone to "Decline" an incoming call=20 > > > > dynamically. > > > > You can have a DND feature (either permanent or dynamic). > > > > > > Which entity to ebable the feature is different from=20 > which entity to=20 > > > execute the service logic. Sure you can enable it on the=20 > phone, but=20 > > > the service logic can still be executed on proxy servers. For=20 > > > example, Microsoft Office Communicator uses CSTA to transmit the=20 > > > "Set DND" > > > command to a gateway, the "Set DND" is not executed on the=20 > > > Communicator itself. > > > > > > -Xiaotao > > > > > > > > > > > This is widely done today. > > > > > > > > > -----Original Message----- > > > > > From: xiaotaow@gmail.com [mailto:xiaotaow@gmail.com]=20 > On Behalf=20 > > > > > Of Xiaotao Wu > > > > > Sent: Thursday, August 03, 2006 4:18 AM > > > > > To: Audet, Francois (SC100:3055) > > > > > Cc: Paul Kyzivat; Scott Lawrence; Elwell, John;=20 > sipping@ietf.org > > > > > Subject: Re: [Sipping] Question on do not disturb indication > > > > > > > > > > On 8/2/06, Francois Audet wrote: > > > > > > On the 603 issue, I'd like to disagree with=20 > everybody else... > > > > > > Sometimes being able to decline the call everywhere is > > > > > actually what > > > > > > you want. > > > > > > As an example, I have multiple forked device. If I send > > > a 603, it > > > > > > triggers the proxy to use whatever alternate routing is > > > sees fit > > > > > > (e.g., voicemail, attendant, etc.). That's the=20 > desired behavior. > > > > > > > > > > > > Problem with a 480/486 with that scenario is that it won't > > > > > trigger the > > > > > > alternate routing and will keep on ringing the other fork.=20 > > > > > > That may not be the desired behavior. > > > > > > > > > > I would expect the response should be sent by the proxy, > > > instead by > > > > > endpoints. If it's send by an endpoint, things may get > > > complicated. > > > > > For example, Bob has a desk phone, a cell phone, a > > > voicemail, and a > > > > > secretary. It's hard to compose a response from his=20 > desk phone=20 > > > > > to indicate that he does not want to ring his desk phone, and=20 > > > > > cell phone, but ring his secretary phone and voicemail. > > > > > > > > > > So, I think the problem is more on what's the expected > > > code for the > > > > > caller, not on the routing behavior of the request/response. > > > > > > > > > > -Xiaotao > > > > > > > > > > > > > > > > > My 2 cents. > > > > > > > > > > > > > -----Original Message----- > > > > > > > From: Paul Kyzivat [mailto:pkyzivat@cisco.com] > > > > > > > Sent: Wednesday, August 02, 2006 3:31 PM > > > > > > > To: Scott Lawrence > > > > > > > Cc: sipping@ietf.org; Elwell, John > > > > > > > Subject: Re: [Sipping] Question on do not disturb=20 > indication > > > > > > > > > > > > > > I pretty much agree with Scott. In many/most cases I > > > > > won't want the > > > > > > > caller to know why his call is going unanswered. > > > > > > > Both 480 and 486 are in effect "white lies". Which one to=20 > > > > > > > use may depend on which impression you want to give to the > > > > > caller. Another > > > > > > > one - if you want to be more rude - is 403 Forbidden. > > > > > > > > > > > > > > I don't see a need for a *special* code that is > > > unique to this > > > > > > > service. > > > > > > > > > > > > > > Paul > > > > > > > > > > > > > > Scott Lawrence wrote: > > > > > > > > On Wed, 2006-08-02 at 21:09 +0100, Elwell, John wrote: > > > > > > > >> RFC 3261 suggests the use of response code 480=20 > > > > > > > >> Temporarily Unavailable for indicating do not disturb=20 > > > > > > > >> (DND), but it is > > > > > > > not used > > > > > > > >> exclusively for this purpose. There are other=20 > documents=20 > > > > > > > >> (e.g., RFC > > > > > > > >> 4504) that suggest the use of 486 Busy Here, but > > > again this > > > > > > > >> is not helpful if the receiver wants to > > > distinguish between > > > > > > > >> DND > > > > > > > and ordinary > > > > > > > >> busy. Presence can be used to indicate this condition > > > > > through the > > > > > > > >> RPID note element, but that does not say why a > > > > > particular session > > > > > > > >> establishment attempt has failed. What are the best > > > > > practices for > > > > > > > >> indicating the DND condition? Does anyone see=20 > a need for > > > > > > > an explicit > > > > > > > >> indicator for this purpose, e.g., a new SIP=20 > response code? > > > > > > > > > > > > > > > > Personally, I like 480 for this. > > > > > > > > > > > > > > > > If the phone wanted to be sneaky, it could just mute the > > > > > > > ringing and > > > > > > > > allow the transaction to time out without explicitly > > > > > rejecting it. > > > > > > > > That prevents the caller from being able to=20 > tell they are=20 > > > > > > > > being disregarded, but some users might prefer that > > > forwarding > > > > > > > take effect > > > > > > > > sooner and not like that. > > > > > > > > > > > > > > > > The one thing that I _don't_ like to see is a 603 being > > > > > > > used for this; > > > > > > > > that messes up any upstream forking. > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > _______________________________________________ > > > > > > > Sipping mailing list > > > > > https://www1.ietf.org/mailman/listinfo/sipping > > > > > > > This list is for NEW development of the application > > > of SIP Use > > > > > > > sip-implementors@cs.columbia.edu for questions on current=20 > > > > > > > sip Use sip@ietf.org for new developments of core SIP > > > > > > > > > > > > > > > > > > > _______________________________________________ > > > > > > Sipping mailing list > > > > > > https://www1.ietf.org/mailman/listinfo/sipping > > > > > > This list is for NEW development of the application=20 > of SIP Use=20 > > > > > > sip-implementors@cs.columbia.edu for questions on > > > current sip Use > > > > > > sip@ietf.org for new developments of core SIP > > > > > > > > > > > > > > > > > > > > > -- > > > > > -Xiaotao > > > > > > > > > > = =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D= =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D= =3D=3D=3D=3D=3D=3D=3D=3D=3D > > > > > Name : Xiaotao Wu > > > > > Email : xwu@avaya.com, xiaotaow@gmail.com > > > > > Web : http://www.cs.columbia.edu/~xiaotaow > > > > > Phone : (732) 852-2133 > > > > > SIP : sip:xiaotaow@cs.columbia.edu > > > > > Address : 1M-240, 307 Middletown Lincroft Road > > > > > Lincroft, NJ 07738-1526, U.S.A. > > > > > > > > > > > > > _______________________________________________ > > > > Sipping mailing list =20 > > > > https://www1.ietf.org/mailman/listinfo/sipping > > > > This list is for NEW development of the application of SIP Use=20 > > > > sip-implementors@cs.columbia.edu for questions on=20 > current sip Use=20 > > > > sip@ietf.org for new developments of core SIP > > > > > > > > > > > > > -- > > > -Xiaotao > > > > > > = =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D= =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D= =3D=3D=3D=3D=3D=3D=3D=3D=3D > > > Name : Xiaotao Wu > > > Email : xwu@avaya.com, xiaotaow@gmail.com > > > Web : http://www.cs.columbia.edu/~xiaotaow > > > Phone : (732) 852-2133 > > > SIP : sip:xiaotaow@cs.columbia.edu > > > Address : 1M-240, 307 Middletown Lincroft Road > > > Lincroft, NJ 07738-1526, U.S.A. > > > > > > > _______________________________________________ > > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > > This list is for NEW development of the application of SIP Use=20 > > sip-implementors@cs.columbia.edu for questions on current sip Use=20 > > sip@ietf.org for new developments of core SIP > > >=20 >=20 > -- > -Xiaotao >=20 > = =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D= =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D= =3D=3D=3D=3D=3D=3D=3D=3D=3D > Name : Xiaotao Wu > Email : xwu@avaya.com, xiaotaow@gmail.com > Web : http://www.cs.columbia.edu/~xiaotaow > Phone : (732) 852-2133 > SIP : sip:xiaotaow@cs.columbia.edu > Address : 1M-240, 307 Middletown Lincroft Road > Lincroft, NJ 07738-1526, U.S.A. >=20 _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Fri Aug 04 12:26:50 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G92Ve-0005LM-C2; Fri, 04 Aug 2006 12:26:50 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G92Vc-0005EJ-Sk for sipping@ietf.org; Fri, 04 Aug 2006 12:26:48 -0400 Received: from zcars04f.nortel.com ([47.129.242.57]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G92Pu-0001AM-JP for sipping@ietf.org; Fri, 04 Aug 2006 12:20:55 -0400 Received: from zrc2hxm0.corp.nortel.com (zrc2hxm0.corp.nortel.com [47.103.123.71]) by zcars04f.nortel.com (Switch-2.2.6/Switch-2.2.0) with ESMTP id k74GKoQ08986; Fri, 4 Aug 2006 12:20:50 -0400 (EDT) X-MimeOLE: Produced By Microsoft Exchange V6.5.7226.0 Content-class: urn:content-classes:message MIME-Version: 1.0 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: quoted-printable Subject: RE: [Sipping] Question on do not disturb indication Date: Fri, 4 Aug 2006 11:20:47 -0500 Message-ID: <1ECE0EB50388174790F9694F77522CCF0BFA33C6@zrc2hxm0.corp.nortel.com> In-Reply-To: <50B1CBA96870A34799A506B2313F2667098D7D21@ntht201e.siemenscomms.co.uk> X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: [Sipping] Question on do not disturb indication Thread-Index: Aca3K2dFlxuhqq6DREyhPVu6nqE+fQAtjcnA From: "Francois Audet" To: "Elwell, John" X-Spam-Score: 0.0 (/) X-Scan-Signature: 4adaf050708fb13be3316a9eee889caa Cc: sipping@ietf.org X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Hi John, Can you provide me an example of where a proxy would react different to "temporarily unavailable" versus "Do not disturb"? I sense that this is kind of similar to the ambiguity of 302 problem... > -----Original Message----- > From: Elwell, John [mailto:john.elwell@siemens.com]=20 > Sent: Thursday, August 03, 2006 11:33 AM > To: Audet, Francois (SC100:3055); Paul Kyzivat; Scott Lawrence > Cc: sipping@ietf.org > Subject: RE: [Sipping] Question on do not disturb indication >=20 > Francois, >=20 > It is not only that I want the caller to know that I am in=20 > DND - also my proxy, if it is scripted to perform special=20 > behaviour on DND (e.g., retarget), then I would like it to=20 > know. I am not sure that 480 is sufficiently precise. >=20 > John=20 >=20 > > -----Original Message----- > > From: Francois Audet [mailto:audet@nortel.com] > > Sent: 03 August 2006 19:10 > > To: Elwell, John; Paul Kyzivat; Scott Lawrence > > Cc: sipping@ietf.org > > Subject: RE: [Sipping] Question on do not disturb indication > >=20 > > =20 > > > [JRE] I agree that "make busy" is a separate feature and is best=20 > > > dealt with by 486. But where I explicitly want the caller to know=20 > > > that I am in DND (possibly with a retry-after time), do you think=20 > > > that 480 is sufficient, bearing in mind that it gets used=20 > for other=20 > > > purposes too, e.g., in RFC3398? > >=20 > > Personally, I think it is sufficient, and I also think that mapping=20 > > DND to 480 is pretty much established by now. > >=20 > > My perspective is Enterprise-centric however: there might be some=20 > > Service Provider services that need a more precise indication. > > I could be convinced that it is needed... > >=20 >=20 _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Fri Aug 04 12:49:34 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G92rc-00052K-Dc; Fri, 04 Aug 2006 12:49:32 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G92ra-00050D-P5 for sipping@ietf.org; Fri, 04 Aug 2006 12:49:30 -0400 Received: from zrtps0kp.nortel.com ([47.140.192.56]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G92lZ-0002kP-JS for sipping@ietf.org; Fri, 04 Aug 2006 12:43:18 -0400 Received: from zrc2hxm0.corp.nortel.com (zrc2hxm0.corp.nortel.com [47.103.123.71]) by zrtps0kp.nortel.com (Switch-2.2.6/Switch-2.2.0) with ESMTP id k74GhCM18840; Fri, 4 Aug 2006 12:43:12 -0400 (EDT) X-MimeOLE: Produced By Microsoft Exchange V6.5.7226.0 Content-class: urn:content-classes:message MIME-Version: 1.0 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: quoted-printable Subject: RE: [Sipping] Question on do not disturb indication Date: Fri, 4 Aug 2006 11:43:11 -0500 Message-ID: <1ECE0EB50388174790F9694F77522CCF0BFA3444@zrc2hxm0.corp.nortel.com> In-Reply-To: <44D367B6.3060903@cisco.com> X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: [Sipping] Question on do not disturb indication Thread-Index: Aca32zJ6QUBKIWcnRUaAqQ2SrPFC/wAB9R8g From: "Francois Audet" To: "Jonathan Rosenberg" , "Paul Kyzivat" X-Spam-Score: 0.0 (/) X-Scan-Signature: bdc523f9a54890b8a30dd6fd53d5d024 Cc: sipping@ietf.org X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Agreed: it does seem like a perfect case for MIG next meeting.=20 > -----Original Message----- > From: Jonathan Rosenberg [mailto:jdrosen@cisco.com]=20 > Sent: Friday, August 04, 2006 8:29 AM > To: Paul Kyzivat > Cc: sipping@ietf.org > Subject: Re: [Sipping] Question on do not disturb indication >=20 > I agree with Paul and Dale that the current prescribed=20 > behavior for 6xx makes it hard to apply to the use case we're=20 > talking about here. >=20 > I think its worth taking a step back here and discussing the=20 > broader problem, which is one I recall we've touched on a few=20 > times but has never really been resolved. >=20 > The general problem is how a UA can impact routing logic in a=20 > proxy at the time of call setup. We have several use cases=20 > here that are important: >=20 > 1. when a user has multiple devices, and a receives a call,=20 > they would like to press a button on their phone called 'send=20 > to voicemail' which causes the call to go to voicemail. This=20 > should terminate any other branches for other UAs for that=20 > user if any are in progress, and halt any sequential search=20 > for othe UAs for that user if any were planned, and go to voicemail. >=20 > 2. Like use case 1, except that the user wants the call to be=20 > rejected outright, and not even sent to voicemail. >=20 > 3. Like use case 1, except that the user wants the caller to=20 > hear ringing for a while more and then get voicemail (sort of=20 > a polite block, where the caller can't differentiate this=20 > case from one where the called party wasn't there. Dean=20 > alluded to this). >=20 > 4. send-to-X: the user wants the caller to be immediately=20 > redirected to somewhere else >=20 >=20 > These cases all share the common problem that, its not that a=20 > response=20 > is being returend from a particular UA instance (in which case proxy=20 > logic should try others), but rather, a response is being=20 > returned from=20 > the *user* that is meant to ask the proxy to definitiely do=20 > something else. >=20 > There are sevearl ways one might solve this problem: >=20 > 1. The UA uploads a CPL script when it registers; that CPL=20 > script tells=20 > the proxy what to do based on specific response codes. For=20 > example, it=20 > can tell the proxy that a 476 means forward to voicemail.=20 > This has the=20 > benefit of not requiring standardized definitions of codes, and of=20 > leaving a lot of flexibility for different services. However,=20 > it falls=20 > apart when there are multiple UA in use and requires CPL=20 > support which=20 > has not seen wide success for many reasons. >=20 > 2. We define one or more new response codes which have the desired=20 > effects. I think this is going to be hard actually, given=20 > that the four=20 > use cases above each might require a new response code >=20 > 3. We have a header field in the response which tells the proxy that=20 > this response shuold be treated as if it came from *all* UA instances=20 > forked to from the AOR, thus cancelling all other forks for that AOR. >=20 >=20 > It feels to me like we need some requirements work and problem=20 > definition here. I'd argue that this is perfect fodder for=20 > our MIG BoF=20 > next IETF. _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Fri Aug 04 13:15:17 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G93GV-0004SN-L2; Fri, 04 Aug 2006 13:15:15 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G93GU-0004SH-Ck for sipping@ietf.org; Fri, 04 Aug 2006 13:15:14 -0400 Received: from zrtps0kp.nortel.com ([47.140.192.56]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G93GT-0004nl-09 for sipping@ietf.org; Fri, 04 Aug 2006 13:15:14 -0400 Received: from zrc2hxm0.corp.nortel.com (zrc2hxm0.corp.nortel.com [47.103.123.71]) by zrtps0kp.nortel.com (Switch-2.2.6/Switch-2.2.0) with ESMTP id k74HF8M01063; Fri, 4 Aug 2006 13:15:08 -0400 (EDT) X-MimeOLE: Produced By Microsoft Exchange V6.5.7226.0 Content-class: urn:content-classes:message MIME-Version: 1.0 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: quoted-printable Subject: RE: [Sipping] Question on do not disturb indication Date: Fri, 4 Aug 2006 12:15:07 -0500 Message-ID: <1ECE0EB50388174790F9694F77522CCF0BFA34CA@zrc2hxm0.corp.nortel.com> In-Reply-To: <200608040215.k742FicH025962@dragon.ariadne.com> X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: [Sipping] Question on do not disturb indication Thread-Index: Aca3bGiotTS6KsBeSo2KJ0twsLvIlgAe5S2w From: "Francois Audet" To: , X-Spam-Score: 0.0 (/) X-Scan-Signature: f4c2cf0bccc868e4cc88dace71fb3f44 Cc: X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Actually, now that I re-read it, I would say that it is "unpleasantly UNclear"... I was focusing on the paragraph of 21.6, which relates 6XX to a SPECIFIC USER (i.e., not a GRUU, or "particular instance"). That, I like, and implies to me that retargeting (i.e., going to a different user) is a valid action upon receipt of 6XX. If you call audet@nortel.com, and I reject it with 6XX, it could be retargeted by one of our proxies to operator@nortel.com for example or operator@comcast.com. However, the subsections for 600/603/606 talk of "no other endpoint" as opposed to the SPECIFIC user. That's the ones that I think you were focussing on. It would prohibit retargeting to operator@nortel.com/operator@comcast.net. Interestingly enough, 604 however is related to a specific URI, not "any endpoints" (and consequently could be retargeted). In short, this whole section contradicts itself by not being clear about what 6XX implies: a failure of target, or a failure of the=20 session... > -----Original Message----- > From: Dale.Worley@comcast.net [mailto:Dale.Worley@comcast.net]=20 > Sent: Thursday, August 03, 2006 7:16 PM > To: sipping@ietf.org > Subject: Re: [Sipping] Question on do not disturb indication >=20 > From: "Francois Audet" >=20 > But I don't think RFC 3261 necessarily needs to be amended. > It's always been legal for a proxy to retarget however it pleases. >=20 > I beg to differ -- RFC 3261 is unpleasantly clear on the=20 > effect of 6xx responses. Handling *other* responses leaves=20 > the proxy with a lot of flexibility in deciding the "target=20 > set", but for 6xx, the target set is effectively emptied. >=20 > Though I see no reason not to change that definition -- I see=20 > the current definition of 6xx as unusable due to evil=20 > interaction with any system that uses "interesting"=20 > retargeting, so "there are no current uses of the feature". >=20 > Dale >=20 > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP=20 > Use sip-implementors@cs.columbia.edu for questions on current=20 > sip Use sip@ietf.org for new developments of core SIP >=20 _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Fri Aug 04 13:54:04 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G93rw-0000HC-0S; Fri, 04 Aug 2006 13:53:56 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G93ru-00009v-17 for sipping@ietf.org; Fri, 04 Aug 2006 13:53:54 -0400 Received: from rtp-iport-2.cisco.com ([64.102.122.149]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G93rs-0007Nr-K5 for sipping@ietf.org; Fri, 04 Aug 2006 13:53:54 -0400 Received: from rtp-dkim-2.cisco.com ([64.102.121.159]) by rtp-iport-2.cisco.com with ESMTP; 04 Aug 2006 13:53:52 -0400 X-IronPort-AV: i="4.07,212,1151899200"; d="scan'208"; a="95168910:sNHT27324672" Received: from rtp-core-1.cisco.com (rtp-core-1.cisco.com [64.102.124.12]) by rtp-dkim-2.cisco.com (8.12.11.20060308/8.12.11) with ESMTP id k74HrqhC015359; Fri, 4 Aug 2006 13:53:52 -0400 Received: from xbh-rtp-211.amer.cisco.com (xbh-rtp-211.cisco.com [64.102.31.102]) by rtp-core-1.cisco.com (8.12.10/8.12.6) with ESMTP id k74Hrldp025569; Fri, 4 Aug 2006 13:53:47 -0400 (EDT) Received: from xfe-rtp-201.amer.cisco.com ([64.102.31.38]) by xbh-rtp-211.amer.cisco.com with Microsoft SMTPSVC(6.0.3790.1830); Fri, 4 Aug 2006 13:53:47 -0400 Received: from [161.44.79.104] ([161.44.79.104]) by xfe-rtp-201.amer.cisco.com with Microsoft SMTPSVC(6.0.3790.1830); Fri, 4 Aug 2006 13:53:47 -0400 Message-ID: <44D389AA.2090106@cisco.com> Date: Fri, 04 Aug 2006 13:53:46 -0400 From: Paul Kyzivat User-Agent: Thunderbird 1.5.0.5 (Windows/20060719) MIME-Version: 1.0 To: "Elwell, John" Subject: Re: [Sipping] Question on do not disturb indication References: <50B1CBA96870A34799A506B2313F2667098D83CE@ntht201e.siemenscomms.co.uk> In-Reply-To: <50B1CBA96870A34799A506B2313F2667098D83CE@ntht201e.siemenscomms.co.uk> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Content-Transfer-Encoding: 7bit X-OriginalArrivalTime: 04 Aug 2006 17:53:47.0304 (UTC) FILETIME=[EF63EA80:01C6B7EE] DKIM-Signature: a=rsa-sha1; q=dns; l=3627; t=1154714032; x=1155578032; c=relaxed/simple; s=rtpdkim2001; h=Content-Type:From:Subject:Content-Transfer-Encoding:MIME-Version; d=cisco.com; i=pkyzivat@cisco.com; z=From:Paul=20Kyzivat=20 |Subject:Re=3A=20[Sipping]=20Question=20on=20do=20not=20disturb=20indication |To:=22Elwell,=20John=22=20; X=v=3Dcisco.com=3B=20h=3Dyu9ohsv1kFqX2uC2CMjguD3sTnY=3D; b=eTH8HivG6CNQhfJvhKxkui4bt1/FFp8MxUbZLdEpSNcASeMtUpxLRXdx06ONDNlgLHUJD1SQ lK0lMxmnA25IR7LfzQ8N6hHg2T07pr1/xkUrDjdTb6VVJBEKZ8iJZJyi; Authentication-Results: rtp-dkim-2.cisco.com; header.From=pkyzivat@cisco.com; dkim=pass ( sig from cisco.com verified; ); X-Spam-Score: 0.0 (/) X-Scan-Signature: cf3becbbd6d1a45acbe2ffd4ab88bdc2 Cc: "Dolly, Martin C, ALABS" , Scott Lawrence , sipping@ietf.org, Dean Willis X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Elwell, John wrote: > Paul, > > If it doesn't allow the proxy or UAC to take automated action, I find it > hard to agree with your opinion that Error-Info seems a reasonable approach. An Error-Info URI wouldn't permit a proxy to automatically know what to do for all possible responses from all possible UASs. But it would allow local conventions to be established. If TISPAN wanted to establish a convention about an E-I URI that means whatever they think DND is, then the proxy can be taught how to react to that. That could be done with URNs that have no other utility. Or it could be done with standardized URLs that could also be dereferenced and rendered by something that doesn't understand the convention. > A new response code would allow an automated action (particularly by a > proxy), and a UAC could also put up a suitable display. Admittedly this > would not be extensible to other conditions (except by defining yet more > response codes), but adding response codes for a small number of common > conditions ought to be feasible. I'm with Jonathan that trying to define them that way is futile. There will not be sufficient agreement on what they mean. I don't want to see SIP locked down to a small fixed set of services. Paul > John > >> -----Original Message----- >> From: Paul Kyzivat [mailto:pkyzivat@cisco.com] >> Sent: 04 August 2006 14:13 >> To: Dean Willis >> Cc: Elwell, John; Dolly, Martin C, ALABS; Scott Lawrence; >> sipping@ietf.org >> Subject: Re: [Sipping] Question on do not disturb indication >> >> >> Dean Willis wrote: >> >>> If we believe that we'd like to convey a distinction between >>> "Temporarily unavailable" and "Temporarily unavailable due >> to subscriber >>> choice" to a human, perhaps a reason phrase on a 480 would >> be adequate? >>> Like "480 Do Not Disturb", maybe. >> I have never understood the purpose of reason phrases, or the >> value of >> sending different ones with the same response code. >> >> Doing so implies an expectation that the UAC will do >> something different >> as a result. Its hard to do something different >> algorithmically, because >> there are no useful specs on what can be there. (Some UASs >> may choose to >> send the reason phrases translated into the language of the UAS >> developer - say Klingon.) >> >> About the best one could hope for would be for some UACs to *display* >> the value, uninterpretted. But that only works for UACs that have the >> capability to do so, and that in fact do so. Realistically, providing >> that is only useful to geeks. >> >>> Now, I will say that Martin is right in that there might be >> many reasons >>> to reject a call, that DND is just the tip of the iceberg, and that >>> there's no way we can define them all. >> Right. >> >> There *is* a mechanism defined for this: Error-Info. There is >> good and >> bad to it: >> >> - good: because the value is a URI (or several) it is arbitrarily >> extensible to new reasons >> >> - good: it can make it possible for the UAC to obtain >> something that can >> be rendered to the calling user, without the UAC needing to know all >> possible reasons. >> >> - bad: its still hard for the UAC or a proxy to take >> programatic action >> based on the reason because there are no standards for what >> the reasons >> might be. It is possible that two different UASs might choose >> different >> URIs to represent the same reason. >> >> On balance, this seems like a reasonable approach to me. >> >> Paul >> > _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Fri Aug 04 13:54:08 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G93s8-00012d-Mj; Fri, 04 Aug 2006 13:54:08 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G93s7-000105-52 for sipping@ietf.org; Fri, 04 Aug 2006 13:54:07 -0400 Received: from exprod6og55.obsmtp.com ([64.18.1.191]) by ietf-mx.ietf.org with smtp (Exim 4.43) id 1G93s5-0007O3-JP for sipping@ietf.org; Fri, 04 Aug 2006 13:54:07 -0400 Received: from source ([192.150.11.134]) by exprod6ob55.postini.com ([64.18.5.12]) with SMTP; Fri, 04 Aug 2006 10:54:00 PDT Received: from inner-relay-1.corp.adobe.com ([153.32.1.51]) by outbound-smtp-1.corp.adobe.com (8.12.10/8.12.10) with ESMTP id k74HqABl019447; Fri, 4 Aug 2006 10:52:10 -0700 (PDT) Received: from fe1.corp.adobe.com (fe1.corp.adobe.com [10.8.192.70]) by inner-relay-1.corp.adobe.com (8.12.10/8.12.10) with ESMTP id k74Hri4Z012701; Fri, 4 Aug 2006 10:53:44 -0700 (PDT) Received: from namail5.corp.adobe.com ([10.8.192.88]) by fe1.corp.adobe.com with Microsoft SMTPSVC(6.0.3790.1830); Fri, 4 Aug 2006 10:53:44 -0700 X-MimeOLE: Produced By Microsoft Exchange V6.5 Content-class: urn:content-classes:message MIME-Version: 1.0 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: quoted-printable Subject: RE: [Sipping] Question on do not disturb indication Date: Fri, 4 Aug 2006 10:53:40 -0700 Message-ID: <24CCCC428EFEA2469BF046DB3C7A8D220C3732@namail5.corp.adobe.com> In-Reply-To: <44D363C4.5040708@cisco.com> X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: [Sipping] Question on do not disturb indication Thread-Index: Aca32LAzyAkK+PrcTGulbjdEiWLfyAAFBIdQ From: "Henry Sinnreich" To: "Jonathan Rosenberg" , "Dean Willis" X-OriginalArrivalTime: 04 Aug 2006 17:53:44.0995 (UTC) FILETIME=[EE039730:01C6B7EE] X-Spam-Score: 0.0 (/) X-Scan-Signature: b132cb3ed2d4be2017585bf6859e1ede Cc: "Dolly, Martin C, ALABS" , Paul Kyzivat , "Elwell, John" , sipping@ietf.org, Scott Lawrence X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org This is also a good time to remember once more there is in a SIP a capability called PRESENCE that is much more useful than DND and many other legacy telephony features. :-) Why ignore Presence and just stick with the "old proven way"? Thanks, Henry -----Original Message----- From: Jonathan Rosenberg [mailto:jdrosen@cisco.com]=20 Sent: Friday, August 04, 2006 8:12 AM To: Dean Willis Cc: Dolly, Martin C, ALABS; Paul Kyzivat; Elwell,John; sipping@ietf.org; Scott Lawrence Subject: Re: [Sipping] Question on do not disturb indication I agree with Dean here. Why do you want to know, at the calling UA side, that the call was busy=20 because of DND? Whether you need a new code depends on the answer: 1. if you want the human user to know that this was DND so they can do=20 something appropriate, a reason phrase is exactly the right thing 2. if you want to have the calling UA to do something different, what=20 would it do differently on a 'your call was rejected by DND' vs. 'your=20 call was rejected because the user is on another call'? If the answer is nothing, then you don't need a new response code. I'm also leary about defining new response codes to do a specific=20 terminating application that is running. Saying that this was rejected=20 due to 'DND' requiers you to have a standard definition for DND. I don't want to do that. As others have pointed out, a reasonable implementation for DND is that the termianting network simply lets the phone ring and=20 eventually responds with a 480. That should be allowed. Thanks, Jonathan R. Dean Willis wrote: >=20 > On Aug 3, 2006, at 1:28 AM, Elwell, John wrote: >=20 >> Dean, >> >> So what would you like the IETF to recommend? Do you believe 480 mostly >> works, or do you think something new is needed? >> >> John >=20 >=20 >=20 > Hey, I'm just a consumer. I don't want to know the technical details. I=20 > just want it to work, mostly. >=20 > If it were up to me, I'd be asking the question: >=20 > What part of DND is important for a calling human to understand, vs. a=20 > calling automata? Is there a distinction needed? >=20 > If we believe that we'd like to convey a distinction between =20 > "Temporarily unavailable" and "Temporarily unavailable due to =20 > subscriber choice" to a human, perhaps a reason phrase on a 480 would > be adequate? Like "480 Do Not Disturb", maybe. >=20 >=20 > An automata is just likely to retry the call periodically anyhow, but a=20 > human might stop and think "Oh yeah, it IS the middle of the night in > Singapore . . .". >=20 > Question: How beneficial would a Retry-After value be in rate- limiting=20 > the retries of automata? >=20 > Now, I will say that Martin is right in that there might be many =20 > reasons to reject a call, that DND is just the tip of the iceberg, and=20 > that there's no way we can define them all. >=20 >=20 > Now, here's an interesting thought. If I use a reason phrase on 480, =20 > and couple it to the right software, I can use SIP INVITE requests as a=20 > covert-channel replacement for HTTP GET operations (or file transfer, > or whatever) , but peer-to-peer using the SIP infrastructure as a=20 > rendezvous service. Don't try this at home, kids. >=20 > --=20 > Dean >=20 > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP > Use sip-implementors@cs.columbia.edu for questions on current sip > Use sip@ietf.org for new developments of core SIP >=20 --=20 Jonathan D. Rosenberg, Ph.D. 600 Lanidex Plaza Cisco Fellow Parsippany, NJ 07054-2711 Cisco Systems jdrosen@cisco.com FAX: (973) 952-5050 http://www.jdrosen.net PHONE: (973) 952-5000 http://www.cisco.com _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Fri Aug 04 13:54:12 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G93sC-00017j-K1; Fri, 04 Aug 2006 13:54:12 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G93sB-00017e-4Z for sipping@ietf.org; Fri, 04 Aug 2006 13:54:11 -0400 Received: from exprod6og54.obsmtp.com ([64.18.1.189]) by ietf-mx.ietf.org with smtp (Exim 4.43) id 1G93s8-0007Np-9i for sipping@ietf.org; Fri, 04 Aug 2006 13:54:11 -0400 Received: from source ([192.150.20.142]) by exprod6ob54.postini.com ([64.18.5.12]) with SMTP; Fri, 04 Aug 2006 10:53:49 PDT Received: from inner-relay-3.eur.adobe.com (inner-relay-3b [10.128.4.236]) by outbound-smtp-2.corp.adobe.com (8.12.10/8.12.10) with ESMTP id k74Hs0uv004837; Fri, 4 Aug 2006 10:54:00 -0700 (PDT) Received: from fe2.corp.adobe.com (fe2.corp.adobe.com [10.8.192.72]) by inner-relay-3.eur.adobe.com (8.12.10/8.12.9) with ESMTP id k74HrfR0002512; Fri, 4 Aug 2006 10:53:43 -0700 (PDT) Received: from namail5.corp.adobe.com ([10.8.192.88]) by fe2.corp.adobe.com with Microsoft SMTPSVC(6.0.3790.1830); Fri, 4 Aug 2006 10:53:40 -0700 X-MimeOLE: Produced By Microsoft Exchange V6.5 Content-class: urn:content-classes:message MIME-Version: 1.0 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: quoted-printable Subject: RE: [Sipping] Question on do not disturb indication Date: Fri, 4 Aug 2006 10:53:40 -0700 Message-ID: <24CCCC428EFEA2469BF046DB3C7A8D220C3730@namail5.corp.adobe.com> In-Reply-To: <1ECE0EB50388174790F9694F77522CCF0BFA33E6@zrc2hxm0.corp.nortel.com> X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: [Sipping] Question on do not disturb indication Thread-Index: Aca3ukBWJG3aDoOqT7qQJkWQzmlVMQAJ7jdwAAIRrSA= From: "Henry Sinnreich" To: "Francois Audet" , "Xiaotao Wu" X-OriginalArrivalTime: 04 Aug 2006 17:53:40.0910 (UTC) FILETIME=[EB9444E0:01C6B7EE] X-Spam-Score: 0.0 (/) X-Scan-Signature: 9e5c23589e6cce06555030c0194c9e2b Cc: "Elwell, John" , Paul Kyzivat , sipping@ietf.org, Scott Lawrence X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org I would like to add also the (hopeful) scenario where my multi-line SIP UA is registered with competing service providers that don't cooperate too much. In such a scenario DND is a matter of local policy in the UA.=20 Building DND into the network seems to be another Centrex-like pipe dream. It's overly complicated as is already. Thanks, Henry -----Original Message----- From: Francois Audet [mailto:audet@nortel.com]=20 Sent: Friday, August 04, 2006 9:26 AM To: Xiaotao Wu Cc: sipping@ietf.org; Paul Kyzivat; Elwell,John; Scott Lawrence Subject: RE: [Sipping] Question on do not disturb indication That's because you are thinking of a network with very centralized control... If there is not that much coordination between the operator of the proxy, and the manufacturer of the terminal, then you don't really have a choice. For example, if I install a proxy from Open Source, or some other manufacturer, and terminals are not tighly controlled, there will be no SOAP or other interface to do this dynamic DND. An operator of a service, e.g., in an Enterprise, may choose to deploy the simplest most basic SIP proxy possible. We can speculate all we want on config protocols between terminals and proxy, but the=20 fact is all terminals that I'm aware of have the capability to do this type of feature=20 on the terminal itself.=20 Some of them have the ability to choose if these features are done "on the device" or on a centralized proxy. > -----Original Message----- > From: xiaotaow@gmail.com [mailto:xiaotaow@gmail.com] On=20 > Behalf Of Xiaotao Wu > Sent: Friday, August 04, 2006 4:34 AM > To: Audet, Francois (SC100:3055) > Cc: Elwell, John; Paul Kyzivat; sipping@ietf.org; Scott Lawrence > Subject: Re: [Sipping] Question on do not disturb indication >=20 > On 8/3/06, Francois Audet wrote: > > Yeah, sure, but this is IETF (SIP), not ECMA (CSTA). >=20 > CSTA is just one example in the existing application of doing this. > HTTP (as you said), SOAP can all be used to do this. It's not=20 > about whether it's IETF or ECMA. > It's about which place we should put the service at.=20 > Certainly you can handle it on an endpoints, but you then=20 > need to make this endpoint be able to control the routing to=20 > other endpoints. That should not be the task of endpoings=20 > (well, maybe in a untrusted P2P environment, this is useful). > So, I think performing DND on a proxy server (if there is,=20 > and maybe with the help of CPL or something like) is more=20 > appropriate than performing the service on endpoints. >=20 > -Xiaotao >=20 > > > > But I agree with you that if you have some other protocol,=20 > you can do=20 > > DND on the proxy instead of the endpoint. > > > > You could also have a Web Page that allows you to affect routing on=20 > > the proxy in a similar way. > > > > > -----Original Message----- > > > From: xiaotaow@gmail.com [mailto:xiaotaow@gmail.com] On Behalf Of=20 > > > Xiaotao Wu > > > Sent: Thursday, August 03, 2006 6:08 PM > > > To: Audet, Francois (SC100:3055) > > > Cc: sipping@ietf.org; Paul Kyzivat; Elwell, John; Scott Lawrence > > > Subject: Re: [Sipping] Question on do not disturb indication > > > > > > On 8/3/06, Francois Audet wrote: > > > > Sure you can do it on phones. > > > > > > > > You an have a button on the phone to "Decline" an incoming call=20 > > > > dynamically. > > > > You can have a DND feature (either permanent or dynamic). > > > > > > Which entity to ebable the feature is different from=20 > which entity to=20 > > > execute the service logic. Sure you can enable it on the=20 > phone, but=20 > > > the service logic can still be executed on proxy servers. For=20 > > > example, Microsoft Office Communicator uses CSTA to transmit the=20 > > > "Set DND" > > > command to a gateway, the "Set DND" is not executed on the=20 > > > Communicator itself. > > > > > > -Xiaotao > > > > > > > > > > > This is widely done today. > > > > > > > > > -----Original Message----- > > > > > From: xiaotaow@gmail.com [mailto:xiaotaow@gmail.com]=20 > On Behalf=20 > > > > > Of Xiaotao Wu > > > > > Sent: Thursday, August 03, 2006 4:18 AM > > > > > To: Audet, Francois (SC100:3055) > > > > > Cc: Paul Kyzivat; Scott Lawrence; Elwell, John;=20 > sipping@ietf.org > > > > > Subject: Re: [Sipping] Question on do not disturb indication > > > > > > > > > > On 8/2/06, Francois Audet wrote: > > > > > > On the 603 issue, I'd like to disagree with=20 > everybody else... > > > > > > Sometimes being able to decline the call everywhere is > > > > > actually what > > > > > > you want. > > > > > > As an example, I have multiple forked device. If I send > > > a 603, it > > > > > > triggers the proxy to use whatever alternate routing is > > > sees fit > > > > > > (e.g., voicemail, attendant, etc.). That's the=20 > desired behavior. > > > > > > > > > > > > Problem with a 480/486 with that scenario is that it won't > > > > > trigger the > > > > > > alternate routing and will keep on ringing the other fork.=20 > > > > > > That may not be the desired behavior. > > > > > > > > > > I would expect the response should be sent by the proxy, > > > instead by > > > > > endpoints. If it's send by an endpoint, things may get > > > complicated. > > > > > For example, Bob has a desk phone, a cell phone, a > > > voicemail, and a > > > > > secretary. It's hard to compose a response from his=20 > desk phone=20 > > > > > to indicate that he does not want to ring his desk phone, and=20 > > > > > cell phone, but ring his secretary phone and voicemail. > > > > > > > > > > So, I think the problem is more on what's the expected > > > code for the > > > > > caller, not on the routing behavior of the request/response. > > > > > > > > > > -Xiaotao > > > > > > > > > > > > > > > > > My 2 cents. > > > > > > > > > > > > > -----Original Message----- > > > > > > > From: Paul Kyzivat [mailto:pkyzivat@cisco.com] > > > > > > > Sent: Wednesday, August 02, 2006 3:31 PM > > > > > > > To: Scott Lawrence > > > > > > > Cc: sipping@ietf.org; Elwell, John > > > > > > > Subject: Re: [Sipping] Question on do not disturb=20 > indication > > > > > > > > > > > > > > I pretty much agree with Scott. In many/most cases I > > > > > won't want the > > > > > > > caller to know why his call is going unanswered. > > > > > > > Both 480 and 486 are in effect "white lies". Which one to=20 > > > > > > > use may depend on which impression you want to give to the > > > > > caller. Another > > > > > > > one - if you want to be more rude - is 403 Forbidden. > > > > > > > > > > > > > > I don't see a need for a *special* code that is > > > unique to this > > > > > > > service. > > > > > > > > > > > > > > Paul > > > > > > > > > > > > > > Scott Lawrence wrote: > > > > > > > > On Wed, 2006-08-02 at 21:09 +0100, Elwell, John wrote: > > > > > > > >> RFC 3261 suggests the use of response code 480=20 > > > > > > > >> Temporarily Unavailable for indicating do not disturb=20 > > > > > > > >> (DND), but it is > > > > > > > not used > > > > > > > >> exclusively for this purpose. There are other=20 > documents=20 > > > > > > > >> (e.g., RFC > > > > > > > >> 4504) that suggest the use of 486 Busy Here, but > > > again this > > > > > > > >> is not helpful if the receiver wants to > > > distinguish between > > > > > > > >> DND > > > > > > > and ordinary > > > > > > > >> busy. Presence can be used to indicate this condition > > > > > through the > > > > > > > >> RPID note element, but that does not say why a > > > > > particular session > > > > > > > >> establishment attempt has failed. What are the best > > > > > practices for > > > > > > > >> indicating the DND condition? Does anyone see=20 > a need for > > > > > > > an explicit > > > > > > > >> indicator for this purpose, e.g., a new SIP=20 > response code? > > > > > > > > > > > > > > > > Personally, I like 480 for this. > > > > > > > > > > > > > > > > If the phone wanted to be sneaky, it could just mute the > > > > > > > ringing and > > > > > > > > allow the transaction to time out without explicitly > > > > > rejecting it. > > > > > > > > That prevents the caller from being able to=20 > tell they are=20 > > > > > > > > being disregarded, but some users might prefer that > > > forwarding > > > > > > > take effect > > > > > > > > sooner and not like that. > > > > > > > > > > > > > > > > The one thing that I _don't_ like to see is a 603 being > > > > > > > used for this; > > > > > > > > that messes up any upstream forking. > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > _______________________________________________ > > > > > > > Sipping mailing list > > > > > https://www1.ietf.org/mailman/listinfo/sipping > > > > > > > This list is for NEW development of the application > > > of SIP Use > > > > > > > sip-implementors@cs.columbia.edu for questions on current=20 > > > > > > > sip Use sip@ietf.org for new developments of core SIP > > > > > > > > > > > > > > > > > > > _______________________________________________ > > > > > > Sipping mailing list > > > > > > https://www1.ietf.org/mailman/listinfo/sipping > > > > > > This list is for NEW development of the application=20 > of SIP Use=20 > > > > > > sip-implementors@cs.columbia.edu for questions on > > > current sip Use > > > > > > sip@ietf.org for new developments of core SIP > > > > > > > > > > > > > > > > > > > > > -- > > > > > -Xiaotao > > > > > > > > > > = =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D= =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D= =3D=3D=3D=3D=3D=3D=3D=3D=3D > > > > > Name : Xiaotao Wu > > > > > Email : xwu@avaya.com, xiaotaow@gmail.com > > > > > Web : http://www.cs.columbia.edu/~xiaotaow > > > > > Phone : (732) 852-2133 > > > > > SIP : sip:xiaotaow@cs.columbia.edu > > > > > Address : 1M-240, 307 Middletown Lincroft Road > > > > > Lincroft, NJ 07738-1526, U.S.A. > > > > > > > > > > > > > _______________________________________________ > > > > Sipping mailing list =20 > > > > https://www1.ietf.org/mailman/listinfo/sipping > > > > This list is for NEW development of the application of SIP Use=20 > > > > sip-implementors@cs.columbia.edu for questions on=20 > current sip Use=20 > > > > sip@ietf.org for new developments of core SIP > > > > > > > > > > > > > -- > > > -Xiaotao > > > > > > = =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D= =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D= =3D=3D=3D=3D=3D=3D=3D=3D=3D > > > Name : Xiaotao Wu > > > Email : xwu@avaya.com, xiaotaow@gmail.com > > > Web : http://www.cs.columbia.edu/~xiaotaow > > > Phone : (732) 852-2133 > > > SIP : sip:xiaotaow@cs.columbia.edu > > > Address : 1M-240, 307 Middletown Lincroft Road > > > Lincroft, NJ 07738-1526, U.S.A. > > > > > > > _______________________________________________ > > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > > This list is for NEW development of the application of SIP Use=20 > > sip-implementors@cs.columbia.edu for questions on current sip Use=20 > > sip@ietf.org for new developments of core SIP > > >=20 >=20 > -- > -Xiaotao >=20 > = =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D= =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D= =3D=3D=3D=3D=3D=3D=3D=3D=3D > Name : Xiaotao Wu > Email : xwu@avaya.com, xiaotaow@gmail.com > Web : http://www.cs.columbia.edu/~xiaotaow > Phone : (732) 852-2133 > SIP : sip:xiaotaow@cs.columbia.edu > Address : 1M-240, 307 Middletown Lincroft Road > Lincroft, NJ 07738-1526, U.S.A. >=20 _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Fri Aug 04 14:49:05 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G94jH-0008I4-BZ; Fri, 04 Aug 2006 14:49:03 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G94jF-0008H9-RR for sipping@ietf.org; Fri, 04 Aug 2006 14:49:01 -0400 Received: from ihemail2.lucent.com ([192.11.222.163]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G94jE-0002oZ-DA for sipping@ietf.org; Fri, 04 Aug 2006 14:49:01 -0400 Received: from ihmail.ih.lucent.com (h135-1-218-70.lucent.com [135.1.218.70]) by ihemail2.lucent.com (8.13.6/IER-o) with ESMTP id k74Ims2h007061; Fri, 4 Aug 2006 13:48:54 -0500 (CDT) Received: from [135.185.244.90] (il0015vkg1.ih.lucent.com [135.185.244.90]) by ihmail.ih.lucent.com (8.11.7p1+Sun/EMS-1.5 sol2) id k74ImrW15423; Fri, 4 Aug 2006 13:48:54 -0500 (CDT) Message-ID: <44D39695.3000208@lucent.com> Date: Fri, 04 Aug 2006 13:48:53 -0500 From: "Vijay K. Gurbani" Organization: Bell Labs Security Technology Research Group User-Agent: Mozilla Thunderbird 1.0.2 (Windows/20050317) X-Accept-Language: en-us, en MIME-Version: 1.0 To: hsinnrei@adobe.com Subject: Re: [Sipping] Question on do not disturb indication References: <24CCCC428EFEA2469BF046DB3C7A8D220C3732@namail5.corp.adobe.com> In-Reply-To: <24CCCC428EFEA2469BF046DB3C7A8D220C3732@namail5.corp.adobe.com> X-Spam-Score: 0.0 (/) X-Scan-Signature: 223e3c753032a50d5dc4443c921c3fcd Cc: SIPPING LIST X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Content-Type: multipart/mixed; boundary="===============0175058789==" Errors-To: sipping-bounces@ietf.org This is a cryptographically signed message in MIME format. --===============0175058789== Content-Type: multipart/signed; protocol="application/x-pkcs7-signature"; micalg=sha1; boundary="------------ms050600070301070903040001" This is a cryptographically signed message in MIME format. --------------ms050600070301070903040001 Content-Type: text/plain; charset=ISO-8859-1; format=flowed Content-Transfer-Encoding: 7bit Henry Sinnreich wrote: > This is also a good time to remember once more there is in a SIP a > capability called PRESENCE that is much more useful than DND and many > other legacy telephony features. :-) > > Why ignore Presence and just stick with the "old proven way"? Unfortunately, it is not that easy (or clean). The original question from John Elwell (I think) was: > What are the best practices for indicating the DND condition? Does > anyone see a need for an explicit indicator for this purpose, e.g., a > new SIP response code? We had a discussion on DND way back in January 2003 in the SIMPLE WG (links to the thread follow in a bit). If you follow that thread, you will conclude that DND is more of a state of mind that cannot be actively captured by a response code. When coupled with presence, it means that all devices that are representing my presence should show a state of CLOSED (and this was debated hotly back in 2003), otherwise if they don't then we simply train the watcher to call us even though we are not in a state to accept that call. As things stand today, I don't think I can push a DND button and make all my devices show a state of CLOSED -- some will after no activity (computer), others will if they are shut off (PDAs, cell phones, with representable avatars on a buddy list), and so on. For some institutional memory, the folks who are active in this thread should really look at the thread we had way back in January 2003 on this subject. The link is: http://www1.ietf.org/mail-archive/web/simple/current/msg00144.html The rest of the thread can be followed from http://www1.ietf.org/mail-archive/web/simple/current/mail265.html Thanks. - vijay -- Vijay K. 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questions on current sip Use sip@ietf.org for new developments of core SIP --===============0175058789==-- From sipping-bounces@ietf.org Fri Aug 04 15:13:56 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G957K-0001xQ-Cy; Fri, 04 Aug 2006 15:13:54 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G957J-0001tn-FY for sipping@ietf.org; Fri, 04 Aug 2006 15:13:53 -0400 Received: from zcars04f.nortel.com ([47.129.242.57]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G957H-0005Di-2I for sipping@ietf.org; Fri, 04 Aug 2006 15:13:53 -0400 Received: from zrc2hxm2.corp.nortel.com (zrc2hxm2.corp.nortel.com [47.103.123.73]) by zcars04f.nortel.com (Switch-2.2.6/Switch-2.2.0) with ESMTP id k74JDkQ22249; Fri, 4 Aug 2006 15:13:47 -0400 (EDT) X-MimeOLE: Produced By Microsoft Exchange V6.5.7226.0 Content-class: urn:content-classes:message MIME-Version: 1.0 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: quoted-printable Subject: RE: [Sipping] Question on do not disturb indication Date: Fri, 4 Aug 2006 14:13:37 -0500 Message-ID: <6FC4416DDE56C44DA0AEE67BC7CA43710FC3AEED@zrc2hxm2.corp.nortel.com> In-Reply-To: <1ECE0EB50388174790F9694F77522CCF0BFA3444@zrc2hxm0.corp.nortel.com> X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: [Sipping] Question on do not disturb indication thread-index: Aca32zJ6QUBKIWcnRUaAqQ2SrPFC/wAB9R8gAAW+4EA= From: "Brian Stucker" To: "Francois Audet" , "Jonathan Rosenberg" , "Paul Kyzivat" X-Spam-Score: 0.0 (/) X-Scan-Signature: a7d2e37451f7f22841e3b6f40c67db0f Cc: sipping@ietf.org X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org I was thinking this would be something for MiG to look at. The cat is not without skin, it's just in how we wish to remove it. Regards, Brian=20 > -----Original Message----- > From: Audet, Francois (SC100:3055)=20 > Sent: Friday, August 04, 2006 12:43 PM > To: Jonathan Rosenberg; Paul Kyzivat > Cc: sipping@ietf.org > Subject: RE: [Sipping] Question on do not disturb indication >=20 > Agreed: it does seem like a perfect case for MIG next meeting.=20 >=20 >=20 >=20 > > -----Original Message----- > > From: Jonathan Rosenberg [mailto:jdrosen@cisco.com] > > Sent: Friday, August 04, 2006 8:29 AM > > To: Paul Kyzivat > > Cc: sipping@ietf.org > > Subject: Re: [Sipping] Question on do not disturb indication > >=20 > > I agree with Paul and Dale that the current prescribed behavior for=20 > > 6xx makes it hard to apply to the use case we're talking about here. > >=20 > > I think its worth taking a step back here and discussing=20 > the broader=20 > > problem, which is one I recall we've touched on a few times but has=20 > > never really been resolved. > >=20 > > The general problem is how a UA can impact routing logic in=20 > a proxy at=20 > > the time of call setup. We have several use cases here that are=20 > > important: > >=20 > > 1. when a user has multiple devices, and a receives a call,=20 > they would=20 > > like to press a button on their phone called 'send to=20 > voicemail' which=20 > > causes the call to go to voicemail. This should terminate any other=20 > > branches for other UAs for that user if any are in=20 > progress, and halt=20 > > any sequential search for othe UAs for that user if any=20 > were planned,=20 > > and go to voicemail. > >=20 > > 2. Like use case 1, except that the user wants the call to=20 > be rejected=20 > > outright, and not even sent to voicemail. > >=20 > > 3. Like use case 1, except that the user wants the caller to hear=20 > > ringing for a while more and then get voicemail (sort of a polite=20 > > block, where the caller can't differentiate this case from=20 > one where=20 > > the called party wasn't there. Dean alluded to this). > >=20 > > 4. send-to-X: the user wants the caller to be immediately=20 > redirected=20 > > to somewhere else > >=20 > >=20 > > These cases all share the common problem that, its not that=20 > a response=20 > > is being returend from a particular UA instance (in which=20 > case proxy=20 > > logic should try others), but rather, a response is being returned=20 > > from the *user* that is meant to ask the proxy to definitiely do=20 > > something else. > >=20 > > There are sevearl ways one might solve this problem: > >=20 > > 1. The UA uploads a CPL script when it registers; that CPL script=20 > > tells the proxy what to do based on specific response codes. For=20 > > example, it can tell the proxy that a 476 means forward to=20 > voicemail. > > This has the > > benefit of not requiring standardized definitions of codes, and of=20 > > leaving a lot of flexibility for different services.=20 > However, it falls=20 > > apart when there are multiple UA in use and requires CPL=20 > support which=20 > > has not seen wide success for many reasons. > >=20 > > 2. We define one or more new response codes which have the desired=20 > > effects. I think this is going to be hard actually, given that the=20 > > four use cases above each might require a new response code > >=20 > > 3. We have a header field in the response which tells the=20 > proxy that=20 > > this response shuold be treated as if it came from *all* UA=20 > instances=20 > > forked to from the AOR, thus cancelling all other forks for=20 > that AOR. > >=20 > >=20 > > It feels to me like we need some requirements work and problem=20 > > definition here. I'd argue that this is perfect fodder for=20 > our MIG BoF=20 > > next IETF. >=20 > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP=20 > Use sip-implementors@cs.columbia.edu for questions on current=20 > sip Use sip@ietf.org for new developments of core SIP >=20 _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Fri Aug 04 15:16:02 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G959N-0003Gl-TD; Fri, 04 Aug 2006 15:16:01 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G959M-0003Gb-6I for sipping@ietf.org; Fri, 04 Aug 2006 15:16:00 -0400 Received: from nylon.softarmor.com ([66.135.38.164]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G959K-0005W7-Qv for sipping@ietf.org; Fri, 04 Aug 2006 15:16:00 -0400 Received: from [64.101.149.124] (mikwils2-w2k.cisco.com [64.101.149.124]) (authenticated bits=0) by nylon.softarmor.com (8.13.1/8.13.1) with ESMTP id k74JI44S012957 (version=TLSv1/SSLv3 cipher=RC4-SHA bits=128 verify=NO); Fri, 4 Aug 2006 14:18:05 -0500 In-Reply-To: <44D39695.3000208@lucent.com> References: <24CCCC428EFEA2469BF046DB3C7A8D220C3732@namail5.corp.adobe.com> <44D39695.3000208@lucent.com> Mime-Version: 1.0 (Apple Message framework v752.2) Content-Type: text/plain; charset=US-ASCII; delsp=yes; format=flowed Message-Id: <6C2A6DF5-2057-43D5-B6F0-86060A66BBE6@softarmor.com> Content-Transfer-Encoding: 7bit From: Dean Willis Subject: Re: [Sipping] Question on do not disturb indication Date: Fri, 4 Aug 2006 14:15:54 -0500 To: "Vijay K. Gurbani" X-Mailer: Apple Mail (2.752.2) X-Spam-Score: 0.0 (/) X-Scan-Signature: d6b246023072368de71562c0ab503126 Cc: SIPPING LIST X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org On Aug 4, 2006, at 1:48 PM, Vijay K. Gurbani wrote: > When > coupled with presence, it means that all devices that are > representing my presence should show a state of CLOSED (and this > was debated hotly back in 2003), otherwise if they don't then > we simply train the watcher to call us even though we are not in > a state to accept that call. As things stand today, I don't > think I can push a DND button and make all my devices show a > state of CLOSED -- some will after no activity (computer), > others will if they are shut off (PDAs, cell phones, with > representable avatars on a buddy list), and so on. So when a request shows up at a device that is CLOSED, how should the device respond? -- Dean _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Fri Aug 04 15:17:16 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G95AZ-00040r-UL; Fri, 04 Aug 2006 15:17:15 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G95AY-00040l-L4 for sipping@ietf.org; Fri, 04 Aug 2006 15:17:14 -0400 Received: from stsc1260-eth-s1-s1p1-vip.va.neustar.com ([156.154.16.129] helo=chiedprmail1.ietf.org) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G8zey-0007TV-Vr for sipping@ietf.org; Fri, 04 Aug 2006 09:24:17 -0400 Received: from rtp-iport-2.cisco.com ([64.102.122.149]) by chiedprmail1.ietf.org with esmtp (Exim 4.43) id 1G8zTo-0008KT-NQ for sipping@ietf.org; Fri, 04 Aug 2006 09:12:49 -0400 Received: from rtp-dkim-2.cisco.com ([64.102.121.159]) by rtp-iport-2.cisco.com with ESMTP; 04 Aug 2006 09:12:44 -0400 X-IronPort-AV: i="4.07,211,1151899200"; d="scan'208"; a="95114047:sNHT26648084" Received: from rtp-core-2.cisco.com (rtp-core-2.cisco.com [64.102.124.13]) by rtp-dkim-2.cisco.com (8.12.11.20060308/8.12.11) with ESMTP id k74DCi2g029274; Fri, 4 Aug 2006 09:12:44 -0400 Received: from xbh-rtp-201.amer.cisco.com (xbh-rtp-201.cisco.com [64.102.31.12]) by rtp-core-2.cisco.com (8.12.10/8.12.6) with ESMTP id k74DCde2012043; Fri, 4 Aug 2006 09:12:39 -0400 (EDT) Received: from xfe-rtp-202.amer.cisco.com ([64.102.31.21]) by xbh-rtp-201.amer.cisco.com with Microsoft SMTPSVC(6.0.3790.1830); Fri, 4 Aug 2006 09:12:39 -0400 Received: from [161.44.79.104] ([161.44.79.104]) by xfe-rtp-202.amer.cisco.com with Microsoft SMTPSVC(6.0.3790.1830); Fri, 4 Aug 2006 09:12:39 -0400 Message-ID: <44D347C6.7070502@cisco.com> Date: Fri, 04 Aug 2006 09:12:38 -0400 From: Paul Kyzivat User-Agent: Thunderbird 1.5.0.5 (Windows/20060719) MIME-Version: 1.0 To: Dean Willis Subject: Re: [Sipping] Question on do not disturb indication References: <50B1CBA96870A34799A506B2313F266709897F68@ntht201e.siemenscomms.co.uk> <1B5A1FDA-AA89-4B8E-807E-2F170054D828@softarmor.com> In-Reply-To: <1B5A1FDA-AA89-4B8E-807E-2F170054D828@softarmor.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Content-Transfer-Encoding: 7bit X-OriginalArrivalTime: 04 Aug 2006 13:12:39.0236 (UTC) FILETIME=[A93D0040:01C6B7C7] DKIM-Signature: a=rsa-sha1; q=dns; l=1859; t=1154697164; x=1155561164; c=relaxed/simple; s=rtpdkim2001; h=Content-Type:From:Subject:Content-Transfer-Encoding:MIME-Version; d=cisco.com; i=pkyzivat@cisco.com; z=From:Paul=20Kyzivat=20 |Subject:Re=3A=20[Sipping]=20Question=20on=20do=20not=20disturb=20indication |To:Dean=20Willis=20; X=v=3Dcisco.com=3B=20h=3Dyu9ohsv1kFqX2uC2CMjguD3sTnY=3D; b=pvCfy9w7lBVaYKEmN9VVGvvesAjfwpruBYCV6rH4ih+U3ckCodB8doB0nTQ2L3HzUBcNf1Ea k0WPQhMS3/xkHI813RvSwO2gJTFBKHnL/VskYxsA2n4UA3NrnWuTnh/t; Authentication-Results: rtp-dkim-2.cisco.com; header.From=pkyzivat@cisco.com; dkim=pass ( sig from cisco.com verified; ); X-Spam-Score: -2.3 (--) X-Scan-Signature: 4adaf050708fb13be3316a9eee889caa Cc: "Dolly, Martin C, ALABS" , sipping@ietf.org, "Elwell, John" , Scott Lawrence X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Dean Willis wrote: > If we believe that we'd like to convey a distinction between > "Temporarily unavailable" and "Temporarily unavailable due to subscriber > choice" to a human, perhaps a reason phrase on a 480 would be adequate? > Like "480 Do Not Disturb", maybe. I have never understood the purpose of reason phrases, or the value of sending different ones with the same response code. Doing so implies an expectation that the UAC will do something different as a result. Its hard to do something different algorithmically, because there are no useful specs on what can be there. (Some UASs may choose to send the reason phrases translated into the language of the UAS developer - say Klingon.) About the best one could hope for would be for some UACs to *display* the value, uninterpretted. But that only works for UACs that have the capability to do so, and that in fact do so. Realistically, providing that is only useful to geeks. > Now, I will say that Martin is right in that there might be many reasons > to reject a call, that DND is just the tip of the iceberg, and that > there's no way we can define them all. Right. There *is* a mechanism defined for this: Error-Info. There is good and bad to it: - good: because the value is a URI (or several) it is arbitrarily extensible to new reasons - good: it can make it possible for the UAC to obtain something that can be rendered to the calling user, without the UAC needing to know all possible reasons. - bad: its still hard for the UAC or a proxy to take programatic action based on the reason because there are no standards for what the reasons might be. It is possible that two different UASs might choose different URIs to represent the same reason. On balance, this seems like a reasonable approach to me. Paul _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Fri Aug 04 15:24:59 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G95I3-00049S-G6; Fri, 04 Aug 2006 15:24:59 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G95I1-00049N-JV for sipping@ietf.org; Fri, 04 Aug 2006 15:24:57 -0400 Received: from ihemail3.lucent.com ([135.245.0.37]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G95I0-0006vA-AK for sipping@ietf.org; Fri, 04 Aug 2006 15:24:57 -0400 Received: from ihmail.ih.lucent.com (h135-1-218-70.lucent.com [135.1.218.70]) by ihemail3.lucent.com (8.13.6/IER-o) with ESMTP id k74JOtiU009735; Fri, 4 Aug 2006 14:24:56 -0500 (CDT) Received: from [135.185.244.90] (il0015vkg1.ih.lucent.com [135.185.244.90]) by ihmail.ih.lucent.com (8.11.7p1+Sun/EMS-1.5 sol2) id k74JOtW08657; Fri, 4 Aug 2006 14:24:55 -0500 (CDT) Message-ID: <44D39F07.6050705@lucent.com> Date: Fri, 04 Aug 2006 14:24:55 -0500 From: "Vijay K. Gurbani" Organization: Bell Labs Security Technology Research Group User-Agent: Mozilla Thunderbird 1.0.2 (Windows/20050317) X-Accept-Language: en-us, en MIME-Version: 1.0 To: Dean Willis Subject: Re: [Sipping] Question on do not disturb indication References: <24CCCC428EFEA2469BF046DB3C7A8D220C3732@namail5.corp.adobe.com> <44D39695.3000208@lucent.com> <6C2A6DF5-2057-43D5-B6F0-86060A66BBE6@softarmor.com> In-Reply-To: <6C2A6DF5-2057-43D5-B6F0-86060A66BBE6@softarmor.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Content-Transfer-Encoding: 7bit X-Spam-Score: 0.0 (/) X-Scan-Signature: 2409bba43e9c8d580670fda8b695204a Cc: SIPPING LIST X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Dean Willis wrote: > So when a request shows up at a device that is CLOSED, how should the > device respond? Dean: Are you trying to bait me, for I am sure you know the answer to that question... In an ideal world, a request would not even show up when a device is CLOSED. In our non-ideal world, a 4xx type response seems appropriate. But DND also means that a request will not succeed on other devices either; so a 6xx is more appropriate. But it is one thing to have a response code, quite another to uniformly apply the DND directive across all the devices that contribute to my presence -- and that was what I was trying to point out in my email regarding institutional memory. Thanks. - vijay -- Vijay K. Gurbani vkg@{lucent.com,research.bell-labs.com,acm.org} Bell Laboratories, Lucent Technologies, Inc. 2701 Lucent Lane, Rm. 9F-546, Lisle, Illinois 60532 (USA) _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Fri Aug 04 16:00:46 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G95qf-0000Ai-4w; Fri, 04 Aug 2006 16:00:45 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G95qd-0000AZ-3e for sipping@ietf.org; Fri, 04 Aug 2006 16:00:43 -0400 Received: from zcars04f.nortel.com ([47.129.242.57]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G95qa-0005mS-L7 for sipping@ietf.org; Fri, 04 Aug 2006 16:00:43 -0400 Received: from zrc2hxm0.corp.nortel.com (zrc2hxm0.corp.nortel.com [47.103.123.71]) by zcars04f.nortel.com (Switch-2.2.6/Switch-2.2.0) with ESMTP id k74K0XQ08035; Fri, 4 Aug 2006 16:00:33 -0400 (EDT) X-MimeOLE: Produced By Microsoft Exchange V6.5.7226.0 Content-class: urn:content-classes:message MIME-Version: 1.0 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: quoted-printable Subject: RE: [Sipping] Question on do not disturb indication Date: Fri, 4 Aug 2006 15:00:31 -0500 Message-ID: <1ECE0EB50388174790F9694F77522CCF0BFA37D9@zrc2hxm0.corp.nortel.com> In-Reply-To: <24CCCC428EFEA2469BF046DB3C7A8D220C3732@namail5.corp.adobe.com> X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: [Sipping] Question on do not disturb indication Thread-Index: Aca32LAzyAkK+PrcTGulbjdEiWLfyAAFBIdQAATTFmA= From: "Francois Audet" To: "Henry Sinnreich" , "Jonathan Rosenberg" , "Dean Willis" X-Spam-Score: 0.0 (/) X-Scan-Signature: 1449ead51a2ff026dcb23465f5379250 Cc: "Dolly, Martin C, ALABS" , Paul Kyzivat , "Elwell, John" , sipping@ietf.org, Scott Lawrence X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Actually, this is a very good point. There are really 2 aspects to "Do Not Disturb": 1 - On aspect for the terminating side to "reject" a session. That's the one I was focussing on. That's the one where many of us are not convinced that we need something beyond what is already allowed with SIP (480, 486, 6XX, etc.). 2 - The other aspect is conveying a DND indication to the other end. I guess John was thinking about that. Henry is right that this aspect is definitively=20 something appropriate for presence... > -----Original Message----- > From: Henry Sinnreich [mailto:hsinnrei@adobe.com]=20 > Sent: Friday, August 04, 2006 10:54 AM > To: Jonathan Rosenberg; Dean Willis > Cc: Dolly, Martin C, ALABS; Paul Kyzivat; Elwell, John;=20 > sipping@ietf.org; Scott Lawrence > Subject: RE: [Sipping] Question on do not disturb indication >=20 > This is also a good time to remember once more there is in a=20 > SIP a capability called PRESENCE that is much more useful=20 > than DND and many other legacy telephony features. :-) >=20 > Why ignore Presence and just stick with the "old proven way"? >=20 > Thanks, Henry >=20 > -----Original Message----- > From: Jonathan Rosenberg [mailto:jdrosen@cisco.com] > Sent: Friday, August 04, 2006 8:12 AM > To: Dean Willis > Cc: Dolly, Martin C, ALABS; Paul Kyzivat; Elwell,John;=20 > sipping@ietf.org; Scott Lawrence > Subject: Re: [Sipping] Question on do not disturb indication >=20 > I agree with Dean here. >=20 > Why do you want to know, at the calling UA side, that the=20 > call was busy because of DND? Whether you need a new code=20 > depends on the answer: >=20 > 1. if you want the human user to know that this was DND so=20 > they can do something appropriate, a reason phrase is exactly=20 > the right thing >=20 > 2. if you want to have the calling UA to do something=20 > different, what would it do differently on a 'your call was=20 > rejected by DND' vs. 'your call was rejected because the user=20 > is on another call'? If the answer is >=20 > nothing, then you don't need a new response code. >=20 > I'm also leary about defining new response codes to do a=20 > specific terminating application that is running. Saying that=20 > this was rejected due to 'DND' requiers you to have a=20 > standard definition for DND. I don't >=20 > want to do that. As others have pointed out, a reasonable=20 > implementation >=20 > for DND is that the termianting network simply lets the phone=20 > ring and eventually responds with a 480. That should be allowed. >=20 > Thanks, > Jonathan R. >=20 > Dean Willis wrote: >=20 > >=20 > > On Aug 3, 2006, at 1:28 AM, Elwell, John wrote: > >=20 > >> Dean, > >> > >> So what would you like the IETF to recommend? Do you believe 480 > mostly > >> works, or do you think something new is needed? > >> > >> John > >=20 > >=20 > >=20 > > Hey, I'm just a consumer. I don't want to know the=20 > technical details. > I=20 > > just want it to work, mostly. > >=20 > > If it were up to me, I'd be asking the question: > >=20 > > What part of DND is important for a calling human to understand, vs. > a=20 > > calling automata? Is there a distinction needed? > >=20 > > If we believe that we'd like to convey a distinction between=20 > > "Temporarily unavailable" and "Temporarily unavailable due to=20 > > subscriber choice" to a human, perhaps a reason phrase on a=20 > 480 would >=20 > > be adequate? Like "480 Do Not Disturb", maybe. > >=20 > >=20 > > An automata is just likely to retry the call periodically=20 > anyhow, but > a=20 > > human might stop and think "Oh yeah, it IS the middle of=20 > the night in >=20 > > Singapore . . .". > >=20 > > Question: How beneficial would a Retry-After value be in rate- > limiting=20 > > the retries of automata? > >=20 > > Now, I will say that Martin is right in that there might be many=20 > > reasons to reject a call, that DND is just the tip of the iceberg, > and=20 > > that there's no way we can define them all. > >=20 > >=20 > > Now, here's an interesting thought. If I use a reason=20 > phrase on 480,=20 > > and couple it to the right software, I can use SIP INVITE=20 > requests as > a=20 > > covert-channel replacement for HTTP GET operations (or file=20 > transfer, >=20 > > or whatever) , but peer-to-peer using the SIP infrastructure as a=20 > > rendezvous service. Don't try this at home, kids. > >=20 > > -- > > Dean > >=20 > > _______________________________________________ > > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > > This list is for NEW development of the application of SIP Use=20 > > sip-implementors@cs.columbia.edu for questions on current sip Use=20 > > sip@ietf.org for new developments of core SIP > >=20 >=20 > --=20 > Jonathan D. Rosenberg, Ph.D. 600 Lanidex Plaza > Cisco Fellow Parsippany, NJ=20 > 07054-2711 > Cisco Systems > jdrosen@cisco.com FAX: (973) 952-5050 > http://www.jdrosen.net PHONE: (973) 952-5000 > http://www.cisco.com >=20 > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP=20 > Use sip-implementors@cs.columbia.edu for questions on current=20 > sip Use sip@ietf.org for new developments of core SIP >=20 > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP=20 > Use sip-implementors@cs.columbia.edu for questions on current=20 > sip Use sip@ietf.org for new developments of core SIP >=20 _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Fri Aug 04 16:04:58 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G95uk-0002mc-Ia; Fri, 04 Aug 2006 16:04:58 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G95uj-0002mX-PO for sipping@ietf.org; Fri, 04 Aug 2006 16:04:57 -0400 Received: from exprod6og50.obsmtp.com ([64.18.1.181]) by ietf-mx.ietf.org with smtp (Exim 4.43) id 1G95ui-0006K1-9f for sipping@ietf.org; Fri, 04 Aug 2006 16:04:57 -0400 Received: from source ([192.150.11.134]) by exprod6ob50.postini.com ([64.18.5.12]) with SMTP; Fri, 04 Aug 2006 13:04:51 PDT Received: from inner-relay-1.corp.adobe.com ([153.32.1.51]) by outbound-smtp-1.corp.adobe.com (8.12.10/8.12.10) with ESMTP id k74K2vBl004751; Fri, 4 Aug 2006 13:03:01 -0700 (PDT) Received: from fe2.corp.adobe.com (fe2.corp.adobe.com [10.8.192.72]) by inner-relay-1.corp.adobe.com (8.12.10/8.12.10) with ESMTP id k74K4k4b008148; Fri, 4 Aug 2006 13:04:46 -0700 (PDT) Received: from namail5.corp.adobe.com ([10.8.192.88]) by fe2.corp.adobe.com with Microsoft SMTPSVC(6.0.3790.1830); Fri, 4 Aug 2006 13:04:45 -0700 X-MimeOLE: Produced By Microsoft Exchange V6.5 Content-class: urn:content-classes:message MIME-Version: 1.0 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: quoted-printable Subject: RE: [Sipping] Question on do not disturb indication Date: Fri, 4 Aug 2006 13:03:30 -0700 Message-ID: <24CCCC428EFEA2469BF046DB3C7A8D220C3765@namail5.corp.adobe.com> In-Reply-To: <44D39695.3000208@lucent.com> X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: [Sipping] Question on do not disturb indication Thread-Index: Aca39qhBnerc1Vy7SUq3sLkUPBC6aAAAEBIA From: "Henry Sinnreich" To: "Vijay K. Gurbani" X-OriginalArrivalTime: 04 Aug 2006 20:04:45.0258 (UTC) FILETIME=[3B188AA0:01C6B801] X-Spam-Score: 0.0 (/) X-Scan-Signature: e8a67952aa972b528dd04570d58ad8fe Cc: SIPPING LIST X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org > DND is more of a state of > mind that cannot be actively captured by a response code. =20 Exactly! That why all commercial IM's have presence icons and the SIMPLE WG is developing RPID, see http://www3.ietf.org/proceedings/06mar/IDs/draft-ietf-simple-rpid-10.txt One of the authors is V. Gurbani :-) Thanks, Henry -----Original Message----- From: Vijay K. Gurbani [mailto:vkg@lucent.com]=20 Sent: Friday, August 04, 2006 11:49 AM To: Henry Sinnreich Cc: SIPPING LIST Subject: Re: [Sipping] Question on do not disturb indication Henry Sinnreich wrote: > This is also a good time to remember once more there is in a SIP a=20 > capability called PRESENCE that is much more useful than DND and many > other legacy telephony features. :-) >=20 > Why ignore Presence and just stick with the "old proven way"? Unfortunately, it is not that easy (or clean). The original question from John Elwell (I think) was: > What are the best practices for indicating the DND condition? Does > anyone see a need for an explicit indicator for this purpose, e.g., a > new SIP response code? We had a discussion on DND way back in January 2003 in the SIMPLE WG (links to the thread follow in a bit). If you follow that thread, you will conclude that DND is more of a state of mind that cannot be actively captured by a response code. When coupled with presence, it means that all devices that are representing my presence should show a state of CLOSED (and this was debated hotly back in 2003), otherwise if they don't then we simply train the watcher to call us even though we are not in a state to accept that call. As things stand today, I don't think I can push a DND button and make all my devices show a state of CLOSED -- some will after no activity (computer), others will if they are shut off (PDAs, cell phones, with representable avatars on a buddy list), and so on. For some institutional memory, the folks who are active in this thread should really look at the thread we had way back in January 2003 on this subject. The link is: http://www1.ietf.org/mail-archive/web/simple/current/msg00144.html The rest of the thread can be followed from http://www1.ietf.org/mail-archive/web/simple/current/mail265.html Thanks. - vijay --=20 Vijay K. Gurbani vkg@{lucent.com,research.bell-labs.com,acm.org} Bell Laboratories, Lucent Technologies, Inc. 2701 Lucent Lane, Rm. 9F-546, Lisle, Illinois 60532 (USA) _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Fri Aug 04 17:22:04 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G977G-0000E9-62; Fri, 04 Aug 2006 17:21:58 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G977F-0000A0-7n for sipping@ietf.org; Fri, 04 Aug 2006 17:21:57 -0400 Received: from ihemail2.lucent.com ([192.11.222.163]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G977B-00072o-UQ for sipping@ietf.org; Fri, 04 Aug 2006 17:21:57 -0400 Received: from ihmail.ih.lucent.com (h135-1-218-70.lucent.com [135.1.218.70]) by ihemail2.lucent.com (8.13.6/IER-o) with ESMTP id k74LLrnk014268; Fri, 4 Aug 2006 16:21:53 -0500 (CDT) Received: from [135.185.244.90] (il0015vkg1.ih.lucent.com [135.185.244.90]) by ihmail.ih.lucent.com (8.11.7p1+Sun/EMS-1.5 sol2) id k74LLqW24047; Fri, 4 Aug 2006 16:21:52 -0500 (CDT) Message-ID: <44D3BA70.8040208@lucent.com> Date: Fri, 04 Aug 2006 16:21:52 -0500 From: "Vijay K. Gurbani" Organization: Bell Labs Security Technology Research Group User-Agent: Mozilla Thunderbird 1.0.2 (Windows/20050317) X-Accept-Language: en-us, en MIME-Version: 1.0 To: Paul Kyzivat Subject: Re: [Sipping] Question on do not disturb indication References: <50B1CBA96870A34799A506B2313F266709897F68@ntht201e.siemenscomms.co.uk> <1B5A1FDA-AA89-4B8E-807E-2F170054D828@softarmor.com> <44D347C6.7070502@cisco.com> In-Reply-To: <44D347C6.7070502@cisco.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Content-Transfer-Encoding: 7bit X-Spam-Score: 0.0 (/) X-Scan-Signature: 8b431ad66d60be2d47c7bfeb879db82c Cc: sipping@ietf.org X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Paul Kyzivat wrote: > I have never understood the purpose of reason phrases, or the value of > sending different ones with the same response code. Paul: From a development point of view, reason phrases are simply fantastic. Why ... inline. > Doing so implies an expectation that the UAC will do something different > as a result. I don't view it that way. Take for instance the 400 Bad Request response. In and of itself, it only says that the request was malformed. But, when coupled with an descriptive phrase, the intent of why the response was issued becomes clearer: 400 Missing Call-ID Header 400 Missing Contact Header 400 Multiple To Headers and so on. Here, to the receiving UAC, the 4xx puts it in a certain state. The state would be the same regardless of the reason phrase. But if the UAC has display capabilities (and many, if not all do), then this becomes another tool for imparting information to the user (or the developer, as the case may be). Another example is the following 200 OK for a REG request: 200 OK 200 Registration Expired The first 200 OK tells the UAC that the registration was accepted. If the UAC later sends a REG w/ Expires:0, then the reason phrase "Registration Expired" conveys appropriate semantic information that to me is better than a simple "OK". > About the best one could hope for would be for some UACs to *display* > the value, uninterpretted. But that only works for UACs that have the > capability to do so, and that in fact do so. Realistically, providing > that is only useful to geeks. Or implementors [(who are geeks by definition, anyway ;-)] Thanks. - vijay -- Vijay K. Gurbani vkg@{lucent.com,research.bell-labs.com,acm.org} Bell Laboratories, Lucent Technologies, Inc. 2701 Lucent Lane, Rm. 9F-546, Lisle, Illinois 60532 (USA) _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Fri Aug 04 17:42:22 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G97Qw-0002t0-K2; Fri, 04 Aug 2006 17:42:18 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G97Qv-0002sW-35 for sipping@ietf.org; Fri, 04 Aug 2006 17:42:17 -0400 Received: from stsc1260-eth-s1-s1p1-vip.va.neustar.com ([156.154.16.129] helo=chiedprmail1.ietf.org) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G96Fx-0008Va-G7 for sipping@ietf.org; Fri, 04 Aug 2006 16:26:53 -0400 Received: from ihemail1.lucent.com ([192.11.222.161]) by chiedprmail1.ietf.org with esmtp (Exim 4.43) id 1G961E-0007xh-Lz for sipping@ietf.org; Fri, 04 Aug 2006 16:11:42 -0400 Received: from ihmail.ih.lucent.com (h135-1-218-70.lucent.com [135.1.218.70]) by ihemail1.lucent.com (8.13.6/IER-o) with ESMTP id k74KBZMX004941; Fri, 4 Aug 2006 15:11:35 -0500 (CDT) Received: from [135.185.244.90] (il0015vkg1.ih.lucent.com [135.185.244.90]) by ihmail.ih.lucent.com (8.11.7p1+Sun/EMS-1.5 sol2) id k74KBYW09236; Fri, 4 Aug 2006 15:11:34 -0500 (CDT) Message-ID: <44D3A9F6.8070004@lucent.com> Date: Fri, 04 Aug 2006 15:11:34 -0500 From: "Vijay K. Gurbani" Organization: Bell Labs Security Technology Research Group User-Agent: Mozilla Thunderbird 1.0.2 (Windows/20050317) X-Accept-Language: en-us, en MIME-Version: 1.0 To: Henry Sinnreich Subject: Re: [Sipping] Question on do not disturb indication References: <24CCCC428EFEA2469BF046DB3C7A8D220C3765@namail5.corp.adobe.com> In-Reply-To: <24CCCC428EFEA2469BF046DB3C7A8D220C3765@namail5.corp.adobe.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Content-Transfer-Encoding: 7bit X-Spam-Score: -2.6 (--) X-Scan-Signature: 7655788c23eb79e336f5f8ba8bce7906 Cc: SIPPING LIST X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Henry Sinnreich wrote: >>DND is more of a state of >>mind that cannot be actively captured by a response code. > > Exactly! > > That why all commercial IM's have presence icons and the SIMPLE WG is > developing RPID, see > http://www3.ietf.org/proceedings/06mar/IDs/draft-ietf-simple-rpid-10.txt > > One of the authors is V. Gurbani :-) Thanks, Henry; yes, I do remember. The thread I pointed out was the successor of another thread that eventually lead to the creation of RPID (which became an RFC recently -- RFC 4480 ;-) ) Ciao. - vijay -- Vijay K. Gurbani vkg@{lucent.com,research.bell-labs.com,acm.org} Bell Laboratories, Lucent Technologies, Inc. 2701 Lucent Lane, Rm. 9F-546, Lisle, Illinois 60532 (USA) _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Sat Aug 05 15:41:36 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G9S1T-0002mI-OX; Sat, 05 Aug 2006 15:41:23 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G9S1R-0002ls-Kt for sipping@ietf.org; Sat, 05 Aug 2006 15:41:21 -0400 Received: from rat.iptel.org ([213.192.59.67] helo=mail.iptel.org) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G9S1Q-0003VI-5B for sipping@ietf.org; Sat, 05 Aug 2006 15:41:21 -0400 Received: from uk1lt10071.iptel.org (shell.iptel.org [213.192.59.74]) by mail.iptel.org (Postfix) with ESMTP id 6B66920A2E1; Sat, 5 Aug 2006 19:40:57 +0000 (UTC) Message-Id: <7.0.1.0.0.20060805213927.0241ce48@iptel.org> X-Mailer: QUALCOMM Windows Eudora Version 7.0.1.0 Date: Sat, 05 Aug 2006 21:40:50 +0200 To: "Vijay K. Gurbani" , Paul Kyzivat From: Jiri Kuthan Subject: Re: [Sipping] Question on do not disturb indication In-Reply-To: <44D3BA70.8040208@lucent.com> References: <50B1CBA96870A34799A506B2313F266709897F68@ntht201e.siemenscomms.co.uk> <1B5A1FDA-AA89-4B8E-807E-2F170054D828@softarmor.com> <44D347C6.7070502@cisco.com> <44D3BA70.8040208@lucent.com> Mime-Version: 1.0 Content-Type: text/plain; charset="us-ascii" X-Spam-Score: 0.0 (/) X-Scan-Signature: e8a67952aa972b528dd04570d58ad8fe Cc: sipping@ietf.org X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Other example is 403 -- I don't think there is enough error code space for all possible reasons why a request is prohibited from execution. If not for other reasons, than for troubleshooting reason phrases seem a great deal to me. -jiri At 23:21 04/08/2006, Vijay K. Gurbani wrote: >Paul Kyzivat wrote: >>I have never understood the purpose of reason phrases, or the value of sending different ones with the same response code. > >Paul: From a development point of view, reason phrases are >simply fantastic. Why ... inline. > >>Doing so implies an expectation that the UAC will do something different as a result. > >I don't view it that way. Take for instance the 400 Bad Request >response. In and of itself, it only says that the request was >malformed. But, when coupled with an descriptive phrase, the >intent of why the response was issued becomes clearer: > > 400 Missing Call-ID Header > 400 Missing Contact Header > 400 Multiple To Headers > >and so on. > >Here, to the receiving UAC, the 4xx puts it in a certain state. >The state would be the same regardless of the reason phrase. >But if the UAC has display capabilities (and many, if not all >do), then this becomes another tool for imparting information >to the user (or the developer, as the case may be). > >Another example is the following 200 OK for a REG request: > > 200 OK > 200 Registration Expired > >The first 200 OK tells the UAC that the registration was >accepted. If the UAC later sends a REG w/ Expires:0, then >the reason phrase "Registration Expired" conveys appropriate >semantic information that to me is better than a simple "OK". > >>About the best one could hope for would be for some UACs to *display* the value, uninterpretted. But that only works for UACs that have the capability to do so, and that in fact do so. Realistically, providing that is only useful to geeks. > >Or implementors [(who are geeks by definition, anyway ;-)] > >Thanks. > >- vijay >-- >Vijay K. Gurbani vkg@{lucent.com,research.bell-labs.com,acm.org} >Bell Laboratories, Lucent Technologies, Inc. >2701 Lucent Lane, Rm. 9F-546, Lisle, Illinois 60532 (USA) > >_______________________________________________ >Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping >This list is for NEW development of the application of SIP >Use sip-implementors@cs.columbia.edu for questions on current sip >Use sip@ietf.org for new developments of core SIP -- Jiri Kuthan http://iptel.org/~jiri/ _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Sun Aug 06 09:53:10 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G9j3g-0001jw-8r; Sun, 06 Aug 2006 09:52:48 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G9j3e-0001jr-54 for sipping@ietf.org; Sun, 06 Aug 2006 09:52:46 -0400 Received: from mailgate.siemenscomms.co.uk ([195.171.110.225] helo=bemg01.siemenscomms.co.uk) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G9j3c-0004Zx-Qk for sipping@ietf.org; Sun, 06 Aug 2006 09:52:46 -0400 Received: from CONVERSION-DAEMON.siemenscomms.co.uk by siemenscomms.co.uk (PMDF V6.0-24 #40642) id <0J3K00H01X8F57@siemenscomms.co.uk> for sipping@ietf.org; Sun, 06 Aug 2006 14:53:03 +0100 (BST) Received: from ntht207e.uksgcs.siemenscomms.co.uk ([137.223.247.82]) by siemenscomms.co.uk (PMDF V6.0-24 #40642) with ESMTP id <0J3K00H03X8F56@siemenscomms.co.uk>; Sun, 06 Aug 2006 14:53:03 +0100 (BST) Received: by ntht207e.uksgcs.siemenscomms.co.uk with Internet Mail Service (5.5.2657.72) id ; Sun, 06 Aug 2006 14:52:42 +0100 Content-return: allowed Date: Sun, 06 Aug 2006 14:52:33 +0100 From: "Elwell, John" Subject: RE: [Sipping] Question on do not disturb indication To: Francois Audet Message-id: <50B1CBA96870A34799A506B2313F2667098D8527@ntht201e.siemenscomms.co.uk> MIME-version: 1.0 X-Mailer: Internet Mail Service (5.5.2657.72) Content-type: text/plain Content-transfer-encoding: 7BIT X-Spam-Score: 0.0 (/) X-Scan-Signature: 41c17b4b16d1eedaa8395c26e9a251c4 Cc: sipping@ietf.org X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Francois, If the use of 480 temporarily available were restricted to that described in RFC 3261, then that means pretty much DND. Unfortunately current practice seems to use 480 for other purposes too, such as mapping from Q.850 cause code 19 No Answer From User in RFC 3398. John > -----Original Message----- > From: Francois Audet [mailto:audet@nortel.com] > Sent: 04 August 2006 17:21 > To: Elwell, John > Cc: sipping@ietf.org > Subject: RE: [Sipping] Question on do not disturb indication > > Hi John, > > Can you provide me an example of where a proxy would react > different to > "temporarily unavailable" versus "Do not disturb"? > > I sense that this is kind of similar to the ambiguity of 302 > problem... > > > -----Original Message----- > > From: Elwell, John [mailto:john.elwell@siemens.com] > > Sent: Thursday, August 03, 2006 11:33 AM > > To: Audet, Francois (SC100:3055); Paul Kyzivat; Scott Lawrence > > Cc: sipping@ietf.org > > Subject: RE: [Sipping] Question on do not disturb indication > > > > Francois, > > > > It is not only that I want the caller to know that I am in > > DND - also my proxy, if it is scripted to perform special > > behaviour on DND (e.g., retarget), then I would like it to > > know. I am not sure that 480 is sufficiently precise. > > > > John > > > > > -----Original Message----- > > > From: Francois Audet [mailto:audet@nortel.com] > > > Sent: 03 August 2006 19:10 > > > To: Elwell, John; Paul Kyzivat; Scott Lawrence > > > Cc: sipping@ietf.org > > > Subject: RE: [Sipping] Question on do not disturb indication > > > > > > > > > > [JRE] I agree that "make busy" is a separate feature > and is best > > > > dealt with by 486. But where I explicitly want the > caller to know > > > > that I am in DND (possibly with a retry-after time), do > you think > > > > that 480 is sufficient, bearing in mind that it gets used > > for other > > > > purposes too, e.g., in RFC3398? > > > > > > Personally, I think it is sufficient, and I also think > that mapping > > > DND to 480 is pretty much established by now. > > > > > > My perspective is Enterprise-centric however: there might be some > > > Service Provider services that need a more precise indication. > > > I could be convinced that it is needed... > > > > > > _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Sun Aug 06 09:54:07 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1G9j4x-00024Y-1J; Sun, 06 Aug 2006 09:54:07 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1G9j4v-00024Q-Nu for sipping@ietf.org; Sun, 06 Aug 2006 09:54:05 -0400 Received: from mailgate.siemenscomms.co.uk ([195.171.110.225] helo=bemg01.siemenscomms.co.uk) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1G9j4u-0004kG-6c for sipping@ietf.org; Sun, 06 Aug 2006 09:54:05 -0400 Received: from CONVERSION-DAEMON.siemenscomms.co.uk by siemenscomms.co.uk (PMDF V6.0-24 #40642) id <0J3K00H01XANBM@siemenscomms.co.uk> for sipping@ietf.org; Sun, 06 Aug 2006 14:54:23 +0100 (BST) Received: from ntht207e.uksgcs.siemenscomms.co.uk ([137.223.247.82]) by siemenscomms.co.uk (PMDF V6.0-24 #40642) with ESMTP id <0J3K00H1YXAN56@siemenscomms.co.uk>; Sun, 06 Aug 2006 14:54:23 +0100 (BST) Received: by ntht207e.uksgcs.siemenscomms.co.uk with Internet Mail Service (5.5.2657.72) id ; Sun, 06 Aug 2006 14:54:02 +0100 Content-return: allowed Date: Sun, 06 Aug 2006 14:54:01 +0100 From: "Elwell, John" Subject: RE: [Sipping] Question on do not disturb indication To: Francois Audet , Henry Sinnreich , Jonathan Rosenberg , Dean Willis Message-id: <50B1CBA96870A34799A506B2313F2667098D8529@ntht201e.siemenscomms.co.uk> MIME-version: 1.0 X-Mailer: Internet Mail Service (5.5.2657.72) Content-type: text/plain Content-transfer-encoding: 7BIT X-Spam-Score: 0.0 (/) X-Scan-Signature: ec7c6dab5a62df223002ae71b5179d41 Cc: "Dolly, Martin C, ALABS" , Paul Kyzivat , sipping@ietf.org, Scott Lawrence X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Francois, See below. John > -----Original Message----- > From: Francois Audet [mailto:audet@nortel.com] > Sent: 04 August 2006 21:01 > To: Henry Sinnreich; Jonathan Rosenberg; Dean Willis > Cc: Dolly, Martin C, ALABS; Paul Kyzivat; Elwell, John; > sipping@ietf.org; Scott Lawrence > Subject: RE: [Sipping] Question on do not disturb indication > > Actually, this is a very good point. > > There are really 2 aspects to "Do Not Disturb": > > 1 - On aspect for the terminating side to "reject" a session. > That's the > one I was > focussing on. That's the one where many of us are not > convinced that > we need > something beyond what is already allowed with SIP (480, 486, 6XX, > etc.). [JRE]But extending this, there is the aspect of how the indication given by the terminating side can be used by its proxy to take appropriate action. I think this is the most important aspect, perhaps more important than aspect 2 below. > > 2 - The other aspect is conveying a DND indication to the other end. I > guess > John was thinking about that. Henry is right that this aspect is > definitively > something appropriate for presence... [JRE] In the very first message in the thread I acknowledged that presence is an alternative. Presence, if availabe and authorised, can give you status independently of any call attempt. But I still feel it would be good to have a proper indication of why a call fails. > > > -----Original Message----- > > From: Henry Sinnreich [mailto:hsinnrei@adobe.com] > > Sent: Friday, August 04, 2006 10:54 AM > > To: Jonathan Rosenberg; Dean Willis > > Cc: Dolly, Martin C, ALABS; Paul Kyzivat; Elwell, John; > > sipping@ietf.org; Scott Lawrence > > Subject: RE: [Sipping] Question on do not disturb indication > > > > This is also a good time to remember once more there is in a > > SIP a capability called PRESENCE that is much more useful > > than DND and many other legacy telephony features. :-) > > > > Why ignore Presence and just stick with the "old proven way"? > > > > Thanks, Henry > > > > -----Original Message----- > > From: Jonathan Rosenberg [mailto:jdrosen@cisco.com] > > Sent: Friday, August 04, 2006 8:12 AM > > To: Dean Willis > > Cc: Dolly, Martin C, ALABS; Paul Kyzivat; Elwell,John; > > sipping@ietf.org; Scott Lawrence > > Subject: Re: [Sipping] Question on do not disturb indication > > > > I agree with Dean here. > > > > Why do you want to know, at the calling UA side, that the > > call was busy because of DND? Whether you need a new code > > depends on the answer: > > > > 1. if you want the human user to know that this was DND so > > they can do something appropriate, a reason phrase is exactly > > the right thing > > > > 2. if you want to have the calling UA to do something > > different, what would it do differently on a 'your call was > > rejected by DND' vs. 'your call was rejected because the user > > is on another call'? If the answer is > > > > nothing, then you don't need a new response code. > > > > I'm also leary about defining new response codes to do a > > specific terminating application that is running. Saying that > > this was rejected due to 'DND' requiers you to have a > > standard definition for DND. I don't > > > > want to do that. As others have pointed out, a reasonable > > implementation > > > > for DND is that the termianting network simply lets the phone > > ring and eventually responds with a 480. That should be allowed. > > > > Thanks, > > Jonathan R. > > > > Dean Willis wrote: > > > > > > > > On Aug 3, 2006, at 1:28 AM, Elwell, John wrote: > > > > > >> Dean, > > >> > > >> So what would you like the IETF to recommend? Do you believe 480 > > mostly > > >> works, or do you think something new is needed? > > >> > > >> John > > > > > > > > > > > > Hey, I'm just a consumer. I don't want to know the > > technical details. > > I > > > just want it to work, mostly. > > > > > > If it were up to me, I'd be asking the question: > > > > > > What part of DND is important for a calling human to > understand, vs. > > a > > > calling automata? Is there a distinction needed? > > > > > > If we believe that we'd like to convey a distinction between > > > "Temporarily unavailable" and "Temporarily unavailable due to > > > subscriber choice" to a human, perhaps a reason phrase on a > > 480 would > > > > > be adequate? Like "480 Do Not Disturb", maybe. > > > > > > > > > An automata is just likely to retry the call periodically > > anyhow, but > > a > > > human might stop and think "Oh yeah, it IS the middle of > > the night in > > > > > Singapore . . .". > > > > > > Question: How beneficial would a Retry-After value be in rate- > > limiting > > > the retries of automata? > > > > > > Now, I will say that Martin is right in that there might be many > > > reasons to reject a call, that DND is just the tip of the iceberg, > > and > > > that there's no way we can define them all. > > > > > > > > > Now, here's an interesting thought. If I use a reason > > phrase on 480, > > > and couple it to the right software, I can use SIP INVITE > > requests as > > a > > > covert-channel replacement for HTTP GET operations (or file > > transfer, > > > > > or whatever) , but peer-to-peer using the SIP > infrastructure as a > > > rendezvous service. Don't try this at home, kids. > > > > > > -- > > > Dean > > > > > > _______________________________________________ > > > Sipping mailing list > https://www1.ietf.org/mailman/listinfo/sipping > > > This list is for NEW development of the application of SIP Use > > > sip-implementors@cs.columbia.edu for questions on current sip Use > > > sip@ietf.org for new developments of core SIP > > > > > > > -- > > Jonathan D. Rosenberg, Ph.D. 600 Lanidex Plaza > > Cisco Fellow Parsippany, NJ > > 07054-2711 > > Cisco Systems > > jdrosen@cisco.com FAX: (973) 952-5050 > > http://www.jdrosen.net PHONE: (973) 952-5000 > > http://www.cisco.com > > > > _______________________________________________ > > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > > This list is for NEW development of the application of SIP > > Use sip-implementors@cs.columbia.edu for questions on current > > sip Use sip@ietf.org for new developments of core SIP > > > > _______________________________________________ > > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > > This list is for NEW development of the application of SIP > > Use sip-implementors@cs.columbia.edu for questions on current > > sip Use sip@ietf.org for new developments of core SIP > > > _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Mon Aug 07 09:10:22 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GA4s0-00072F-0O; Mon, 07 Aug 2006 09:10:12 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GA4ry-00071E-Bb for sipping@ietf.org; Mon, 07 Aug 2006 09:10:10 -0400 Received: from rtp-iport-2.cisco.com ([64.102.122.149]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GA4rx-00071A-17 for sipping@ietf.org; Mon, 07 Aug 2006 09:10:10 -0400 Received: from rtp-dkim-1.cisco.com ([64.102.121.158]) by rtp-iport-2.cisco.com with ESMTP; 07 Aug 2006 09:10:08 -0400 X-IronPort-AV: i="4.07,217,1151899200"; d="scan'208"; a="95456849:sNHT27815112" Received: from rtp-core-1.cisco.com (rtp-core-1.cisco.com [64.102.124.12]) by rtp-dkim-1.cisco.com (8.12.11.20060308/8.12.11) with ESMTP id k77DA8BL008089; Mon, 7 Aug 2006 09:10:08 -0400 Received: from xbh-rtp-201.amer.cisco.com (xbh-rtp-201.cisco.com [64.102.31.12]) by rtp-core-1.cisco.com (8.12.10/8.12.6) with ESMTP id k77DA8dp016569; Mon, 7 Aug 2006 09:10:08 -0400 (EDT) Received: from xfe-rtp-201.amer.cisco.com ([64.102.31.38]) by xbh-rtp-201.amer.cisco.com with Microsoft SMTPSVC(6.0.3790.1830); Mon, 7 Aug 2006 09:10:08 -0400 Received: from [161.44.79.104] ([161.44.79.104]) by xfe-rtp-201.amer.cisco.com with Microsoft SMTPSVC(6.0.3790.1830); Mon, 7 Aug 2006 09:10:08 -0400 Message-ID: <44D73BAF.7030907@cisco.com> Date: Mon, 07 Aug 2006 09:10:07 -0400 From: Paul Kyzivat User-Agent: Thunderbird 1.5.0.5 (Windows/20060719) MIME-Version: 1.0 To: "Vijay K. Gurbani" Subject: Re: [Sipping] Question on do not disturb indication References: <50B1CBA96870A34799A506B2313F266709897F68@ntht201e.siemenscomms.co.uk> <1B5A1FDA-AA89-4B8E-807E-2F170054D828@softarmor.com> <44D347C6.7070502@cisco.com> <44D3BA70.8040208@lucent.com> In-Reply-To: <44D3BA70.8040208@lucent.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Content-Transfer-Encoding: 7bit X-OriginalArrivalTime: 07 Aug 2006 13:10:08.0301 (UTC) FILETIME=[CE8365D0:01C6BA22] DKIM-Signature: a=rsa-sha1; q=dns; l=2174; t=1154956208; x=1155820208; c=relaxed/simple; s=rtpdkim1001; h=Content-Type:From:Subject:Content-Transfer-Encoding:MIME-Version; d=cisco.com; i=pkyzivat@cisco.com; z=From:Paul=20Kyzivat=20 |Subject:Re=3A=20[Sipping]=20Question=20on=20do=20not=20disturb=20indication |To:=22Vijay=20K.=20Gurbani=22=20; X=v=3Dcisco.com=3B=20h=3Dyu9ohsv1kFqX2uC2CMjguD3sTnY=3D; b=T+fvZ9CaMJaCnAla+/FBWkHCKbjeJHDhoUzhjB1BCcrq39J8zJQBy6tEfH1pk2D1yIJtNUmE 0o+b5aiht3wFSoC2ztdCf0EGKX0/CzBKfZ6Z77QVq5Y/+stOjhhBbpFN; Authentication-Results: rtp-dkim-1.cisco.com; header.From=pkyzivat@cisco.com; dkim=pass ( sig from cisco.com verified; ); X-Spam-Score: 0.0 (/) X-Scan-Signature: 0a7aa2e6e558383d84476dc338324fab Cc: sipping@ietf.org X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Vijay K. Gurbani wrote: > Paul Kyzivat wrote: >> I have never understood the purpose of reason phrases, or the value of >> sending different ones with the same response code. > > Paul: From a development point of view, reason phrases are > simply fantastic. Why ... inline. OK. I do accept that they can be useful for debugging. But I don't reason phrases being useful for communicating reasons *operationally*. >> Doing so implies an expectation that the UAC will do something >> different as a result. > > I don't view it that way. Take for instance the 400 Bad Request > response. In and of itself, it only says that the request was > malformed. But, when coupled with an descriptive phrase, the > intent of why the response was issued becomes clearer: > > 400 Missing Call-ID Header > 400 Missing Contact Header > 400 Multiple To Headers > > and so on. > > Here, to the receiving UAC, the 4xx puts it in a certain state. > The state would be the same regardless of the reason phrase. > But if the UAC has display capabilities (and many, if not all > do), then this becomes another tool for imparting information > to the user (or the developer, as the case may be). For developer, yes. For the user, I don't think so. It requires the UAC to give up too much control over the UI, potentially leading to support calls, etc. because the user is confused over what is displayed. Paul > Another example is the following 200 OK for a REG request: > > 200 OK > 200 Registration Expired > > The first 200 OK tells the UAC that the registration was > accepted. If the UAC later sends a REG w/ Expires:0, then > the reason phrase "Registration Expired" conveys appropriate > semantic information that to me is better than a simple "OK". > >> About the best one could hope for would be for some UACs to *display* >> the value, uninterpretted. But that only works for UACs that have the >> capability to do so, and that in fact do so. Realistically, providing >> that is only useful to geeks. > > Or implementors [(who are geeks by definition, anyway ;-)] > > Thanks. > > - vijay _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Mon Aug 07 14:13:35 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GA9bN-0006LA-GF; Mon, 07 Aug 2006 14:13:21 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GA9bM-0006Gw-9Y for sipping@ietf.org; Mon, 07 Aug 2006 14:13:20 -0400 Received: from zcars04f.nortel.com ([47.129.242.57]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GA9bK-0006Db-QD for sipping@ietf.org; Mon, 07 Aug 2006 14:13:20 -0400 Received: from zrc2hxm0.corp.nortel.com (zrc2hxm0.corp.nortel.com [47.103.123.71]) by zcars04f.nortel.com (Switch-2.2.6/Switch-2.2.0) with ESMTP id k77ID8I02747; Mon, 7 Aug 2006 14:13:08 -0400 (EDT) X-MimeOLE: Produced By Microsoft Exchange V6.5 Content-class: urn:content-classes:message MIME-Version: 1.0 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: quoted-printable Subject: RE: [Sipping] Question on do not disturb indication Date: Mon, 7 Aug 2006 13:12:38 -0500 Message-ID: <1ECE0EB50388174790F9694F77522CCF0C0462B9@zrc2hxm0.corp.nortel.com> In-Reply-To: <50B1CBA96870A34799A506B2313F2667098D8529@ntht201e.siemenscomms.co.uk> X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: [Sipping] Question on do not disturb indication thread-index: Aca5X85y4+ZTAyhcTtqX+acsGLHx7AA7AVFQ From: "Francois Audet" To: "Elwell, John" , "Henry Sinnreich" , "Jonathan Rosenberg" , "Dean Willis" X-Spam-Score: 0.0 (/) X-Scan-Signature: a1f9797ba297220533cb8c3f4bc709a8 Cc: "Dolly, Martin C, ALABS" , Paul Kyzivat , sipping@ietf.org, Scott Lawrence X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org John, I guess the reason for the different opinions we are seeing is that some products (even in PSTN-land) do not care about the reason for an "unvailable" (except for 'busy'). Many, for example, don't treat things differently if DND was the result of the user pressing a DND key, or if there is a temporary reason why the user not available (network issue for example). In other words, while I believe that the concept of "busy" is quasi-universal, I do not=20 believe that the concept of "explicit DND" is necessarily universal.=20 That may explain why some of us don't believe that we need to distinguish between "Temporarily Unavailable" versus "Do not Disturb". > -----Original Message----- > From: Elwell, John [mailto:john.elwell@siemens.com]=20 > Sent: Sunday, August 06, 2006 6:54 AM > To: Audet, Francois (SC100:3055); Henry Sinnreich; Jonathan=20 > Rosenberg; Dean Willis > Cc: Dolly, Martin C, ALABS; Paul Kyzivat; sipping@ietf.org;=20 > Scott Lawrence > Subject: RE: [Sipping] Question on do not disturb indication >=20 > Francois, >=20 > See below. >=20 > John=20 >=20 > > -----Original Message----- > > From: Francois Audet [mailto:audet@nortel.com] > > Sent: 04 August 2006 21:01 > > To: Henry Sinnreich; Jonathan Rosenberg; Dean Willis > > Cc: Dolly, Martin C, ALABS; Paul Kyzivat; Elwell, John;=20 > > sipping@ietf.org; Scott Lawrence > > Subject: RE: [Sipping] Question on do not disturb indication > >=20 > > Actually, this is a very good point. > >=20 > > There are really 2 aspects to "Do Not Disturb": > >=20 > > 1 - On aspect for the terminating side to "reject" a session.=20 > > That's the > > one I was > > focussing on. That's the one where many of us are not convinced=20 > > that we need > > something beyond what is already allowed with SIP (480,=20 > 486, 6XX,=20 > > etc.). > [JRE]But extending this, there is the aspect of how the=20 > indication given by the terminating side can be used by its=20 > proxy to take appropriate action. I think this is the most=20 > important aspect, perhaps more important than aspect > 2 below. >=20 > >=20 > > 2 - The other aspect is conveying a DND indication to the=20 > other end. I=20 > > guess > > John was thinking about that. Henry is right that this=20 > aspect is=20 > > definitively > > something appropriate for presence... > [JRE] In the very first message in the thread I acknowledged=20 > that presence is an alternative. Presence, if availabe and=20 > authorised, can give you status independently of any call=20 > attempt. But I still feel it would be good to have a proper=20 > indication of why a call fails. >=20 > >=20 > > > -----Original Message----- > > > From: Henry Sinnreich [mailto:hsinnrei@adobe.com] > > > Sent: Friday, August 04, 2006 10:54 AM > > > To: Jonathan Rosenberg; Dean Willis > > > Cc: Dolly, Martin C, ALABS; Paul Kyzivat; Elwell, John;=20 > > > sipping@ietf.org; Scott Lawrence > > > Subject: RE: [Sipping] Question on do not disturb indication > > >=20 > > > This is also a good time to remember once more there is=20 > in a SIP a=20 > > > capability called PRESENCE that is much more useful than DND and=20 > > > many other legacy telephony features. :-) > > >=20 > > > Why ignore Presence and just stick with the "old proven way"? > > >=20 > > > Thanks, Henry > > >=20 > > > -----Original Message----- > > > From: Jonathan Rosenberg [mailto:jdrosen@cisco.com] > > > Sent: Friday, August 04, 2006 8:12 AM > > > To: Dean Willis > > > Cc: Dolly, Martin C, ALABS; Paul Kyzivat; Elwell,John;=20 > > > sipping@ietf.org; Scott Lawrence > > > Subject: Re: [Sipping] Question on do not disturb indication > > >=20 > > > I agree with Dean here. > > >=20 > > > Why do you want to know, at the calling UA side, that the=20 > call was=20 > > > busy because of DND? Whether you need a new code depends on the=20 > > > answer: > > >=20 > > > 1. if you want the human user to know that this was DND=20 > so they can=20 > > > do something appropriate, a reason phrase is exactly the=20 > right thing > > >=20 > > > 2. if you want to have the calling UA to do something different,=20 > > > what would it do differently on a 'your call was rejected by DND'=20 > > > vs. 'your call was rejected because the user is on=20 > another call'? If=20 > > > the answer is > > >=20 > > > nothing, then you don't need a new response code. > > >=20 > > > I'm also leary about defining new response codes to do a specific=20 > > > terminating application that is running. Saying that this was=20 > > > rejected due to 'DND' requiers you to have a standard=20 > definition for=20 > > > DND. I don't > > >=20 > > > want to do that. As others have pointed out, a reasonable=20 > > > implementation > > >=20 > > > for DND is that the termianting network simply lets the=20 > phone ring=20 > > > and eventually responds with a 480. That should be allowed. > > >=20 > > > Thanks, > > > Jonathan R. > > >=20 > > > Dean Willis wrote: > > >=20 > > > >=20 > > > > On Aug 3, 2006, at 1:28 AM, Elwell, John wrote: > > > >=20 > > > >> Dean, > > > >> > > > >> So what would you like the IETF to recommend? Do you=20 > believe 480 > > > mostly > > > >> works, or do you think something new is needed? > > > >> > > > >> John > > > >=20 > > > >=20 > > > >=20 > > > > Hey, I'm just a consumer. I don't want to know the > > > technical details. > > > I > > > > just want it to work, mostly. > > > >=20 > > > > If it were up to me, I'd be asking the question: > > > >=20 > > > > What part of DND is important for a calling human to > > understand, vs. > > > a > > > > calling automata? Is there a distinction needed? > > > >=20 > > > > If we believe that we'd like to convey a distinction between=20 > > > > "Temporarily unavailable" and "Temporarily unavailable due to=20 > > > > subscriber choice" to a human, perhaps a reason phrase on a > > > 480 would > > >=20 > > > > be adequate? Like "480 Do Not Disturb", maybe. > > > >=20 > > > >=20 > > > > An automata is just likely to retry the call periodically > > > anyhow, but > > > a > > > > human might stop and think "Oh yeah, it IS the middle of > > > the night in > > >=20 > > > > Singapore . . .". > > > >=20 > > > > Question: How beneficial would a Retry-After value be in rate- > > > limiting > > > > the retries of automata? > > > >=20 > > > > Now, I will say that Martin is right in that there=20 > might be many=20 > > > > reasons to reject a call, that DND is just the tip of=20 > the iceberg, > > > and > > > > that there's no way we can define them all. > > > >=20 > > > >=20 > > > > Now, here's an interesting thought. If I use a reason > > > phrase on 480, > > > > and couple it to the right software, I can use SIP INVITE > > > requests as > > > a > > > > covert-channel replacement for HTTP GET operations (or file > > > transfer, > > >=20 > > > > or whatever) , but peer-to-peer using the SIP > > infrastructure as a > > > > rendezvous service. Don't try this at home, kids. > > > >=20 > > > > -- > > > > Dean > > > >=20 > > > > _______________________________________________ > > > > Sipping mailing list > > https://www1.ietf.org/mailman/listinfo/sipping > > > > This list is for NEW development of the application of SIP Use=20 > > > > sip-implementors@cs.columbia.edu for questions on=20 > current sip Use=20 > > > > sip@ietf.org for new developments of core SIP > > > >=20 > > >=20 > > > --=20 > > > Jonathan D. Rosenberg, Ph.D. 600 Lanidex Plaza > > > Cisco Fellow Parsippany, NJ=20 > > > 07054-2711 > > > Cisco Systems > > > jdrosen@cisco.com FAX: =20 > (973) 952-5050 > > > http://www.jdrosen.net PHONE:=20 > (973) 952-5000 > > > http://www.cisco.com > > >=20 > > > _______________________________________________ > > > Sipping mailing list =20 > https://www1.ietf.org/mailman/listinfo/sipping > > > This list is for NEW development of the application of SIP Use=20 > > > sip-implementors@cs.columbia.edu for questions on current sip Use=20 > > > sip@ietf.org for new developments of core SIP > > >=20 > > > _______________________________________________ > > > Sipping mailing list =20 > https://www1.ietf.org/mailman/listinfo/sipping > > > This list is for NEW development of the application of SIP Use=20 > > > sip-implementors@cs.columbia.edu for questions on current sip Use=20 > > > sip@ietf.org for new developments of core SIP > > >=20 > >=20 >=20 _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Tue Aug 08 09:07:30 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GARIk-0003n2-De; Tue, 08 Aug 2006 09:07:18 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GARIi-0003mx-Rw for sipping@ietf.org; Tue, 08 Aug 2006 09:07:17 -0400 Received: from mx12.bbn.com ([128.33.0.81]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GARIg-0002Y6-Ct for sipping@ietf.org; Tue, 08 Aug 2006 09:07:16 -0400 Received: from dommiel.bbn.com ([192.1.122.15] helo=[127.0.0.1]) by mx12.bbn.com with esmtp (Exim 4.60) (envelope-from ) id 1GARIW-0002gb-43; Tue, 08 Aug 2006 09:07:04 -0400 Message-ID: <44D88C72.2020507@bbn.com> Date: Tue, 08 Aug 2006 09:06:58 -0400 From: "Richard L. Barnes" User-Agent: Thunderbird 1.5.0.5 (Windows/20060719) MIME-Version: 1.0 To: ravishankar.shiroor@wipro.com Subject: Re: [Sipping] Some thoughts on fixing early media References: <532B18E13CF9E64380EF5FDDE265E071384C58@blr-m2-msg.wipro.com> In-Reply-To: <532B18E13CF9E64380EF5FDDE265E071384C58@blr-m2-msg.wipro.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Content-Transfer-Encoding: 7bit X-Spam-Score: 0.0 (/) X-Scan-Signature: bcd240e64c427d3d3617cfc704e7fd7f Cc: pkyzivat@cisco.com, christer.holmberg@ericsson.com, bstucker@nortel.com, dean.willis@softarmor.com, sipping@ietf.org X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org This is a little bit of a tangent, but the idea that "hang-up or be billed" messages are a driver for allowing early media is a little distasteful to me -- Such messages aren't media (they're not the "content" of the call), but rather signaling messages sent as an audio stream. It seems to me like ideally, such a message would be sent as a 1XX response (possibly with the audio included in the body?). This would take a little more smarts in a PSTN gateway, but it doesn't seem like an insurmountable problem. Does this line of thinking seem reasonable? Thanks, --Richard ravishankar.shiroor@wipro.com wrote: > Hi Sayan, > > Would it not be the UAS's responsibility to decide that the user should > be alerted only after the Early Media is played? > I think that the UAS which has early media to play, can decide to > postpone the alerting till it has been sent to the UAC. UAC need not > have to control this - I think this should satisfy all the use cases > needed. (for example "please hang-up in ten seconds from now unless you > wish to be billed a huge amount" or something like that should be played > before UAS alerts the user.) > > Regards, > Ravi. > > > -- > > Ravishankar. G. Shiroor > Wipro Technologies, Bangalore. > > ravishankar.shiroor@wipro.com > -- > > >> -----Original Message----- >> From: sayan.chowdhury@wipro.com [mailto:sayan.chowdhury@wipro.com] >> Sent: Tuesday, July 25, 2006 11:21 PM >> To: bstucker@nortel.com; pkyzivat@cisco.com >> Cc: christer.holmberg@ericsson.com; sipping@ietf.org; >> dean.willis@softarmor.com >> Subject: RE: [Sipping] Some thoughts on fixing early media >> >> Hello Brian , >> I am trying to put a call flow here , (I will not try to draw >> as my mail client is messing it up) >> >> 1>Callee sends an offer with emflow=none >> 2>gets back say two 183 responses (establishing early dialog) >> both with >> early media class as two-way. >> 3>it selects the 183 with the higher q-value and more >> critical em class >> and adds new offer in the PRACK with emflow as sendrecv. >> 4>PRACKs the other 183 as well but with emflow=none and gets >> back 200 OK >> for that. >> 4>Gets back the answer in the 200 OK to PRACK for the 183 with higher >> q-value. >> 5>Does early media. >> 6>Sends an UPDATE to the next 183 with lower q-value with emflow as >> sendrcv >> ... continues >> >> If my understanding of the flow is correct , then in the >> above scenario how does the UAC indicate to the UAS(s) not to >> alert the user until early media is over ? >> >> Do we need a confirm-status mechanism as well wherein the UAC >> can tell when to alert the UAS ? >> There may be use cases when the UAC wants to defer the >> alerting until the UAC has heard all the early media "flows". >> For eg , when the UAC chooses which UAS it wants to have a >> regular media session with after hearing to all the early >> media sessions. >> >> Regards , >> Sayan >> >> -----Original Message----- >> From: Brian Stucker [mailto:bstucker@nortel.com] >> Sent: Tuesday, July 25, 2006 9:02 PM >> To: Paul Kyzivat >> Cc: Dean Willis; sipping; Christer Holmberg (JO/LMF) >> Subject: RE: [Sipping] Some thoughts on fixing early media >> >> Yes, this is pretty much what's going to happen. I'll add >> that to the text, actually, since I think it deserves being >> called out. >> >> In cases where network interworking is not present (ie. no >> SIP-to-PSTN type gateways are present in the routing tree) >> things will basically behave as they always have been because >> most SIP endpoints do not generate early media. It's services >> like CRBT and PSTN gateways that get us into trouble. For the >> other endpoints we simply don't wish to clip the media if >> possible. For the ones that do generate early media, we can >> fork to them, but don't necessarily want to hear what they >> all have to say in as a chorus. >> >> Regards, >> Brian >> >>> -----Original Message----- >>> From: Paul Kyzivat [mailto:pkyzivat@cisco.com] >>> Sent: Tuesday, July 25, 2006 8:03 AM >>> To: Stucker, Brian (RICH1:B620) >>> Cc: Christer Holmberg (JO/LMF); Dean Willis; Stephen Sprunk; sipping >>> Subject: Re: [Sipping] Some thoughts on fixing early media >>> >>> If the UAC only gives permission to one tine of the fork to send >>> media, and preconditions are used to prevent alerting until >> permission >> >>> to send is granted, then parallel forking has been turned >> into serial >>> forking under the control of the UAC. >>> >>> While it seems a little odd, it is perhaps the best that >> can be done >>> under the circumstances. It works well in that those >> destinations that >> >>> don't require early media can all alert in parallel, and >> only the ones >> >>> requiring early media are serialized. That is a huge >> improvement over >>> the proxy making the choice, because it doesn't know which >> ones will >>> require early media. >>> >>> Paul >>> >>> Brian Stucker wrote: >>>> Yes, I think you understand what I am suggesting. >>>> >>>> Regards, >>>> Brian >>>> >>>> -----Original Message----- >>>> From: Christer Holmberg (JO/LMF) >>>> [mailto:christer.holmberg@ericsson.com] >>>> >>>> Sent: Sunday, July 23, 2006 5:20 PM >>>> To: Stucker, Brian (RICH1:B620); Dean Willis; Stephen Sprunk >>>> Cc: sipping >>>> Subject: VS: [Sipping] Some thoughts on fixing early media >>>> >>>> Hi, >>>> >>>>> I'm proposing that the far end would not alert until >> permission to >>>>> send >>>>> media had been granted. >>>> That's called pre-conditions, isn't it? :) >>>> >>>> Or, you could also send an INVITE WITH SDP, but the UAS would not >>>> altert the user until some other type of indicator has been >>> sent to it. >>>> >>>> Regards, >>>> >>>> Christer >>>> >>>> >>>> >>>> -----Original Message----- >>>> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com] >>>> Sent: Sunday, July 23, 2006 12:14 AM >>>> To: Stucker, Brian (RICH1:B620); Dean Willis; Stephen Sprunk >>>> Cc: sipping >>>> Subject: RE: [Sipping] Some thoughts on fixing early media >>>> >>>> Hi Brian, >>>> >>>> What if an UAS wants to send 200 OK (and media) before >> the UAC has >>>> told him it's ok to send media? Someone would have to send >>> an UPDATE, >>>> to switch the media to sendrecv. >>>> >>>> Or, are you proposing that the UAS, until it's given >> permission to >>>> send media, wouldn't even alert the user behind it? >>>> >>>> Regards, >>>> >>>> Christer >>>> >>>> >>>> >>>> >>>> _______________________________________________ >>>> Sipping mailing list >> https://www1.ietf.org/mailman/listinfo/sipping >>>> This list is for NEW development of the application of SIP Use >>>> sip-implementors@cs.columbia.edu for questions on current sip Use >>>> sip@ietf.org for new developments of core SIP >>>> >> _______________________________________________ >> Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping >> This list is for NEW development of the application of SIP >> Use sip-implementors@cs.columbia.edu for questions on current >> sip Use sip@ietf.org for new developments of core SIP >> >> _______________________________________________ >> Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping >> This list is for NEW development of the application of SIP >> Use sip-implementors@cs.columbia.edu for questions on current >> sip Use sip@ietf.org for new developments of core SIP >> > > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP > Use sip-implementors@cs.columbia.edu for questions on current sip > Use sip@ietf.org for new developments of core SIP > > _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Tue Aug 08 09:13:51 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GARP3-0007fc-Hr; Tue, 08 Aug 2006 09:13:49 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GARP1-0007aw-KJ for sipping@ietf.org; Tue, 08 Aug 2006 09:13:47 -0400 Received: from wip-ec-wd.wipro.com ([203.91.193.32]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GARP0-0004dy-5k for sipping@ietf.org; Tue, 08 Aug 2006 09:13:47 -0400 Received: from wip-ec-wd.wipro.com (localhost.wipro.com [127.0.0.1]) by localhost (Postfix) with ESMTP id 3965F20634 for ; Tue, 8 Aug 2006 18:40:22 +0530 (IST) Received: from blr-ec-bh01.wipro.com (blr-ec-bh01.wipro.com [10.201.50.91]) by wip-ec-wd.wipro.com (Postfix) with ESMTP id 229B9205DB for ; Tue, 8 Aug 2006 18:40:22 +0530 (IST) Received: from blr-k1-msg.wipro.com ([10.117.50.99]) by blr-ec-bh01.wipro.com with Microsoft SMTPSVC(6.0.3790.1830); Tue, 8 Aug 2006 18:43:43 +0530 X-MimeOLE: Produced By Microsoft Exchange V6.5 Content-class: urn:content-classes:message MIME-Version: 1.0 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: quoted-printable Subject: RE: [Sipping] Some thoughts on fixing early media Date: Tue, 8 Aug 2006 18:43:41 +0530 Message-ID: <897DA809E857CD40A8419CF517948E2101FFF783@blr-k1-msg.wipro.com> In-Reply-To: <44D88C72.2020507@bbn.com> X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: [Sipping] Some thoughts on fixing early media Thread-Index: Aca66xQkdD1nf5IgQPun37Bf6/tjqQAAMatQ From: To: , X-OriginalArrivalTime: 08 Aug 2006 13:13:43.0604 (UTC) FILETIME=[7941C740:01C6BAEC] X-Spam-Score: 0.2 (/) X-Scan-Signature: 223e3c753032a50d5dc4443c921c3fcd Cc: pkyzivat@cisco.com, dean.willis@softarmor.com, sipping@ietf.org, christer.holmberg@ericsson.com, bstucker@nortel.com X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Hello Richard ,=20 I think there are a a few difficulty in such an approach For one thing the early media may not be only "hang up or be billed" kind of thing ,=20 It can be a a two way stream collecting some user input as in-band DTMF ... i.e. two way=20 There can be several other kinds as Brian mentions in his draft.=20 Also at the PSTN gateway it may be very difficult(if not impossible) to interpret and/or include the audio stream in some kind of signaling message , specialy if the gateway is "distributed" in an MGC-MG architecture. In that case the signaling entity will not be on the media plane at all ...=20 Regards , Sayan=20 =20 -----Original Message----- From: Richard L. Barnes [mailto:rbarnes@bbn.com]=20 Sent: Tuesday, August 08, 2006 6:37 PM To: RaviShankar Shiroor (WT01 - Product Engineering Solutions) Cc: Sayan Chowdhury (WT01 - IP-Multimedia Carrier & Ent Networks); bstucker@nortel.com; pkyzivat@cisco.com; dean.willis@softarmor.com; sipping@ietf.org; christer.holmberg@ericsson.com Subject: Re: [Sipping] Some thoughts on fixing early media This is a little bit of a tangent, but the idea that "hang-up or be billed" messages are a driver for allowing early media is a little distasteful to me -- Such messages aren't media (they're not the "content" of the call), but rather signaling messages sent as an audio stream. It seems to me like ideally, such a message would be sent as a 1XX response (possibly with the audio included in the body?). This would take a little more smarts in a PSTN gateway, but it doesn't seem like an insurmountable problem. Does this line of thinking seem reasonable? Thanks, --Richard ravishankar.shiroor@wipro.com wrote: > Hi Sayan, >=20 > Would it not be the UAS's responsibility to decide that the user=20 > should be alerted only after the Early Media is played? > I think that the UAS which has early media to play, can decide to=20 > postpone the alerting till it has been sent to the UAC. UAC need not=20 > have to control this - I think this should satisfy all the use cases=20 > needed. (for example "please hang-up in ten seconds from now unless=20 > you wish to be billed a huge amount" or something like that should be=20 > played before UAS alerts the user.) >=20 > Regards, > Ravi. >=20 >=20 > -- > =20 > Ravishankar. G. Shiroor > Wipro Technologies, Bangalore. > =20 > ravishankar.shiroor@wipro.com > -- > =20 >=20 >> -----Original Message----- >> From: sayan.chowdhury@wipro.com [mailto:sayan.chowdhury@wipro.com] >> Sent: Tuesday, July 25, 2006 11:21 PM >> To: bstucker@nortel.com; pkyzivat@cisco.com >> Cc: christer.holmberg@ericsson.com; sipping@ietf.org;=20 >> dean.willis@softarmor.com >> Subject: RE: [Sipping] Some thoughts on fixing early media >> >> Hello Brian , >> I am trying to put a call flow here , (I will not try to draw as my=20 >> mail client is messing it up) >> >> 1>Callee sends an offer with emflow=3Dnone >> 2>gets back say two 183 responses (establishing early dialog) >> both with >> early media class as two-way. >> 3>it selects the 183 with the higher q-value and more >> critical em class >> and adds new offer in the PRACK with emflow as sendrecv. >> 4>PRACKs the other 183 as well but with emflow=3Dnone and gets >> back 200 OK >> for that. >> 4>Gets back the answer in the 200 OK to PRACK for the 183 with higher >> q-value. >> 5>Does early media.=20 >> 6>Sends an UPDATE to the next 183 with lower q-value with emflow as >> sendrcv >> ... continues >> >> If my understanding of the flow is correct , then in the above=20 >> scenario how does the UAC indicate to the UAS(s) not to alert the=20 >> user until early media is over ? >> >> Do we need a confirm-status mechanism as well wherein the UAC can=20 >> tell when to alert the UAS ? >> There may be use cases when the UAC wants to defer the alerting until >> the UAC has heard all the early media "flows". >> For eg , when the UAC chooses which UAS it wants to have a regular=20 >> media session with after hearing to all the early media sessions. >> >> Regards , >> Sayan >> >> -----Original Message----- >> From: Brian Stucker [mailto:bstucker@nortel.com] >> Sent: Tuesday, July 25, 2006 9:02 PM >> To: Paul Kyzivat >> Cc: Dean Willis; sipping; Christer Holmberg (JO/LMF) >> Subject: RE: [Sipping] Some thoughts on fixing early media >> >> Yes, this is pretty much what's going to happen. I'll add that to the >> text, actually, since I think it deserves being called out. >> >> In cases where network interworking is not present (ie. no=20 >> SIP-to-PSTN type gateways are present in the routing tree) things=20 >> will basically behave as they always have been because most SIP=20 >> endpoints do not generate early media. It's services like CRBT and=20 >> PSTN gateways that get us into trouble. For the other endpoints we=20 >> simply don't wish to clip the media if possible. For the ones that do >> generate early media, we can fork to them, but don't necessarily want >> to hear what they all have to say in as a chorus. >> >> Regards, >> Brian >> >>> -----Original Message----- >>> From: Paul Kyzivat [mailto:pkyzivat@cisco.com] >>> Sent: Tuesday, July 25, 2006 8:03 AM >>> To: Stucker, Brian (RICH1:B620) >>> Cc: Christer Holmberg (JO/LMF); Dean Willis; Stephen Sprunk; sipping >>> Subject: Re: [Sipping] Some thoughts on fixing early media >>> >>> If the UAC only gives permission to one tine of the fork to send=20 >>> media, and preconditions are used to prevent alerting until >> permission >> >>> to send is granted, then parallel forking has been turned >> into serial >>> forking under the control of the UAC. >>> >>> While it seems a little odd, it is perhaps the best that >> can be done >>> under the circumstances. It works well in that those >> destinations that >> >>> don't require early media can all alert in parallel, and >> only the ones >> >>> requiring early media are serialized. That is a huge >> improvement over >>> the proxy making the choice, because it doesn't know which >> ones will >>> require early media. >>> >>> Paul >>> >>> Brian Stucker wrote: >>>> Yes, I think you understand what I am suggesting. >>>> >>>> Regards, >>>> Brian >>>> >>>> -----Original Message----- >>>> From: Christer Holmberg (JO/LMF) >>>> [mailto:christer.holmberg@ericsson.com] >>>> >>>> Sent: Sunday, July 23, 2006 5:20 PM >>>> To: Stucker, Brian (RICH1:B620); Dean Willis; Stephen Sprunk >>>> Cc: sipping >>>> Subject: VS: [Sipping] Some thoughts on fixing early media >>>> >>>> Hi, >>>> >>>>> I'm proposing that the far end would not alert until >> permission to >>>>> send >>>>> media had been granted. >>>> That's called pre-conditions, isn't it? :) >>>> =20 >>>> Or, you could also send an INVITE WITH SDP, but the UAS would not=20 >>>> altert the user until some other type of indicator has been >>> sent to it. >>>> =20 >>>> Regards, >>>> =20 >>>> Christer >>>> =20 >>>> >>>> >>>> -----Original Message----- >>>> From: Christer Holmberg [mailto:christer.holmberg@ericsson.com] >>>> Sent: Sunday, July 23, 2006 12:14 AM >>>> To: Stucker, Brian (RICH1:B620); Dean Willis; Stephen Sprunk >>>> Cc: sipping >>>> Subject: RE: [Sipping] Some thoughts on fixing early media >>>> >>>> Hi Brian, >>>> >>>> What if an UAS wants to send 200 OK (and media) before >> the UAC has >>>> told him it's ok to send media? Someone would have to send >>> an UPDATE, >>>> to switch the media to sendrecv. >>>> >>>> Or, are you proposing that the UAS, until it's given >> permission to >>>> send media, wouldn't even alert the user behind it? >>>> >>>> Regards, >>>> >>>> Christer >>>> >>>> >>>> >>>> >>>> _______________________________________________ >>>> Sipping mailing list >> https://www1.ietf.org/mailman/listinfo/sipping >>>> This list is for NEW development of the application of SIP Use=20 >>>> sip-implementors@cs.columbia.edu for questions on current sip Use=20 >>>> sip@ietf.org for new developments of core SIP >>>> >> _______________________________________________ >> Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping >> This list is for NEW development of the application of SIP Use=20 >> sip-implementors@cs.columbia.edu for questions on current sip Use=20 >> sip@ietf.org for new developments of core SIP >> >> _______________________________________________ >> Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping >> This list is for NEW development of the application of SIP Use=20 >> sip-implementors@cs.columbia.edu for questions on current sip Use=20 >> sip@ietf.org for new developments of core SIP >> >=20 > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP Use=20 > sip-implementors@cs.columbia.edu for questions on current sip Use=20 > sip@ietf.org for new developments of core SIP >=20 >=20 _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Tue Aug 08 11:40:23 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GATgo-0007fQ-BO; Tue, 08 Aug 2006 11:40:18 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GATgn-0007fL-0F for sipping@ietf.org; Tue, 08 Aug 2006 11:40:17 -0400 Received: from rtp-iport-1.cisco.com ([64.102.122.148]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GATgj-0001r1-Aa for sipping@ietf.org; Tue, 08 Aug 2006 11:40:16 -0400 Received: from rtp-dkim-2.cisco.com ([64.102.121.159]) by rtp-iport-1.cisco.com with ESMTP; 08 Aug 2006 08:40:10 -0700 X-BrightmailFiltered: true X-Brightmail-Tracker: AAAAAA== X-IronPort-AV: i="4.07,222,1151910000"; d="scan'208"; a="35148779:sNHT28294060" Received: from rtp-core-1.cisco.com (rtp-core-1.cisco.com [64.102.124.12]) by rtp-dkim-2.cisco.com (8.12.11.20060308/8.12.11) with ESMTP id k78FeAxf022611; Tue, 8 Aug 2006 11:40:10 -0400 Received: from xbh-rtp-201.amer.cisco.com (xbh-rtp-201.cisco.com [64.102.31.12]) by rtp-core-1.cisco.com (8.12.10/8.12.6) with ESMTP id k78FeAdp011407; Tue, 8 Aug 2006 11:40:10 -0400 (EDT) Received: from xmb-rtp-20b.amer.cisco.com ([64.102.31.53]) by xbh-rtp-201.amer.cisco.com with Microsoft SMTPSVC(6.0.3790.1830); Tue, 8 Aug 2006 11:40:10 -0400 X-MimeOLE: Produced By Microsoft Exchange V6.5 Content-class: urn:content-classes:message MIME-Version: 1.0 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: quoted-printable Subject: RE: [Sipping] Question on do not disturb indication Date: Tue, 8 Aug 2006 11:40:08 -0400 Message-ID: <072C5B76F7CEAB488172C6F64B30B5E301C6E402@xmb-rtp-20b.amer.cisco.com> X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: [Sipping] Question on do not disturb indication Thread-Index: Aca34Cx4m+z1EjWfTMaZ2tM4yTkTswDIEqQg From: "Michael Hammer \(mhammer\)" To: , "Jonathan Rosenberg \(jdrosen\)" X-OriginalArrivalTime: 08 Aug 2006 15:40:10.0496 (UTC) FILETIME=[EEA71800:01C6BB00] DKIM-Signature: a=rsa-sha1; q=dns; l=8500; t=1155051610; x=1155915610; c=relaxed/simple; s=rtpdkim2001; h=Content-Type:From:Subject:Content-Transfer-Encoding:MIME-Version; d=cisco.com; i=mhammer@cisco.com; z=From:=22Michael=20Hammer=20\(mhammer\)=22=20 |Subject:RE=3A=20[Sipping]=20Question=20on=20do=20not=20disturb=20indication |To:, =20=22Jonathan=20Rosenberg=20\(jdrosen\)=22=20; X=v=3Dcisco.com=3B=20h=3DtckYbUB4bmNYQXrJH2n4cDA572g=3D; b=B1C8USBs+A4kNtzcgnd7uyGzGLN2SOFgASa99+qSIbDRl31UhHWVFv2fQ5VRMjhZQwmQRf16 Y+XrI0Cg2PRFpc12cT5K2+BtmtPqjP4Ehg1EbWUrTfE5Za5ePGeQEKcA; Authentication-Results: rtp-dkim-2.cisco.com; header.From=mhammer@cisco.com; dkim=pass ( sig from cisco.com verified; ); X-Spam-Score: 0.0 (/) X-Scan-Signature: ff0adf256e4dd459cc25215cfa732ac1 Cc: "Paul Kyzivat \(pkyzivat\)" , sipping@ietf.org X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org There seems to be a distinction between "what just happened" (hit a busy/DND user) and "what should happen next" (go to voicemail or stop calling me). Is MIME and CPL needed or just some token in a new header: Instruction: or Instruction: Mike > -----Original Message----- > From: Xiaotao Wu [mailto:xwu@avaya.com]=20 > Sent: Friday, August 04, 2006 12:08 PM > To: Jonathan Rosenberg (jdrosen) > Cc: Paul Kyzivat (pkyzivat); sipping@ietf.org > Subject: Re: [Sipping] Question on do not disturb indication >=20 > > > > There are sevearl ways one might solve this problem: > > > > 1. The UA uploads a CPL script when it registers; that CPL script=20 > > tells the proxy what to do based on specific response codes. For=20 > > example, it can tell the proxy that a 476 means forward to=20 > voicemail.=20 > > This has the benefit of not requiring standardized definitions of=20 > > codes, and of leaving a lot of flexibility for different services.=20 > > However, it falls apart when there are multiple UA in use=20 > and requires=20 > > CPL support which has not seen wide success for many reasons. > > > > 2. We define one or more new response codes which have the desired=20 > > effects. I think this is going to be hard actually, given that the=20 > > four use cases above each might require a new response code > > > > 3. We have a header field in the response which tells the=20 > proxy that=20 > > this response shuold be treated as if it came from *all* UA=20 > instances=20 > > forked to from the AOR, thus cancelling all other forks for=20 > that AOR. >=20 > 4. Putting a script (e.g., a CPL-like script) as a MIME part=20 > in the response to instruct the proxy what to do. This is=20 > different from option 1, which invokes service logic by a=20 > specific response code. For example, for DND, we can still=20 > use 480 response code, but have a script instructing the=20 > proxy to proxy the call to voicemail. >=20 > -Xiaotao >=20 > > > > > > It feels to me like we need some requirements work and problem=20 > > definition here. I'd argue that this is perfect fodder for=20 > our MIG BoF=20 > > next IETF. > > > > -Jonathan R. > > > > > > > > Paul Kyzivat wrote: > > > > > I went at reviewed 3261 on this and found some interesting things: > > > > > > - Global Failures 6xx > > > > > > 6xx responses indicate that a server has definitive=20 > information about > > > a particular user, not just the particular instance=20 > indicated in the > > > Request-URI. > > > > > > (This refers to a "particular user", but how that=20 > relates to a call > > > in progress isn't clear. Do two extensions on one AOR=20 > both belong > > > to the same "user", or not? Should the phone and a VM=20 > server backing > > > up the phone be viewed as the same user? If a proxy is=20 > configured > > > to offer calls first to Alice's phone, and if no answer then to > > > Alice's assistant Bob, does a 6xx response by Alice's phone say > > > anything about Bob?) > > > > > > - 603 Decline: > > > > > > This status response is returned only if the client knows > > > that no other end point will answer the request. > > > > > > (This is in conflict with the language in 6xx because it talks > > > about end point rather than user. It doesn't appear that the > > > difference is because of anything special about 603 vs the > > > general 6xx.) > > > > > > - 604 Does Not Exist Anywhere: > > > > > > The server has authoritative information that the user=20 > indicated in > > > the *** Request-URI *** does not exist anywhere. > > > > > > (This again differs from the 6xx description in a way that > > > doesn't seem related to the specifics of 604 vs 6xx. This one > > > does talk about *user*, and is more specific about=20 > what it means > > > by user. But not in a very helpful way. It explicitly calls out > > > the R-URI, and not say the To-URI. The R-URI typically only > > > identifies one UA, which means this response may not=20 > technically > > > mean anything different than 404.} > > > > > > This to me indicates a general lack of specificity in=20 > 3261 about the=20 > > > scope of applicability of the 6xx codes. As we are=20 > finding with many=20 > > > aspects of 3261, as we progress we find a need to tighten=20 > things up=20 > > > - clarify things that ambiguous or just wrong. But before=20 > that can=20 > > > be done, we need to figure out how we thing it ought to work. > > > > > > IMO there has to be some discretion in how 6xx codes are handled=20 > > > upstream - this interacts with what the upstream node is=20 > trying to=20 > > > accomplish. For instance: > > > > > > - if a proxy intends to try one or more contacts of a single AOR, > > > and it receives a 6xx from one of them, it normally shouldn't > > > try the others. > > > > > > - if a proxy intends to try one or more contacts of AOR-1 and then > > > try one or more contacts of AOR-2, and it receives a 6xx from > > > one of the AOR-1 contacts, then it should have the discretion > > > to still try AOR-2. Whether it does this or not may depend on > > > what it is trying to achieve by trying both. > > > > > > There clearly is a fine line here. The nodes receiving the 6xx=20 > > > responses need some discretion. But the nodes sending the=20 > 6xx need=20 > > > to have enough definition about what the codes mean to=20 > choose which=20 > > > one to return, so that the right thing might happen. > > > > > > Paul > > > > > > > > > > > > Dale.Worley@comcast.net wrote: > > > > > >> From: Scott Lawrence > > >> > > >> The one thing that I _don't_ like to see is a 603=20 > being used for this; > > >> that messes up any upstream forking. > > >> > > >> Personally, I think we should file a bug against the=20 > *existence* of=20 > > >> 6xx codes. They just can't be made to work correctly in=20 > the light=20 > > >> of multiple routings and forwardings of a URI. > > >> > > >> Dale > > >> > > >> _______________________________________________ > > >> Sipping mailing list =20 > > >> https://www1.ietf.org/mailman/listinfo/sipping > > >> This list is for NEW development of the application of SIP Use=20 > > >> sip-implementors@cs.columbia.edu for questions on=20 > current sip Use=20 > > >> sip@ietf.org for new developments of core SIP > > >> > > > > > > _______________________________________________ > > > Sipping mailing list =20 > https://www1.ietf.org/mailman/listinfo/sipping > > > This list is for NEW development of the application of SIP Use=20 > > > sip-implementors@cs.columbia.edu for questions on current sip Use=20 > > > sip@ietf.org for new developments of core SIP > > > > > > > -- > > Jonathan D. Rosenberg, Ph.D. 600 Lanidex Plaza > > Cisco Fellow Parsippany,=20 > NJ 07054-2711 > > Cisco Systems > > jdrosen@cisco.com FAX: (973) 952-5050 > > http://www.jdrosen.net PHONE: (973) 952-5000 > > http://www.cisco.com > > > > _______________________________________________ > > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > > This list is for NEW development of the application of SIP Use=20 > > sip-implementors@cs.columbia.edu for questions on current sip Use=20 > > sip@ietf.org for new developments of core SIP > > >=20 >=20 > -- > -Xiaotao >=20 > = =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D= =3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D=3D= =3D=3D=3D=3D=3D=3D=3D=3D=3D > Name : Xiaotao Wu > Email : xwu@avaya.com, xiaotaow@gmail.com > Web : http://www.cs.columbia.edu/~xiaotaow > Phone : (732) 852-2133 > SIP : sip:xiaotaow@cs.columbia.edu > Address : 1M-240, 307 Middletown Lincroft Road > Lincroft, NJ 07738-1526, U.S.A. >=20 > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP=20 > Use sip-implementors@cs.columbia.edu for questions on current=20 > sip Use sip@ietf.org for new developments of core SIP >=20 _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Tue Aug 08 18:04:25 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GAZgP-0000SG-5O; Tue, 08 Aug 2006 18:04:17 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GAZgN-0000SA-VV for sipping@ietf.org; Tue, 08 Aug 2006 18:04:15 -0400 Received: from web31503.mail.mud.yahoo.com ([68.142.198.132]) by ietf-mx.ietf.org with smtp (Exim 4.43) id 1GAZgM-00061A-MA for sipping@ietf.org; Tue, 08 Aug 2006 18:04:15 -0400 Received: (qmail 70196 invoked by uid 60001); 8 Aug 2006 22:04:14 -0000 DomainKey-Signature: a=rsa-sha1; q=dns; c=nofws; s=s1024; d=yahoo.com; h=Message-ID:Received:Date:From:Subject:To:Cc:In-Reply-To:MIME-Version:Content-Type:Content-Transfer-Encoding; b=ZCbA1gufQyXc8nYW9EIq1fXu8EK9hK1BB7MqAKfh/CBJxEP2/2bZN+8v/y2t2Ycjy2sbBtEDQ+E0UOQM41M0BpF7Xh9veBbIUYE5r8rQFMQcfpAOazqCbYGTIUSA18pdPF6BFd+8GkHSuD+tGB641ANm7WMOHJZmaHQMtUG9BZQ= ; Message-ID: <20060808220414.70194.qmail@web31503.mail.mud.yahoo.com> Received: from [198.152.12.67] by web31503.mail.mud.yahoo.com via HTTP; Tue, 08 Aug 2006 23:04:14 BST Date: Tue, 8 Aug 2006 23:04:14 +0100 (BST) From: Kapil Nayar Subject: Re: [Sipping] Clarification on RFC 3959 - early media To: Paul Kyzivat In-Reply-To: <44D349A9.4010100@cisco.com> MIME-Version: 1.0 Content-Type: text/plain; charset=iso-8859-1 Content-Transfer-Encoding: 8bit X-Spam-Score: 0.0 (/) X-Scan-Signature: 5a9a1bd6c2d06a21d748b7d0070ddcb8 Cc: sipping@ietf.org X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Thanks Paul for the response...the draft does clarify a lot of bits and pieces on the offer-answer model spread out in various RFCs. When do we expect a formal RFC? 1. Although, not specifically mentioned but 100rel would be a must to support the RFC3959 for early media? 2. How does this RFC address the forking problem w.r.t. early media - could you throw in some more views? 3. Are there any other early media models being used by products out there? thanks, -kapil nayar --- Paul Kyzivat wrote: > > > Kapil Nayar wrote: > > Hi > > > > The RFC mentions the 183-session progress to have > both > > the early-offer and the answer which is replied > with > > the early-answer in PRACK. > > > > The RFC seems to imply that we DONOT need an > answer > > (SDP) in the 200-OK. > > Is it correct? > > *IF* the offer/answer cycle has been *completed* > prior to the sending of > the 200 for the INVITE, then SDP need not be > conveyed in the 200. To get > in this situation requires the use of reliable > provisional responses. > > This is a complex subject. Aspects of it are spread > across many RFCs, > and much is underspecified. See > draft-sawada-sipping-sip-offeranswer-01 > for a more complete specification. > > Paul > ___________________________________________________________ Try the all-new Yahoo! Mail. "The New Version is radically easier to use" – The Wall Street Journal http://uk.docs.yahoo.com/nowyoucan.html _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Tue Aug 08 18:10:57 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GAZmr-0006vm-BQ; Tue, 08 Aug 2006 18:10:57 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GAZmq-0006vh-32 for sipping@ietf.org; Tue, 08 Aug 2006 18:10:56 -0400 Received: from zcars04f.nortel.com ([47.129.242.57]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GAZmo-0007Vw-Oj for sipping@ietf.org; Tue, 08 Aug 2006 18:10:56 -0400 Received: from zrc2hxm2.corp.nortel.com (zrc2hxm2.corp.nortel.com [47.103.123.73]) by zcars04f.nortel.com (Switch-2.2.6/Switch-2.2.0) with ESMTP id k78MAdm15092; Tue, 8 Aug 2006 18:10:39 -0400 (EDT) x-mimeole: Produced By Microsoft Exchange V6.5 Content-class: urn:content-classes:message MIME-Version: 1.0 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: quoted-printable Subject: RE: [Sipping] Clarification on RFC 3959 - early media Date: Tue, 8 Aug 2006 17:10:38 -0500 Message-ID: <6FC4416DDE56C44DA0AEE67BC7CA43710FD327A8@zrc2hxm2.corp.nortel.com> In-Reply-To: <20060808220414.70194.qmail@web31503.mail.mud.yahoo.com> X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: [Sipping] Clarification on RFC 3959 - early media Thread-Index: Aca7Nqpw/V5YL8MAQvix4Q5+8BY73AAAKmLA From: "Brian Stucker" To: "Kapil Nayar" , "Paul Kyzivat" X-Spam-Score: 0.0 (/) X-Scan-Signature: 0fa76816851382eb71b0a882ccdc29ac Cc: sipping@ietf.org X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org You may wish to take a look at the following: http://www.ietf.org/internet-drafts/draft-stucker-sipping-early-media-co ping-01.txt Regards, Brian=20 > -----Original Message----- > From: Kapil Nayar [mailto:kapilnayar@yahoo.com]=20 > Sent: Tuesday, August 08, 2006 5:04 PM > To: Paul Kyzivat > Cc: sipping@ietf.org > Subject: Re: [Sipping] Clarification on RFC 3959 - early media >=20 > Thanks Paul for the response...the draft does clarify a lot=20 > of bits and pieces on the offer-answer model spread out in=20 > various RFCs. When do we expect a formal RFC? >=20 > 1. Although, not specifically mentioned but 100rel would be a=20 > must to support the RFC3959 for early media?=20 > 2. How does this RFC address the forking problem w.r.t. early=20 > media - could you throw in some more views? > 3. Are there any other early media models being used by=20 > products out there? >=20 > thanks, > -kapil nayar >=20 > --- Paul Kyzivat wrote: >=20 > >=20 > >=20 > > Kapil Nayar wrote: > > > Hi > > >=20 > > > The RFC mentions the 183-session progress to have > > both > > > the early-offer and the answer which is replied > > with > > > the early-answer in PRACK. > > >=20 > > > The RFC seems to imply that we DONOT need an > > answer > > > (SDP) in the 200-OK.=20 > > > Is it correct? > >=20 > > *IF* the offer/answer cycle has been *completed* prior to=20 > the sending=20 > > of the 200 for the INVITE, then SDP need not be conveyed in=20 > the 200.=20 > > To get in this situation requires the use of reliable provisional=20 > > responses. > >=20 > > This is a complex subject. Aspects of it are spread across=20 > many RFCs,=20 > > and much is underspecified. See > > draft-sawada-sipping-sip-offeranswer-01 > > for a more complete specification. > >=20 > > Paul > >=20 >=20 >=20 >=20 > =09 > ___________________________________________________________ > Try the all-new Yahoo! Mail. "The New Version is radically=20 > easier to use" - The Wall Street Journal=20 > http://uk.docs.yahoo.com/nowyoucan.html >=20 > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP=20 > Use sip-implementors@cs.columbia.edu for questions on current=20 > sip Use sip@ietf.org for new developments of core SIP >=20 _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Tue Aug 08 18:13:47 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GAZpb-0000Ad-5w; Tue, 08 Aug 2006 18:13:47 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GAZpZ-0000AY-KY for sipping@ietf.org; Tue, 08 Aug 2006 18:13:45 -0400 Received: from zrtps0kn.nortel.com ([47.140.192.55]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GAZpY-00082Y-94 for sipping@ietf.org; Tue, 08 Aug 2006 18:13:45 -0400 Received: from zrc2hxm2.corp.nortel.com (zrc2hxm2.corp.nortel.com [47.103.123.73]) by zrtps0kn.nortel.com (Switch-2.2.6/Switch-2.2.0) with ESMTP id k78MBru02001; Tue, 8 Aug 2006 18:11:54 -0400 (EDT) x-mimeole: Produced By Microsoft Exchange V6.5 Content-class: urn:content-classes:message MIME-Version: 1.0 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: quoted-printable Subject: RE: [Sipping] Clarification on RFC 3959 - early media Date: Tue, 8 Aug 2006 17:11:53 -0500 Message-ID: <6FC4416DDE56C44DA0AEE67BC7CA43710FD327AF@zrc2hxm2.corp.nortel.com> In-Reply-To: <44D349A9.4010100@cisco.com> X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: [Sipping] Clarification on RFC 3959 - early media Thread-Index: Aca3yShmHse1r61+SimygoFPlcuGXQDblmVw From: "Brian Stucker" To: "Paul Kyzivat" , "Kapil Nayar" X-Spam-Score: 0.0 (/) X-Scan-Signature: 4adaf050708fb13be3316a9eee889caa Cc: sipping@ietf.org X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Although if the offer/answer cycle has been completed, technically, SDP in the 200 response is not required, due to various mechanisms that try to deal (poorly) with early media that exist in the wild, NOT doing so may frequently cause one-way voice to occur in the presence of forking. Regards, Brian=20 > -----Original Message----- > From: Paul Kyzivat [mailto:pkyzivat@cisco.com]=20 > Sent: Friday, August 04, 2006 8:21 AM > To: Kapil Nayar > Cc: sipping@ietf.org > Subject: Re: [Sipping] Clarification on RFC 3959 - early media >=20 >=20 >=20 > Kapil Nayar wrote: > > Hi > >=20 > > The RFC mentions the 183-session progress to have both the=20 > early-offer=20 > > and the answer which is replied with the early-answer in PRACK. > >=20 > > The RFC seems to imply that we DONOT need an answer > > (SDP) in the 200-OK.=20 > > Is it correct? >=20 > *IF* the offer/answer cycle has been *completed* prior to the=20 > sending of the 200 for the INVITE, then SDP need not be=20 > conveyed in the 200. To get in this situation requires the=20 > use of reliable provisional responses. >=20 > This is a complex subject. Aspects of it are spread across=20 > many RFCs, and much is underspecified. See=20 > draft-sawada-sipping-sip-offeranswer-01 > for a more complete specification. >=20 > Paul >=20 > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP=20 > Use sip-implementors@cs.columbia.edu for questions on current=20 > sip Use sip@ietf.org for new developments of core SIP >=20 _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Tue Aug 08 19:13:06 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GAaky-0003RJ-Nc; Tue, 08 Aug 2006 19:13:04 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GAakw-0003M9-KP for sipping@ietf.org; Tue, 08 Aug 2006 19:13:02 -0400 Received: from sj-iport-4.cisco.com ([171.68.10.86]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GAaku-0002w8-6x for sipping@ietf.org; Tue, 08 Aug 2006 19:13:02 -0400 Received: from sj-dkim-6.cisco.com ([171.68.10.81]) by sj-iport-4.cisco.com with ESMTP; 08 Aug 2006 16:12:59 -0700 X-IronPort-AV: i="4.07,223,1151910000"; d="scan'208"; a="1845363997:sNHT28084358" Received: from sj-core-4.cisco.com (sj-core-4.cisco.com [171.68.223.138]) by sj-dkim-6.cisco.com (8.12.11.20060308/8.12.11) with ESMTP id k78NCxE7005926; Tue, 8 Aug 2006 16:12:59 -0700 Received: from xbh-rtp-201.amer.cisco.com (xbh-rtp-201.cisco.com [64.102.31.12]) by sj-core-4.cisco.com (8.12.10/8.12.6) with ESMTP id k78NCwIZ006314; Tue, 8 Aug 2006 16:12:59 -0700 (PDT) Received: from xfe-rtp-202.amer.cisco.com ([64.102.31.21]) by xbh-rtp-201.amer.cisco.com with Microsoft SMTPSVC(6.0.3790.1830); Tue, 8 Aug 2006 19:12:58 -0400 Received: from [161.44.79.104] ([161.44.79.104]) by xfe-rtp-202.amer.cisco.com with Microsoft SMTPSVC(6.0.3790.1830); Tue, 8 Aug 2006 19:12:58 -0400 Message-ID: <44D91A79.8070605@cisco.com> Date: Tue, 08 Aug 2006 19:12:57 -0400 From: Paul Kyzivat User-Agent: Thunderbird 1.5.0.5 (Windows/20060719) MIME-Version: 1.0 To: Brian Stucker Subject: Re: [Sipping] Clarification on RFC 3959 - early media References: <6FC4416DDE56C44DA0AEE67BC7CA43710FD327AF@zrc2hxm2.corp.nortel.com> In-Reply-To: <6FC4416DDE56C44DA0AEE67BC7CA43710FD327AF@zrc2hxm2.corp.nortel.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Content-Transfer-Encoding: 7bit X-OriginalArrivalTime: 08 Aug 2006 23:12:58.0297 (UTC) FILETIME=[2FEC8A90:01C6BB40] DKIM-Signature: a=rsa-sha1; q=dns; l=3291; t=1155078779; x=1155942779; c=relaxed/simple; s=sjdkim6002; h=Content-Type:From:Subject:Content-Transfer-Encoding:MIME-Version; d=cisco.com; i=pkyzivat@cisco.com; z=From:Paul=20Kyzivat=20 |Subject:Re=3A=20[Sipping]=20Clarification=20on=20RFC=203959=20-=20early=20media; X=v=3Dcisco.com=3B=20h=3D/ddYJqkrd9IxqsyFuDtUIsYBJE0=3D; b=VHB9BHMm/UFerbuUGmpRGfJioehdLjDrHV7FpptSPcguOA4nMgONe1kpaxC9Lq2+hywTVGTJ 47RfWtS+9icSg3Ri4a6Pnoc00KBXG3949wfCIOZu2FJNeuYay+nb7CdM; Authentication-Results: sj-dkim-6.cisco.com; header.From=pkyzivat@cisco.com; dkim=pass ( sig from cisco.com verified; ); X-Spam-Score: 0.0 (/) X-Scan-Signature: 3002fc2e661cd7f114cb6bae92fe88f1 Cc: sipping@ietf.org X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Brian Stucker wrote: > Although if the offer/answer cycle has been completed, technically, SDP > in the 200 response is not required, due to various mechanisms that try > to deal (poorly) with early media that exist in the wild, NOT doing so > may frequently cause one-way voice to occur in the presence of forking. Brian, You are talking about some UAC that is trying to cope with forking, chooses one fork to listen to, which turns out not to be the one that answers? In that case the UAC *should* still maintain the proper offer/answer state machine for each fork, so that when it learns which fork *did* answer it will know what answer to work with henceforth. If it doesn't, it can depend on the "answer" in the 200 only if that really is the answer - when there was no reliable provisional with the answer. When the UAS has sent an answer in a reliable provisional, it *can* put a copy of it in the 200. That will probably do no harm if there has only been one offer and one answer. But if there has been another offer/answer exchange before the 200, then putting SDP in the 200 is really problematic. As we have tried to interpret 3261, in that circumstance the SDP in the 200 must be the *first* answer, which is not the correct one to be used at that time. The dumb UAC that depends on the SDP in the 200 will then malfunction. To make it work correctly, the UAS would have to put the final answer into the 200, which violates 3261. In that case it seems safer to put nothing into the 200. So, what do you suggest here? I guess we could recommend that if there was an offer in the invite, and it was answered with a reliable provisional, and if there has been no subsequent offer/answer, then the 200 should contain a copy of the answer. Would that be better than recommending that the 200 never contain SDP when the offer/answer cycle has been completed? Paul > Regards, > Brian > >> -----Original Message----- >> From: Paul Kyzivat [mailto:pkyzivat@cisco.com] >> Sent: Friday, August 04, 2006 8:21 AM >> To: Kapil Nayar >> Cc: sipping@ietf.org >> Subject: Re: [Sipping] Clarification on RFC 3959 - early media >> >> >> >> Kapil Nayar wrote: >>> Hi >>> >>> The RFC mentions the 183-session progress to have both the >> early-offer >>> and the answer which is replied with the early-answer in PRACK. >>> >>> The RFC seems to imply that we DONOT need an answer >>> (SDP) in the 200-OK. >>> Is it correct? >> *IF* the offer/answer cycle has been *completed* prior to the >> sending of the 200 for the INVITE, then SDP need not be >> conveyed in the 200. To get in this situation requires the >> use of reliable provisional responses. >> >> This is a complex subject. Aspects of it are spread across >> many RFCs, and much is underspecified. See >> draft-sawada-sipping-sip-offeranswer-01 >> for a more complete specification. >> >> Paul >> >> _______________________________________________ >> Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping >> This list is for NEW development of the application of SIP >> Use sip-implementors@cs.columbia.edu for questions on current >> sip Use sip@ietf.org for new developments of core SIP >> > _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Tue Aug 08 19:26:19 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GAaxl-0004N6-0B; Tue, 08 Aug 2006 19:26:17 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GAaxj-0004N0-B5 for sipping@ietf.org; Tue, 08 Aug 2006 19:26:15 -0400 Received: from rtp-iport-1.cisco.com ([64.102.122.148]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GAaxh-0004uc-1O for sipping@ietf.org; Tue, 08 Aug 2006 19:26:15 -0400 Received: from rtp-dkim-1.cisco.com ([64.102.121.158]) by rtp-iport-1.cisco.com with ESMTP; 08 Aug 2006 16:26:11 -0700 X-BrightmailFiltered: true X-Brightmail-Tracker: AAAAAA== X-IronPort-AV: i="4.07,223,1151910000"; d="scan'208"; a="35218400:sNHT25451088" Received: from rtp-core-1.cisco.com (rtp-core-1.cisco.com [64.102.124.12]) by rtp-dkim-1.cisco.com (8.12.11.20060308/8.12.11) with ESMTP id k78NQBIU031992; Tue, 8 Aug 2006 19:26:11 -0400 Received: from xbh-rtp-201.amer.cisco.com (xbh-rtp-201.cisco.com [64.102.31.12]) by rtp-core-1.cisco.com (8.12.10/8.12.6) with ESMTP id k78NQBdp022646; Tue, 8 Aug 2006 19:26:11 -0400 (EDT) Received: from xfe-rtp-202.amer.cisco.com ([64.102.31.21]) by xbh-rtp-201.amer.cisco.com with Microsoft SMTPSVC(6.0.3790.1830); Tue, 8 Aug 2006 19:26:11 -0400 Received: from [161.44.79.104] ([161.44.79.104]) by xfe-rtp-202.amer.cisco.com with Microsoft SMTPSVC(6.0.3790.1830); Tue, 8 Aug 2006 19:26:11 -0400 Message-ID: <44D91D92.4010708@cisco.com> Date: Tue, 08 Aug 2006 19:26:10 -0400 From: Paul Kyzivat User-Agent: Thunderbird 1.5.0.5 (Windows/20060719) MIME-Version: 1.0 To: Kapil Nayar Subject: Re: [Sipping] Clarification on RFC 3959 - early media References: <20060808220414.70194.qmail@web31503.mail.mud.yahoo.com> In-Reply-To: <20060808220414.70194.qmail@web31503.mail.mud.yahoo.com> Content-Type: text/plain; charset=windows-1252; format=flowed Content-Transfer-Encoding: 8bit X-OriginalArrivalTime: 08 Aug 2006 23:26:11.0089 (UTC) FILETIME=[08770010:01C6BB42] DKIM-Signature: a=rsa-sha1; q=dns; l=2726; t=1155079571; x=1155943571; c=relaxed/simple; s=rtpdkim1001; h=Content-Type:From:Subject:Content-Transfer-Encoding:MIME-Version; d=cisco.com; i=pkyzivat@cisco.com; z=From:Paul=20Kyzivat=20 |Subject:Re=3A=20[Sipping]=20Clarification=20on=20RFC=203959=20-=20early=20media |To:Kapil=20Nayar=20; X=v=3Dcisco.com=3B=20h=3DWPFI1pcFqNj5gSaJ0bGBFkoLv7A=3D; b=X4sw8m9QEPhXXSAf+5FTCIfDazQsvdDxxqhTuvS5afuDD7ZcOshbj+f4i7VA6FOS6pmwEfOG 5tHtqQ2DLgyk9wFsBzdvCLo0r8Wf01GOptb1IpFCEmcsriVnk2ahqt6W; Authentication-Results: rtp-dkim-1.cisco.com; header.From=pkyzivat@cisco.com; dkim=pass ( sig from cisco.com verified; ); X-Spam-Score: 0.0 (/) X-Scan-Signature: 10d3e4e3c32e363f129e380e644649be Cc: sipping@ietf.org X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Kapil Nayar wrote: > Thanks Paul for the response...the draft does clarify > a lot of bits and pieces on the offer-answer model > spread out in various RFCs. When do we expect a formal > RFC? > > 1. Although, not specifically mentioned but 100rel > would be a must to support the RFC3959 for early > media? What we have been discussing is all the gateway model described in 3960, not the application model addressed by 3959. In theory I think you could use 3959 without using 100rel - using only UPDATE. In practice I doubt anyone supports only UPDATE and not 100rel. (In practice, I wonder if anybody supports 3959.) > 2. How does this RFC address the forking problem > w.r.t. early media - could you throw in some more > views? Offer/answer is complicated. Early media is complicated. And they are related. We are trying to keep the problem factored as best we can to retain some sanity. (Not sure how successful we are being in preserving sanity.) The offer/answer draft addresses part of the early media problem when the gateway model is used, in that it helps understand how the offer/answer exchange progresses on each fork. It does not say anything about how the media from the multiple sessions, that might be negotiated because of forking, is rendered to the user. > 3. Are there any other early media models being used > by products out there? I fear there are as many models as products. That is why we are trying to get this stuff more formalized. There is long discussion thread on early media, and there is another draft on early media coping. Paul > thanks, > -kapil nayar > > --- Paul Kyzivat wrote: > >> >> Kapil Nayar wrote: >>> Hi >>> >>> The RFC mentions the 183-session progress to have >> both >>> the early-offer and the answer which is replied >> with >>> the early-answer in PRACK. >>> >>> The RFC seems to imply that we DONOT need an >> answer >>> (SDP) in the 200-OK. >>> Is it correct? >> *IF* the offer/answer cycle has been *completed* >> prior to the sending of >> the 200 for the INVITE, then SDP need not be >> conveyed in the 200. To get >> in this situation requires the use of reliable >> provisional responses. >> >> This is a complex subject. Aspects of it are spread >> across many RFCs, >> and much is underspecified. See >> draft-sawada-sipping-sip-offeranswer-01 >> for a more complete specification. >> >> Paul >> > > > > > ___________________________________________________________ > Try the all-new Yahoo! Mail. "The New Version is radically easier to use" – The Wall Street Journal > http://uk.docs.yahoo.com/nowyoucan.html > _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Wed Aug 09 00:04:55 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GAfJL-0001dk-4Q; Wed, 09 Aug 2006 00:04:51 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GAfJJ-0001bR-As for sipping@ietf.org; Wed, 09 Aug 2006 00:04:49 -0400 Received: from rwcrmhc11.comcast.net ([204.127.192.81]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GAfJH-0005ym-24 for sipping@ietf.org; Wed, 09 Aug 2006 00:04:49 -0400 Received: from dragon.ariadne.com (dworley.hsd1.ma.comcast.net[24.34.79.42]) by comcast.net (rwcrmhc11) with ESMTP id <20060809040445m1100l3uthe>; Wed, 9 Aug 2006 04:04:46 +0000 Received: from dragon.ariadne.com (dragon.ariadne.com [127.0.0.1]) by dragon.ariadne.com (8.12.8/8.12.8) with ESMTP id k7944ieY013596 for ; Wed, 9 Aug 2006 00:04:44 -0400 Received: (from worley@localhost) by dragon.ariadne.com (8.12.8/8.12.8/Submit) id k7944itk013592; Wed, 9 Aug 2006 00:04:44 -0400 Date: Wed, 9 Aug 2006 00:04:44 -0400 Message-Id: <200608090404.k7944itk013592@dragon.ariadne.com> To: sipping@ietf.org From: Dale.Worley@comcast.net In-reply-to: <44D367B6.3060903@cisco.com> (jdrosen@cisco.com) Subject: Re: [Sipping] Question on do not disturb indication References: <50B1CBA96870A34799A506B2313F266709897F3A@ntht201e.siemenscomms.co.uk> <1154553270.2866.86.camel@sukothai.pingtel.com> <200608022251.k72Mpvkp020060@dragon.ariadne.com> <44D20161.9050506@cisco.com> <44D367B6.3060903@cisco.com> X-Spam-Score: 0.2 (/) X-Scan-Signature: 8abaac9e10c826e8252866cbe6766464 X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org From: Jonathan Rosenberg 1. when a user has multiple devices, and a receives a call, they would like to press a button on their phone called 'send to voicemail' which causes the call to go to voicemail. This should terminate any other branches for other UAs for that user if any are in progress, and halt any sequential search for othe UAs for that user if any were planned, and go to voicemail. 2. Like use case 1, except that the user wants the call to be rejected outright, and not even sent to voicemail. 3. Like use case 1, except that the user wants the caller to hear ringing for a while more and then get voicemail (sort of a polite block, where the caller can't differentiate this case from one where the called party wasn't there. Dean alluded to this). 4. send-to-X: the user wants the caller to be immediately redirected to somewhere else 1 and 4 can be accomplished by the callee UA either proxying the INVITE on to the desired destination, or returning a 302 to make the next upstream proxy do so. 3 can be accomplished by the UI ceasing user notification while the UA continues to handle the SIP transaction as it would if the user did not answer. 2 is trickier, because it wants to modify the forking results, when the UA isn't fully aware of what those forking results might be. Dale _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Wed Aug 09 02:46:37 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GAhps-0005Tw-4p; Wed, 09 Aug 2006 02:46:36 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GAhpr-0005Tr-AD for sipping@ietf.org; Wed, 09 Aug 2006 02:46:35 -0400 Received: from mailgw4.ericsson.se ([193.180.251.62]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GAhpp-0007vZ-DV for sipping@ietf.org; Wed, 09 Aug 2006 02:46:35 -0400 Received: from esealmw129.eemea.ericsson.se (unknown [153.88.254.120]) by mailgw4.ericsson.se (Symantec Mail Security) with ESMTP id AD9526E0002; Wed, 9 Aug 2006 08:46:30 +0200 (CEST) Received: from esealmw127.eemea.ericsson.se ([153.88.254.171]) by esealmw129.eemea.ericsson.se with Microsoft SMTPSVC(6.0.3790.1830); Wed, 9 Aug 2006 08:46:30 +0200 Received: from esealmw113.eemea.ericsson.se ([153.88.200.4]) by esealmw127.eemea.ericsson.se with Microsoft SMTPSVC(6.0.3790.1830); Wed, 9 Aug 2006 08:46:29 +0200 X-MimeOLE: Produced By Microsoft Exchange V6.5 Content-class: urn:content-classes:message MIME-Version: 1.0 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: quoted-printable Subject: RE: [Sipping] Clarification on RFC 3959 - early media Date: Wed, 9 Aug 2006 08:46:28 +0200 Message-ID: <5EB80D22825EEE42872083AD5BFFB594017F12E7@esealmw113.eemea.ericsson.se> In-Reply-To: <44D91A79.8070605@cisco.com> X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: [Sipping] Clarification on RFC 3959 - early media Thread-Index: Aca7QEAZe18Bm+/3Qwadigbio7F6sQAPt3Xg From: "Christer Holmberg \(JO/LMF\)" To: "Paul Kyzivat" , "Brian Stucker" X-OriginalArrivalTime: 09 Aug 2006 06:46:29.0771 (UTC) FILETIME=[8B39F5B0:01C6BB7F] X-Brightmail-Tracker: AAAAAA== X-Spam-Score: 0.0 (/) X-Scan-Signature: 6d95a152022472c7d6cdf886a0424dc6 Cc: sipping@ietf.org X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Hi,=20 >>Although if the offer/answer cycle has been completed, technically,=20 >>SDP in the 200 response is not required, due to various mechanisms=20 >>that try to deal (poorly) with early media that exist in the wild, NOT >>doing so may frequently cause one-way voice to occur in the presence of forking. >> >> >You are talking about some UAC that is trying to cope with forking, chooses one fork to listen to, which turns out not to=20 >be the one that answers? > >In that case the UAC *should* still maintain the proper offer/answer state machine for each fork, so that when it learns=20 >which fork *did* answer it will know what answer to work with henceforth. > >If it doesn't, it can depend on the "answer" in the 200 only if that really is the answer - when there was no reliable=20 >provisional with the answer. > >When the UAS has sent an answer in a reliable provisional, it *can* put a copy of it in the 200. That will probably do no=20 >harm if there has only been one offer and one answer. But if there has been another offer/answer exchange before the 200,=20 >then putting SDP in the 200 is really problematic. As we have tried to interpret 3261, in that circumstance the SDP in=20 >the 200 must be the *first* answer, which is not the correct one to be used at that time. The dumb UAC that depends on=20 >the SDP in the 200 will then malfunction. To make it work correctly, the UAS would have to put the final answer into the=20 >200, which violates 3261. In that case it seems safer to put nothing into the 200. > >So, what do you suggest here? > >I guess we could recommend that if there was an offer in the invite, and it was answered with a reliable provisional, and=20 >if there has been no subsequent offer/answer, then the 200 should contain a copy of the answer. Would that be better than=20 >recommending that the 200 never contain SDP when the offer/answer cycle has been completed? My proposal would be to say that the UAC, if it has already received an answer, simply doesn't care about the SDP in the 200 OK. Then it doesn't matter what the UAS sends, what recommendations the UAS follows, etc... Regards, Christer Paul > Regards, > Brian >=20 >> -----Original Message----- >> From: Paul Kyzivat [mailto:pkyzivat@cisco.com] >> Sent: Friday, August 04, 2006 8:21 AM >> To: Kapil Nayar >> Cc: sipping@ietf.org >> Subject: Re: [Sipping] Clarification on RFC 3959 - early media >> >> >> >> Kapil Nayar wrote: >>> Hi >>> >>> The RFC mentions the 183-session progress to have both the >> early-offer >>> and the answer which is replied with the early-answer in PRACK. >>> >>> The RFC seems to imply that we DONOT need an answer >>> (SDP) in the 200-OK.=20 >>> Is it correct? >> *IF* the offer/answer cycle has been *completed* prior to the sending >> of the 200 for the INVITE, then SDP need not be conveyed in the 200.=20 >> To get in this situation requires the use of reliable provisional=20 >> responses. >> >> This is a complex subject. Aspects of it are spread across many RFCs, >> and much is underspecified. See >> draft-sawada-sipping-sip-offeranswer-01 >> for a more complete specification. >> >> Paul >> >> _______________________________________________ >> Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping >> This list is for NEW development of the application of SIP Use=20 >> sip-implementors@cs.columbia.edu for questions on current sip Use=20 >> sip@ietf.org for new developments of core SIP >> >=20 _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Wed Aug 09 09:08:34 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GAnnD-000212-Iq; Wed, 09 Aug 2006 09:08:15 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GAnnC-00020t-92 for sipping@ietf.org; Wed, 09 Aug 2006 09:08:14 -0400 Received: from rtp-iport-1.cisco.com ([64.102.122.148]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GAnnA-0001BY-TM for sipping@ietf.org; Wed, 09 Aug 2006 09:08:14 -0400 Received: from rtp-dkim-2.cisco.com ([64.102.121.159]) by rtp-iport-1.cisco.com with ESMTP; 09 Aug 2006 06:08:11 -0700 X-BrightmailFiltered: true X-Brightmail-Tracker: AAAAAA== X-IronPort-AV: i="4.07,224,1151910000"; d="scan'208"; a="35296291:sNHT43766302" Received: from rtp-core-2.cisco.com (rtp-core-2.cisco.com [64.102.124.13]) by rtp-dkim-2.cisco.com (8.12.11.20060308/8.12.11) with ESMTP id k79D8ABl032152; Wed, 9 Aug 2006 09:08:10 -0400 Received: from xbh-rtp-201.amer.cisco.com (xbh-rtp-201.cisco.com [64.102.31.12]) by rtp-core-2.cisco.com (8.12.10/8.12.6) with ESMTP id k79D8Ae2018125; Wed, 9 Aug 2006 09:08:10 -0400 (EDT) Received: from xfe-rtp-202.amer.cisco.com ([64.102.31.21]) by xbh-rtp-201.amer.cisco.com with Microsoft SMTPSVC(6.0.3790.1830); Wed, 9 Aug 2006 09:08:10 -0400 Received: from [161.44.79.104] ([161.44.79.104]) by xfe-rtp-202.amer.cisco.com with Microsoft SMTPSVC(6.0.3790.1830); Wed, 9 Aug 2006 09:08:10 -0400 Message-ID: <44D9DE39.8010006@cisco.com> Date: Wed, 09 Aug 2006 09:08:09 -0400 From: Paul Kyzivat User-Agent: Thunderbird 1.5.0.5 (Windows/20060719) MIME-Version: 1.0 To: Dale.Worley@comcast.net Subject: Re: [Sipping] Question on do not disturb indication References: <50B1CBA96870A34799A506B2313F266709897F3A@ntht201e.siemenscomms.co.uk> <1154553270.2866.86.camel@sukothai.pingtel.com> <200608022251.k72Mpvkp020060@dragon.ariadne.com> <44D20161.9050506@cisco.com> <44D367B6.3060903@cisco.com> <200608090404.k7944itk013592@dragon.ariadne.com> In-Reply-To: <200608090404.k7944itk013592@dragon.ariadne.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Content-Transfer-Encoding: 7bit X-OriginalArrivalTime: 09 Aug 2006 13:08:10.0243 (UTC) FILETIME=[DCF8AD30:01C6BBB4] DKIM-Signature: a=rsa-sha1; q=dns; l=3375; t=1155128890; x=1155992890; c=relaxed/simple; s=rtpdkim2001; h=Content-Type:From:Subject:Content-Transfer-Encoding:MIME-Version; d=cisco.com; i=pkyzivat@cisco.com; z=From:Paul=20Kyzivat=20 |Subject:Re=3A=20[Sipping]=20Question=20on=20do=20not=20disturb=20indication |To:Dale.Worley@comcast.net; X=v=3Dcisco.com=3B=20h=3Dyu9ohsv1kFqX2uC2CMjguD3sTnY=3D; b=UkfgiOBC/EwFSu+HxxYZFS2M/jkmRKf7K0EktrY0RzaK/vMRi9qvT/xQzb77vLgeDeQ+Dy/3 jjrY/U3AEe7X36030eXx4ZmbgzGbclAH++KwQoDpQ1gwMy7KkPl8s13z; Authentication-Results: rtp-dkim-2.cisco.com; header.From=pkyzivat@cisco.com; dkim=pass ( sig from cisco.com verified; ); X-Spam-Score: 0.0 (/) X-Scan-Signature: 00e94c813bef7832af255170dca19e36 Cc: sipping@ietf.org X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org The trouble with these scenarios is that they are only part of a bigger picture. Consider the following: - Alice calls Bob. - Bob has configured his proxy with some policies about how to handle the call - One of Bob's policies is to parallel fork calls to Bob's own phones (several) and to Charlie's AOR - Charlie has configured his proxy with policies about how to handle the call. - One of Charlie's policies is to parallel fork calls to his own phones. Now assume that all of the phones above have buttons with meanings something like what Jonathan mentions. One question is whether those buttons imply invocation of some new policy that competes with the proxy based policies, or if the buttons should simply provide input to the policy decisions. That isn't a question we can or should answer here. I think we must recognize that there are many ways of doing this. What we can do is work through a variety of use cases and convince ourselves there is sufficient mechanism to facilitate the use cases we think important. It seems to me that for a good result you must assume that the actions of the phone need to be coordinated with the actions of the proxy in front of the phone. (E.g. if the proxy has policy to fail over to voicemail, then the phone probably ought not be returning 3xx when it is busy or doesn't answer.) That extends to the case where the call is delegated to another AOR. But in that case it is much harder to ensure the behaviors are coordinated. Paul Dale.Worley@comcast.net wrote: > From: Jonathan Rosenberg > > 1. when a user has multiple devices, and a receives a call, they would > like to press a button on their phone called 'send to voicemail' which > causes the call to go to voicemail. This should terminate any other > branches for other UAs for that user if any are in progress, and halt > any sequential search for othe UAs for that user if any were planned, > and go to voicemail. > > 2. Like use case 1, except that the user wants the call to be rejected > outright, and not even sent to voicemail. > > 3. Like use case 1, except that the user wants the caller to hear > ringing for a while more and then get voicemail (sort of a polite block, > where the caller can't differentiate this case from one where the called > party wasn't there. Dean alluded to this). > > 4. send-to-X: the user wants the caller to be immediately redirected to > somewhere else > > 1 and 4 can be accomplished by the callee UA either proxying the > INVITE on to the desired destination, or returning a 302 to make the > next upstream proxy do so. > > 3 can be accomplished by the UI ceasing user notification while the UA > continues to handle the SIP transaction as it would if the user did > not answer. > > 2 is trickier, because it wants to modify the forking results, when > the UA isn't fully aware of what those forking results might be. > > Dale > > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP > Use sip-implementors@cs.columbia.edu for questions on current sip > Use sip@ietf.org for new developments of core SIP > _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Wed Aug 09 09:57:24 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GAoYj-0003Hp-V3; Wed, 09 Aug 2006 09:57:21 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GAoYi-0003Hc-2d for sipping@ietf.org; Wed, 09 Aug 2006 09:57:20 -0400 Received: from nf-out-0910.google.com ([64.233.182.188]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GAoYg-0005eG-NS for sipping@ietf.org; Wed, 09 Aug 2006 09:57:20 -0400 Received: by nf-out-0910.google.com with SMTP id p48so190369nfa for ; Wed, 09 Aug 2006 06:57:16 -0700 (PDT) DomainKey-Signature: a=rsa-sha1; q=dns; c=nofws; s=beta; d=gmail.com; h=received:message-id:date:from:reply-to:sender:to:subject:cc:in-reply-to:mime-version:content-type:content-transfer-encoding:content-disposition:references:x-google-sender-auth; b=riW84RSy1A3271A1ulLKVXDpSss0RZ//XPx0q1x5+9ro9Hgb3C8JfdhCTbY4y3LmkhucUI9p35GmH0XJGI7GkxKdfPvA4EnFKmcx3mRB1VDQr3YBkXYVR0pD+UlOjR9rHEM3jGq0p0qNmAA7uLo0koBF6HsyE1U6pGQzvHU5xQs= Received: by 10.49.43.11 with SMTP id v11mr1524361nfj; Wed, 09 Aug 2006 06:57:16 -0700 (PDT) Received: by 10.48.164.7 with HTTP; Wed, 9 Aug 2006 06:57:16 -0700 (PDT) Message-ID: Date: Wed, 9 Aug 2006 09:57:16 -0400 From: "Xiaotao Wu" To: "Dale.Worley@comcast.net" Subject: Re: [Sipping] Question on do not disturb indication In-Reply-To: <200608090404.k7944itk013592@dragon.ariadne.com> MIME-Version: 1.0 Content-Type: text/plain; charset=ISO-8859-1; format=flowed Content-Transfer-Encoding: 7bit Content-Disposition: inline References: <50B1CBA96870A34799A506B2313F266709897F3A@ntht201e.siemenscomms.co.uk> <1154553270.2866.86.camel@sukothai.pingtel.com> <200608022251.k72Mpvkp020060@dragon.ariadne.com> <44D20161.9050506@cisco.com> <44D367B6.3060903@cisco.com> <200608090404.k7944itk013592@dragon.ariadne.com> X-Google-Sender-Auth: d98aaf9dae70a563 X-Spam-Score: 0.5 (/) X-Scan-Signature: f4c2cf0bccc868e4cc88dace71fb3f44 Cc: sipping@ietf.org X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list Reply-To: xiaotaow@gmail.com List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org On 8/9/06, Dale.Worley@comcast.net wrote: > From: Jonathan Rosenberg > > 1. when a user has multiple devices, and a receives a call, they would > like to press a button on their phone called 'send to voicemail' which > causes the call to go to voicemail. This should terminate any other > branches for other UAs for that user if any are in progress, and halt > any sequential search for othe UAs for that user if any were planned, > and go to voicemail. > > 2. Like use case 1, except that the user wants the call to be rejected > outright, and not even sent to voicemail. > > 3. Like use case 1, except that the user wants the caller to hear > ringing for a while more and then get voicemail (sort of a polite block, > where the caller can't differentiate this case from one where the called > party wasn't there. Dean alluded to this). > > 4. send-to-X: the user wants the caller to be immediately redirected to > somewhere else > > 1 and 4 can be accomplished by the callee UA either proxying the > INVITE on to the desired destination, or returning a 302 to make the > next upstream proxy do so. 302 may not terminate the other branches. -Xiaotao > > 3 can be accomplished by the UI ceasing user notification while the UA > continues to handle the SIP transaction as it would if the user did > not answer. > > 2 is trickier, because it wants to modify the forking results, when > the UA isn't fully aware of what those forking results might be. > > Dale > > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP > Use sip-implementors@cs.columbia.edu for questions on current sip > Use sip@ietf.org for new developments of core SIP > -- -Xiaotao =========================================================== Name : Xiaotao Wu Email : xwu@avaya.com, xiaotaow@gmail.com Web : http://www.cs.columbia.edu/~xiaotaow Phone : (732) 852-2133 SIP : sip:xiaotaow@cs.columbia.edu Address : 1M-240, 307 Middletown Lincroft Road Lincroft, NJ 07738-1526, U.S.A. _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Wed Aug 09 10:40:30 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GApEP-0000St-8i; Wed, 09 Aug 2006 10:40:25 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GApEO-0000So-K7 for sipping@ietf.org; Wed, 09 Aug 2006 10:40:24 -0400 Received: from rwcrmhc12.comcast.net ([204.127.192.82]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GApEL-00018d-BM for sipping@ietf.org; Wed, 09 Aug 2006 10:40:24 -0400 Received: from dragon.ariadne.com (dworley.hsd1.ma.comcast.net[24.34.79.42]) by comcast.net (rwcrmhc12) with ESMTP id <20060809144020m1200bon2ie>; Wed, 9 Aug 2006 14:40:20 +0000 Received: from dragon.ariadne.com (dragon.ariadne.com [127.0.0.1]) by dragon.ariadne.com (8.12.8/8.12.8) with ESMTP id k79EeIeY028648 for ; Wed, 9 Aug 2006 10:40:19 -0400 Received: (from worley@localhost) by dragon.ariadne.com (8.12.8/8.12.8/Submit) id k79EeI76028644; Wed, 9 Aug 2006 10:40:18 -0400 Date: Wed, 9 Aug 2006 10:40:18 -0400 Message-Id: <200608091440.k79EeI76028644@dragon.ariadne.com> To: sipping@ietf.org From: Dale.Worley@comcast.net In-reply-to: (xwu@avaya.com) Subject: Re: [Sipping] Question on do not disturb indication References: <50B1CBA96870A34799A506B2313F266709897F3A@ntht201e.siemenscomms.co.uk> <1154553270.2866.86.camel@sukothai.pingtel.com> <200608022251.k72Mpvkp020060@dragon.ariadne.com> <44D20161.9050506@cisco.com> <44D367B6.3060903@cisco.com> <200608090404.k7944itk013592@dragon.ariadne.com> X-Spam-Score: 0.2 (/) X-Scan-Signature: 1ac7cc0a4cd376402b85bc1961a86ac2 X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org From: "Xiaotao Wu" 302 may not terminate the other branches. True. But if it goes to a voicemail system, which presumably answers immediately, then it will. But some of these concepts (particularly, "reject"), assume that the UA that is ringing has the knowledge/authority to affect the routing for all branches of the call. But is not always the case, since the original request-URI may not be the AOR for the UA, but rather some other AOR that has its own policies for routing. Dale _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Wed Aug 09 21:45:11 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GAzbS-00060S-UC; Wed, 09 Aug 2006 21:44:54 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GAzbS-00060N-4c for sipping@ietf.org; Wed, 09 Aug 2006 21:44:54 -0400 Received: from buraja.hst.terra.com.br ([200.176.10.198]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GAzbP-0006XP-Gu for sipping@ietf.org; Wed, 09 Aug 2006 21:44:54 -0400 Received: from cosmoledo.terra.com.br (cosmoledo.hst.terra.com.br [200.176.10.14]) by buraja.hst.terra.com.br (Postfix) with ESMTP id CF14122A4073 for ; Wed, 9 Aug 2006 22:44:49 -0300 (BRT) X-Terra-Karma: -2% X-Terra-Hash: c232b83eee2c8629bad7dca7c05bfece Received-SPF: pass (cosmoledo.terra.com.br: domain of terra.com.br designates 200.176.10.14 as permitted sender) client-ip=200.176.10.14; envelope-from=marchisotti@terra.com.br; helo=GUSTAVO; Received: from GUSTAVO (201-0-83-126.dsl.telesp.net.br [201.0.83.126]) (authenticated user marchisotti) by cosmoledo.terra.com.br (Postfix) with ESMTP id 24EC02A2806D for ; Wed, 9 Aug 2006 22:44:49 -0300 (BRT) From: "marchisotti" To: Date: Wed, 9 Aug 2006 22:44:45 -0300 Organization: Gustavo G. Marchisotti Message-ID: <011201c6bc1e$8ee968e0$0301010a@GUSTAVO> MIME-Version: 1.0 X-Mailer: Microsoft Office Outlook 11 X-MimeOLE: Produced By Microsoft MimeOLE V6.00.2900.2962 Thread-Index: Aca8Ho5mKfNRcwhTQdyz7P3MN6LM9A== X-Spam-Score: 0.2 (/) X-Scan-Signature: a8041eca2a724d631b098c15e9048ce9 Subject: [Sipping] Translation ISDN (PRI) and R2D to SIP X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list Reply-To: marchisotti@terra.com.br List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Content-Type: multipart/mixed; boundary="===============2094566427==" Errors-To: sipping-bounces@ietf.org This is a multi-part message in MIME format. --===============2094566427== Content-Type: multipart/alternative; boundary="----=_NextPart_000_0113_01C6BC05.699C30E0" This is a multi-part message in MIME format. ------=_NextPart_000_0113_01C6BC05.699C30E0 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: 7bit Hi, Anybody knows the correctly translation of messages between ISDN to SIP and R2D to SIP? I need the correctly mapping in all common situations: - Normal call; - Busy call; - No attempt call; - Etc. Is very difficult to find material about that (I found only the mapping between SS7 and SIP). Thanks! Gustavo ------=_NextPart_000_0113_01C6BC05.699C30E0 Content-Type: text/html; charset="us-ascii" Content-Transfer-Encoding: quoted-printable

Hi,

 

Anybody knows the correctly translation of messages = between ISDN to SIP and R2D to SIP? I need the correctly mapping in all common situations:

-          Normal = call;

-          Busy = call;

-          No attempt = call;

-          Etc.

 

Is very difficult to find material about that (I = found only the mapping between SS7 and SIP).

 

Thanks!

Gustavo

------=_NextPart_000_0113_01C6BC05.699C30E0-- --===============2094566427== Content-Type: text/plain; charset="us-ascii" MIME-Version: 1.0 Content-Transfer-Encoding: 7bit Content-Disposition: inline _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP --===============2094566427==-- From sipping-bounces@ietf.org Wed Aug 09 22:47:45 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GB0aC-0008P1-Dr; Wed, 09 Aug 2006 22:47:40 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GB0aA-0008IN-KT for sipping@ietf.org; Wed, 09 Aug 2006 22:47:38 -0400 Received: from rtp-iport-2.cisco.com ([64.102.122.149]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GB0a9-0000zD-9h for sipping@ietf.org; Wed, 09 Aug 2006 22:47:38 -0400 Received: from rtp-dkim-2.cisco.com ([64.102.121.159]) by rtp-iport-2.cisco.com with ESMTP; 09 Aug 2006 22:47:37 -0400 X-IronPort-AV: i="4.08,108,1154923200"; d="scan'208,217"; a="95961449:sNHT52557120" Received: from rtp-core-1.cisco.com (rtp-core-1.cisco.com [64.102.124.12]) by rtp-dkim-2.cisco.com (8.12.11.20060308/8.12.11) with ESMTP id k7A2lbhu000677; Wed, 9 Aug 2006 22:47:37 -0400 Received: from xbh-rtp-201.amer.cisco.com (xbh-rtp-201.cisco.com [64.102.31.12]) by rtp-core-1.cisco.com (8.12.10/8.12.6) with ESMTP id k7A2ladp009019; Wed, 9 Aug 2006 22:47:36 -0400 (EDT) Received: from xmb-rtp-215.amer.cisco.com ([64.102.31.124]) by xbh-rtp-201.amer.cisco.com with Microsoft SMTPSVC(6.0.3790.1830); Wed, 9 Aug 2006 22:47:36 -0400 X-MimeOLE: Produced By Microsoft Exchange V6.5 Content-class: urn:content-classes:message MIME-Version: 1.0 Subject: RE: [Sipping] Translation ISDN (PRI) and R2D to SIP Date: Wed, 9 Aug 2006 22:47:35 -0400 Message-ID: <8983EC086A9D954BA74D9763E853CF3E01F177F0@xmb-rtp-215.amer.cisco.com> X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: [Sipping] Translation ISDN (PRI) and R2D to SIP Thread-Index: Aca8Ho5mKfNRcwhTQdyz7P3MN6LM9AACLf1A From: "Sanjay Sinha \(sanjsinh\)" To: , X-OriginalArrivalTime: 10 Aug 2006 02:47:36.0711 (UTC) FILETIME=[56785170:01C6BC27] DKIM-Signature: a=rsa-sha1; q=dns; l=6872; t=1155178057; x=1156042057; c=relaxed/simple; s=rtpdkim2001; h=Content-Type:From:Subject:Content-Transfer-Encoding:MIME-Version; d=cisco.com; i=sanjsinh@cisco.com; z=From:=22Sanjay=20Sinha=20\(sanjsinh\)=22=20 |Subject:RE=3A=20[Sipping]=20Translation=20ISDN=20(PRI)=20and=20R2D=20to=20SIP=20 |To:,=20; X=v=3Dcisco.com=3B=20h=3DK8j4KOLrqzQTwW97xjhhsfup1r8=3D; b=L1a1FQRfWHYak9i5a3Ai67EmY7BQmAXFsvN8sijSCh12J32fZPp0nbLkCaB7JYobfPNmBYNW 6RSQTbI4P90CmKjAb120xpbLTekfnA0syotgP1v094P2t7TkbAkYmdz7; Authentication-Results: rtp-dkim-2.cisco.com; header.From=sanjsinh@cisco.com; dkim=pass ( 59 extraneous bytes; sig from cisco.com verified; ); X-Spam-Score: 0.0 (/) X-Scan-Signature: 88b11fc64c1bfdb4425294ef5374ca07 Cc: X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Content-Type: multipart/mixed; boundary="===============1782436703==" Errors-To: sipping-bounces@ietf.org This is a multi-part message in MIME format. --===============1782436703== Content-class: urn:content-classes:message Content-Type: multipart/alternative; boundary="----_=_NextPart_001_01C6BC27.5657A39A" This is a multi-part message in MIME format. ------_=_NextPart_001_01C6BC27.5657A39A Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: quoted-printable You may want to look at RFC 3666 =20 Sanjay ________________________________ From: marchisotti [mailto:marchisotti@terra.com.br]=20 Sent: Wednesday, August 09, 2006 9:45 PM To: sipping@ietf.org Subject: [Sipping] Translation ISDN (PRI) and R2D to SIP=20 Hi, =20 Anybody knows the correctly translation of messages between ISDN to SIP and R2D to SIP? I need the correctly mapping in all common situations: - Normal call; - Busy call; - No attempt call; - Etc. =20 Is very difficult to find material about that (I found only the mapping between SS7 and SIP).=20 =20 Thanks! Gustavo ------_=_NextPart_001_01C6BC27.5657A39A Content-Type: text/html; charset="us-ascii" Content-Transfer-Encoding: quoted-printable
You may want to look at RFC = 3666
 
Sanjay


From: marchisotti=20 [mailto:marchisotti@terra.com.br]
Sent: Wednesday, August 09, = 2006=20 9:45 PM
To: sipping@ietf.org
Subject: [Sipping] = Translation=20 ISDN (PRI) and R2D to SIP

Hi,

 

Anybody knows the = correctly=20 translation of messages between ISDN to SIP and R2D to SIP? I need the = correctly=20 mapping in all common situations:

-         =20 Normal=20 call;

-         =20 Busy=20 call;

-         =20 No attempt=20 call;

-         =20 Etc.

 

Is very difficult to find = material=20 about that (I found only the mapping between SS7 and SIP).=20

 

Thanks!

Gustavo

------_=_NextPart_001_01C6BC27.5657A39A-- --===============1782436703== Content-Type: text/plain; charset="us-ascii" MIME-Version: 1.0 Content-Transfer-Encoding: 7bit Content-Disposition: inline _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP --===============1782436703==-- From sipping-bounces@ietf.org Thu Aug 10 02:44:06 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GB4Gr-0000Ei-8O; Thu, 10 Aug 2006 02:43:57 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GB4Gq-0000Ed-Av for sipping@ietf.org; Thu, 10 Aug 2006 02:43:56 -0400 Received: from mailrelay2.alcatel.de ([194.113.59.96]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GB4Go-0006Ya-Tb for sipping@ietf.org; Thu, 10 Aug 2006 02:43:56 -0400 Received: from demail05.netfr.alcatel.fr (demail05.netfr.alcatel.fr [155.132.182.205]) by mailrelay2.alcatel.de (8.12.11.20060308/8.12.11/ICT TSC MAIL 2005) with ESMTP id k7A6hpRt016880; Thu, 10 Aug 2006 08:43:51 +0200 In-Reply-To: <8983EC086A9D954BA74D9763E853CF3E01F177F0@xmb-rtp-215.amer.cisco.com> Subject: RE: [Sipping] Translation ISDN (PRI) and R2D to SIP To: "Sanjay Sinha \(sanjsinh\)" X-Mailer: Lotus Notes Release 6.5 September 26, 2003 Message-ID: From: Albrecht.Schwarz@alcatel.de Date: Thu, 10 Aug 2006 08:43:49 +0200 X-MIMETrack: Serialize by Router on DEMAIL05/DE/ALCATEL(Release 5.0.13aHF163 | June 23, 2005) at 08/10/2006 08:43:51 MIME-Version: 1.0 Content-type: text/plain; charset=us-ascii X-Scanned-By: MIMEDefang 2.49 on 149.204.45.73 X-Spam-Score: 0.2 (/) X-Scan-Signature: 244a2fd369eaf00ce6820a760a3de2e8 Cc: sipping@ietf.org, marchisotti@terra.com.br X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Gustavo, * public ISDN NNI => SS7 ISUP to SIP => see ITU-T Rec. Q.1912.5 * public ISDN UNI => DSS1 (Q.931) to SIP => not yet any interworking standard available to my knowledge * privat ISDN UNI => e.g. QSIG to SIP => see RFC 4497 Guess you look at the UNI cases (due to your PRI indication). - Albrecht "Sanjay Sinha \(sanjsinh\)" To: , Subject: RE: [Sipping] Translation ISDN (PRI) and R2D to SIP 10.08.2006 04:47 You may want to look at RFC 3666 Sanjay From: marchisotti [mailto:marchisotti@terra.com.br] Sent: Wednesday, August 09, 2006 9:45 PM To: sipping@ietf.org Subject: [Sipping] Translation ISDN (PRI) and R2D to SIP Hi, Anybody knows the correctly translation of messages between ISDN to SIP and R2D to SIP? I need the correctly mapping in all common situations: - Normal call; - Busy call; - No attempt call; - Etc. Is very difficult to find material about that (I found only the mapping between SS7 and SIP). Thanks! Gustavo_______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Thu Aug 10 10:54:22 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GBBvA-0005iK-Il; Thu, 10 Aug 2006 10:54:04 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GBBv6-0005bK-Ot; Thu, 10 Aug 2006 10:54:00 -0400 Received: from ns1.neustar.com ([2001:503:c779:1a::9c9a:108a]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GBBv5-0000m0-J9; Thu, 10 Aug 2006 10:54:00 -0400 Received: from stiedprstage1.ietf.org (stiedprstage1.va.neustar.com [10.31.47.10]) by ns1.neustar.com (Postfix) with ESMTP id 8A52626E22; Thu, 10 Aug 2006 14:53:59 +0000 (GMT) Received: from ietf by stiedprstage1.ietf.org with local (Exim 4.43) id 1GBBv5-0003zs-Es; Thu, 10 Aug 2006 10:53:59 -0400 X-test-idtracker: no To: IETF-Announce From: The IESG Message-Id: Date: Thu, 10 Aug 2006 10:53:59 -0400 X-Spam-Score: -2.8 (--) X-Scan-Signature: 1ac7cc0a4cd376402b85bc1961a86ac2 Cc: sipping@ietf.org Subject: [Sipping] Last Call: 'Registration Event Package Extension for Session Initiation Protocol (SIP) Globally Routable User Agent URIs (GRUU)' to Proposed Standard (draft-ietf-sipping-gruu-reg-event) X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list Reply-To: iesg@ietf.org List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org The IESG has received a request from the Session Initiation Proposal Investigation WG to consider the following document: - 'Registration Event Package Extension for Session Initiation Protocol (SIP) Globally Routable User Agent URIs (GRUU) ' as a Proposed Standard The IESG plans to make a decision in the next few weeks, and solicits final comments on this action. Please send any comments to the iesg@ietf.org or ietf@ietf.org mailing lists by 2006-08-24. The file can be obtained via http://www.ietf.org/internet-drafts/draft-ietf-sipping-gruu-reg-event-06.txt _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Thu Aug 10 17:57:29 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GBIWm-0006Xa-I8; Thu, 10 Aug 2006 17:57:20 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GBIWk-0006XV-R1 for sipping@ietf.org; Thu, 10 Aug 2006 17:57:18 -0400 Received: from rtp-iport-2.cisco.com ([64.102.122.149]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GBIWj-0007tL-Gb for sipping@ietf.org; Thu, 10 Aug 2006 17:57:18 -0400 Received: from rtp-dkim-1.cisco.com ([64.102.121.158]) by rtp-iport-2.cisco.com with ESMTP; 10 Aug 2006 17:57:18 -0400 X-IronPort-AV: i="4.08,112,1154923200"; d="scan'208"; a="96121776:sNHT26823672" Received: from rtp-core-2.cisco.com (rtp-core-2.cisco.com [64.102.124.13]) by rtp-dkim-1.cisco.com (8.12.11.20060308/8.12.11) with ESMTP id k7ALvHiR028124; Thu, 10 Aug 2006 17:57:17 -0400 Received: from xbh-rtp-211.amer.cisco.com (xbh-rtp-211.cisco.com [64.102.31.102]) by rtp-core-2.cisco.com (8.12.10/8.12.6) with ESMTP id k7ALvHe2021400; Thu, 10 Aug 2006 17:57:17 -0400 (EDT) Received: from xfe-rtp-202.amer.cisco.com ([64.102.31.21]) by xbh-rtp-211.amer.cisco.com with Microsoft SMTPSVC(6.0.3790.1830); Thu, 10 Aug 2006 17:57:16 -0400 Received: from [161.44.79.104] ([161.44.79.104]) by xfe-rtp-202.amer.cisco.com with Microsoft SMTPSVC(6.0.3790.1830); Thu, 10 Aug 2006 17:57:16 -0400 Message-ID: <44DBABBC.8000808@cisco.com> Date: Thu, 10 Aug 2006 17:57:16 -0400 From: Paul Kyzivat User-Agent: Thunderbird 1.5.0.5 (Windows/20060719) MIME-Version: 1.0 To: "Christer Holmberg (JO/LMF)" Subject: Re: [Sipping] Clarification on RFC 3959 - early media References: <5EB80D22825EEE42872083AD5BFFB594017F12E7@esealmw113.eemea.ericsson.se> In-Reply-To: <5EB80D22825EEE42872083AD5BFFB594017F12E7@esealmw113.eemea.ericsson.se> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Content-Transfer-Encoding: 7bit X-OriginalArrivalTime: 10 Aug 2006 21:57:16.0514 (UTC) FILETIME=[F1A2F420:01C6BCC7] DKIM-Signature: a=rsa-sha1; q=dns; l=2653; t=1155247037; x=1156111037; c=relaxed/simple; s=rtpdkim1001; h=Content-Type:From:Subject:Content-Transfer-Encoding:MIME-Version; d=cisco.com; i=pkyzivat@cisco.com; z=From:Paul=20Kyzivat=20 |Subject:Re=3A=20[Sipping]=20Clarification=20on=20RFC=203959=20-=20early=20media |To:=22Christer=20Holmberg=20(JO/LMF)=22=20; X=v=3Dcisco.com=3B=20h=3D/ddYJqkrd9IxqsyFuDtUIsYBJE0=3D; b=UnryJ6+HL02xM315WuVaJp7GGtGRBBjTIbWZHHP/tpW3459cjgWYrmgAcpi/NDP77pXMSuRA WXsUUgtssSidDblwp3ZU63WgETJ6fYRz/ahprOvmTHrUEoEe+saj7zWV; Authentication-Results: rtp-dkim-1.cisco.com; header.From=pkyzivat@cisco.com; dkim=pass ( sig from cisco.com verified; ); X-Spam-Score: 0.0 (/) X-Scan-Signature: f4c2cf0bccc868e4cc88dace71fb3f44 Cc: sipping@ietf.org, Brian Stucker X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Christer Holmberg (JO/LMF) wrote: > Hi, > >>> Although if the offer/answer cycle has been completed, technically, >>> SDP in the 200 response is not required, due to various mechanisms >>> that try to deal (poorly) with early media that exist in the wild, NOT > >>> doing so may frequently cause one-way voice to occur in the presence > of forking. >>> >> You are talking about some UAC that is trying to cope with forking, > chooses one fork to listen to, which turns out not to >> be the one that answers? >> >> In that case the UAC *should* still maintain the proper offer/answer > state machine for each fork, so that when it learns >> which fork *did* answer it will know what answer to work with > henceforth. >> If it doesn't, it can depend on the "answer" in the 200 only if that > really is the answer - when there was no reliable >> provisional with the answer. >> >> When the UAS has sent an answer in a reliable provisional, it *can* put > a copy of it in the 200. That will probably do no >> harm if there has only been one offer and one answer. But if there has > been another offer/answer exchange before the 200, >> then putting SDP in the 200 is really problematic. As we have tried to > interpret 3261, in that circumstance the SDP in >> the 200 must be the *first* answer, which is not the correct one to be > used at that time. The dumb UAC that depends on >> the SDP in the 200 will then malfunction. To make it work correctly, > the UAS would have to put the final answer into the >> 200, which violates 3261. In that case it seems safer to put nothing > into the 200. >> So, what do you suggest here? >> >> I guess we could recommend that if there was an offer in the invite, > and it was answered with a reliable provisional, and >> if there has been no subsequent offer/answer, then the 200 should > contain a copy of the answer. Would that be better than >> recommending that the 200 never contain SDP when the offer/answer cycle > has been completed? > > My proposal would be to say that the UAC, if it has already received an > answer, simply doesn't care about the SDP in the 200 OK. Then it doesn't > matter what the UAS sends, what recommendations the UAS follows, etc... Yes, but there is a lot of broken stuff out there in the wild. And its not particularly surprising because this stuff is pretty confusing. What the UAS does can enhance interop with UACs that are confused. And visa versa. Of course no single set of rules is going to make interop possible among all the confused and all the different ways they may be confused. Paul _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Thu Aug 10 18:29:03 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GBJ1Q-00056p-3d; Thu, 10 Aug 2006 18:29:00 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GBJ1O-00056k-Kz for sipping@ietf.org; Thu, 10 Aug 2006 18:28:58 -0400 Received: from rtp-iport-2.cisco.com ([64.102.122.149]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GBJ1N-0004Br-CJ for sipping@ietf.org; Thu, 10 Aug 2006 18:28:58 -0400 Received: from rtp-dkim-2.cisco.com ([64.102.121.159]) by rtp-iport-2.cisco.com with ESMTP; 10 Aug 2006 18:28:57 -0400 X-IronPort-AV: i="4.08,112,1154923200"; d="scan'208"; a="96125455:sNHT722595206" Received: from rtp-core-1.cisco.com (rtp-core-1.cisco.com [64.102.124.12]) by rtp-dkim-2.cisco.com (8.12.11.20060308/8.12.11) with ESMTP id k7AMSuMG004178; Thu, 10 Aug 2006 18:28:56 -0400 Received: from xbh-rtp-211.amer.cisco.com (xbh-rtp-211.cisco.com [64.102.31.102]) by rtp-core-1.cisco.com (8.12.10/8.12.6) with ESMTP id k7AMSsdp011767; Thu, 10 Aug 2006 18:28:54 -0400 (EDT) Received: from xfe-rtp-201.amer.cisco.com ([64.102.31.38]) by xbh-rtp-211.amer.cisco.com with Microsoft SMTPSVC(6.0.3790.1830); Thu, 10 Aug 2006 18:28:54 -0400 Received: from [161.44.79.104] ([161.44.79.104]) by xfe-rtp-201.amer.cisco.com with Microsoft SMTPSVC(6.0.3790.1830); Thu, 10 Aug 2006 18:28:53 -0400 Message-ID: <44DBB324.3080307@cisco.com> Date: Thu, 10 Aug 2006 18:28:52 -0400 From: Paul Kyzivat User-Agent: Thunderbird 1.5.0.5 (Windows/20060719) MIME-Version: 1.0 To: Brian Stucker Subject: Re: [Sipping] Clarification on RFC 3959 - early media References: <6FC4416DDE56C44DA0AEE67BC7CA43710FD327A8@zrc2hxm2.corp.nortel.com> In-Reply-To: <6FC4416DDE56C44DA0AEE67BC7CA43710FD327A8@zrc2hxm2.corp.nortel.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Content-Transfer-Encoding: 7bit X-OriginalArrivalTime: 10 Aug 2006 22:28:54.0017 (UTC) FILETIME=[5CA2EF10:01C6BCCC] DKIM-Signature: a=rsa-sha1; q=dns; l=739; t=1155248936; x=1156112936; c=relaxed/simple; s=rtpdkim2001; h=Content-Type:From:Subject:Content-Transfer-Encoding:MIME-Version; d=cisco.com; i=pkyzivat@cisco.com; z=From:Paul=20Kyzivat=20 |Subject:Re=3A=20[Sipping]=20Clarification=20on=20RFC=203959=20-=20early=20media |To:Brian=20Stucker=20; X=v=3Dcisco.com=3B=20h=3D/ddYJqkrd9IxqsyFuDtUIsYBJE0=3D; b=r4iUsWwX04YzKSBAuWttaxJJAdW2yoGdG5AH5Rqb8f3JFaHIrXT/ZJZ0WodaMHWL1W99dW7j Tz2w6pUyh48tlH55WIDLNmDRVocCRhwq7CsoVFVWrjFCby0gbVrpeUGE; Authentication-Results: rtp-dkim-2.cisco.com; header.From=pkyzivat@cisco.com; dkim=pass ( sig from cisco.com verified; ); X-Spam-Score: 0.0 (/) X-Scan-Signature: 7d33c50f3756db14428398e2bdedd581 Cc: sipping@ietf.org X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Brian, I just got around to looking at this. I need to think about the techniques you propose at the end. But I am having trouble getting past the first half of the document. I have never heard mention of the things you describe as Pre-routing early media or Pre-presentation early media. Nor have I heard of the Proxy-side coping mechanisms you describe. These are all positively *obscene*. You suggest that these are common. Is that really true??? (Maybe you live in a particularly dangerous neighborhood.) Tell me it isn't true! Please! Paul Brian Stucker wrote: > You may wish to take a look at the following: > > http://www.ietf.org/internet-drafts/draft-stucker-sipping-early-media-co > ping-01.txt _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Thu Aug 10 19:58:38 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GBKPz-0005cz-0O; Thu, 10 Aug 2006 19:58:27 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GBKPx-0005cu-PV for sipping@ietf.org; Thu, 10 Aug 2006 19:58:25 -0400 Received: from sj-iport-5.cisco.com ([171.68.10.87]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GBKPw-0004DI-FA for sipping@ietf.org; Thu, 10 Aug 2006 19:58:25 -0400 Received: from sj-dkim-5.cisco.com ([171.68.10.79]) by sj-iport-5.cisco.com with ESMTP; 10 Aug 2006 16:58:24 -0700 X-IronPort-AV: i="4.08,112,1154934000"; d="scan'208"; a="311071533:sNHT31011540" Received: from sj-core-4.cisco.com (sj-core-4.cisco.com [171.68.223.138]) by sj-dkim-5.cisco.com (8.12.11.20060308/8.12.11) with ESMTP id k7ANwN3o019531; Thu, 10 Aug 2006 16:58:23 -0700 Received: from dwingwxp ([10.32.130.33]) by sj-core-4.cisco.com (8.12.10/8.12.6) with ESMTP id k7ANwNIZ008080; Thu, 10 Aug 2006 16:58:23 -0700 (PDT) From: "Dan Wing" To: "'Paul Kyzivat'" , "'Brian Stucker'" Subject: RE: [Sipping] Clarification on RFC 3959 - early media Date: Thu, 10 Aug 2006 16:58:22 -0700 Message-ID: <10b301c6bcd8$dd3da3c0$2182200a@amer.cisco.com> MIME-Version: 1.0 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: 7bit X-Mailer: Microsoft Office Outlook 11 x-mimeole: Produced By Microsoft MimeOLE V6.00.2900.2869 In-Reply-To: <44DBB324.3080307@cisco.com> Thread-Index: Aca8zG8aKy3H0f1PSp6x5kdMKA5PRQADE9ow DKIM-Signature: a=rsa-sha1; q=dns; l=1507; t=1155254303; x=1156118303; c=relaxed/simple; s=sjdkim5002; h=Content-Type:From:Subject:Content-Transfer-Encoding:MIME-Version; d=cisco.com; i=dwing@cisco.com; z=From:=22Dan=20Wing=22=20 |Subject:RE=3A=20[Sipping]=20Clarification=20on=20RFC=203959=20-=20early=20media; X=v=3Dcisco.com=3B=20h=3DKT3kGL/JJQbMa9Ia7/jkma0+1NQ=3D; b=WgbWSHPBUkaJfsf9ahj+woirY5+9QiT5hxNd76jajbOMjP3/aAxUXbrkvbaujGrYIBkhRSby wIhNEoY+EynEN1sQIRLmG25RMA2mIog5iJR+kKB3cLPEFjoTVOcEWjuq; Authentication-Results: sj-dkim-5.cisco.com; header.From=dwing@cisco.com; dkim=pass ( sig from cisco.com verified; ); X-Spam-Score: 0.0 (/) X-Scan-Signature: cab78e1e39c4b328567edb48482b6a69 Cc: sipping@ietf.org X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Not to mention they're really nasty with SRTP, and create another reason for endpoints to allow RTP to be played while SRTP is being set up. -d > -----Original Message----- > From: Paul Kyzivat [mailto:pkyzivat@cisco.com] > Sent: Thursday, August 10, 2006 3:29 PM > To: Brian Stucker > Cc: sipping@ietf.org > Subject: Re: [Sipping] Clarification on RFC 3959 - early media > > Brian, > > I just got around to looking at this. I need to think about the > techniques you propose at the end. But I am having trouble > getting past > the first half of the document. I have never heard mention of > the things > you describe as Pre-routing early media or Pre-presentation > early media. > Nor have I heard of the Proxy-side coping mechanisms you describe. > > These are all positively *obscene*. You suggest that these > are common. > Is that really true??? (Maybe you live in a particularly dangerous > neighborhood.) > > Tell me it isn't true! Please! > > Paul > > Brian Stucker wrote: > > You may wish to take a look at the following: > > > > > http://www.ietf.org/internet-drafts/draft-stucker-sipping-earl > y-media-co > > ping-01.txt > > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP > Use sip-implementors@cs.columbia.edu for questions on current sip > Use sip@ietf.org for new developments of core SIP _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Fri Aug 11 10:56:46 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GBYQz-0007qR-R7; Fri, 11 Aug 2006 10:56:25 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GBYQy-0007qL-8W for sipping@ietf.org; Fri, 11 Aug 2006 10:56:24 -0400 Received: from stsc1260-eth-s1-s1p1-vip.va.neustar.com ([156.154.16.129] helo=chiedprmail1.ietf.org) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GBV66-0001gy-FT for sipping@ietf.org; Fri, 11 Aug 2006 07:22:38 -0400 Received: from sj-iport-5.cisco.com ([171.68.10.87]) by chiedprmail1.ietf.org with esmtp (Exim 4.43) id 1GBUyH-0004jz-8Z for sipping@ietf.org; Fri, 11 Aug 2006 07:14:34 -0400 Received: from sj-dkim-8.cisco.com ([171.68.10.93]) by sj-iport-5.cisco.com with ESMTP; 11 Aug 2006 04:14:31 -0700 X-IronPort-AV: i="4.08,114,1154934000"; d="scan'208,217"; a="311161858:sNHT58362856" Received: from sj-core-3.cisco.com (sj-core-3.cisco.com [171.68.223.137]) by sj-dkim-8.cisco.com (8.12.11.20060308/8.12.11) with ESMTP id k7BBEUss005167; Fri, 11 Aug 2006 04:14:30 -0700 Received: from xbh-rtp-211.amer.cisco.com (xbh-rtp-211.cisco.com [64.102.31.102]) by sj-core-3.cisco.com (8.12.10/8.12.6) with ESMTP id k7BBETuD012881; Fri, 11 Aug 2006 04:14:29 -0700 (PDT) Received: from xmb-rtp-20b.amer.cisco.com ([64.102.31.53]) by xbh-rtp-211.amer.cisco.com with Microsoft SMTPSVC(6.0.3790.1830); Fri, 11 Aug 2006 07:14:29 -0400 X-MimeOLE: Produced By Microsoft Exchange V6.5 Content-class: urn:content-classes:message MIME-Version: 1.0 Date: Fri, 11 Aug 2006 07:11:45 -0400 Message-ID: <15B86BC7352F864BB53A47B540C089B602044010@xmb-rtp-20b.amer.cisco.com> X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: b2bua/call control aspect for rfc4244 (History-Info header) Thread-Index: Aca9Nu5QiUeP+JGhR1WpV0IaHjwzpQ== From: "Ganesh Jayadevan \(gjayadev\)" To: X-OriginalArrivalTime: 11 Aug 2006 11:14:29.0017 (UTC) FILETIME=[5007B090:01C6BD37] DKIM-Signature: a=rsa-sha1; q=dns; l=16356; t=1155294870; x=1156158870; c=relaxed/relaxed; s=sjdkim8002; h=Content-Type:From:Subject:Content-Transfer-Encoding:MIME-Version; d=cisco.com; i=gjayadev@cisco.com; z=From:=22Ganesh=20Jayadevan=20\(gjayadev\)=22=20 |Subject:b2bua/call=20control=20aspect=20for=20rfc4244=20(History-Info=20header); X=v=3Dcisco.com=3B=20h=3Ds+46YAUXcVBLPH5A7ZdhEoaAUW8=3D; b=j7IZDCqwj8HrhwSwiImOgtLN93NcABhexq1tRoob8cqQoYkxE0iH1mmgYaNLmY5FSGer9iuo CGzHVTi6XoZNOl/fa+2KdBiZwN/yvbyefb3vKM/FKyT7+bm1Ajk/dxc/; Authentication-Results: sj-dkim-8.cisco.com; header.DKIM-Signature=gjayadev@cisco.com; dkim=fail ( 57 extraneous bytes; RSA-96 err: "Ganesh Jayadevan \(gjayadev\)" ; cisco.com/sjdkim8002 fail; ); header.From=gjayadev@cisco.com; dkim=neutral X-Spam-Score: -2.6 (--) X-Scan-Signature: 4166dd0e0c668adc975c3d3e0f1bce3b Cc: "Cullen Jennings \(fluffy\)" , "Sudipto Mukherjee \(sudiptom\)" , Jon.Peterson@NeuStar.biz, mark@digitalfountain.com, mary.barnes@nortel.com, "Mohammad Al-Taraireh \(maltarai\)" Subject: [Sipping] b2bua/call control aspect for rfc4244 (History-Info header) X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Content-Type: multipart/mixed; boundary="===============2052488190==" Errors-To: sipping-bounces@ietf.org This is a multi-part message in MIME format. --===============2052488190== Content-class: urn:content-classes:message Content-Type: multipart/alternative; boundary="----_=_NextPart_001_01C6BD37.4FBFB3A2" This is a multi-part message in MIME format. ------_=_NextPart_001_01C6BD37.4FBFB3A2 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: quoted-printable Folks, =20 rfc4244 (History-Info) addresses what the UAC, UAS, and=20 the proxy must do with the History-Info header. True to the tradition of SIP, a b2bua is not given any special status but is seen as a concat of a UAS-UAC. =20 I am wondering if this leaves a gap and hinders achieving the objectives of the rfc. =20 If the requirements were interpreted strictly, the UAC in a b2bua will follow the UAC rules as stated in=20 Section 4.3.1. ie to essentially insert a H-I and add and index of "1". =20 This may be insufficient.=20 =20 B2BUAs play an important role in Call control and a call/session attempt across a B2BUA should be viewed as a single piece. =20 For example, if we had: =20 1 2 UAC (A)--------B2BUA or CC--------UAS (B) | | 3 \---------------------------- Voice Mail =20 Lets say A calls B via a huntgroup. So, the=20 It's the History-Info in the leg between B2BUA and B (ie 2) that is likely to contain the correct identity of B and therefore has to be used when the call gets retargeted to Voice Mail. Which means the History-Info header from leg 2 has to make it back to leg 1. =20 For this to happen the History-Info header must be copied over from leg 1 to leg 2 using the=20 "Basic Forwarding" Rule in 4.3.3.1.3 ie: =20 Also:=20 =20 "In the case of a Request that is being forwarded, the index is determined by adding another level of indexing since the depth/length of the branch is increasing." =20 Reworking the index is important because the tree structure has to be maintained. =20 I can think of two ways of dealing with this: =20 1. The Application is an acknowledged entity in the rfc. Treat UAC/UAS not as an endpoint but as a component that may be passed a list of History-Info headers by the Application. =20 The reworking of the index can either be Application reposibility or UAC responsibility.=20 =20 We have two options: =20 1.1) UAC is probably where it should reworked as this fits with the proxy reqts of forwarding..and we are doing forwarding in a larger sense. =20 1.2) Let Application rework the index and pass it down to the UAC. =20 Also, the headers can be passed into the UAS from the Application to address the voice-mail use-case identified above. =20 In either case, I suggest we add some language to clarify this. Because currently the UAC is instructed to add a History-Info header with an index value of "1". And the UAS is asked to simply copy back the H-I=20 headers from the request to response. I find this insufficient. =20 2. Specify a model in which B2BUA plays a role and define responsibilities. =20 What do you folks think? =20 Thanks, Ganesh =20 ------_=_NextPart_001_01C6BD37.4FBFB3A2 Content-Type: text/html; charset="us-ascii" Content-Transfer-Encoding: quoted-printable
Folks,
 
rfc4244=20 (History-Info) addresses what the UAC, UAS, and
the proxy=20 must do with the History-Info=20 header.
True to the=20 tradition of SIP, a b2bua is not given any special
status but=20 is seen as a concat of a UAS-UAC.
 
I am=20 wondering if this leaves a gap and hinders achieving = the
objectives=20 of the rfc.
 
If the=20 requirements were interpreted strictly, the UAC
in a b2bua=20 will follow the UAC rules as stated in
Section=20 4.3.1. ie to essentially insert a H-I and add
and index=20 of "1".
 
This may be=20 insufficient.
 
B2BUAs play=20 an important role in Call control and a = call/session
attempt=20 across a B2BUA should be viewed as a single piece.
 
For example,=20 if we had:
 
       &nbs= p; =20 1            =       =20 2
UAC=20 (A)--------B2BUA or CC--------UAS (B)
|
|       &nb= sp;      =20 3
 \---------------------------- Voice= =20 Mail
 
Lets say A=20 calls B via a huntgroup. So, the
It's the=20 History-Info in the leg between
B2BUA and B=20 (ie 2) that is likely to contain the
correct=20 identity of B and therefore has to
be used when=20 the call gets retargeted to Voice Mail.
Which means=20 the History-Info header from leg 2 has
to make it=20 back to leg 1.
 
For this to=20 happen the History-Info header must
be copied=20 over from leg 1 to leg 2 using the
"Basic=20 Forwarding" Rule in 4.3.3.1.3 ie:
 
Also:
 
"In the case of a Request that is = being
forwarded, the=20 index is determined by adding another level of
indexing since the=20 depth/length of the branch is increasing."
 
Reworking the index is important because the = tree=20 structure
has to be = maintained.
 
I can think of two ways of dealing = with=20 this:
 
1. The Application is an acknowledged entity = in the=20 rfc.
Treat UAC/UAS not as an endpoint but as a=20 component
that may be passed a list of History-Info = headers by the=20 Application.
 
The reworking of the index can either be = Application=20 reposibility
or UAC responsibility. =
 
We have two = options:
 
1.1) UAC is probably where it should=20 reworked
     as this fits with = the proxy=20 reqts of forwarding..and we are doing
     forwarding in a = larger=20 sense.
 
1.2) Let Application rework the index and = pass it down to=20 the UAC.
 
Also, the headers can be passed into the UAS = from the=20 Application to
address the voice-mail use-case identified=20 above.
 
In either case, I suggest we add some = language to clarify=20 this. Because
currently the UAC is instructed to add a = History-Info=20 header with an
index value of "1". And the UAS is asked to = simply copy=20 back the H-I
headers from the request to response. I find = this=20 insufficient.
 
2. Specify a model in which B2BUA plays a = role and define=20 responsibilities.
 
What do you folks = think?
 
Thanks,
Ganesh
 
------_=_NextPart_001_01C6BD37.4FBFB3A2-- --===============2052488190== Content-Type: text/plain; charset="us-ascii" MIME-Version: 1.0 Content-Transfer-Encoding: 7bit Content-Disposition: inline _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP --===============2052488190==-- From sipping-bounces@ietf.org Fri Aug 11 11:38:11 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GBZ5O-0005pC-Hg; Fri, 11 Aug 2006 11:38:10 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GBZ5N-0005p7-Iz for sipping@ietf.org; Fri, 11 Aug 2006 11:38:09 -0400 Received: from zcars04f.nortel.com ([47.129.242.57]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GBZ5M-0006nd-7j for sipping@ietf.org; Fri, 11 Aug 2006 11:38:09 -0400 Received: from zrc2hxm2.corp.nortel.com (zrc2hxm2.corp.nortel.com [47.103.123.73]) by zcars04f.nortel.com (Switch-2.2.6/Switch-2.2.0) with ESMTP id k7BFc0R22348; Fri, 11 Aug 2006 11:38:00 -0400 (EDT) x-mimeole: Produced By Microsoft Exchange V6.5 Content-class: urn:content-classes:message MIME-Version: 1.0 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: quoted-printable Subject: RE: [Sipping] Clarification on RFC 3959 - early media Date: Fri, 11 Aug 2006 10:37:59 -0500 Message-ID: <6FC4416DDE56C44DA0AEE67BC7CA43710FDDECC4@zrc2hxm2.corp.nortel.com> In-Reply-To: <44DBB324.3080307@cisco.com> X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: [Sipping] Clarification on RFC 3959 - early media Thread-Index: Aca8zGXHzO90gODhTDuYZdU+w3bazwAj1r4A From: "Brian Stucker" To: "Paul Kyzivat" X-Spam-Score: 0.0 (/) X-Scan-Signature: 21c69d3cfc2dd19218717dbe1d974352 Cc: sipping@ietf.org X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Pre-routing, pre-presentation, etc. are monikers that I came up with to try to explain at which points early media can arise during processing. The idea here is that all early media is not equal, some early media is more evil than others. I've seen all of the techniques mentioned in the draft in the wild. Many of them are undetectable to the originating UA. Regards, Brian=20 > -----Original Message----- > From: Paul Kyzivat [mailto:pkyzivat@cisco.com]=20 > Sent: Thursday, August 10, 2006 5:29 PM > To: Stucker, Brian (RICH1:B620) > Cc: Kapil Nayar; sipping@ietf.org > Subject: Re: [Sipping] Clarification on RFC 3959 - early media >=20 > Brian, >=20 > I just got around to looking at this. I need to think about=20 > the techniques you propose at the end. But I am having=20 > trouble getting past the first half of the document. I have=20 > never heard mention of the things you describe as Pre-routing=20 > early media or Pre-presentation early media.=20 > Nor have I heard of the Proxy-side coping mechanisms you describe. >=20 > These are all positively *obscene*. You suggest that these=20 > are common.=20 > Is that really true??? (Maybe you live in a particularly dangerous > neighborhood.) >=20 > Tell me it isn't true! Please! >=20 > Paul >=20 > Brian Stucker wrote: > > You may wish to take a look at the following: > >=20 > >=20 > http://www.ietf.org/internet-drafts/draft-stucker-sipping-early-media- > > co > > ping-01.txt >=20 _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Fri Aug 11 14:25:36 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GBbhH-0000sy-SH; Fri, 11 Aug 2006 14:25:27 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GBbhF-0000s6-Qx for sipping@ietf.org; Fri, 11 Aug 2006 14:25:25 -0400 Received: from rtp-iport-2.cisco.com ([64.102.122.149]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GBbhE-0003L3-GM for sipping@ietf.org; Fri, 11 Aug 2006 14:25:25 -0400 Received: from rtp-dkim-2.cisco.com ([64.102.121.159]) by rtp-iport-2.cisco.com with ESMTP; 11 Aug 2006 14:25:24 -0400 X-IronPort-AV: i="4.08,115,1154923200"; d="scan'208"; a="96279103:sNHT30775270" Received: from rtp-core-2.cisco.com (rtp-core-2.cisco.com [64.102.124.13]) by rtp-dkim-2.cisco.com (8.12.11.20060308/8.12.11) with ESMTP id k7BIPOdI027634; Fri, 11 Aug 2006 14:25:24 -0400 Received: from xbh-rtp-201.amer.cisco.com (xbh-rtp-201.cisco.com [64.102.31.12]) by rtp-core-2.cisco.com (8.12.10/8.12.6) with ESMTP id k7BIPOe2006872; Fri, 11 Aug 2006 14:25:24 -0400 (EDT) Received: from xfe-rtp-202.amer.cisco.com ([64.102.31.21]) by xbh-rtp-201.amer.cisco.com with Microsoft SMTPSVC(6.0.3790.1830); Fri, 11 Aug 2006 14:25:24 -0400 Received: from [161.44.79.104] ([161.44.79.104]) by xfe-rtp-202.amer.cisco.com with Microsoft SMTPSVC(6.0.3790.1830); Fri, 11 Aug 2006 14:25:23 -0400 Message-ID: <44DCCB93.6090802@cisco.com> Date: Fri, 11 Aug 2006 14:25:23 -0400 From: Paul Kyzivat User-Agent: Thunderbird 1.5.0.5 (Windows/20060719) MIME-Version: 1.0 To: Brian Stucker Subject: Re: [Sipping] Clarification on RFC 3959 - early media References: <6FC4416DDE56C44DA0AEE67BC7CA43710FDDECC4@zrc2hxm2.corp.nortel.com> In-Reply-To: <6FC4416DDE56C44DA0AEE67BC7CA43710FDDECC4@zrc2hxm2.corp.nortel.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Content-Transfer-Encoding: 7bit X-OriginalArrivalTime: 11 Aug 2006 18:25:23.0548 (UTC) FILETIME=[828809C0:01C6BD73] DKIM-Signature: a=rsa-sha1; q=dns; l=1857; t=1155320724; x=1156184724; c=relaxed/simple; s=rtpdkim2001; h=Content-Type:From:Subject:Content-Transfer-Encoding:MIME-Version; d=cisco.com; i=pkyzivat@cisco.com; z=From:Paul=20Kyzivat=20 |Subject:Re=3A=20[Sipping]=20Clarification=20on=20RFC=203959=20-=20early=20media |To:Brian=20Stucker=20; X=v=3Dcisco.com=3B=20h=3D/ddYJqkrd9IxqsyFuDtUIsYBJE0=3D; b=aUb+gJ50w8BMnhYIGsgbja3dkxEm4ZsAf3eJonMfi9t2+y6tAdg8sRn1pgU101+3DVoV06ds xXmARjUS5czS795MdfDRx25SqUvcpaY5HCjyuvkUw0L3KRgsoYlbwbYq; Authentication-Results: rtp-dkim-2.cisco.com; header.From=pkyzivat@cisco.com; dkim=pass ( sig from cisco.com verified; ); X-Spam-Score: 0.0 (/) X-Scan-Signature: 02ec665d00de228c50c93ed6b5e4fc1a Cc: sipping@ietf.org X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Brian Stucker wrote: > Pre-routing, pre-presentation, etc. are monikers that I came up with to > try to explain at which points early media can arise during processing. It isn't the names that bother me. (It's obvious you coined those.) It's the behaviors described that are disturbing. > The idea here is that all early media is not equal, some early media is > more evil than others. > > I've seen all of the techniques mentioned in the draft in the wild. Time to declare open season! (Kill those nasty beasts before they hurt the children.) (I wouldn't be killing any of my kin would I?) > Many of them are undetectable to the originating UA. Except that things work funny, or sometimes don't work. Paul > Regards, > Brian > >> -----Original Message----- >> From: Paul Kyzivat [mailto:pkyzivat@cisco.com] >> Sent: Thursday, August 10, 2006 5:29 PM >> To: Stucker, Brian (RICH1:B620) >> Cc: Kapil Nayar; sipping@ietf.org >> Subject: Re: [Sipping] Clarification on RFC 3959 - early media >> >> Brian, >> >> I just got around to looking at this. I need to think about >> the techniques you propose at the end. But I am having >> trouble getting past the first half of the document. I have >> never heard mention of the things you describe as Pre-routing >> early media or Pre-presentation early media. >> Nor have I heard of the Proxy-side coping mechanisms you describe. >> >> These are all positively *obscene*. You suggest that these >> are common. >> Is that really true??? (Maybe you live in a particularly dangerous >> neighborhood.) >> >> Tell me it isn't true! Please! >> >> Paul >> >> Brian Stucker wrote: >>> You may wish to take a look at the following: >>> >>> >> http://www.ietf.org/internet-drafts/draft-stucker-sipping-early-media- >>> co >>> ping-01.txt > _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Fri Aug 11 14:45:48 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GBc0t-00087P-Lk; Fri, 11 Aug 2006 14:45:43 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GBc0s-000865-G5 for sipping@ietf.org; Fri, 11 Aug 2006 14:45:42 -0400 Received: from rtp-iport-2.cisco.com ([64.102.122.149]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GBbpj-0004Op-D3 for sipping@ietf.org; Fri, 11 Aug 2006 14:34:13 -0400 Received: from rtp-dkim-2.cisco.com ([64.102.121.159]) by rtp-iport-2.cisco.com with ESMTP; 11 Aug 2006 14:34:11 -0400 X-IronPort-AV: i="4.08,115,1154923200"; d="scan'208"; a="96280867:sNHT41991036" Received: from rtp-core-2.cisco.com (rtp-core-2.cisco.com [64.102.124.13]) by rtp-dkim-2.cisco.com (8.12.11.20060308/8.12.11) with ESMTP id k7BIYBAQ032329; Fri, 11 Aug 2006 14:34:11 -0400 Received: from xbh-rtp-211.amer.cisco.com (xbh-rtp-211.cisco.com [64.102.31.102]) by rtp-core-2.cisco.com (8.12.10/8.12.6) with ESMTP id k7BIY6eA009475; Fri, 11 Aug 2006 14:34:10 -0400 (EDT) Received: from xmb-rtp-20b.amer.cisco.com ([64.102.31.53]) by xbh-rtp-211.amer.cisco.com with Microsoft SMTPSVC(6.0.3790.1830); Fri, 11 Aug 2006 14:34:10 -0400 X-MimeOLE: Produced By Microsoft Exchange V6.5 Content-class: urn:content-classes:message MIME-Version: 1.0 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: quoted-printable Subject: RE: [Sipping] b2bua/call control aspect for rfc4244 (History-Info header) Date: Fri, 11 Aug 2006 14:31:26 -0400 Message-ID: <15B86BC7352F864BB53A47B540C089B6020441C8@xmb-rtp-20b.amer.cisco.com> X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: [Sipping] b2bua/call control aspect for rfc4244 (History-Info header) Thread-Index: Aca9cqWc4cyu9FhXQQiUzGtu4vxtIgAALHZg From: "Ganesh Jayadevan \(gjayadev\)" To: "Dean Willis" X-OriginalArrivalTime: 11 Aug 2006 18:34:10.0108 (UTC) FILETIME=[BC62B7C0:01C6BD74] DKIM-Signature: a=rsa-sha1; q=dns; l=3095; t=1155321251; x=1156185251; c=relaxed/simple; s=rtpdkim2001; h=Content-Type:From:Subject:Content-Transfer-Encoding:MIME-Version; d=cisco.com; i=gjayadev@cisco.com; z=From:=22Ganesh=20Jayadevan=20\(gjayadev\)=22=20 |Subject:RE=3A=20[Sipping]=20b2bua/call=20control=20aspect=20for=20rfc4244=20(His tory-Info=20header) |To:=22Dean=20Willis=22=20; X=v=3Dcisco.com=3B=20h=3D1cLGn2DD5TC9ybLvkwjBKeHi0Jo=3D; b=VZW06tC/5aD98cPnbxo4G/CuKb6lLV/EPZvKb9T91cJK9VWIsn1o61UImZBAffac9i8sxRel tq5rjWfMHvW+F9f+4r2YtaUPXezM/Rjpfn5dxZCAms06UGg67uTBAwy1; Authentication-Results: rtp-dkim-2.cisco.com; header.From=gjayadev@cisco.com; dkim=pass ( sig from cisco.com verified; ); X-Spam-Score: 0.5 (/) X-Scan-Signature: d185fa790257f526fedfd5d01ed9c976 Cc: "Cullen Jennings \(fluffy\)" , "Sudipto Mukherjee \(sudiptom\)" , Jon.Peterson@NeuStar.biz, mark@digitalfountain.com, mary.barnes@nortel.com, "Mohammad Al-Taraireh \(maltarai\)" , sipping@ietf.org X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Hi Dean, I see what you are saying about B2BUAs in general. I also proposed another alternative of an Application being able to pass a HI tuple to a UAC (or UAS) that can forward (or backward) propogate. This interface and function can be defined in the limited context of this rfc. Without this call control b2buas will implement rfc4244 in=20 an undefined way leading to inconsistent infomration models and the purpose of the rfc will be defeated. thanks, Ganesh -----Original Message----- From: Dean Willis [mailto:dean.willis@softarmor.com]=20 Sent: Friday, August 11, 2006 2:19 PM To: Ganesh Jayadevan (gjayadev) Cc: sipping@ietf.org; Cullen Jennings (fluffy); Sudipto Mukherjee (sudiptom); Jon.Peterson@NeuStar.biz; mark@digitalfountain.com; mary.barnes@nortel.com; Mohammad Al-Taraireh (maltarai) Subject: Re: [Sipping] b2bua/call control aspect for rfc4244 (History-Info header) On Aug 11, 2006, at 6:11 AM, Ganesh Jayadevan ((gjayadev)) wrote: > Folks, > > rfc4244 (History-Info) addresses what the UAC, UAS, and the proxy must > do with the History-Info header. > True to the tradition of SIP, a b2bua is not given any special status=20 > but is seen as a concat of a UAS-UAC. > > I am wondering if this leaves a gap and hinders achieving the=20 > objectives of the rfc. > The problem is that defining what a B2BUA does, in general, seems to be an intractable problem. You have one specific sub-model of B2BUA that is fairly proxy-like in its behavior in mind. However, another model of B2BUA serves as an anonymizer. It's entire role in life is to keep one side from knowing anything about the "real UA" on the other side of the B2BUA. This probably includes disclosing that there's even a B2BUA in the loop -- the "other side" =20 should see the anonymizing B2BUA as the terminal node. So by this logic, we should specify that B2BUAs do not propagate History-Info at all. Now, clearly that's also wrong for the general case. There may not even be a "general case" for B2BUA behavior. So we end up with a situation where each application of a B2BUA requires an analysis of the appropriate transparency behavior for that specific application. Given that there are arguably a large number (if not infinite set) of B2BUA applications, this means that a thorough description of the behavior of B2BUAs with respect to any specific extension is sufficiently difficult as to be infeasible. Contrast this with the bounded problem space of describing the extensions behavior wrt to UAC, UAS, proxy (possibly stateless, transaction stateful, and dialog stateful) and registrar. Now it may be that there are a finite set of design patterns for B2BUAs that can be cleanly defined, each of which has an obvious set of transparency requirements. I'm certainly not convinced that this set is small or the problem tractable, but if you'd like to try and establish an appropriate taxonomy along with discussion of the transparency behavior of each pattern, I'll be happy to read your results. -- Dean _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Fri Aug 11 14:56:48 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GBcBb-0000o6-St; Fri, 11 Aug 2006 14:56:47 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GBcBa-0000nF-K5 for sipping@ietf.org; Fri, 11 Aug 2006 14:56:46 -0400 Received: from zrtps0kp.nortel.com ([47.140.192.56]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GBcBY-0006XZ-8m for sipping@ietf.org; Fri, 11 Aug 2006 14:56:46 -0400 Received: from zrc2hxm2.corp.nortel.com (zrc2hxm2.corp.nortel.com [47.103.123.73]) by zrtps0kp.nortel.com (Switch-2.2.6/Switch-2.2.0) with ESMTP id k7BIucV00985; Fri, 11 Aug 2006 14:56:38 -0400 (EDT) x-mimeole: Produced By Microsoft Exchange V6.5 Content-class: urn:content-classes:message MIME-Version: 1.0 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: quoted-printable Subject: RE: [Sipping] Clarification on RFC 3959 - early media Date: Fri, 11 Aug 2006 13:56:37 -0500 Message-ID: <6FC4416DDE56C44DA0AEE67BC7CA43710FDDF059@zrc2hxm2.corp.nortel.com> In-Reply-To: <44DCCB93.6090802@cisco.com> X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: [Sipping] Clarification on RFC 3959 - early media Thread-Index: Aca9c4iqU4Jn7U/2RlONfvuWa+HJ/QAA8PXA From: "Brian Stucker" To: "Paul Kyzivat" X-Spam-Score: 0.0 (/) X-Scan-Signature: 386e0819b1192672467565a524848168 Cc: sipping@ietf.org X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org =20 > -----Original Message----- > From: Paul Kyzivat [mailto:pkyzivat@cisco.com]=20 > Sent: Friday, August 11, 2006 1:25 PM > To: Stucker, Brian (RICH1:B620) > Cc: Kapil Nayar; sipping@ietf.org > Subject: Re: [Sipping] Clarification on RFC 3959 - early media >=20 >=20 >=20 > Brian Stucker wrote: > > Pre-routing, pre-presentation, etc. are monikers that I=20 > came up with=20 > > to try to explain at which points early media can arise=20 > during processing. >=20 > It isn't the names that bother me. (It's obvious you coined=20 > those.) It's the behaviors described that are disturbing. >=20 > > The idea here is that all early media is not equal, some=20 > early media=20 > > is more evil than others. > >=20 > > I've seen all of the techniques mentioned in the draft in the wild. >=20 > Time to declare open season! > (Kill those nasty beasts before they hurt the children.) >=20 > (I wouldn't be killing any of my kin would I?) :) I'm not interested in pointing fingers, I just want to get the problem under control. To others who may have experience with early media, and know of a mechanism that they've seen used that is not currently documented in the draft, please email me so I can include it in the discussion. >=20 > > Many of them are undetectable to the originating UA. >=20 > Except that things work funny, or sometimes don't work. True, but you have to keep in mind that many of the things that work funny or don't work with early media are things that happen on behalf of the terminator. How is the originator going to tell that the early media the network is trying to provide from a PSTN gateway or some service for the terminator is missing? >=20 > Paul >=20 > > Regards, > > Brian > >=20 > >> -----Original Message----- > >> From: Paul Kyzivat [mailto:pkyzivat@cisco.com] > >> Sent: Thursday, August 10, 2006 5:29 PM > >> To: Stucker, Brian (RICH1:B620) > >> Cc: Kapil Nayar; sipping@ietf.org > >> Subject: Re: [Sipping] Clarification on RFC 3959 - early media > >> > >> Brian, > >> > >> I just got around to looking at this. I need to think about the=20 > >> techniques you propose at the end. But I am having trouble getting=20 > >> past the first half of the document. I have never heard mention of=20 > >> the things you describe as Pre-routing early media or=20 > >> Pre-presentation early media. > >> Nor have I heard of the Proxy-side coping mechanisms you describe. > >> > >> These are all positively *obscene*. You suggest that these are=20 > >> common. > >> Is that really true??? (Maybe you live in a particularly dangerous > >> neighborhood.) > >> > >> Tell me it isn't true! Please! > >> > >> Paul > >> > >> Brian Stucker wrote: > >>> You may wish to take a look at the following: > >>> > >>> > >>=20 > http://www.ietf.org/internet-drafts/draft-stucker-sipping-early-media > >> - > >>> co > >>> ping-01.txt > >=20 >=20 _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Fri Aug 11 14:57:37 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GBcCP-0002Je-QQ; Fri, 11 Aug 2006 14:57:37 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GBcCN-0002JU-R1 for sipping@ietf.org; Fri, 11 Aug 2006 14:57:35 -0400 Received: from rtp-iport-2.cisco.com ([64.102.122.149]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GBcCM-0006a6-G7 for sipping@ietf.org; Fri, 11 Aug 2006 14:57:35 -0400 Received: from rtp-dkim-2.cisco.com ([64.102.121.159]) by rtp-iport-2.cisco.com with ESMTP; 11 Aug 2006 14:57:34 -0400 X-IronPort-AV: i="4.08,115,1154923200"; d="scan'208"; a="96284342:sNHT1933423756" Received: from rtp-core-2.cisco.com (rtp-core-2.cisco.com [64.102.124.13]) by rtp-dkim-2.cisco.com (8.12.11.20060308/8.12.11) with ESMTP id k7BIvWML009801; Fri, 11 Aug 2006 14:57:32 -0400 Received: from xbh-rtp-211.amer.cisco.com (xbh-rtp-211.cisco.com [64.102.31.102]) by rtp-core-2.cisco.com (8.12.10/8.12.6) with ESMTP id k7BIvWe2015504; Fri, 11 Aug 2006 14:57:32 -0400 (EDT) Received: from xfe-rtp-202.amer.cisco.com ([64.102.31.21]) by xbh-rtp-211.amer.cisco.com with Microsoft SMTPSVC(6.0.3790.1830); Fri, 11 Aug 2006 14:57:31 -0400 Received: from [161.44.79.104] ([161.44.79.104]) by xfe-rtp-202.amer.cisco.com with Microsoft SMTPSVC(6.0.3790.1830); Fri, 11 Aug 2006 14:57:31 -0400 Message-ID: <44DCD31A.1070906@cisco.com> Date: Fri, 11 Aug 2006 14:57:30 -0400 From: Paul Kyzivat User-Agent: Thunderbird 1.5.0.5 (Windows/20060719) MIME-Version: 1.0 To: "Ganesh Jayadevan (gjayadev)" Subject: Re: [Sipping] b2bua/call control aspect for rfc4244 (History-Info header) References: <15B86BC7352F864BB53A47B540C089B6020441C8@xmb-rtp-20b.amer.cisco.com> In-Reply-To: <15B86BC7352F864BB53A47B540C089B6020441C8@xmb-rtp-20b.amer.cisco.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Content-Transfer-Encoding: 7bit X-OriginalArrivalTime: 11 Aug 2006 18:57:31.0435 (UTC) FILETIME=[FFA43FB0:01C6BD77] DKIM-Signature: a=rsa-sha1; q=dns; l=3786; t=1155322652; x=1156186652; c=relaxed/simple; s=rtpdkim2001; h=Content-Type:From:Subject:Content-Transfer-Encoding:MIME-Version; d=cisco.com; i=pkyzivat@cisco.com; z=From:Paul=20Kyzivat=20 |Subject:Re=3A=20[Sipping]=20b2bua/call=20control=20aspect=20for=20rfc4244=20(His tory-Info=0A=20header) |To:=22Ganesh=20Jayadevan=20(gjayadev)=22=20; X=v=3Dcisco.com=3B=20h=3DDkXAXVd5uBATvEKBBNaAn9uAFoo=3D; b=vEhf/lQVz+MNSHl98J7QF90K28Tbtz53GnTE6uBV715h9A2damvdwgOSx/XIfq00VmhYTlc8 Jgp1JVDYUVCYpvzJAVJpCc83uNj065KqbOW2iaeI/fg7eBsai/euDn3V; Authentication-Results: rtp-dkim-2.cisco.com; header.From=pkyzivat@cisco.com; dkim=pass ( sig from cisco.com verified; ); X-Spam-Score: 0.5 (/) X-Scan-Signature: cf3becbbd6d1a45acbe2ffd4ab88bdc2 Cc: "Cullen Jennings \(fluffy\)" , "Sudipto Mukherjee \(sudiptom\)" , Jon.Peterson@NeuStar.biz, mark@digitalfountain.com, "Mohammad Al-Taraireh \(maltarai\)" , mary.barnes@nortel.com, Dean Willis , sipping@ietf.org X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Ganesh Jayadevan (gjayadev) wrote: > Hi Dean, > > I see what you are saying about B2BUAs in general. > > I also proposed another alternative of an Application > being able to pass a HI tuple to a UAC (or UAS) that can forward > (or backward) propogate. This interface and function can be > defined in the limited context of this rfc. Yes, I can see how it would make sense to permit the application to do this. I agree with dean on the general intractability of specifying anything useful in general for a B2BUA. Paul > Without this call control b2buas will implement rfc4244 in > an undefined way leading to inconsistent infomration models > and the purpose of the rfc will be defeated. > > thanks, > Ganesh > > -----Original Message----- > From: Dean Willis [mailto:dean.willis@softarmor.com] > Sent: Friday, August 11, 2006 2:19 PM > To: Ganesh Jayadevan (gjayadev) > Cc: sipping@ietf.org; Cullen Jennings (fluffy); Sudipto Mukherjee > (sudiptom); Jon.Peterson@NeuStar.biz; mark@digitalfountain.com; > mary.barnes@nortel.com; Mohammad Al-Taraireh (maltarai) > Subject: Re: [Sipping] b2bua/call control aspect for rfc4244 > (History-Info header) > > > On Aug 11, 2006, at 6:11 AM, Ganesh Jayadevan ((gjayadev)) wrote: > >> Folks, >> >> rfc4244 (History-Info) addresses what the UAC, UAS, and the proxy must > >> do with the History-Info header. >> True to the tradition of SIP, a b2bua is not given any special status >> but is seen as a concat of a UAS-UAC. >> >> I am wondering if this leaves a gap and hinders achieving the >> objectives of the rfc. >> > > The problem is that defining what a B2BUA does, in general, seems to be > an intractable problem. > > You have one specific sub-model of B2BUA that is fairly proxy-like in > its behavior in mind. > > However, another model of B2BUA serves as an anonymizer. It's entire > role in life is to keep one side from knowing anything about the "real > UA" on the other side of the B2BUA. This probably includes disclosing > that there's even a B2BUA in the loop -- the "other side" > should see the anonymizing B2BUA as the terminal node. > > So by this logic, we should specify that B2BUAs do not propagate > History-Info at all. > > Now, clearly that's also wrong for the general case. There may not even > be a "general case" for B2BUA behavior. > > So we end up with a situation where each application of a B2BUA requires > an analysis of the appropriate transparency behavior for that specific > application. > > Given that there are arguably a large number (if not infinite set) of > B2BUA applications, this means that a thorough description of the > behavior of B2BUAs with respect to any specific extension is > sufficiently difficult as to be infeasible. > > Contrast this with the bounded problem space of describing the > extensions behavior wrt to UAC, UAS, proxy (possibly stateless, > transaction stateful, and dialog stateful) and registrar. > > Now it may be that there are a finite set of design patterns for B2BUAs > that can be cleanly defined, each of which has an obvious set of > transparency requirements. I'm certainly not convinced that this set is > small or the problem tractable, but if you'd like to try and establish > an appropriate taxonomy along with discussion of the transparency > behavior of each pattern, I'll be happy to read your results. > > -- > Dean > > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP > Use sip-implementors@cs.columbia.edu for questions on current sip > Use sip@ietf.org for new developments of core SIP > _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Fri Aug 11 18:59:37 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GBfyU-0004fF-0m; Fri, 11 Aug 2006 18:59:30 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GBfyT-0004ea-2g for sipping@ietf.org; Fri, 11 Aug 2006 18:59:29 -0400 Received: from stsc1260-eth-s1-s1p1-vip.va.neustar.com ([156.154.16.129] helo=chiedprmail1.ietf.org) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GBblX-0003hG-3Y for sipping@ietf.org; Fri, 11 Aug 2006 14:29:51 -0400 Received: from nylon.softarmor.com ([66.135.38.164]) by chiedprmail1.ietf.org with esmtp (Exim 4.43) id 1GBbb8-0002e1-8I for sipping@ietf.org; Fri, 11 Aug 2006 14:19:06 -0400 Received: from [64.101.149.196] (dwillis-mb.cisco.com [64.101.149.196]) (authenticated bits=0) by nylon.softarmor.com (8.13.1/8.13.1) with ESMTP id k7BIL6OG010734 (version=TLSv1/SSLv3 cipher=RC4-SHA bits=128 verify=NO); Fri, 11 Aug 2006 13:21:07 -0500 In-Reply-To: <15B86BC7352F864BB53A47B540C089B602044010@xmb-rtp-20b.amer.cisco.com> References: <15B86BC7352F864BB53A47B540C089B602044010@xmb-rtp-20b.amer.cisco.com> Mime-Version: 1.0 (Apple Message framework v752.2) Content-Type: text/plain; charset=US-ASCII; delsp=yes; format=flowed Message-Id: <1840314F-CC26-4CA7-ADD8-B9B14DE4CF28@softarmor.com> Content-Transfer-Encoding: 7bit From: Dean Willis Subject: Re: [Sipping] b2bua/call control aspect for rfc4244 (History-Info header) Date: Fri, 11 Aug 2006 13:18:49 -0500 To: Ganesh Jayadevan ((gjayadev)) X-Mailer: Apple Mail (2.752.2) X-Spam-Score: -2.6 (--) X-Scan-Signature: 3e15cc4fdc61d7bce84032741d11c8e5 Cc: "Cullen Jennings \(fluffy\)" , "Sudipto Mukherjee \(sudiptom\)" , Jon.Peterson@NeuStar.biz, mark@digitalfountain.com, mary.barnes@nortel.com, "Mohammad Al-Taraireh \(maltarai\)" , sipping@ietf.org X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org On Aug 11, 2006, at 6:11 AM, Ganesh Jayadevan ((gjayadev)) wrote: > Folks, > > rfc4244 (History-Info) addresses what the UAC, UAS, and > the proxy must do with the History-Info header. > True to the tradition of SIP, a b2bua is not given any special > status but is seen as a concat of a UAS-UAC. > > I am wondering if this leaves a gap and hinders achieving the > objectives of the rfc. > The problem is that defining what a B2BUA does, in general, seems to be an intractable problem. You have one specific sub-model of B2BUA that is fairly proxy-like in its behavior in mind. However, another model of B2BUA serves as an anonymizer. It's entire role in life is to keep one side from knowing anything about the "real UA" on the other side of the B2BUA. This probably includes disclosing that there's even a B2BUA in the loop -- the "other side" should see the anonymizing B2BUA as the terminal node. So by this logic, we should specify that B2BUAs do not propagate History-Info at all. Now, clearly that's also wrong for the general case. There may not even be a "general case" for B2BUA behavior. So we end up with a situation where each application of a B2BUA requires an analysis of the appropriate transparency behavior for that specific application. Given that there are arguably a large number (if not infinite set) of B2BUA applications, this means that a thorough description of the behavior of B2BUAs with respect to any specific extension is sufficiently difficult as to be infeasible. Contrast this with the bounded problem space of describing the extensions behavior wrt to UAC, UAS, proxy (possibly stateless, transaction stateful, and dialog stateful) and registrar. Now it may be that there are a finite set of design patterns for B2BUAs that can be cleanly defined, each of which has an obvious set of transparency requirements. I'm certainly not convinced that this set is small or the problem tractable, but if you'd like to try and establish an appropriate taxonomy along with discussion of the transparency behavior of each pattern, I'll be happy to read your results. -- Dean _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Mon Aug 14 05:21:53 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GCYdf-0004QV-Ic; Mon, 14 Aug 2006 05:21:39 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GCYdd-0004QQ-R8 for sipping@ietf.org; Mon, 14 Aug 2006 05:21:37 -0400 Received: from sonussf1.sonusnet.com ([208.45.178.26]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GCYdc-0003m3-Hk for sipping@ietf.org; Mon, 14 Aug 2006 05:21:37 -0400 Received: from sonusmail06.sonusnet.com (sonusmail06.sonusnet.com [10.128.32.156]) by sonussf1.sonusnet.com (8.13.3/8.13.3) with ESMTP id k7E9LaHT029743 for ; Mon, 14 Aug 2006 05:21:36 -0400 Received: from SONUSINMAIL01.sonusnet.com ([10.128.254.7]) by sonusmail06.sonusnet.com with Microsoft SMTPSVC(6.0.3790.1830); Mon, 14 Aug 2006 05:21:36 -0400 X-MimeOLE: Produced By Microsoft Exchange V6.0.6603.0 content-class: urn:content-classes:message MIME-Version: 1.0 Date: Mon, 14 Aug 2006 14:51:32 +0530 Message-ID: <70798CF421F00F4DA018059E5B7EEB8C7A5C26@sonusinmail01.sonusnet.com> X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: bug in rfc-3261 Thread-Index: Aca/gwfiixc33VgoReSnMHo4UQN/Ig== From: "B, Nataraju" To: X-OriginalArrivalTime: 14 Aug 2006 09:21:36.0259 (UTC) FILETIME=[0A641530:01C6BF83] X-Spam-Score: 0.0 (/) X-Scan-Signature: f49c97ce49302a02285a2d36a99eef8c Subject: [Sipping] bug in rfc-3261 X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Content-Type: multipart/mixed; boundary="===============1040513359==" Errors-To: sipping-bounces@ietf.org This is a multi-part message in MIME format. --===============1040513359== content-class: urn:content-classes:message Content-Type: multipart/alternative; boundary="----_=_NextPart_001_01C6BF83.08172448" This is a multi-part message in MIME format. ------_=_NextPart_001_01C6BF83.08172448 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: quoted-printable Header field where proxy ACK BYE CAN INV OPT REG WWW-Authenticate 401 ar - m - m m m WWW-Authenticate 407 ar - o - o o o Contact 3xx d - o - o o o Why proxy column is mentioned as "ar" for WWW-Authenticate in 401/407. It should have been mentioned as "r" only. In no case proxy would add the WWW-Authenticate header for responses 401/407 ?. Does the proxy aggregation considered for this case ? Why proxy is allowed to "delete only" for contact in 3xx response? Even it should be allowed to read the contacts in 3xx replies. BTW, in which scenario it can delete a contact in 3xx response. This option should have been either "dr" or "r" only depending on answer to my previous question.=20 Thanks, Nataraju A B Sonus Networks Ind Pvt Ltd ------_=_NextPart_001_01C6BF83.08172448 Content-Type: text/html; charset="us-ascii" Content-Transfer-Encoding: quoted-printable bug in rfc-3261

Header = field           &n= bsp;  where       proxy ACK BYE CAN = INV OPT REG

WWW-Authenticate        &nbs= p;  401         = ar    -   m   -   = m   m   m

WWW-Authenticate        &nbs= p;  407         = ar    -   o   -   = o   o   o

Contact          &= nbsp;          3xx         d    -   = o   -   o   o   = o

Why proxy column is mentioned as “ar” = for WWW-Authenticate in = 401/407. It should have been mentioned as “r” only. In no = case proxy would add the WWW-Authenticate header for responses 401/407 = ?. Does the proxy aggregation considered for this = case ?

Why proxy is allowed to “delete only” = for contact in 3xx response? Even it should be allowed to read the = contacts in 3xx replies. BTW, in which scenario it can delete a contact in 3xx response. = This option should have = been = either “dr” or “r” only depending on answer to = my previous question.

Thanks,

Nataraju A B

Sonus Networks Ind Pvt Ltd

------_=_NextPart_001_01C6BF83.08172448-- --===============1040513359== Content-Type: text/plain; charset="us-ascii" MIME-Version: 1.0 Content-Transfer-Encoding: 7bit Content-Disposition: inline _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP --===============1040513359==-- From sipping-bounces@ietf.org Mon Aug 14 05:47:55 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GCZ34-0006FW-4A; Mon, 14 Aug 2006 05:47:54 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GCZ32-0006Ad-Da for sipping@ietf.org; Mon, 14 Aug 2006 05:47:52 -0400 Received: from szxga02-in.huawei.com ([61.144.161.54]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GCZ2z-00061M-IF for sipping@ietf.org; Mon, 14 Aug 2006 05:47:52 -0400 Received: from huawei.com (szxga02-in [172.24.2.6]) by szxga02-in.huawei.com (iPlanet Messaging Server 5.2 HotFix 1.25 (built Mar 3 2004)) with ESMTP id <0J3Z006R6FZOE9@szxga02-in.huawei.com> for sipping@ietf.org; Mon, 14 Aug 2006 18:04:37 +0800 (CST) Received: from huawei.com ([172.24.1.24]) by szxga02-in.huawei.com (iPlanet Messaging Server 5.2 HotFix 1.25 (built Mar 3 2004)) with ESMTP id <0J3Z00KUOFZOS6@szxga02-in.huawei.com> for sipping@ietf.org; Mon, 14 Aug 2006 18:04:36 +0800 (CST) Received: from k19120I ([10.70.145.109]) by szxml04-in.huawei.com (iPlanet Messaging Server 5.2 HotFix 1.25 (built Mar 3 2004)) with ESMTPA id <0J3Z00DTJFQDSY@szxml04-in.huawei.com> for sipping@ietf.org; Mon, 14 Aug 2006 17:59:01 +0800 (CST) Date: Mon, 14 Aug 2006 17:47:02 +0800 From: Manjunath Warad Subject: RE: [Sipping] bug in rfc-3261 In-reply-to: <70798CF421F00F4DA018059E5B7EEB8C7A5C26@sonusinmail01.sonusnet.com> To: "'B, Nataraju'" , sipping@ietf.org Message-id: <000b01c6bf86$9825e120$6d91460a@china.huawei.com> MIME-version: 1.0 X-MIMEOLE: Produced By Microsoft MimeOLE V6.00.2800.1106 X-Mailer: Microsoft Outlook, Build 10.0.3416 Importance: Normal X-Priority: 3 (Normal) X-MSMail-priority: Normal X-Spam-Score: 0.0 (/) X-Scan-Signature: 156eddb66af16eef49a76ae923b15b92 Cc: X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Content-Type: multipart/mixed; boundary="===============1317105639==" Errors-To: sipping-bounces@ietf.org This is a multi-part message in MIME format. --===============1317105639== Content-type: multipart/alternative; boundary="Boundary_(ID_Kh+9gaVQDmUDIkVYmKUoOA)" This is a multi-part message in MIME format. --Boundary_(ID_Kh+9gaVQDmUDIkVYmKUoOA) Content-type: text/plain; charset=us-ascii Content-transfer-encoding: 7BIT Hi, Please see inline... Rgds, Manju -----Original Message----- From: B, Nataraju [mailto:bnataraju@sonusnet.com] Sent: Monday, August 14, 2006 5:22 PM To: sipping@ietf.org Subject: [Sipping] bug in rfc-3261 Header field where proxy ACK BYE CAN INV OPT REG WWW-Authenticate 401 ar - m - m m m WWW-Authenticate 407 ar - o - o o o Contact 3xx d - o - o o o Why proxy column is mentioned as "ar" for WWW-Authenticate in 401/407. It should have been mentioned as "r" only. In no case proxy would add the WWW-Authenticate header for responses 401/407 ?. Does the proxy aggregation considered for this case ? Yes you are right! Proxy aggregation is considered here. Why proxy is allowed to "delete only" for contact in 3xx response? Even it should be allowed to read the contacts in 3xx replies. BTW, in which scenario it can delete a contact in 3xx response. This option should have been either "dr" or "r" only depending on answer to my previous question. Here again, proxy can remove the contact, for which it has already tried, so that upstream proxies need not try on those contacts again. This technique is used when proxy chose 3xx as best response to be forwarded for upstream proxies. So, I think "adr" need to be used here as this header can be aggregated to form the final 3xx response if more than 1 is present. Please let me know, if my interpretation is wrong. Thanks, Nataraju A B Sonus Networks Ind Pvt Ltd --Boundary_(ID_Kh+9gaVQDmUDIkVYmKUoOA) Content-type: text/html; charset=us-ascii Content-transfer-encoding: 7BIT Message
Hi,
    Please see inline...
 
Rgds,
Manju
-----Original Message-----
From: B, Nataraju [mailto:bnataraju@sonusnet.com]
Sent: Monday, August 14, 2006 5:22 PM
To: sipping@ietf.org
Subject: [Sipping] bug in rfc-3261

Header field              where       proxy ACK BYE CAN INV OPT REG

WWW-Authenticate           401         ar    -   m   -   m   m   m

WWW-Authenticate           407         ar    -   o   -   o   o   o

Contact                     3xx         d    -   o   -   o   o   o

Why proxy column is mentioned as “ar” for WWW-Authenticate in 401/407. It should have been mentioned as “r” only. In no case proxy would add the WWW-Authenticate header for responses 401/407 ?. Does the proxy aggregation considered for this case ? 

Yes you are right! Proxy aggregation is considered here. 

Why proxy is allowed to “delete only” for contact in 3xx response? Even it should be allowed to read the contacts in 3xx replies. BTW, in which scenario it can delete a contact in 3xx response. This option should have been either “dr” or “r” only depending on answer to my previous question.  

Here again, proxy can remove the contact, for which it has already tried, so that upstream proxies need not try on those contacts again. This technique is used when proxy chose 3xx as best response to be forwarded for upstream proxies. So, I think "adr" need to be used here as this header can be aggregated to form the final 3xx response if more than 1 is present.

Please let me know, if my interpretation is wrong.

Thanks,

Nataraju A B

Sonus Networks Ind Pvt Ltd

--Boundary_(ID_Kh+9gaVQDmUDIkVYmKUoOA)-- --===============1317105639== Content-Type: text/plain; charset="us-ascii" MIME-Version: 1.0 Content-Transfer-Encoding: 7bit Content-Disposition: inline _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP --===============1317105639==-- From sipping-bounces@ietf.org Tue Aug 15 08:43:24 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GCyGB-0007me-E7; Tue, 15 Aug 2006 08:43:07 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GCyG9-0007m1-Im for Sipping@ietf.org; Tue, 15 Aug 2006 08:43:05 -0400 Received: from web8704.mail.in.yahoo.com ([203.84.221.125]) by ietf-mx.ietf.org with smtp (Exim 4.43) id 1GCyG7-0007XS-Pn for Sipping@ietf.org; Tue, 15 Aug 2006 08:43:05 -0400 Received: (qmail 63704 invoked by uid 60001); 15 Aug 2006 12:43:02 -0000 DomainKey-Signature: a=rsa-sha1; q=dns; c=nofws; s=s1024; d=yahoo.co.in; h=Message-ID:Received:Date:From:Subject:To:MIME-Version:Content-Type:Content-Transfer-Encoding; b=2CxlJs8/5Eb8AdVWhKSPGOjfK1B+pEhavVlGoPt9EhwMdlfzC8uuGqFKJAzXbNKvDy7ab9qpHmuWUTX1/6qNSrk5VKG72+lMXZ4qQ10jJ2yVMVDAslvh7g5qEMEC0+oHtTt3q8kR/jZg7Ow3pPfOpyHo6OutD66ju4fRCAxHvzE= ; Message-ID: <20060815124301.63702.qmail@web8704.mail.in.yahoo.com> Received: from [220.224.36.49] by web8704.mail.in.yahoo.com via HTTP; Tue, 15 Aug 2006 13:43:01 BST Date: Tue, 15 Aug 2006 13:43:01 +0100 (BST) From: AMIT KUNDRA To: Sipping@ietf.org MIME-Version: 1.0 X-Spam-Score: 0.6 (/) X-Scan-Signature: 69a74e02bbee44ab4f8eafdbcedd94a1 Cc: Subject: [Sipping] Example of interworking between a Wireline Softswitch & a CDMA Softswitch X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Content-Type: multipart/mixed; boundary="===============0003355693==" Errors-To: sipping-bounces@ietf.org --===============0003355693== Content-Type: multipart/alternative; boundary="0-341132688-1155645781=:62485" Content-Transfer-Encoding: 8bit --0-341132688-1155645781=:62485 Content-Type: text/plain; charset=iso-8859-1 Content-Transfer-Encoding: 8bit Hi , I have one doubt. For one of our clients , having CDMA Mobile services & Wireline services If we are to have interworking between a Wireline Softswitch & a CDMA Softswitch then do we use SIP or BICC ? as many people think that BICC is more clearly defines than SIP . From this I have 3 more queries: 1. Is SIP & BICC the same kind of protocol with just the difference that BICC is part of SS& while SIP is part of VoIP ? 2. If different then what is this talk about integarting BICC into SIP ? 3. Is there any SIP variant, called SIP-I as I only know the SIP & SIP-T ? 4. Pls clearly distinguish between SIP,SIP-T & SIP-I with practical examples as to where which to be used ? Pls respond urgently --------------------------------- Here's a new way to find what you're looking for - Yahoo! Answers Send FREE SMS to your friend's mobile from Yahoo! Messenger Version 8. Get it NOW --0-341132688-1155645781=:62485 Content-Type: text/html; charset=iso-8859-1 Content-Transfer-Encoding: 8bit
Hi ,
 
I have one doubt. For one of our clients , having CDMA Mobile services & Wireline services If we are to have interworking between a Wireline Softswitch & a CDMA Softswitch then do we use SIP or BICC ? as many people think that BICC is more clearly defines than SIP . From this I have 3 more queries:
 
1. Is SIP & BICC the same kind of protocol with just the difference that BICC is part of SS& while SIP is part of VoIP ?
 
2. If different then what is this talk about integarting BICC into SIP ?
 
3. Is there any SIP variant, called SIP-I as I only know the SIP & SIP-T ?
 
4. Pls clearly distinguish between SIP,SIP-T & SIP-I with practical examples as to where which to be used ?
 
Pls respond urgently
 
 
 


Here's a new way to find what you're looking for - Yahoo! Answers
Send FREE SMS to your friend's mobile from Yahoo! Messenger Version 8. Get it NOW --0-341132688-1155645781=:62485-- --===============0003355693== Content-Type: text/plain; charset="us-ascii" MIME-Version: 1.0 Content-Transfer-Encoding: 7bit Content-Disposition: inline _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP --===============0003355693==-- From sipping-bounces@ietf.org Tue Aug 15 09:00:58 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GCyXR-0004QM-Ev; Tue, 15 Aug 2006 09:00:57 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GCyXP-0004Oo-DE for sipping@ietf.org; Tue, 15 Aug 2006 09:00:55 -0400 Received: from fw.polycom.co.il ([212.179.41.2] helo=isrexch01.israel.polycom.com) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GCyRq-0000Mj-MC for sipping@ietf.org; Tue, 15 Aug 2006 08:55:12 -0400 X-MimeOLE: Produced By Microsoft Exchange V6.5 Content-class: urn:content-classes:message MIME-Version: 1.0 Content-Type: text/plain; charset="iso-8859-1" Content-Transfer-Encoding: quoted-printable Date: Tue, 15 Aug 2006 15:55:07 +0300 Message-ID: <144ED8561CE90C41A3E5908EDECE315C03CB5A9C@IsrExch01.israel.polycom.com> X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: Review of draft-ietf-sipping-session-policy-framework-01 Thread-Index: AcbAagjl7qVpGySTTsWf2FbXn+ouzw== From: "Even, Roni" To: X-Spam-Score: 1.2 (+) X-Scan-Signature: 9ed51c9d1356100bce94f1ae4ec616a9 Cc: volkerh@bell-labs.com, Mary Barnes , Gonzalo.Camarillo@ericsson.com Subject: [Sipping] Review of draft-ietf-sipping-session-policy-framework-01 X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Draft: draft-ietf-sipping-session-policy-framework-01 Reviewer: Roni Even Review Date: 15/8/06 Review Deadline: 25/8/06 Status: Initial review=20 Summary: This draft is basically ready for publication, but has nits = that should be fixed before publication. The document looks good and being asked a lot about bandwidth management = in SIP I find it very important. I have a question: In the session specific policy call flows the UA that send the invite = insert a header field (Policy-Id) indicating to the proxy that it = contacted the Policy server. How does the proxy on the other side know = that the UA contacted the policy server when it gets the 200 with the = answer. A nit Introduction section - says "In contrast to other methods for inserting = a media intermediary, the use of session policies does not require the = inspection or modification of SIP message bodies". Looking at section = 4.3.1 for example shows that at least the two proxies PA and PB will = change the SIP message. I also assume that policy enforcement will = require inspection of the messages. Roni Even ******************************************************************* Roni Even Polycom NSD =A0 Tel: +972-3-9251200 Mobile: +972-54-5481099 Fax: +972-3-9211571 email: roni.even@polycom.co.il _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Tue Aug 15 09:09:52 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GCyg3-0007bD-2c; Tue, 15 Aug 2006 09:09:51 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GCyg1-0007b6-GJ for sipping@ietf.org; Tue, 15 Aug 2006 09:09:49 -0400 Received: from fw.polycom.co.il ([212.179.41.2] helo=isrexch01.israel.polycom.com) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GCyg0-0001UG-3I for sipping@ietf.org; Tue, 15 Aug 2006 09:09:49 -0400 X-MimeOLE: Produced By Microsoft Exchange V6.5 Content-class: urn:content-classes:message MIME-Version: 1.0 Content-Type: text/plain; charset="iso-8859-1" Content-Transfer-Encoding: quoted-printable Date: Tue, 15 Aug 2006 16:09:45 +0300 Message-ID: <144ED8561CE90C41A3E5908EDECE315C03CB5AA2@IsrExch01.israel.polycom.com> X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: Review of draft-ietf-sipping-policy-package-01 Thread-Index: AcbAagjl7qVpGySTTsWf2FbXn+ouzw== From: "Even, Roni" To: X-Spam-Score: 0.1 (/) X-Scan-Signature: 9ed51c9d1356100bce94f1ae4ec616a9 Cc: volkerh@bell-labs.com, Mary Barnes , Gonzalo.Camarillo@ericsson.com Subject: [Sipping] Review of draft-ietf-sipping-policy-package-01 X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Draft: draft-ietf-sipping-policy-package-01 Reviewer: Roni Even Review Date: 15/8/06 Review Deadline: 25/8/06 Status: Initial review=20 Summary: This draft is on the right track but has open issues, described = in the review. In section 3.3 on the open issue, I think that XML is good direction, = using it will enable to define media policy not in terms of SDP = architecture. Still there is a need to be able to map from SDP to XML. = The mapping should allow specifying per session and per media policy. In section 3.6 on the open issue, I think that there should be a basic = set at least the media and bandwidth. The subscriber may update this = information, for example, if the peer sends a re-invite. This can be = very useful for bandwidth management and even provide a tool for = resource scheduling application that will work also with the non session = specific policy.=20 In section 3.9 why SHOULD is used and not SHALL, if a session policy is = in place and a UA subscribed to the policy server then it must work = according to the policy. Roni Even ******************************************************************* Roni Even Polycom NSD =A0 Tel: +972-3-9251200 Mobile: +972-54-5481099 Fax: +972-3-9211571 email: roni.even@polycom.co.il _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Tue Aug 15 09:12:37 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GCyij-000053-Jw; Tue, 15 Aug 2006 09:12:37 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GCyih-00004y-VM for sipping@ietf.org; Tue, 15 Aug 2006 09:12:35 -0400 Received: from fw.polycom.co.il ([212.179.41.2] helo=isrexch01.israel.polycom.com) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GCyig-0001iM-Kt for sipping@ietf.org; Tue, 15 Aug 2006 09:12:35 -0400 X-MimeOLE: Produced By Microsoft Exchange V6.5 Content-class: urn:content-classes:message MIME-Version: 1.0 Content-Type: text/plain; charset="iso-8859-1" Content-Transfer-Encoding: quoted-printable Date: Tue, 15 Aug 2006 16:12:32 +0300 Message-ID: <144ED8561CE90C41A3E5908EDECE315C03CB5AA3@IsrExch01.israel.polycom.com> X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: Review of draft-ietf-sipping-media-policy-dataset-01 Thread-Index: AcbAagjl7qVpGySTTsWf2FbXn+ouzw== From: "Even, Roni" To: X-Spam-Score: 0.1 (/) X-Scan-Signature: 0bc60ec82efc80c84b8d02f4b0e4de22 Cc: volkerh@bell-labs.com, Mary Barnes , Gonzalo.Camarillo@ericsson.com Subject: [Sipping] Review of draft-ietf-sipping-media-policy-dataset-01 X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Draft: draft-ietf-sipping-media-policy-dataset-01 Reviewer: Roni Even Review Date: 15/8/06 Review Deadline: 25/8/06 Status: Initial review=20 Summary: This draft is basically ready for publication, but has nits = that should be fixed before publication. One nit The draft has media-type both as attribute and element maybe rename one = of them for clarity. Roni Even ******************************************************************* Roni Even Polycom NSD =A0 Tel: +972-3-9251200 Mobile: +972-54-5481099 Fax: +972-3-9211571 email: roni.even@polycom.co.il _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Tue Aug 15 11:49:30 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GD18z-0002Ki-7H; Tue, 15 Aug 2006 11:47:53 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GD18x-0002KX-W2 for Sipping@ietf.org; Tue, 15 Aug 2006 11:47:52 -0400 Received: from rtp-iport-2.cisco.com ([64.102.122.149]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GD18w-0003wa-D9 for Sipping@ietf.org; Tue, 15 Aug 2006 11:47:51 -0400 Received: from rtp-dkim-1.cisco.com ([64.102.121.158]) by rtp-iport-2.cisco.com with ESMTP; 15 Aug 2006 11:47:49 -0400 X-IronPort-AV: i="4.08,128,1154923200"; d="scan'208,217"; a="96919498:sNHT3717105090" Received: from rtp-core-2.cisco.com (rtp-core-2.cisco.com [64.102.124.13]) by rtp-dkim-1.cisco.com (8.12.11.20060308/8.12.11) with ESMTP id k7FFlmZM021960; Tue, 15 Aug 2006 11:47:48 -0400 Received: from xbh-rtp-201.amer.cisco.com (xbh-rtp-201.cisco.com [64.102.31.12]) by rtp-core-2.cisco.com (8.12.10/8.12.6) with ESMTP id k7FFlmKH023767; Tue, 15 Aug 2006 11:47:48 -0400 (EDT) Received: from xmb-rtp-20b.amer.cisco.com ([64.102.31.53]) by xbh-rtp-201.amer.cisco.com with Microsoft SMTPSVC(6.0.3790.1830); Tue, 15 Aug 2006 11:47:48 -0400 X-MimeOLE: Produced By Microsoft Exchange V6.5 Content-class: urn:content-classes:message MIME-Version: 1.0 Subject: RE: [Sipping] Example of interworking between a Wireline Softswitch & aCDMA Softswitch Date: Tue, 15 Aug 2006 11:47:47 -0400 Message-ID: <072C5B76F7CEAB488172C6F64B30B5E301CE9D55@xmb-rtp-20b.amer.cisco.com> X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: [Sipping] Example of interworking between a Wireline Softswitch & aCDMA Softswitch Thread-Index: AcbAaHsTb5aV4NkTRRut9pTTGgWsuwAGXz9g From: "Michael Hammer \(mhammer\)" To: "AMIT KUNDRA" , X-OriginalArrivalTime: 15 Aug 2006 15:47:48.0914 (UTC) FILETIME=[28C86120:01C6C082] DKIM-Signature: a=rsa-sha1; q=dns; l=4921; t=1155656868; x=1156520868; c=relaxed/simple; s=rtpdkim1001; h=Content-Type:From:Subject:Content-Transfer-Encoding:MIME-Version; d=cisco.com; i=mhammer@cisco.com; z=From:=22Michael=20Hammer=20\(mhammer\)=22=20 |Subject:RE=3A=20[Sipping]=20Example=20of=20interworking=20between=20a=20Wireline =20Softswitch=20&=20aCDMA=20Softswitch |To:=22AMIT=20KUNDRA=22=20,=20; X=v=3Dcisco.com=3B=20h=3D4gHgzvaFm7DVB/y3RgURGjGZqoI=3D; b=joOBXbBENU2oxP+huKBs8drDNAT3gYve3vRrGfpSUh1LkY4d5aOTrcD7PpZwZ14CnYaZkBfF fHPq69rfvmY1/v3wko1nueHHduCA/bIOIYurB0S/fTwK1vt46RUTJ8JF; Authentication-Results: rtp-dkim-1.cisco.com; header.From=mhammer@cisco.com; dkim=pass ( 59 extraneous bytes; sig from cisco.com verified; ); X-Spam-Score: 0.1 (/) X-Scan-Signature: c54bc2f42d02429833c0ca4b8725abd7 Cc: X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Content-Type: multipart/mixed; boundary="===============0501206506==" Errors-To: sipping-bounces@ietf.org This is a multi-part message in MIME format. --===============0501206506== Content-class: urn:content-classes:message Content-Type: multipart/alternative; boundary="----_=_NextPart_001_01C6C082.282A356A" This is a multi-part message in MIME format. ------_=_NextPart_001_01C6C082.282A356A Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: quoted-printable Read Q.1912.5 =20 If you want to be more forward thinking, focus on SIP. =20 Mike =20 ________________________________ From: AMIT KUNDRA [mailto:akundra_1609@yahoo.co.in]=20 Sent: Tuesday, August 15, 2006 8:43 AM To: Sipping@ietf.org Subject: [Sipping] Example of interworking between a Wireline Softswitch & aCDMA Softswitch =09 =09 Hi , =20 I have one doubt. For one of our clients , having CDMA Mobile services & Wireline services If we are to have interworking between a Wireline Softswitch & a CDMA Softswitch then do we use SIP or BICC ? as many people think that BICC is more clearly defines than SIP . From this I have 3 more queries: =20 1. Is SIP & BICC the same kind of protocol with just the difference that BICC is part of SS& while SIP is part of VoIP ? =20 2. If different then what is this talk about integarting BICC into SIP ? =20 3. Is there any SIP variant, called SIP-I as I only know the SIP & SIP-T ? =20 4. Pls clearly distinguish between SIP,SIP-T & SIP-I with practical examples as to where which to be used ? =20 Pls respond urgently=20 =20 =20 =20 =09 ________________________________ Here's a new way to find what you're looking for - Yahoo! Answers Send FREE SMS to your friend's mobile from Yahoo! Messenger Version 8. Get it NOW ------_=_NextPart_001_01C6C082.282A356A Content-Type: text/html; charset="us-ascii" Content-Transfer-Encoding: quoted-printable
Read Q.1912.5
 
If you want to be more forward thinking, focus on=20 SIP.
 
Mike
 


From: AMIT KUNDRA=20 [mailto:akundra_1609@yahoo.co.in]
Sent: Tuesday, August 15, = 2006=20 8:43 AM
To: Sipping@ietf.org
Subject: [Sipping] = Example of=20 interworking between a Wireline Softswitch & aCDMA=20 Softswitch

Hi ,
 
I have one doubt. For one of our clients , having CDMA Mobile = services=20 & Wireline services If we are to have interworking between a = Wireline=20 Softswitch & a CDMA Softswitch then do we use SIP or BICC ? as = many people=20 think that BICC is more clearly defines than SIP . From this I = have 3=20 more queries:
 
1. Is SIP & BICC the same kind of protocol with just the = difference=20 that BICC is part of SS& while SIP is part of VoIP ?
 
2. If different then what is this talk about integarting BICC = into SIP=20 ?
 
3. Is there any SIP variant, called SIP-I as I only know the SIP = &=20 SIP-T ?
 
4. Pls clearly distinguish between SIP,SIP-T & SIP-I with = practical=20 examples as to where which to be used ?
 
Pls respond urgently
 
 
 


Here's a new way to find what you're looking for - Yahoo!=20 Answers
Send FREE SMS to your friend's mobile from Yahoo! = Messenger=20 Version 8. Get=20 it NOW ------_=_NextPart_001_01C6C082.282A356A-- --===============0501206506== Content-Type: text/plain; charset="us-ascii" MIME-Version: 1.0 Content-Transfer-Encoding: 7bit Content-Disposition: inline _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP --===============0501206506==-- From sipping-bounces@ietf.org Tue Aug 15 13:37:48 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GD2rE-0002P1-3r; Tue, 15 Aug 2006 13:37:40 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GD2rC-0002Ov-M4 for Sipping@ietf.org; Tue, 15 Aug 2006 13:37:38 -0400 Received: from mail2.spherecom.com ([66.237.126.105]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GD2rB-00023x-6J for Sipping@ietf.org; Tue, 15 Aug 2006 13:37:38 -0400 Content-class: urn:content-classes:message MIME-Version: 1.0 X-MimeOLE: Produced By Microsoft Exchange V6.5 Date: Tue, 15 Aug 2006 12:34:54 -0500 Message-ID: <5E6F57726155C84481FA244AE5C6DBA45C5EBE@stratosphere.spherecom.com> X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: Questions on draft-anil-sipping-bla-03.txt Thread-Index: AcaprrYe/LcUQwAGS/WiryiJlYj7RQW4RGmw From: "Agrawal, Vishal" To: X-Spam-Score: 0.2 (/) X-Scan-Signature: 3f3e54d3c03ed638c06aa9fa6861237e Cc: Subject: [Sipping] Questions on draft-anil-sipping-bla-03.txt X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Content-Type: multipart/mixed; boundary="===============0487679238==" Errors-To: sipping-bounces@ietf.org This is a multi-part message in MIME format. --===============0487679238== Content-class: urn:content-classes:message Content-Type: multipart/alternative; boundary="----_=_NextPart_001_01C6C091.7F341E0D" This is a multi-part message in MIME format. ------_=_NextPart_001_01C6C091.7F341E0D Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: quoted-printable In section 6.3 "Call offered to a BLA Group", it's possible that more than one user agents answer the incoming call simultaneously. What should be the responses from the Proxy? Should the proxy send an ACK followed by a BYE request? How would the agent receiving BYE know that a race condition just occurred at the Proxy? I think that the section 8 should cover this race condition.=20 =20 -Vishal ------_=_NextPart_001_01C6C091.7F341E0D Content-Type: text/html; charset="us-ascii" Content-Transfer-Encoding: quoted-printable

In section 6.3 “Call offered to a BLA = Group”, it’s possible that more than one user agents answer the incoming = call simultaneously. What should be the responses from the Proxy? Should the proxy send an = ACK followed by a BYE request? How would the agent receiving BYE know that a = race condition just occurred at the Proxy? I think that the section 8 should = cover this race condition.

 

-Vishal

------_=_NextPart_001_01C6C091.7F341E0D-- --===============0487679238== Content-Type: text/plain; charset="us-ascii" MIME-Version: 1.0 Content-Transfer-Encoding: 7bit Content-Disposition: inline _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP --===============0487679238==-- From sipping-bounces@ietf.org Tue Aug 15 13:47:30 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GD30a-0006ep-2E; Tue, 15 Aug 2006 13:47:20 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GD30Y-0006ek-El for Sipping@ietf.org; Tue, 15 Aug 2006 13:47:18 -0400 Received: from mail2.spherecom.com ([66.237.126.105]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GD30X-0002Yv-09 for Sipping@ietf.org; Tue, 15 Aug 2006 13:47:18 -0400 Content-class: urn:content-classes:message MIME-Version: 1.0 X-MimeOLE: Produced By Microsoft Exchange V6.5 Date: Tue, 15 Aug 2006 12:44:35 -0500 Message-ID: <5E6F57726155C84481FA244AE5C6DBA45C5EC0@stratosphere.spherecom.com> X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: draft-sawada-sipping-sip-offeranswer-01.txt Thread-Index: AcbAknjdtYV8KD40T9mccih+pIn9vg== From: "Agrawal, Vishal" To: X-Spam-Score: 0.2 (/) X-Scan-Signature: 7698d1420ecbbce1995432e99bb6d1a1 Cc: Subject: [Sipping] draft-sawada-sipping-sip-offeranswer-01.txt X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Content-Type: multipart/mixed; boundary="===============2128258383==" Errors-To: sipping-bounces@ietf.org This is a multi-part message in MIME format. --===============2128258383== Content-class: urn:content-classes:message Content-Type: multipart/alternative; boundary="----_=_NextPart_001_01C6C092.D900E8AE" This is a multi-part message in MIME format. ------_=_NextPart_001_01C6C092.D900E8AE Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: quoted-printable In section 3.2 must both UA - A and B reject the offer 1 and offer 2 with 491 responses? The draft says that offer 2 must be rejected, how does the UA B know not to reject offer 1? =20 Vishal ------_=_NextPart_001_01C6C092.D900E8AE Content-Type: text/html; charset="us-ascii" Content-Transfer-Encoding: quoted-printable

In section 3.2 must both UA - A and B reject the = offer 1 and offer 2 with 491 responses? The draft says that offer 2 must be = rejected, how does the UA B know not to reject offer 1?

 

Vishal

------_=_NextPart_001_01C6C092.D900E8AE-- --===============2128258383== Content-Type: text/plain; charset="us-ascii" MIME-Version: 1.0 Content-Transfer-Encoding: 7bit Content-Disposition: inline _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP --===============2128258383==-- From sipping-bounces@ietf.org Tue Aug 15 16:10:08 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GD5Dz-0004uV-2P; Tue, 15 Aug 2006 16:09:19 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GD5Dx-0004uQ-9j for Sipping@ietf.org; Tue, 15 Aug 2006 16:09:17 -0400 Received: from rtp-iport-2.cisco.com ([64.102.122.149]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GD5Dv-0002XK-07 for Sipping@ietf.org; Tue, 15 Aug 2006 16:09:17 -0400 Received: from rtp-dkim-2.cisco.com ([64.102.121.159]) by rtp-iport-2.cisco.com with ESMTP; 15 Aug 2006 16:09:12 -0400 X-IronPort-AV: i="4.08,128,1154923200"; d="scan'208"; a="96965296:sNHT6816722988" Received: from rtp-core-1.cisco.com (rtp-core-1.cisco.com [64.102.124.12]) by rtp-dkim-2.cisco.com (8.12.11.20060308/8.12.11) with ESMTP id k7FK9C5c019302; Tue, 15 Aug 2006 16:09:12 -0400 Received: from xbh-rtp-211.amer.cisco.com (xbh-rtp-211.cisco.com [64.102.31.102]) by rtp-core-1.cisco.com (8.12.10/8.12.6) with ESMTP id k7FK9Ch5006416; Tue, 15 Aug 2006 16:09:12 -0400 (EDT) Received: from xfe-rtp-201.amer.cisco.com ([64.102.31.38]) by xbh-rtp-211.amer.cisco.com with Microsoft SMTPSVC(6.0.3790.1830); Tue, 15 Aug 2006 16:09:11 -0400 Received: from [161.44.79.104] ([161.44.79.104]) by xfe-rtp-201.amer.cisco.com with Microsoft SMTPSVC(6.0.3790.1830); Tue, 15 Aug 2006 16:09:11 -0400 Message-ID: <44E229E7.6030203@cisco.com> Date: Tue, 15 Aug 2006 16:09:11 -0400 From: Paul Kyzivat User-Agent: Thunderbird 1.5.0.5 (Windows/20060719) MIME-Version: 1.0 To: "Agrawal, Vishal" Subject: Re: [Sipping] draft-sawada-sipping-sip-offeranswer-01.txt References: <5E6F57726155C84481FA244AE5C6DBA45C5EC0@stratosphere.spherecom.com> In-Reply-To: <5E6F57726155C84481FA244AE5C6DBA45C5EC0@stratosphere.spherecom.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Content-Transfer-Encoding: 7bit X-OriginalArrivalTime: 15 Aug 2006 20:09:11.0932 (UTC) FILETIME=[AC9703C0:01C6C0A6] DKIM-Signature: a=rsa-sha1; q=dns; l=983; t=1155672552; x=1156536552; c=relaxed/simple; s=rtpdkim2001; h=Content-Type:From:Subject:Content-Transfer-Encoding:MIME-Version; d=cisco.com; i=pkyzivat@cisco.com; z=From:Paul=20Kyzivat=20 |Subject:Re=3A=20[Sipping]=20draft-sawada-sipping-sip-offeranswer-01.txt |To:=22Agrawal,=20Vishal=22=20; X=v=3Dcisco.com=3B=20h=3DsD/eRLp8ibkzyqTiiyJgN4spFkw=3D; b=G9/TFSwVkNukSjUfJOyeEDJB69Sw7joCseBIvRXOMhDCuAg4D/99pj7BRGYbMQ0v9z5845ZJ zg1+lNw+XVz+Sybax0x8Elt92CCfqcmC1Z8CImczIL0WZWTQFgg1KzSD; Authentication-Results: rtp-dkim-2.cisco.com; header.From=pkyzivat@cisco.com; dkim=pass ( sig from cisco.com verified; ); X-Spam-Score: 0.0 (/) X-Scan-Signature: ffa9dfbbe7cc58b3fa6b8ae3e57b0aa3 Cc: Sipping@ietf.org X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Agrawal, Vishal wrote: > In section 3.2 must both UA - A and B reject the offer 1 and offer 2 > with 491 responses? The draft says that offer 2 must be rejected, how > does the UA B know not to reject offer 1? If both offers are carried by UPDATEs then I believe they both fail with 491. Because of the symmetry there is no good way for a winner to be picked. When one is in an UPDATE and the other is in a PRACK the asymmetry allows designating one (the PRACK) as a winner. (Of course there can't be offers in PRACKs going in both directions.) Paul > Vishal > > > ------------------------------------------------------------------------ > > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP > Use sip-implementors@cs.columbia.edu for questions on current sip > Use sip@ietf.org for new developments of core SIP _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Wed Aug 16 18:12:00 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GDTb7-0000GK-IS; Wed, 16 Aug 2006 18:10:49 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GDTb6-0000G9-J5 for Sipping@ietf.org; Wed, 16 Aug 2006 18:10:48 -0400 Received: from ug-out-1314.google.com ([66.249.92.170]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GDTb4-0001Lk-6s for Sipping@ietf.org; Wed, 16 Aug 2006 18:10:48 -0400 Received: by ug-out-1314.google.com with SMTP id m2so336908uge for ; Wed, 16 Aug 2006 15:10:45 -0700 (PDT) DomainKey-Signature: a=rsa-sha1; q=dns; c=nofws; s=beta; d=gmail.com; h=received:message-id:date:from:to:subject:cc:in-reply-to:mime-version:content-type:references; b=f3MSGL/RwuyJaUdbUJuOtHVJlpSZnNhwyYWeb3SmvSfhtDgb1hl1tdSAFoDbQywJTecibLsVR4TUEcHCH2otgJwB8uljtb8Ir2RyvvGjp/IxQ+kR/kw59GzT+u6CdJcJ6ZuFfsPEFwwmw5tORUR+TFctTuI7TIyj2bVmR/Z/TzI= Received: by 10.49.43.2 with SMTP id v2mr1308536nfj; Wed, 16 Aug 2006 15:10:45 -0700 (PDT) Received: by 10.78.139.6 with HTTP; Wed, 16 Aug 2006 15:10:45 -0700 (PDT) Message-ID: <4ff4e7370608161510i772f5303r1a10f151b1e9d06e@mail.gmail.com> Date: Wed, 16 Aug 2006 15:10:45 -0700 From: Venkatesh To: "Agrawal, Vishal" Subject: Re: [Sipping] Questions on draft-anil-sipping-bla-03.txt In-Reply-To: <5E6F57726155C84481FA244AE5C6DBA45C5EBE@stratosphere.spherecom.com> MIME-Version: 1.0 References: <5E6F57726155C84481FA244AE5C6DBA45C5EBE@stratosphere.spherecom.com> X-Spam-Score: 0.1 (/) X-Scan-Signature: 02ec665d00de228c50c93ed6b5e4fc1a Cc: Sipping@ietf.org X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Content-Type: multipart/mixed; boundary="===============0255418268==" Errors-To: sipping-bounces@ietf.org --===============0255418268== Content-Type: multipart/alternative; boundary="----=_Part_48473_16900008.1155766245384" ------=_Part_48473_16900008.1155766245384 Content-Type: text/plain; charset=ISO-8859-1; format=flowed Content-Transfer-Encoding: 7bit Content-Disposition: inline Vishal: Offering calls to a BLA group relies the ability of the proxy to fork the request. The proxy simply forwards all 2xx responses to the calling side in cases where multiple UA's answer the call simultaneously. UA's are expected to handle this condition per section 13.2.2 of RFC 3261. I am not sure there is anything the UA would have to do in addition to what is already mentioned in RFC 3261 for accomodating this condition. Venkatesh On 8/15/06, Agrawal, Vishal wrote: > > In section 6.3 "Call offered to a BLA Group", it's possible that more > than one user agents answer the incoming call simultaneously. What should be > the responses from the Proxy? Should the proxy send an ACK followed by a BYE > request? How would the agent receiving BYE know that a race condition just > occurred at the Proxy? I think that the section 8 should cover this race > condition. > > > > -Vishal > > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP > Use sip-implementors@cs.columbia.edu for questions on current sip > Use sip@ietf.org for new developments of core SIP > > ------=_Part_48473_16900008.1155766245384 Content-Type: text/html; charset=ISO-8859-1 Content-Transfer-Encoding: 7bit Content-Disposition: inline
Vishal:
 
Offering calls to a BLA group relies the ability of the proxy to fork the request. The proxy simply forwards all 2xx responses to the calling side in cases where multiple UA's answer the call simultaneously. UA's are expected to handle this condition per section 13.2.2 of RFC 3261. I am not sure there is anything the UA would have to do in addition to what is already mentioned in RFC 3261 for accomodating this condition.
 
Venkatesh
 
On 8/15/06, Agrawal, Vishal <VAgrawal@spherecom.com> wrote:

In section 6.3 "Call offered to a BLA Group", it's possible that more than one user agents answer the incoming call simultaneously. What should be the responses from the Proxy? Should the proxy send an ACK followed by a BYE request? How would the agent receiving BYE know that a race condition just occurred at the Proxy? I think that the section 8 should cover this race condition.

 

-Vishal


_______________________________________________
Sipping mailing list   https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP


------=_Part_48473_16900008.1155766245384-- --===============0255418268== Content-Type: text/plain; charset="us-ascii" MIME-Version: 1.0 Content-Transfer-Encoding: 7bit Content-Disposition: inline _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP --===============0255418268==-- From sipping-bounces@ietf.org Wed Aug 16 18:33:52 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GDTw5-0001fC-TZ; Wed, 16 Aug 2006 18:32:29 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GDTw4-0001f6-Go for Sipping@ietf.org; Wed, 16 Aug 2006 18:32:28 -0400 Received: from mail2.spherecom.com ([66.237.126.105]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GDTw1-0002N2-J0 for Sipping@ietf.org; Wed, 16 Aug 2006 18:32:28 -0400 Content-class: urn:content-classes:message MIME-Version: 1.0 Subject: RE: [Sipping] Questions on draft-anil-sipping-bla-03.txt X-MimeOLE: Produced By Microsoft Exchange V6.5 Date: Wed, 16 Aug 2006 17:29:33 -0500 Message-ID: <5E6F57726155C84481FA244AE5C6DBA45C5F0B@stratosphere.spherecom.com> X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: [Sipping] Questions on draft-anil-sipping-bla-03.txt Thread-Index: AcbBgNQS4MwN8BHHTIG2JqFaSPeL/wAADcfg From: "Agrawal, Vishal" To: "Venkatesh" X-Spam-Score: 0.0 (/) X-Scan-Signature: 8a7d7b1ff9fdb39b8762b25bc5367e56 Cc: Sipping@ietf.org X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Content-Type: multipart/mixed; boundary="===============1500324034==" Errors-To: sipping-bounces@ietf.org This is a multi-part message in MIME format. --===============1500324034== Content-class: urn:content-classes:message Content-Type: multipart/alternative; boundary="----_=_NextPart_001_01C6C183.D7714683" This is a multi-part message in MIME format. ------_=_NextPart_001_01C6C183.D7714683 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: quoted-printable Hi Venkatesh, =20 I agree that the UAC should follow both section 13.2.2 and 15 in RFC 3261 to determine the next course of action. However, e.g. when all but one UA receive the BYE, do they terminate the SUBSCRIBE dialog usage as well? Or they need to keep the SUBSCRIBE dialog usage till they receive the NOTIFY with terminated subscription state? =20 I've few other questions basically to clear my understanding:=20 1.) Can two simultaneous outbound calls be allowed from two BLA user-agents if they choose different line appearances? 2.) How does the BLA user-agent tell the registrar/proxy, how many line appearances it has configured for the BLA DA? As per section 5.1 in the BLA ID it's recommended to send one REGISTER request per DA on the device, right? Should the proxy not send the INVITE to a user-agent in the BLA group if it has no line appearance available? 3.) The feature doesn't work for a SIP ATA, right? 4.) Are 2 & 3 configuration issues?=20 =20 Vishal =20 ________________________________ From: Venkatesh [mailto:vvenkatar@gmail.com]=20 Sent: Wednesday, August 16, 2006 5:11 PM To: Agrawal, Vishal Cc: Sipping@ietf.org Subject: Re: [Sipping] Questions on draft-anil-sipping-bla-03.txt =20 Vishal: =20 Offering calls to a BLA group relies the ability of the proxy to fork the request. The proxy simply forwards all 2xx responses to the calling side in cases where multiple UA's answer the call simultaneously. UA's are expected to handle this condition per section 13.2.2 of RFC 3261. I am not sure there is anything the UA would have to do in addition to what is already mentioned in RFC 3261 for accomodating this condition. =20 Venkatesh =20 On 8/15/06, Agrawal, Vishal wrote:=20 In section 6.3 "Call offered to a BLA Group", it's possible that more than one user agents answer the incoming call simultaneously. What should be the responses from the Proxy? Should the proxy send an ACK followed by a BYE request? How would the agent receiving BYE know that a race condition just occurred at the Proxy? I think that the section 8 should cover this race condition.=20 =20 -Vishal _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping =20 This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP =20 ------_=_NextPart_001_01C6C183.D7714683 Content-Type: text/html; charset="us-ascii" Content-Transfer-Encoding: quoted-printable

Hi = Venkatesh,

 

I = agree that the UAC should follow both section 13.2.2 and 15 in RFC 3261 to = determine the next course of action. However, e.g. when all but one UA receive the = BYE, do they terminate the SUBSCRIBE dialog usage as well? Or they need to keep = the SUBSCRIBE dialog usage till they receive the NOTIFY with terminated subscription state?

 

I’ve few other questions basically to clear my understanding: =

1.)     Can two simultaneous outbound calls be allowed from two BLA user-agents if they choose different line = appearances?

2.)    How does the BLA user-agent tell the registrar/proxy, how = many line appearances it has configured for the BLA DA? As per section 5.1 in the = BLA ID it’s recommended to send one REGISTER request per DA on the = device, right? Should the proxy not send the INVITE to a user-agent in the BLA = group if it has no line appearance available?

3.)    The feature doesn’t work for a SIP ATA, = right?

4.)    Are 2 & 3 configuration issues? =

 

Vishal

 


From: = Venkatesh [mailto:vvenkatar@gmail.com]
Sent: Wednesday, = August 16, = 2006 5:11 PM
To: Agrawal, Vishal
Cc: Sipping@ietf.org
Subject: Re: [Sipping] = Questions on draft-anil-sipping-bla-03.txt

 

Vishal:

 

Offering calls to a BLA group relies the ability of the proxy to = fork the request. The proxy simply forwards all 2xx responses to the calling = side in cases where multiple UA's answer the call simultaneously. UA's are expected to handle this condition per section 13.2.2 of RFC 3261. I = am not sure there is anything the UA would have to do in addition to what is = already mentioned in RFC 3261 for accomodating this condition.
 

Venkatesh
 

On 8/15/06, Agrawal, Vishal = <VAgrawal@spherecom.com> = wrote:

In section 6.3 "Call offered to a BLA Group", it's possible that = more than one user agents answer the incoming call simultaneously. What = should be the responses from the Proxy? Should the proxy send an ACK followed by a = BYE request? How would the agent receiving BYE know that a race condition = just occurred at the Proxy? I think that the section 8 should cover this race condition.

 

-Vishal


_______________________________________________
Sipping mailing list   https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.colu= mbia.edu for questions on current sip
Use sip@ietf.org for new = developments of core SIP

 

------_=_NextPart_001_01C6C183.D7714683-- --===============1500324034== Content-Type: text/plain; charset="us-ascii" MIME-Version: 1.0 Content-Transfer-Encoding: 7bit Content-Disposition: inline _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP --===============1500324034==-- From sipping-bounces@ietf.org Wed Aug 16 19:57:45 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GDVF6-0000rh-Dt; Wed, 16 Aug 2006 19:56:12 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GDVF3-0000rc-AA for Sipping@ietf.org; Wed, 16 Aug 2006 19:56:09 -0400 Received: from nf-out-0910.google.com ([64.233.182.186]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GDVF2-00005x-M1 for Sipping@ietf.org; Wed, 16 Aug 2006 19:56:09 -0400 Received: by nf-out-0910.google.com with SMTP id n15so886579nfc for ; Wed, 16 Aug 2006 16:56:07 -0700 (PDT) DomainKey-Signature: a=rsa-sha1; q=dns; c=nofws; s=beta; d=gmail.com; h=received:message-id:date:from:to:subject:cc:in-reply-to:mime-version:content-type:references; b=bGiKkubRkvfS4JctXf8dz45dM/hFMA/OenqpVcU84n7owM9fqV/5dqu3BovgjNUEFOKKt0M/QnmYJdNHqw9uj7GzireceaDOg0CCYyA6STj/EFyfWbqTKF4TKyDQCv1cG519SvVlYszcGn5CA0DwhX2UZymuYtMeMRwS0t4FHMc= Received: by 10.49.29.3 with SMTP id g3mr1410941nfj; Wed, 16 Aug 2006 16:56:07 -0700 (PDT) Received: by 10.78.139.6 with HTTP; Wed, 16 Aug 2006 16:56:07 -0700 (PDT) Message-ID: <4ff4e7370608161656i3d815b26mc2b62db3374f54f2@mail.gmail.com> Date: Wed, 16 Aug 2006 16:56:07 -0700 From: Venkatesh To: "Agrawal, Vishal" Subject: Re: [Sipping] Questions on draft-anil-sipping-bla-03.txt In-Reply-To: <5E6F57726155C84481FA244AE5C6DBA45C5F0B@stratosphere.spherecom.com> MIME-Version: 1.0 References: <5E6F57726155C84481FA244AE5C6DBA45C5F0B@stratosphere.spherecom.com> X-Spam-Score: 0.1 (/) X-Scan-Signature: 501044f827b673024f6a4cb1d46e67d2 Cc: Sipping@ietf.org X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Content-Type: multipart/mixed; boundary="===============0727150878==" Errors-To: sipping-bounces@ietf.org --===============0727150878== Content-Type: multipart/alternative; boundary="----=_Part_50638_28150962.1155772567688" ------=_Part_50638_28150962.1155772567688 Content-Type: text/plain; charset=ISO-8859-1; format=flowed Content-Transfer-Encoding: 7bit Content-Disposition: inline Replies inline.... On 8/16/06, Agrawal, Vishal wrote: > > Hi Venkatesh, > > > > I agree that the UAC should follow both section 13.2.2 and 15 in RFC 3261 > to determine the next course of action. However, e.g. when all but one UA > receive the BYE, do they terminate the SUBSCRIBE dialog usage as well? Or > they need to keep the SUBSCRIBE dialog usage till they receive the NOTIFY > with terminated subscription state? > [Venkatesh]: The SUBSCRIBE for dialog-state is an independent dialog by itself. Thus receipt of a BYE does not terminate the subscription. > > I've few other questions basically to clear my understanding: > > 1.) Can two simultaneous outbound calls be allowed from two BLA > user-agents if they choose different line appearances? > [Venkatesh]: Yes. 2.) How does the BLA user-agent tell the registrar/proxy, how many line > appearances it has configured for the BLA DA? As per section 5.1 in the > BLA ID it's recommended to send one REGISTER request per DA on the device, > right? Should the proxy not send the INVITE to a user-agent in the BLA group > if it has no line appearance available? > [Venkatesh]: The BLA user-agent does not need to provide this information to the Registrar/Proxy. This is managed locally by the UA. It is for this reason, the draft recommends that UA's send only one register for the AOR regardless of how many line appearances it is configured with. Since line management is local to the phone, the proxy will always fork an incoming request to a UA regardless of availability of lines at a specific UA. UA's can reject the incoming INVITE with an appropriate response if it does not have any free lines. 3.) The feature doesn't work for a SIP ATA, right? > [Venkatesh]: I am not sure I understand this question. 4.) Are 2 & 3 configuration issues? > > > Vishal > > > ------------------------------ > > *From:* Venkatesh [mailto:vvenkatar@gmail.com] > *Sent:* Wednesday, August 16, 2006 5:11 PM > *To:* Agrawal, Vishal > *Cc:* Sipping@ietf.org > *Subject:* Re: [Sipping] Questions on draft-anil-sipping-bla-03.txt > > > > Vishal: > > > > Offering calls to a BLA group relies the ability of the proxy to fork the > request. The proxy simply forwards all 2xx responses to the calling side in > cases where multiple UA's answer the call simultaneously. UA's are expected > to handle this condition per section 13.2.2 of RFC 3261. I am not sure > there is anything the UA would have to do in addition to what is already > mentioned in RFC 3261 for accomodating this condition. > > > Venkatesh > > > On 8/15/06, *Agrawal, Vishal* wrote: > > In section 6.3 "Call offered to a BLA Group", it's possible that more than > one user agents answer the incoming call simultaneously. What should be the > responses from the Proxy? Should the proxy send an ACK followed by a BYE > request? How would the agent receiving BYE know that a race condition just > occurred at the Proxy? I think that the section 8 should cover this race > condition. > > > > -Vishal > > > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP > Use sip-implementors@cs.columbia.edu for questions on current sip > Use sip@ietf.org for new developments of core SIP > > > ------=_Part_50638_28150962.1155772567688 Content-Type: text/html; charset=ISO-8859-1 Content-Transfer-Encoding: 7bit Content-Disposition: inline Replies inline....

On 8/16/06, Agrawal, Vishal <VAgrawal@spherecom.com> wrote:

Hi Venkatesh,

 

I agree that the UAC should follow both section 13.2.2 and 15 in RFC 3261 to determine the next course of action. However, e.g . when all but one UA receive the BYE, do they terminate the SUBSCRIBE dialog usage as well? Or they need to keep the SUBSCRIBE dialog usage till they receive the NOTIFY with terminated subscription state?

[Venkatesh]: The SUBSCRIBE for dialog-state is an independent dialog by itself. Thus receipt of a BYE does not terminate the subscription.

 

I've few other questions basically to clear my understanding:

1.)     Can two simultaneous outbound calls be allowed from two BLA user-agents if they choose different line appearances?

[Venkatesh]: Yes.

2.)    How does the BLA user-agent tell the registrar/proxy, how many line appearances it has configured for the BLA DA? As per section 5.1 in the BLA ID it's recommended to send one REGISTER request per DA on the device, right? Should the proxy not send the INVITE to a user-agent in the BLA group if it has no line appearance available?

 
[Venkatesh]: The BLA user-agent does not need to provide this information to the Registrar/Proxy. This is managed locally by the UA. It is for this reason, the draft recommends that UA's send only one register for the AOR regardless of how many line appearances it is configured with. Since line management is local to the phone, the proxy will always fork an incoming request to a UA regardless of availability of lines at a specific UA. UA's can reject the incoming INVITE with an appropriate response if it does not have any free lines.

3.)    The feature doesn't work for a SIP ATA, right?


[Venkatesh]: I am not sure I understand this question.


4.)    Are 2 & 3 configuration issues?

 

Vishal

 


From: Venkatesh [mailto: vvenkatar@gmail.com]
Sent: Wednesday, August 16, 2006 5:11 PM
To: Agrawal, Vishal
Cc: Sipping@ietf.org
Subject: Re: [Sipping] Questions on draft-anil-sipping-bla-03.txt

 

Vishal:

 

Offering calls to a BLA group relies the ability of the proxy to fork the request. The proxy simply forwards all 2xx responses to the calling side in cases where multiple UA's answer the call simultaneously. UA's are expected to handle this condition per section 13.2.2 of RFC 3261. I am not sure there is anything the UA would have to do in addition to what is already mentioned in RFC 3261 for accomodating this condition.
 

Venkatesh
 

On 8/15/06, Agrawal, Vishal < VAgrawal@spherecom.com> wrote:

In section 6.3 "Call offered to a BLA Group", it's possible that more than one user agents answer the incoming call simultaneously. What should be the responses from the Proxy? Should the proxy send an ACK followed by a BYE request? How would the agent receiving BYE know that a race condition just occurred at the Proxy? I think that the section 8 should cover this race condition.

 

-Vishal


_______________________________________________
Sipping mailing list   https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions on current sip
Use sip@ietf.org for new developments of core SIP

 


------=_Part_50638_28150962.1155772567688-- --===============0727150878== Content-Type: text/plain; charset="us-ascii" MIME-Version: 1.0 Content-Transfer-Encoding: 7bit Content-Disposition: inline _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP --===============0727150878==-- From sipping-bounces@ietf.org Wed Aug 16 23:48:15 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GDYq2-0000rd-IG; Wed, 16 Aug 2006 23:46:34 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GDYq1-0000rU-Bx for sipping@ietf.org; Wed, 16 Aug 2006 23:46:33 -0400 Received: from sonussf1.sonusnet.com ([208.45.178.26]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GDYpz-0005KX-Ph for sipping@ietf.org; Wed, 16 Aug 2006 23:46:33 -0400 Received: from sonusmail05.sonusnet.com (sonusmail05.sonusnet.com [10.128.32.155]) by sonussf1.sonusnet.com (8.13.3/8.13.3) with ESMTP id k7H3kQ3A027468; Wed, 16 Aug 2006 23:46:26 -0400 Received: from SONUSINMAIL01.sonusnet.com ([10.128.254.7]) by sonusmail05.sonusnet.com with Microsoft SMTPSVC(6.0.3790.1830); Wed, 16 Aug 2006 23:46:26 -0400 X-MimeOLE: Produced By Microsoft Exchange V6.0.6603.0 content-class: urn:content-classes:message MIME-Version: 1.0 Content-Type: multipart/mixed; boundary="----_=_NextPart_001_01C6C1AF.B479695C" Date: Thu, 17 Aug 2006 09:16:21 +0530 Message-ID: <70798CF421F00F4DA018059E5B7EEB8C7A5D17@sonusinmail01.sonusnet.com> X-MS-Has-Attach: yes X-MS-TNEF-Correlator: Thread-Topic: [Sip] comments on draft-ietf-sipping-dialogusage-02.txt Thread-Index: Aca/qnu7TPQm75soSWGV6bpsTk8mSQCBFM3w From: "B, Nataraju" To: "Paul Kyzivat" X-OriginalArrivalTime: 17 Aug 2006 03:46:26.0253 (UTC) FILETIME=[B721EFD0:01C6C1AF] X-Spam-Score: 0.0 (/) X-Scan-Signature: 2beba50d0fcdeee5f091c59f204d4365 Cc: sipping@ietf.org Subject: [Sipping] RE: [Sip] comments on draft-ietf-sipping-dialogusage-02.txt X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org This is a multi-part message in MIME format. ------_=_NextPart_001_01C6C1AF.B479695C Content-Type: text/plain; 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Mon, 14 Aug 2006 12:04:48 +0530 Received: from sonussf1.sonusnet.com ([192.168.1.51]) by sonusmail01.sonusnet.com with Microsoft SMTPSVC(5.0.2195.6713); Mon, 14 Aug 2006 02:34:41 -0400 MIME-Version: 1.0 Content-Type: multipart/mixed; boundary="----_=_NextPart_002_01C6BF6B.BD00F400" Received: from megatron.ietf.org (megatron.ietf.org [156.154.16.145]) by sonussf1.sonusnet.com (8.13.3/8.13.3) with ESMTP id k7E6YeNL003687; Mon, 14 Aug 2006 02:34:40 -0400 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GCW1s-0001fC-7m; Mon, 14 Aug 2006 02:34:28 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GCW1p-0001Zn-JR for sip@ietf.org; Mon, 14 Aug 2006 02:34:25 -0400 Received: from sonussf1.sonusnet.com ([208.45.178.26]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GCW1o-0001bo-6p for sip@ietf.org; Mon, 14 Aug 2006 02:34:25 -0400 Received: from sonusmail06.sonusnet.com (sonusmail06.sonusnet.com [10.128.32.156]) by sonussf1.sonusnet.com (8.13.3/8.13.3) with ESMTP id k7E6YOvH003659 for ; Mon, 14 Aug 2006 02:34:24 -0400 Received: from SONUSINMAIL01.sonusnet.com ([10.128.254.7]) by sonusmail06.sonusnet.com with Microsoft SMTPSVC(6.0.3790.1830); Mon, 14 Aug 2006 02:34:23 -0400 content-class: urn:content-classes:message Subject: [Sip] comments on draft-ietf-sipping-dialogusage-02.txt Date: Mon, 14 Aug 2006 12:04:19 +0530 Message-ID: <70798CF421F00F4DA018059E5B7EEB8C7A5BDA@sonusinmail01.sonusnet.com> X-MS-Has-Attach: yes X-MS-TNEF-Correlator: Thread-Topic: comments on draft-ietf-sipping-dialogusage-02.txt Thread-Index: Aca/a6vhg1oTxT0IQdO3kIXzBLjzNA== List-Help: List-Subscribe: , List-Unsubscribe: , From: "B, Nataraju" To: This is a multi-part message in MIME format. ------_=_NextPart_002_01C6BF6B.BD00F400 Content-Type: text/plain; charset="utf-8" Content-Transfer-Encoding: base64 SGkgU3BhcmtzLCANCg0KIA0KDQpJIGhhZCBnb25lIHRocm91Z2ggdGhlIGRvY3VtZW50LCBoZXJl IGFyZSBmZXcgY29tbWVudHMuIA0KDQogDQoNCk5vdGU6IEkgYXNzdW1lIHRoZSBiZWhhdmlvciBt ZW50aW9uZWQgaW4gc2VjIDQgYXBwbGllcyBib3RoIGF0IGNsaWVudCByZWNlaXZpbmcgdGhlc2Ug cmVzcG9uc2VzIChOb3RpZmllcikgYW5kIHNlcnZlciBnZW5lcmF0aW5nIHRoZXNlIHJlcGxpZXMg KFN1YnNjcmliZXIpLiBJZiB0aGlzIHdhcyBtZWFudCB0byBiZSBvbmx5IEAgc2VuZGVyIG9mIE5P VElGWSBzaGFsbCB3ZSBuZWVkIHRvIG1lbnRpb24gdGhlIHNpbWlsYXIgc2V0IG9mIGFjdGlvbnMg QCBTdWJzY3JpYmVyIHNpZGUgdG9vPyANCg0KIA0KDQoxLiBTZWMgNCBkZXNjcmliZXMgd2hhdCB0 aGUgZWZmZWN0cyBvbiBkaWFsb2cgYXJlIGFuZCBpdHMgdXNhZ2VzIHdoZW4gcGFydGljdWxhciBy ZXNwb25zZXMgYXJlIHJlY2VpdmVkIGZvciBOT1RJRlkgbWVzc2FnZSwgdGhlIHJlc3BvbnNlcyA0 MDQsIDQxMCwgNDE2LCBsZWFkIHRvIGRpYWxvZyBhbmQgaXRzIHVzYWdlIHRlcm1pbmF0aW9uLiAN Cg0KICAgICAgYS4gSSBmZWVsIHdlIHNob3VsZCBoYXZlIG1lbnRpb25lZCBPTkxZIHRoZSBhdXRo ZW50aWNhdGVkIE5PVElGWSByZXF1ZXN0cyB3aWxsIGhhdmUgdGhpcyBlZmZlY3Qgb24gdGhlIGRp YWxvZ3MvdXNhZ2VzLiBPdGhlcndpc2UgYW55IG1pZGRsZSBtZW4gY2FuIHRyYWNlIHRoZSBtZXNz YWdlIGFuZCBzZW5kIHRoZSBOT1RJRlkgbWVzc2FnZXMgd2hpY2ggdGVybWluYXRlIHRoZSBkaWFs b2dzLCB3aGljaCBpcyB1bmRlc2lyYWJsZSDimLkuIElmIHRoZSBlbnRpdHkgc2VuZGluZyBOT1RJ RlkgcmVhbGx5IGludGVuZCB0byBkbyB0aGF0IGl0IGNhbiByZXNlbmQgTk9USUZZIHdpdGggYXV0 aG9yaXphdGlvbiBjcmVkZW50aWFscyB3aGljaCB3aWxsIGhlbHAgdG8gdGVybWluYXRlIHRoZSBh c3NvY2lhdGVkIGRpYWxvZ3MuDQoNCiANCg0KICAgICAgYi4gT3RoZXIgYWx0ZXJuYXRpdmUgd291 bGQgYmUgdGhhdCBsZXQgdGhvc2UgcmVzcG9uc2VzIGhhdmUgbm8gZWZmZWN0IG9uIHRoZSBkaWFs b2cvdXNhZ2VzLCBldmVudHVhbGx5IHRoZSBzdWJzY3JpcHRpb24gZHVyYXRpb24gd291bGQgdGFr ZSBjYXJlIG9mIGRpYWxvZyB0ZXJtaW5hdGlvbiBvciB0aGUgYWN0dWFsIGVudGl0eSB3b3VsZCB0 ZXJtaW5hdGUgb3IgZXh0ZW5kIHRoZSBzdWJzY3JpcHRpb24uIEFueSB3YXkgdGhlIHJlcXVlc3Qg ZnJvbSBhY3R1YWwgbm90aWZpZXIgd291bGQgaGF2ZSB0aGUgcHJvcGVyIFVSSXMgb3IgY3JlZGVu dGlhbHMgd2hpY2ggd2lsbCBuZXZlciBsZWFkIHRvIHRoZXNlIHJlc3BvbnNlcy4gDQoNCiANCg0K VGhlIHNhbWUgbG9naWMgd291bGQgbmVlZCB0byBleHRlbmQgZm9yIHJlcGxpZXMgNDgxLCA0ODMs IDQ4NCwgNDg5IGFuZCA2MDQgYWxzby4NCg0KIA0KDQoyLiBTZWMgNCBvZiB0aGUgZHJhZnQNCg0K ICAgDQoNCiAgICA8PDwgICA0MjMgSW50ZXJ2YWwgVG9vIEJyaWVmOiBUaGlzIHJlc3BvbnNlIHdv bid0IGhhcHBlbiBpbiBvdXIgZXhhbXBsZQ0KDQogICAgICBzY2VuYXJpbywgYnV0IGlmIGl0IGNh bWUgaW4gcmVzcG9uc2UgdG8gYSByZVNVQlNDUklCRSwgdGhlDQoNCiAgICAgIHN1YnNjcmliZSB1 c2FnZSBpcyBub3QgZGVzdHJveWVkIChvciBvdGhlcndpc2UgYWZmZWN0ZWQpLiAgTm8NCg0KICAg ICAgb3RoZXIgdXNhZ2VzIG9mIHRoZSBkaWFsb2cgYXJlIGFmZmVjdGVkLiA+Pj4NCg0KIA0KDQog ICAgICBJbiB0aGUgYWJvdmUgUGFyYSB0aGUgbWVhbmluZyBvZiBsYXN0IHBhcnQgb2YgMXN0IHNl bnRlbmNlIChJIG1lYW4sIHRoZSBzdWJzY3JpYmUgdXNhZ2UgaXMgbm90IGRlc3Ryb3llZCAob3Ig b3RoZXJ3aXNlIGFmZmVjdGVkKSkgYW5kIDJuZCBzZW50ZW5jZSBpcyBub3QgY2xlYXIsIGJvdGgg bWVudGlvbiB0aGUgc2FtZSBtZWFuaW5nPyBDYW4gd2UgbWFrZSBpdCBleHBsaWNpdCBhYm91dCB3 aGF0IHdlIHJlYWxseSBtZWFuIGhlcmUgPyANCg0KIA0KDQpUaGFua3MsIA0KDQpOYXRhcmFqdSBB IEINCg0KU29udXMgTmV0d29ya3MgSW5kaWEgUHZ0IEx0ZA0KDQo= ------_=_NextPart_002_01C6BF6B.BD00F400 Content-Type: text/plain; name="ATT192005.txt" Content-Transfer-Encoding: base64 Content-Description: ATT192005.txt Content-Disposition: inline; filename="ATT192005.txt" X19fX19fX19fX19fX19fX19fX19fX19fX19fX19fX19fX19fX19fX19fX19fX18NClNpcCBtYWls aW5nIGxpc3QgIGh0dHBzOi8vd3d3MS5pZXRmLm9yZy9tYWlsbWFuL2xpc3RpbmZvL3NpcA0KVGhp cyBsaXN0IGlzIGZvciBORVcgZGV2ZWxvcG1lbnQgb2YgdGhlIGNvcmUgU0lQIFByb3RvY29sDQpV c2Ugc2lwLWltcGxlbWVudG9yc0Bjcy5jb2x1bWJpYS5lZHUgZm9yIHF1ZXN0aW9ucyBvbiBjdXJy ZW50IHNpcA0KVXNlIHNpcHBpbmdAaWV0Zi5vcmcgZm9yIG5ldyBkZXZlbG9wbWVudHMgb24gdGhl IGFwcGxpY2F0aW9uIG9mIHNpcA== ------_=_NextPart_002_01C6BF6B.BD00F400-- ------_=_NextPart_001_01C6C1AF.B479695C Content-Type: text/plain; charset="us-ascii" MIME-Version: 1.0 Content-Transfer-Encoding: 7bit Content-Disposition: inline _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP ------_=_NextPart_001_01C6C1AF.B479695C-- From sipping-bounces@ietf.org Thu Aug 17 04:51:10 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GDdZY-0004Q3-Ec; Thu, 17 Aug 2006 04:49:52 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GDdZX-0004Py-Js for sipping@ietf.org; Thu, 17 Aug 2006 04:49:51 -0400 Received: from sonussf1.sonusnet.com ([208.45.178.26]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GDdZW-0006XX-5B for sipping@ietf.org; Thu, 17 Aug 2006 04:49:51 -0400 Received: from sonusmail06.sonusnet.com (sonusmail06.sonusnet.com [10.128.32.156]) by sonussf1.sonusnet.com (8.13.3/8.13.3) with ESMTP id k7H8nhwv019537; Thu, 17 Aug 2006 04:49:43 -0400 Received: from SONUSINMAIL01.sonusnet.com ([10.128.254.7]) by sonusmail06.sonusnet.com with Microsoft SMTPSVC(6.0.3790.1830); Thu, 17 Aug 2006 04:49:43 -0400 X-MimeOLE: Produced By Microsoft Exchange V6.0.6603.0 content-class: urn:content-classes:message MIME-Version: 1.0 Subject: RE: [Sipping] bug in rfc-3261 Date: Thu, 17 Aug 2006 14:19:38 +0530 Message-ID: <70798CF421F00F4DA018059E5B7EEB8C7A5DE9@sonusinmail01.sonusnet.com> X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: [Sipping] bug in rfc-3261 Thread-Index: Aca/ieeygas/2OQ4S2q8R/zZrklZkwCT+TNQ From: "B, Nataraju" To: "Manjunath Warad" , X-OriginalArrivalTime: 17 Aug 2006 08:49:43.0021 (UTC) FILETIME=[154061D0:01C6C1DA] X-Spam-Score: 0.0 (/) X-Scan-Signature: 7d38eb86565b1a4d8c3dba35af39014d Cc: X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Content-Type: multipart/mixed; boundary="===============0463970250==" Errors-To: sipping-bounces@ietf.org This is a multi-part message in MIME format. --===============0463970250== content-class: urn:content-classes:message Content-Type: multipart/alternative; boundary="----_=_NextPart_001_01C6C1DA.129F252F" This is a multi-part message in MIME format. ------_=_NextPart_001_01C6C1DA.129F252F Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: quoted-printable Hi All,=20 =20 I did not get many replies.=20 Should this be logged as bug against 3261 or not ?=20 Thanks,=20 Nataraju A B=20 _____ =20 From: Manjunath Warad [mailto:manjunathwarad@huawei.com]=20 Sent: Monday, August 14, 2006 3:17 PM To: B, Nataraju; sipping@ietf.org Subject: RE: [Sipping] bug in rfc-3261 =20 Hi, Please see inline... =20 Rgds, Manju -----Original Message----- From: B, Nataraju [mailto:bnataraju@sonusnet.com]=20 Sent: Monday, August 14, 2006 5:22 PM To: sipping@ietf.org Subject: [Sipping] bug in rfc-3261 Header field where proxy ACK BYE CAN INV OPT REG WWW-Authenticate 401 ar - m - m m m WWW-Authenticate 407 ar - o - o o o Contact 3xx d - o - o o o Why proxy column is mentioned as "ar" for WWW-Authenticate in 401/407. It should have been mentioned as "r" only. In no case proxy would add the WWW-Authenticate header for responses 401/407 ?. Does the proxy aggregation considered for this case ?=20 Yes you are right! Proxy aggregation is considered here.=20 Why proxy is allowed to "delete only" for contact in 3xx response? Even it should be allowed to read the contacts in 3xx replies. BTW, in which scenario it can delete a contact in 3xx response. This option should have been either "dr" or "r" only depending on answer to my previous question. =20 Here again, proxy can remove the contact, for which it has already tried, so that upstream proxies need not try on those contacts again. This technique is used when proxy chose 3xx as best response to be forwarded for upstream proxies. So, I think "adr" need to be used here as this header can be aggregated to form the final 3xx response if more than 1 is present. [ABN] this one should be considered bug against rfc-3261.=20 Please let me know, if my interpretation is wrong. Thanks, Nataraju A B Sonus Networks Ind Pvt Ltd ------_=_NextPart_001_01C6C1DA.129F252F Content-Type: text/html; charset="us-ascii" Content-Transfer-Encoding: quoted-printable Message

Hi All, =

 

I did = not get many replies.

Should = this be logged as bug against 3261 or not ?

Thanks,
Nataraju A B


From: = Manjunath Warad [mailto:manjunathwarad@huawei.com]
Sent: Monday, August 14, = 2006 3:17 PM
To: B, Nataraju; sipping@ietf.org
Subject: RE: [Sipping] = bug in rfc-3261

 

Hi,

    Please see = inline...

 

Rgds,

Manju

-----Original = Message-----
From: B, Nataraju [mailto:bnataraju@sonusnet.com]
Sent: Monday, August 14, = 2006 5:22 PM
To: sipping@ietf.org
Subject: [Sipping] bug in = rfc-3261

Header field           &n= bsp;  where       proxy ACK BYE CAN INV OPT = REG

WWW-Authenticate      &nbs= p;    401         ar    -   m   -   m   m   = m

WWW-Authenticate      &nbs= p;    407         ar    -   o   -   o   o   = o

Contact        &= nbsp;            = 3xx         d    -   o   = -   o   o   o

Why proxy column is mentioned as “ar” for WWW-Authenticate in 401/407. It should have been = mentioned as “r” only. In no case proxy would add the WWW-Authenticate = header for responses 401/407 ?. Does the = proxy aggregation considered for this case ? 

Yes you are right! Proxy aggregation is considered here. 

Why proxy is allowed to = “delete only” for contact in 3xx response? Even it should be allowed to = read the contacts in 3xx replies. BTW, in which scenario it can = delete a contact in 3xx response. This option should have been = either “dr” or “r” only depending on answer to my = previous question.  

Here again, proxy can remove the contact, for which it has already tried, so that upstream proxies = need not try on those contacts again. This technique is used when proxy = chose 3xx as best response to be forwarded for upstream = proxies. So, I think "adr" need to be used here as this = header can be aggregated to form the final 3xx response if more than 1 is present.

[ABN] this one should be = considered bug against rfc-3261.

Please let me know, if my interpretation is = wrong.

Thanks,

Nataraju A = B

Sonus Networks Ind Pvt = Ltd

------_=_NextPart_001_01C6C1DA.129F252F-- --===============0463970250== Content-Type: text/plain; charset="us-ascii" MIME-Version: 1.0 Content-Transfer-Encoding: 7bit Content-Disposition: inline _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP --===============0463970250==-- From sipping-bounces@ietf.org Thu Aug 17 10:15:28 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GDicy-00061f-Ol; Thu, 17 Aug 2006 10:13:44 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GDicw-00060H-W1 for sipping@ietf.org; Thu, 17 Aug 2006 10:13:42 -0400 Received: from rtp-iport-2.cisco.com ([64.102.122.149]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GDhSj-0003vC-3z for sipping@ietf.org; Thu, 17 Aug 2006 08:59:07 -0400 Received: from rtp-dkim-1.cisco.com ([64.102.121.158]) by rtp-iport-2.cisco.com with ESMTP; 17 Aug 2006 08:59:05 -0400 X-IronPort-AV: i="4.08,136,1154923200"; d="scan'208"; a="97274823:sNHT34395596" Received: from rtp-core-1.cisco.com (rtp-core-1.cisco.com [64.102.124.12]) by rtp-dkim-1.cisco.com (8.12.11.20060308/8.12.11) with ESMTP id k7HCx4RH030068; Thu, 17 Aug 2006 08:59:04 -0400 Received: from xbh-rtp-201.amer.cisco.com (xbh-rtp-201.cisco.com [64.102.31.12]) by rtp-core-1.cisco.com (8.12.10/8.12.6) with ESMTP id k7HCx4dM019454; Thu, 17 Aug 2006 08:59:04 -0400 (EDT) Received: from xfe-rtp-201.amer.cisco.com ([64.102.31.38]) by xbh-rtp-201.amer.cisco.com with Microsoft SMTPSVC(6.0.3790.1830); Thu, 17 Aug 2006 08:59:04 -0400 Received: from [161.44.79.104] ([161.44.79.104]) by xfe-rtp-201.amer.cisco.com with Microsoft SMTPSVC(6.0.3790.1830); Thu, 17 Aug 2006 08:59:04 -0400 Message-ID: <44E46817.9000409@cisco.com> Date: Thu, 17 Aug 2006 08:59:03 -0400 From: Paul Kyzivat User-Agent: Thunderbird 1.5.0.5 (Windows/20060719) MIME-Version: 1.0 To: "B, Nataraju" References: <70798CF421F00F4DA018059E5B7EEB8C7A5D17@sonusinmail01.sonusnet.com> In-Reply-To: <70798CF421F00F4DA018059E5B7EEB8C7A5D17@sonusinmail01.sonusnet.com> Content-Type: text/plain; charset=UTF-8; format=flowed Content-Transfer-Encoding: 8bit X-OriginalArrivalTime: 17 Aug 2006 12:59:04.0099 (UTC) FILETIME=[EABFB730:01C6C1FC] DKIM-Signature: a=rsa-sha1; q=dns; l=4793; t=1155819544; x=1156683544; c=relaxed/simple; s=rtpdkim1001; h=Content-Type:From:Subject:Content-Transfer-Encoding:MIME-Version; d=cisco.com; i=pkyzivat@cisco.com; z=From:Paul=20Kyzivat=20 |Subject:Re=3A=20[Sip]=20comments=20on=20draft-ietf-sipping-dialogusage-02.txt |To:=22B,=20Nataraju=22=20; X=v=3Dcisco.com=3B=20h=3DTH6BmAmf0rRE0cKOJxwAXgTnrbA=3D; b=bhCaCDtowkC9VDLk1bmrCfz0P1kHQJKDGsTfJMQNEEUWw3sqUep7/8FnLbr9pNxaUgmQIy8S xe+sER08NwPkj32gS+iraWvWjFFz6qpOrUkm53UWhFkEuCtCCtzRgOQX; Authentication-Results: rtp-dkim-1.cisco.com; header.From=pkyzivat@cisco.com; dkim=pass ( sig from cisco.com verified; ); X-Spam-Score: 0.0 (/) X-Scan-Signature: ff03b0075c3fc728d7d60a15b4ee1ad2 Cc: sipping@ietf.org Subject: [Sipping] Re: [Sip] comments on draft-ietf-sipping-dialogusage-02.txt X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org B, Nataraju wrote: > Paul, > > What I meant to mention here was that, in the scenarios which might lead to these responses must be authenticated before we call for this Reponses... > I mean if we receive a NOTIFY(un-authenticated) 404 should not be sent out which will terminate the dialog/usage. > > Authenticate the NTFY request first then only it could be rejected with 404 or any other appropriate replies which would result in dialog termination. If I understand you, this is just standard procedure, applicable regardless of dialog usages. Authentication is supposed to be done first, so no extra information is given to unauthenticated callers. Paul > Thanks, > Nataraju A B > >> -----Original Message----- >> From: Paul Kyzivat [mailto:pkyzivat@cisco.com] >> Sent: Monday, August 14, 2006 7:34 PM >> To: B, Nataraju >> Cc: sip@ietf.org >> Subject: Re: [Sip] comments on draft-ietf-sipping-dialogusage-02.txt >> >> >> >> B, Nataraju wrote: >> >>> 1. Sec 4 describes what the effects on dialog are and its usages when >>> particular responses are received for NOTIFY message, the responses 404, >>> 410, 416, lead to dialog and its usage termination. >>> >>> a. I feel we should have mentioned ONLY the authenticated NOTIFY >>> requests will have this effect on the dialogs/usages. Otherwise any >>> middle men can trace the message and send the NOTIFY messages which >>> terminate the dialogs, which is undesirable ☹. If the entity sending >>> NOTIFY really intend to do that it can resend NOTIFY with authorization >>> credentials which will help to terminate the associated dialogs. >> I agree we wouldn't want to enable a MITM attack here. >> >> But I think that is already covered. If the request isn't properly >> authenticated, then the response will be 401, not 404, 410, or 416. >> >> Paul > > > > ------------------------------------------------------------------------ > > Subject: > [Sip] comments on draft-ietf-sipping-dialogusage-02.txt > From: > "B, Nataraju" > Date: > Mon, 14 Aug 2006 12:04:19 +0530 > To: > > > To: > > > > Hi Sparks, > > > > I had gone through the document, here are few comments. > > > > Note: I assume the behavior mentioned in sec 4 applies both at client receiving these responses (Notifier) and server generating these replies (Subscriber). If this was meant to be only @ sender of NOTIFY shall we need to mention the similar set of actions @ Subscriber side too? > > > > 1. Sec 4 describes what the effects on dialog are and its usages when particular responses are received for NOTIFY message, the responses 404, 410, 416, lead to dialog and its usage termination. > > a. I feel we should have mentioned ONLY the authenticated NOTIFY requests will have this effect on the dialogs/usages. Otherwise any middle men can trace the message and send the NOTIFY messages which terminate the dialogs, which is undesirable ☹. If the entity sending NOTIFY really intend to do that it can resend NOTIFY with authorization credentials which will help to terminate the associated dialogs. > > > > b. Other alternative would be that let those responses have no effect on the dialog/usages, eventually the subscription duration would take care of dialog termination or the actual entity would terminate or extend the subscription. Any way the request from actual notifier would have the proper URIs or credentials which will never lead to these responses. > > > > The same logic would need to extend for replies 481, 483, 484, 489 and 604 also. > > > > 2. Sec 4 of the draft > > > > <<< 423 Interval Too Brief: This response won't happen in our example > > scenario, but if it came in response to a reSUBSCRIBE, the > > subscribe usage is not destroyed (or otherwise affected). No > > other usages of the dialog are affected. >>> > > > > In the above Para the meaning of last part of 1st sentence (I mean, the subscribe usage is not destroyed (or otherwise affected)) and 2nd sentence is not clear, both mention the same meaning? Can we make it explicit about what we really mean here ? > > > > Thanks, > > Nataraju A B > > Sonus Networks India Pvt Ltd > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Sip mailing list https://www1.ietf.org/mailman/listinfo/sip > This list is for NEW development of the core SIP Protocol > Use sip-implementors@cs.columbia.edu for questions on current sip > Use sipping@ietf.org for new developments on the application of sip _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Thu Aug 17 13:30:39 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GDlfO-000547-1W; Thu, 17 Aug 2006 13:28:26 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GDlfM-000540-Mh for sipping@ietf.org; Thu, 17 Aug 2006 13:28:24 -0400 Received: from nylon.softarmor.com ([66.135.38.164]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GDlfL-0005vI-BZ for sipping@ietf.org; Thu, 17 Aug 2006 13:28:24 -0400 Received: from [64.101.149.196] (dwillis-mb.cisco.com [64.101.149.196]) (authenticated bits=0) by nylon.softarmor.com (8.13.1/8.13.1) with ESMTP id k7HHUZXd028327 (version=TLSv1/SSLv3 cipher=RC4-SHA bits=128 verify=NO); Thu, 17 Aug 2006 12:30:36 -0500 In-Reply-To: <70798CF421F00F4DA018059E5B7EEB8C7A5DE9@sonusinmail01.sonusnet.com> References: <70798CF421F00F4DA018059E5B7EEB8C7A5DE9@sonusinmail01.sonusnet.com> Mime-Version: 1.0 (Apple Message framework v752.2) Content-Type: text/plain; charset=WINDOWS-1252; delsp=yes; format=flowed Message-Id: Content-Transfer-Encoding: quoted-printable From: Dean Willis Subject: Re: [Sipping] bug in rfc-3261 Date: Thu, 17 Aug 2006 12:28:03 -0500 To: "B, Nataraju" X-Mailer: Apple Mail (2.752.2) X-Spam-Score: 0.0 (/) X-Scan-Signature: a2c12dacc0736f14d6b540e805505a86 Cc: sipping@ietf.org X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Personally, I think that whole misbegotten table should be logged as =20 a bug in 3261. It confuses more people than Hilbert's Paradox. But I've not yet seem a better way to render that information. If =20 anybody has an idea, please bring it to the table. -- dean On Aug 17, 2006, at 3:49 AM, B, Nataraju wrote: > Hi All, > > > > I did not get many replies. > > Should this be logged as bug against 3261 or not ? > > Thanks, > Nataraju A B > > From: Manjunath Warad [mailto:manjunathwarad@huawei.com] > Sent: Monday, August 14, 2006 3:17 PM > To: B, Nataraju; sipping@ietf.org > Subject: RE: [Sipping] bug in rfc-3261 > > > > Hi, > > Please see inline... > > > > Rgds, > > Manju > > -----Original Message----- > From: B, Nataraju [mailto:bnataraju@sonusnet.com] > Sent: Monday, August 14, 2006 5:22 PM > To: sipping@ietf.org > Subject: [Sipping] bug in rfc-3261 > > Header field where proxy ACK BYE CAN INV OPT REG > > WWW-Authenticate 401 ar - m - m m m > > WWW-Authenticate 407 ar - o - o o o > > Contact 3xx d - o - o o o > > Why proxy column is mentioned as =93ar=94 for WWW-Authenticate in =20 > 401/407. It should have been mentioned as =93r=94 only. In no case =20 > proxy would add the WWW-Authenticate header for responses =20 > 401/407 ?. Does the proxy aggregation considered for this case ? > > Yes you are right! Proxy aggregation is considered here. > > Why proxy is allowed to =93delete only=94 for contact in 3xx response? = =20 > Even it should be allowed to read the contacts in 3xx replies. BTW, =20= > in which scenario it can delete a contact in 3xx response. This =20 > option should have been either =93dr=94 or =93r=94 only depending on = answer =20 > to my previous question. > > Here again, proxy can remove the contact, for which it has already =20 > tried, so that upstream proxies need not try on those contacts =20 > again. This technique is used when proxy chose 3xx as best response =20= > to be forwarded for upstream proxies. So, I think "adr" need to be =20 > used here as this header can be aggregated to form the final 3xx =20 > response if more than 1 is present. > > [ABN] this one should be considered bug against rfc-3261. > > Please let me know, if my interpretation is wrong. > > Thanks, > > Nataraju A B > > Sonus Networks Ind Pvt Ltd > > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP > Use sip-implementors@cs.columbia.edu for questions on current sip > Use sip@ietf.org for new developments of core SIP _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Thu Aug 17 16:48:01 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GDolQ-0004rF-Jc; Thu, 17 Aug 2006 16:46:52 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GDolO-0004qS-3r; Thu, 17 Aug 2006 16:46:50 -0400 Received: from stsc1260-eth-s1-s1p1-vip.va.neustar.com ([156.154.16.129] helo=chiedprmail1.ietf.org) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GDo5u-0006og-IA; Thu, 17 Aug 2006 16:03:58 -0400 Received: from ns3.neustar.com ([156.154.24.138]) by chiedprmail1.ietf.org with esmtp (Exim 4.43) id 1GDnsu-0006Bk-Di; Thu, 17 Aug 2006 15:50:35 -0400 Received: from stiedprstage1.ietf.org (stiedprstage1.va.neustar.com [10.31.47.10]) by ns3.neustar.com (Postfix) with ESMTP id 1571817591; Thu, 17 Aug 2006 19:50:02 +0000 (GMT) Received: from ietf by stiedprstage1.ietf.org with local (Exim 4.43) id 1GDnsP-0007fZ-Ln; Thu, 17 Aug 2006 15:50:01 -0400 Content-Type: Multipart/Mixed; Boundary="NextPart" Mime-Version: 1.0 To: i-d-announce@ietf.org From: Internet-Drafts@ietf.org Message-Id: Date: Thu, 17 Aug 2006 15:50:01 -0400 X-Spam-Score: -5.9 (-----) X-Scan-Signature: c3a18ef96977fc9bcc21a621cbf1174b Cc: sipping@ietf.org Subject: [Sipping] I-D ACTION:draft-ietf-sipping-toip-06.txt X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org --NextPart A New Internet-Draft is available from the on-line Internet-Drafts directories. This draft is a work item of the Session Initiation Proposal Investigation Working Group of the IETF. Title : Framework for real-time text over IP using the Session Initiation Protocol (SIP) Author(s) : A. Van Wijk, G. Gybels Filename : draft-ietf-sipping-toip-06.txt Pages : 27 Date : 2006-8-17 This document lists the essential requirements for real-time Text- over-IP (ToIP) and defines a framework for implementation of all required functions based on the Session Initiation Protocol (SIP) and the Real-Time Transport Protocol (RTP). This includes interworking between Text-over-IP and existing text telephony on the PSTN and other networks. A URL for this Internet-Draft is: http://www.ietf.org/internet-drafts/draft-ietf-sipping-toip-06.txt To remove yourself from the I-D Announcement list, send a message to i-d-announce-request@ietf.org with the word unsubscribe in the body of the message. You can also visit https://www1.ietf.org/mailman/listinfo/I-D-announce to change your subscription settings. Internet-Drafts are also available by anonymous FTP. Login with the username "anonymous" and a password of your e-mail address. After logging in, type "cd internet-drafts" and then "get draft-ietf-sipping-toip-06.txt". A list of Internet-Drafts directories can be found in http://www.ietf.org/shadow.html or ftp://ftp.ietf.org/ietf/1shadow-sites.txt Internet-Drafts can also be obtained by e-mail. Send a message to: mailserv@ietf.org. In the body type: "FILE /internet-drafts/draft-ietf-sipping-toip-06.txt". NOTE: The mail server at ietf.org can return the document in MIME-encoded form by using the "mpack" utility. To use this feature, insert the command "ENCODING mime" before the "FILE" command. To decode the response(s), you will need "munpack" or a MIME-compliant mail reader. Different MIME-compliant mail readers exhibit different behavior, especially when dealing with "multipart" MIME messages (i.e. documents which have been split up into multiple messages), so check your local documentation on how to manipulate these messages. Below is the data which will enable a MIME compliant mail reader implementation to automatically retrieve the ASCII version of the Internet-Draft. --NextPart Content-Type: Multipart/Alternative; Boundary="OtherAccess" --OtherAccess Content-Type: Message/External-body; access-type="mail-server"; server="mailserv@ietf.org" Content-Type: text/plain Content-ID: <2006-8-17134336.I-D@ietf.org> ENCODING mime FILE /internet-drafts/draft-ietf-sipping-toip-06.txt --OtherAccess Content-Type: Message/External-body; name="draft-ietf-sipping-toip-06.txt"; site="ftp.ietf.org"; access-type="anon-ftp"; directory="internet-drafts" Content-Type: text/plain Content-ID: <2006-8-17134336.I-D@ietf.org> --OtherAccess-- --NextPart Content-Type: text/plain; charset="us-ascii" MIME-Version: 1.0 Content-Transfer-Encoding: 7bit Content-Disposition: inline _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP --NextPart-- From sipping-bounces@ietf.org Thu Aug 17 19:19:26 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GDr7d-0003pc-06; Thu, 17 Aug 2006 19:17:57 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GDr7c-0003pN-0Z for sipping@ietf.org; Thu, 17 Aug 2006 19:17:56 -0400 Received: from hme1.july.broomfield1.level3.net ([209.245.18.8] helo=f4bb49-05.idc1.level3.com) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GDr7a-0007ut-NL for sipping@ietf.org; Thu, 17 Aug 2006 19:17:55 -0400 Received: from montag.eng.level3.com (montag.eng.l3.com [10.1.68.57]) by f4bb49-05.idc1.level3.com (8.12.10/8.12.10) with ESMTP id k7HNHqUX020944 for ; Thu, 17 Aug 2006 23:17:53 GMT Subject: [Sipping] Updated Draft: SIP Performance Metrics From: Daryl Malas To: sipping@ietf.org Content-Type: text/plain Date: Thu, 17 Aug 2006 17:16:17 -0600 Message-Id: <1155856577.2258.22.camel@montag.eng.level3.com> Mime-Version: 1.0 X-Mailer: Evolution 2.6.0 (2.6.0-1) Content-Transfer-Encoding: 7bit X-Spam-Score: 0.1 (/) X-Scan-Signature: 0bc60ec82efc80c84b8d02f4b0e4de22 X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org To all, I have submitted an updated version of the SIP Performance Metrics draft. I renamed the title to SIP End-to-End Performance Metrics for clarity of scope purposes. The following lists some of the changes to this draft: 1. Clarified scope of the document (see introduction and throughout). 2. Defined new terms such as end-to-end. 3. Removed Average Hops per Session. 4. Completely re-did Average Hops per INVITE to clarify. 5. Updated all flows to be compliant with RFC 3665. 6. Updated terms to be compliant with RFC 3261. 7. I described the metrics for both successful and failed situations. 8. I added flows for redirect scenarios. 9. I added verbiage regarding testing. 10. I added the Registration Request Delay (RRD) metric 11. Finally, I clarified a lot of details throughout the document, trying to alleviate confusion. The draft can be found here: http://www.ietf.org/internet-drafts/draft-malas-performance-metrics-04.txt I would really appreciate feedback and comments on the draft. Thanks...--Daryl _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Thu Aug 17 20:19:55 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GDs4S-0000SM-73; Thu, 17 Aug 2006 20:18:44 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GDs4R-0000Ph-E3 for sipping@ietf.org; Thu, 17 Aug 2006 20:18:43 -0400 Received: from stsc1260-eth-s1-s1p1-vip.va.neustar.com ([156.154.16.129] helo=chiedprmail1.ietf.org) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GDqZD-0003Ng-Fu for sipping@ietf.org; Thu, 17 Aug 2006 18:42:23 -0400 Received: from zrtps0kp.nortel.com ([47.140.192.56]) by chiedprmail1.ietf.org with esmtp (Exim 4.43) id 1GDqOs-0001BQ-5W for sipping@ietf.org; Thu, 17 Aug 2006 18:31:44 -0400 Received: from zrc2hxm1.corp.nortel.com (zrc2hxm1.corp.nortel.com [47.103.123.72]) by zrtps0kp.nortel.com (Switch-2.2.6/Switch-2.2.0) with ESMTP id k7HMVbC05347 for ; Thu, 17 Aug 2006 18:31:38 -0400 (EDT) X-MimeOLE: Produced By Microsoft Exchange V6.5 Content-class: urn:content-classes:message MIME-Version: 1.0 Content-Type: text/plain; charset="US-ASCII" Content-Transfer-Encoding: quoted-printable Date: Thu, 17 Aug 2006 17:31:35 -0500 Message-ID: In-Reply-To: X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: Ready for IESG review?: draft-ietf-sipping-toip-06.txt Thread-Index: AcbCPqMZsf7qrNxhRtO74z1eKlxsIgADJgrg From: "Mary Barnes" To: X-Spam-Score: -2.5 (--) X-Scan-Signature: 10d3e4e3c32e363f129e380e644649be Subject: [Sipping] Ready for IESG review?: draft-ietf-sipping-toip-06.txt X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Hi all, After posting this question about 7 weeks ago, there was a lively discussion that resulted in agreement on additional editorial changes to this document. The proposed changes were summarized on 7 July 2006:=20 http://www1.ietf.org/mail-archive/web/sipping/current/msg11420.html There was one additional change in section 6.2.5.1, related to the discussion of the "txp" media content attribute in the MMUSIC WG, it was decided to remove the explicit reference in this document.=20 The -06 version contains all those updates, with the diff available at: http://tools.ietf.org/wg/sipping/draft-ietf-sipping-toip/draft-ietf-sipp ing-toip-06-from-05.diff.html We'd like to give folks until 25 Aug 2006 to identify any additional concerns on the readiness of this document.=20 Regards, Mary SIPPING WG co-chair -----Original Message----- From: Internet-Drafts@ietf.org [mailto:Internet-Drafts@ietf.org]=20 Sent: Thursday, August 17, 2006 2:50 PM To: i-d-announce@ietf.org Cc: sipping@ietf.org Subject: I-D ACTION:draft-ietf-sipping-toip-06.txt A New Internet-Draft is available from the on-line Internet-Drafts=20 directories. This draft is a work item of the Session Initiation Proposal Investigation Working Group of the IETF. Title : Framework for real-time text over IP using the Session Initiation Protocol (SIP) Author(s) : A. Van Wijk, G. Gybels Filename : draft-ietf-sipping-toip-06.txt Pages : 27 Date : 2006-8-17 =09 This document lists the essential requirements for real-time Text- over-IP (ToIP) and defines a framework for implementation of all required functions based on the Session Initiation Protocol (SIP) and the Real-Time Transport Protocol (RTP). This includes interworking between Text-over-IP and existing text telephony on the PSTN and other networks. A URL for this Internet-Draft is: http://www.ietf.org/internet-drafts/draft-ietf-sipping-toip-06.txt To remove yourself from the I-D Announcement list, send a message to=20 i-d-announce-request@ietf.org with the word unsubscribe in the body of=20 the message.=20 You can also visit https://www1.ietf.org/mailman/listinfo/I-D-announce=20 to change your subscription settings. Internet-Drafts are also available by anonymous FTP. Login with the=20 username "anonymous" and a password of your e-mail address. After=20 logging in, type "cd internet-drafts" and then=20 "get draft-ietf-sipping-toip-06.txt". A list of Internet-Drafts directories can be found in http://www.ietf.org/shadow.html=20 or ftp://ftp.ietf.org/ietf/1shadow-sites.txt Internet-Drafts can also be obtained by e-mail. Send a message to: mailserv@ietf.org. In the body type: "FILE /internet-drafts/draft-ietf-sipping-toip-06.txt". =09 NOTE: The mail server at ietf.org can return the document in MIME-encoded form by using the "mpack" utility. To use this feature, insert the command "ENCODING mime" before the "FILE" command. To decode the response(s), you will need "munpack" or a MIME-compliant mail reader. Different MIME-compliant mail readers exhibit different behavior, especially when dealing with "multipart" MIME messages (i.e. documents which have been split up into multiple messages), so check your local documentation on how to manipulate these messages. Below is the data which will enable a MIME compliant mail reader implementation to automatically retrieve the ASCII version of the Internet-Draft. _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Fri Aug 18 02:29:07 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GDxpG-0000Xt-A9; Fri, 18 Aug 2006 02:27:26 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GDxpE-0000Rd-EO for sipping@ietf.org; Fri, 18 Aug 2006 02:27:24 -0400 Received: from sonussf1.sonusnet.com ([208.45.178.26]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GDxpE-0008Qo-3O for sipping@ietf.org; Fri, 18 Aug 2006 02:27:24 -0400 Received: from sonusmail01.sonusnet.com (sonusmail01.sonusnet.com [10.128.32.75]) by sonussf1.sonusnet.com (8.13.3/8.13.3) with ESMTP id k7I6RMjC007701; Fri, 18 Aug 2006 02:27:23 -0400 Received: from SONUSINMAIL01.sonusnet.com ([10.128.254.7]) by sonusmail01.sonusnet.com with Microsoft SMTPSVC(5.0.2195.6713); Fri, 18 Aug 2006 02:27:22 -0400 X-MimeOLE: Produced By Microsoft Exchange V6.0.6603.0 content-class: urn:content-classes:message MIME-Version: 1.0 Date: Fri, 18 Aug 2006 11:57:18 +0530 Message-ID: <70798CF421F00F4DA018059E5B7EEB8C7A5FD8@sonusinmail01.sonusnet.com> X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: [Sip] comments on draft-ietf-sipping-dialogusage-02.txt Thread-Index: AcbB/O3w2bO5pUKoREK2Hq0XQ6js2QAke1iA From: "B, Nataraju" To: "Paul Kyzivat" X-OriginalArrivalTime: 18 Aug 2006 06:27:22.0826 (UTC) FILETIME=[5D4FCAA0:01C6C28F] X-Spam-Score: 0.0 (/) X-Scan-Signature: cf3becbbd6d1a45acbe2ffd4ab88bdc2 Cc: sipping@ietf.org Subject: [Sipping] RE: [Sip] comments on 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YXBwbGljYXRpb24gb2Ygc2lwDQoNCg0K --===============0578228680== Content-Type: text/plain; charset="us-ascii" MIME-Version: 1.0 Content-Transfer-Encoding: 7bit Content-Disposition: inline _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP --===============0578228680==-- From sipping-bounces@ietf.org Fri Aug 18 03:22:31 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GDygG-0007aX-T9; Fri, 18 Aug 2006 03:22:12 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GDygG-0007aS-AS for sipping@ietf.org; Fri, 18 Aug 2006 03:22:12 -0400 Received: from mx1.shearar.net ([72.21.63.189]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GDygF-0005F5-3u for sipping@ietf.org; Fri, 18 Aug 2006 03:22:12 -0400 Received: from [196.209.116.126] (helo=wsfrank) by mx1.shearar.net with smtp (Exim 4.52) id 1GDyhN-0002eL-TO for sipping@ietf.org; Fri, 18 Aug 2006 09:23:25 +0200 Message-ID: <006701c6c297$02b8e600$0600000a@wsfrank> From: "Frank Shearar" To: Date: Fri, 18 Aug 2006 09:21:56 +0200 MIME-Version: 1.0 Content-Type: text/plain; charset="iso-8859-1" Content-Transfer-Encoding: 7bit X-Priority: 3 X-MSMail-Priority: Normal X-Mailer: Microsoft Outlook Express 6.00.2800.1807 X-MimeOLE: Produced By Microsoft MimeOLE V6.00.2800.1807 X-Spam-Score: -102.5 X-Spam-Score: 0.0 (/) X-Scan-Signature: 798b2e660f1819ae38035ac1d8d5e3ab Subject: [Sipping] Minor grammatical nits in draft-ietf-sipping-toip-06 X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Section 5.2.3.2: They are a) "No-gain" codec solution, b) the Cellular Text Telephony Modem (CTM) solution [8] both collectively called "Baudot mode" solution in the USA. should read "They are the a) ...., and b) ..." Section 6.2.5.1: This means that the other user may be used to formal turn taking during the call. should maybe read This means that the textphone user may be used to formal turn taking during the call. Looks good! frank _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Fri Aug 18 09:13:25 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GE49n-0004Lm-CZ; Fri, 18 Aug 2006 09:13:03 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GE49l-0004Lh-QV for sipping@ietf.org; Fri, 18 Aug 2006 09:13:01 -0400 Received: from rtp-iport-1.cisco.com ([64.102.122.148]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GE49j-0000va-BV for sipping@ietf.org; Fri, 18 Aug 2006 09:13:01 -0400 Received: from rtp-dkim-1.cisco.com ([64.102.121.158]) by rtp-iport-1.cisco.com with ESMTP; 18 Aug 2006 06:12:59 -0700 X-BrightmailFiltered: true X-Brightmail-Tracker: AAAAAA== X-IronPort-AV: i="4.08,143,1154934000"; d="scan'208"; a="36879720:sNHT27120270" Received: from rtp-core-2.cisco.com (rtp-core-2.cisco.com [64.102.124.13]) by rtp-dkim-1.cisco.com (8.12.11.20060308/8.12.11) with ESMTP id k7IDCwjL003530; Fri, 18 Aug 2006 09:12:58 -0400 Received: from xbh-rtp-211.amer.cisco.com (xbh-rtp-211.cisco.com [64.102.31.102]) by rtp-core-2.cisco.com (8.12.10/8.12.6) with ESMTP id k7IDCwuI020871; Fri, 18 Aug 2006 09:12:58 -0400 (EDT) Received: from xfe-rtp-201.amer.cisco.com ([64.102.31.38]) by xbh-rtp-211.amer.cisco.com with Microsoft SMTPSVC(6.0.3790.1830); Fri, 18 Aug 2006 09:12:57 -0400 Received: from [161.44.79.104] ([161.44.79.104]) by xfe-rtp-201.amer.cisco.com with Microsoft SMTPSVC(6.0.3790.1830); Fri, 18 Aug 2006 09:12:57 -0400 Message-ID: <44E5BCD9.9050600@cisco.com> Date: Fri, 18 Aug 2006 09:12:57 -0400 From: Paul Kyzivat User-Agent: Thunderbird 1.5.0.5 (Windows/20060719) MIME-Version: 1.0 To: "B, Nataraju" References: <70798CF421F00F4DA018059E5B7EEB8C7A5FD8@sonusinmail01.sonusnet.com> In-Reply-To: <70798CF421F00F4DA018059E5B7EEB8C7A5FD8@sonusinmail01.sonusnet.com> Content-Type: text/plain; charset=UTF-8; format=flowed Content-Transfer-Encoding: 8bit X-OriginalArrivalTime: 18 Aug 2006 13:12:57.0906 (UTC) FILETIME=[06265120:01C6C2C8] DKIM-Signature: a=rsa-sha1; q=dns; l=6646; t=1155906778; x=1156770778; c=relaxed/simple; s=rtpdkim1001; h=Content-Type:From:Subject:Content-Transfer-Encoding:MIME-Version; d=cisco.com; i=pkyzivat@cisco.com; z=From:Paul=20Kyzivat=20 |Subject:Re=3A=20[Sip]=20comments=20on=20draft-ietf-sipping-dialogusage-02.txt |To:=22B,=20Nataraju=22=20; X=v=3Dcisco.com=3B=20h=3DTH6BmAmf0rRE0cKOJxwAXgTnrbA=3D; b=H06DAAf4Hf0IRQoFMpfz11Joqx6o/+s77Me0ssZdkjpPHad1v/OuOZxfyaZjj+Ys2q5YpuXU 6xCIyeBv1enlVAFcSBp28RYR+NB6y8RgQOmMboeIY9dymfBxI9PPr8Cc; Authentication-Results: rtp-dkim-1.cisco.com; header.From=pkyzivat@cisco.com; dkim=pass ( sig from cisco.com verified; ); X-Spam-Score: 0.0 (/) X-Scan-Signature: a92270ba83d7ead10c5001bb42ec3221 Cc: sipping@ietf.org Subject: [Sipping] Re: [Sip] comments on draft-ietf-sipping-dialogusage-02.txt X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org B, Nataraju wrote: > I was thinking making it explicit would be better. > > At least shall we mention this one as normative information? > Otherwise many more others might think the same way (which I thought earlier) and develop the buggy software... Well, I *thought* it was clear in 3261. But I went back to check, and it is far from clear. Section 8.2 says: UASs SHOULD process the requests in the order of the steps that follow in this section (that is, starting with authentication, then inspecting the method, the header fields, and so on throughout the remainder of this section). which certainly implies that authentication should be done first. Unfortunately, the subsections that follow this intro don't mention authentication at all. :-( And, this section pertains to requests outside a dialog, so it doesn't really apply anyway. Section 12, which defines behavior for dialog establishment and requests within dialogs doesn't mention authentication or authorization at all. So I guess it isn't at all clear that that if 401 is applicable to a request then it should be issued in preference to 404, 410, or 416, and that this should be true inside a dialog as well as outside. It would be good to get some input from others on this. Paul > Thanks, > Nataraju A B > >> -----Original Message----- >> From: Paul Kyzivat [mailto:pkyzivat@cisco.com] >> Sent: Thursday, August 17, 2006 6:29 PM >> To: B, Nataraju >> Cc: sipping@ietf.org >> Subject: Re: [Sip] comments on draft-ietf-sipping-dialogusage-02.txt >> >> >> >> B, Nataraju wrote: >>> Paul, >>> >>> What I meant to mention here was that, in the scenarios which might lead >> to these responses must be authenticated before we call for this >> Reponses... >>> I mean if we receive a NOTIFY(un-authenticated) 404 should not be >> sent out which will terminate the dialog/usage. >>> Authenticate the NTFY request first then only it could be >> rejected with 404 or any other appropriate replies which would result in >> dialog termination. >> >> If I understand you, this is just standard procedure, applicable >> regardless of dialog usages. Authentication is supposed to be done >> first, so no extra information is given to unauthenticated callers. >> >> Paul >> >>> Thanks, >>> Nataraju A B >>> >>>> -----Original Message----- >>>> From: Paul Kyzivat [mailto:pkyzivat@cisco.com] >>>> Sent: Monday, August 14, 2006 7:34 PM >>>> To: B, Nataraju >>>> Cc: sip@ietf.org >>>> Subject: Re: [Sip] comments on draft-ietf-sipping-dialogusage-02.txt >>>> >>>> >>>> >>>> B, Nataraju wrote: >>>> >>>>> 1. Sec 4 describes what the effects on dialog are and its usages when >>>>> particular responses are received for NOTIFY message, the responses >> 404, >>>>> 410, 416, lead to dialog and its usage termination. >>>>> >>>>> a. I feel we should have mentioned ONLY the authenticated NOTIFY >>>>> requests will have this effect on the dialogs/usages. Otherwise any >>>>> middle men can trace the message and send the NOTIFY messages which >>>>> terminate the dialogs, which is undesirable ☹. If the entity sending >>>>> NOTIFY really intend to do that it can resend NOTIFY with >> authorization >>>>> credentials which will help to terminate the associated dialogs. >>>> I agree we wouldn't want to enable a MITM attack here. >>>> >>>> But I think that is already covered. If the request isn't properly >>>> authenticated, then the response will be 401, not 404, 410, or 416. >>>> >>>> Paul >>> >>> >>> ------------------------------------------------------------------------ >>> >>> Subject: >>> [Sip] comments on draft-ietf-sipping-dialogusage-02.txt >>> From: >>> "B, Nataraju" >>> Date: >>> Mon, 14 Aug 2006 12:04:19 +0530 >>> To: >>> >>> >>> To: >>> >>> >>> >>> Hi Sparks, >>> >>> >>> >>> I had gone through the document, here are few comments. >>> >>> >>> >>> Note: I assume the behavior mentioned in sec 4 applies both at client >> receiving these responses (Notifier) and server generating these replies >> (Subscriber). If this was meant to be only @ sender of NOTIFY shall we >> need to mention the similar set of actions @ Subscriber side too? >>> >>> >>> 1. Sec 4 describes what the effects on dialog are and its usages when >> particular responses are received for NOTIFY message, the responses 404, >> 410, 416, lead to dialog and its usage termination. >>> a. I feel we should have mentioned ONLY the authenticated NOTIFY >> requests will have this effect on the dialogs/usages. Otherwise any middle >> men can trace the message and send the NOTIFY messages which terminate the >> dialogs, which is undesirable ☹. If the entity sending NOTIFY really >> intend to do that it can resend NOTIFY with authorization credentials >> which will help to terminate the associated dialogs. >>> >>> >>> b. Other alternative would be that let those responses have no >> effect on the dialog/usages, eventually the subscription duration would >> take care of dialog termination or the actual entity would terminate or >> extend the subscription. Any way the request from actual notifier would >> have the proper URIs or credentials which will never lead to these >> responses. >>> >>> >>> The same logic would need to extend for replies 481, 483, 484, 489 and >> 604 also. >>> >>> >>> 2. Sec 4 of the draft >>> >>> >>> >>> <<< 423 Interval Too Brief: This response won't happen in our >> example >>> scenario, but if it came in response to a reSUBSCRIBE, the >>> >>> subscribe usage is not destroyed (or otherwise affected). No >>> >>> other usages of the dialog are affected. >>> >>> >>> >>> >>> In the above Para the meaning of last part of 1st sentence (I >> mean, the subscribe usage is not destroyed (or otherwise affected)) and >> 2nd sentence is not clear, both mention the same meaning? Can we make it >> explicit about what we really mean here ? >>> >>> >>> Thanks, >>> >>> Nataraju A B >>> >>> Sonus Networks India Pvt Ltd >>> >>> >>> >>> ------------------------------------------------------------------------ >>> >>> _______________________________________________ >>> Sip mailing list https://www1.ietf.org/mailman/listinfo/sip >>> This list is for NEW development of the core SIP Protocol >>> Use sip-implementors@cs.columbia.edu for questions on current sip >>> Use sipping@ietf.org for new developments on the application of sip > > _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Sat Aug 19 23:36:50 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GEe3C-0005k7-NP; Sat, 19 Aug 2006 23:32:38 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GEe3B-0005jz-TA for sipping@ietf.org; Sat, 19 Aug 2006 23:32:37 -0400 Received: from rvil-eframer.radvision.com ([80.74.106.104] helo=eframer.radvision.com) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GEe39-00029K-PJ for sipping@ietf.org; Sat, 19 Aug 2006 23:32:37 -0400 Received: (from root@localhost) by eframer.radvision.com (8.13.4/8.12.9) id k7K3bu0j012634 for sipping@ietf.org; Sun, 20 Aug 2006 06:37:56 +0300 Received: from rvil-mail1.RADVISION.com (rvil-mail1.radvision.com [172.20.2.100]) by eframer.radvision.com (8.13.4/8.12.9) with ESMTP id k7K3btA2012628 for ; Sun, 20 Aug 2006 06:37:55 +0300 x-mimeole: Produced By Microsoft Exchange V6.5.7226.0 Content-class: urn:content-classes:message MIME-Version: 1.0 Date: Sun, 20 Aug 2006 06:32:22 +0300 Message-ID: X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: SDP fmtp video negotiation Thread-Index: Aca7hWc1VCvzL/ZeSWy9gyp0vEKoYgDC5BBAAV4LM2A= From: "Erez Morabia" To: X-Spam-Score: 0.0 (/) X-Scan-Signature: 6b519fb0ef66258f34533f52ff46aedf Subject: [Sipping] SDP fmtp video negotiation X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Content-Type: multipart/mixed; boundary="===============1237231808==" Errors-To: sipping-bounces@ietf.org This is a multi-part message in MIME format. --===============1237231808== Content-class: urn:content-classes:message Content-Type: multipart/alternative; boundary="----_=_NextPart_001_01C6C409.4DA4A945" This is a multi-part message in MIME format. ------_=_NextPart_001_01C6C409.4DA4A945 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: quoted-printable Hi, I'm trying to find here a common base for the negotiation regarding the different codecs on the video media side. It seems that when you advertise fmtp, you declare what you are willing to receive (decoder capabilities) and implicitly what you prefer to send (except for some H264 exceptions). The thumb rule is that when receiving an offer, and there is intersection between the offerer and offeree capabilities, then video channels can be opened in both directions.=20 Is my observation correct? =20 Example: Alice supports QCIF(MPI=3D2) Maximum-Bit=3DRate=3D256Kbps Bob supports CIF(MPI=3D1) QCIF(MPI=3D1) Maximum-Bit=3DRate=3D128Kbps =20 Scenario 1: Alice offers: QCIF(MPI=3D2) Maximum-Bit=3DRate=3D256Kbps Bob answers: QCIF(MPI=3D2) Maximum-Bit=3DRate=3D128Kbps (or CIF(MPI=3D1) QCIF(MPI=3D1) Maximum-Bit=3DRate=3D128Kbps). Is that a correct negotiation? =20 Scenario 2: Bob offers CIF(MPI=3D1) QCIF(MPI=3D1) Maximum-Bit=3DRate=3D128Kbps. Alice answers: QCIF(MPI=3D2) Maximum-Bit=3DRate=3D128Kbps (or = QCIF(MPI=3D2) Maximum-Bit=3DRate=3D256Kbps). Is that a correct negotiation? =20 In both scenarios, both incoming and outgoing channels should be opened correctly with QCIF(MPI=3D2) Maximum-Bit=3DRate=3D128Kbps. =20 BTW The drafts/RFCs I'm using are: H.261: draft-ietf-avt-rfc2032-bis-13.txt H.263: draft-koskelainen-sdp263-01.txt (I'm not sure about that one, does the H.263+ draft overrides it?) H.263+: draft-ietf-avt-rfc2429-bis-09.txt H.264: RFC 3984 =20 Thanks, Erez =20 ------_=_NextPart_001_01C6C409.4DA4A945 Content-Type: text/html; charset="us-ascii" Content-Transfer-Encoding: quoted-printable

Hi,

I’m trying to find here a common base for the negotiation regarding the different codecs on the video media = side.

It seems that when you advertise fmtp, you declare = what you are willing to receive (decoder capabilities) and implicitly what you = prefer to send (except for some H264 exceptions).

The thumb rule is that when receiving an offer, and = there is intersection between the offerer and offeree capabilities, then video = channels can be opened in both directions.

Is my observation = correct?

 

Example:

Alice supports QCIF(MPI=3D2) Maximum-Bit=3DRate=3D256Kbps

Bob supports CIF(MPI=3D1) QCIF(MPI=3D1) = Maximum-Bit=3DRate=3D128Kbps

 

Scenario 1:

Alice offers: QCIF(MPI=3D2) Maximum-Bit=3DRate=3D256Kbps

Bob answers: QCIF(MPI=3D2) = Maximum-Bit=3DRate=3D128Kbps (or CIF(MPI=3D1) QCIF(MPI=3D1) = Maximum-Bit=3DRate=3D128Kbps).

Is that a correct = negotiation?

 

Scenario 2:

Bob offers CIF(MPI=3D1) QCIF(MPI=3D1) = Maximum-Bit=3DRate=3D128Kbps.

Alice answers: QCIF(MPI=3D2) Maximum-Bit=3DRate=3D128Kbps (or QCIF(MPI=3D2) = Maximum-Bit=3DRate=3D256Kbps).

Is that a correct = negotiation?

 

In both scenarios, both incoming and outgoing = channels should be opened correctly with QCIF(MPI=3D2) = Maximum-Bit=3DRate=3D128Kbps.

 

BTW

The drafts/RFCs I’m using = are:

H.261: = draft-ietf-avt-rfc2032-bis-13.txt

H.263: draft-koskelainen-sdp263-01.txt (I’m not = sure about that one, does the H.263+ draft overrides = it?)

H.263+: = draft-ietf-avt-rfc2429-bis-09.txt

H.264: RFC 3984

 

Thanks,

Erez

 

------_=_NextPart_001_01C6C409.4DA4A945-- --===============1237231808== Content-Type: text/plain; charset="us-ascii" MIME-Version: 1.0 Content-Transfer-Encoding: 7bit Content-Disposition: inline _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP --===============1237231808==-- From sipping-bounces@ietf.org Mon Aug 21 17:14:46 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GFH5f-0007gd-1b; Mon, 21 Aug 2006 17:13:47 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GFH5d-0007gR-Eh for sipping@ietf.org; Mon, 21 Aug 2006 17:13:45 -0400 Received: from nat.stoke.com ([209.78.40.100] helo=trio.stoke.com) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GFH5b-0004Dv-Ra for sipping@ietf.org; Mon, 21 Aug 2006 17:13:45 -0400 Received: from us.stoke.com (minsk.us.stoke.com [172.16.4.20]) by trio.stoke.com (8.13.3/8.13.3) with ESMTP id k7LLDhAr033389 for ; Mon, 21 Aug 2006 14:13:43 -0700 (PDT) (envelope-from yafan@stoke.com) X-MimeOLE: Produced By Microsoft Exchange V6.5.7226.0 Content-class: urn:content-classes:message MIME-Version: 1.0 Date: Mon, 21 Aug 2006 14:15:32 -0700 Message-ID: <23EECEC9B06584478B1C9E38C253D35F74ACF4@minsk.us.stoke.com> X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: Fixed Mobile Convergence (draft-yafan-fmc-arch-00.txt) Thread-Index: AcbFZq+8XXo8hqTMRregC7/hP3Ig9gAAAPvw From: "Yafan An" To: X-Virus-Scanned: ClamAV 0.88/1702/Mon Aug 21 08:23:21 2006 on trio.stoke.com X-Virus-Status: Clean X-Spam-Score: 0.0 (/) X-Scan-Signature: ccfb4541e989aa743998098cd315d0fd Subject: [Sipping] Fixed Mobile Convergence (draft-yafan-fmc-arch-00.txt) X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Content-Type: multipart/mixed; boundary="===============0697764581==" Errors-To: sipping-bounces@ietf.org This is a multi-part message in MIME format. --===============0697764581== Content-class: urn:content-classes:message Content-Type: multipart/alternative; boundary="----_=_NextPart_001_01C6C566.EFB5ADF5" This is a multi-part message in MIME format. ------_=_NextPart_001_01C6C566.EFB5ADF5 Content-Type: text/plain; charset="US-ASCII" Content-Transfer-Encoding: quoted-printable Dear Sipping Enthusiasts: If you have not noticed the ietf drafts, I'd like to invite you to read and comment on the submitted draft. You can send your comments to me or the sipping list. Thank you. Yafan Title : An Architecture Framework for Fixed Mobile Convergence Using SIP as Access Call Control Protocol Author(s) : Y. An Filename : draft-yafan-fmc-arch-00.txt Pages : 32 Date : 2006-7-28 =09 This document describes an architecture framework for achieving converged mobile services across alternative radio access networks (GAN, or Generic Access Network), such as WLAN, WiMax, and the cellular network, while utilizing SIP for media session control between the dual mode handset (DMH) and a network SIP proxy. The key benefits of this architecture include (1) fast and flexible service deployment in the IP domain, and (2) allow seamless voice call continuity services across IP and circuit-switched cellular domains. A URL for this Internet-Draft is: http://www.ietf.org/internet-drafts/draft-yafan-fmc-arch-00.txt YAFAN AN - CHIEF ARCHITCET=20 T: 408.855.2957 M: 510.386.2647 F: 408.855.2978 =20 SERVICES WITHOUT BOUNDARIES =09 ------_=_NextPart_001_01C6C566.EFB5ADF5 Content-Type: text/html; charset="US-ASCII" Content-Transfer-Encoding: quoted-printable Fixed Mobile Convergence (draft-yafan-fmc-arch-00.txt)

Dear = Sipping Enthusiasts:

If = you have not noticed the ietf drafts, Id like to = invite you to read and comment on the submitted draft. You can send your = comments to me or the sipping list.

Thank = you.

Yafan


        Title   =         : An Architecture = Framework for Fixed Mobile Convergence Using SIP as = Access Call Control Protocol

        Author(s)       : Y. = An

        Filename        = : draft-yafan-fmc-arch-00.txt

        Pages   =         : 32

        Date    =         : 2006-7-28

       

This document describes an architecture framework for = achieving

converged mobile services across alternative radio access = networks

(GAN, or Generic Access Network), such as WLAN, WiMax, and = the

cellular network, while utilizing SIP for media session = control

between the dual mode handset (DMH) and a network SIP proxy.  The key

benefits of this architecture include (1) fast and flexible = service

deployment in the IP domain, and (2) allow seamless voice = call

continuity services across IP and circuit-switched cellular = domains.


A URL for this Internet-Draft is:

= http://www.ietf.org/internet-drafts/draft-yafan-fmc-arch-00.txt


        YAFAN AN CHIEF ARCHITCET
T: 408.855.2957  M: 510.386.2647  F: = 408.855.2978  
SERVICES = WITHOUT BOUNDARIES    

------_=_NextPart_001_01C6C566.EFB5ADF5-- --===============0697764581== Content-Type: text/plain; charset="us-ascii" MIME-Version: 1.0 Content-Transfer-Encoding: 7bit Content-Disposition: inline _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP --===============0697764581==-- From sipping-bounces@ietf.org Mon Aug 21 20:37:22 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GFKGA-0006me-5q; Mon, 21 Aug 2006 20:36:50 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GFKG8-0006mZ-PN for sipping@ietf.org; Mon, 21 Aug 2006 20:36:48 -0400 Received: from stsc1260-eth-s1-s1p1-vip.va.neustar.com ([156.154.16.129] helo=chiedprmail1.ietf.org) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GFHFn-0007dv-1n for sipping@ietf.org; Mon, 21 Aug 2006 17:24:15 -0400 Received: from nat.stoke.com ([209.78.40.100] helo=trio.stoke.com) by chiedprmail1.ietf.org with esmtp (Exim 4.43) id 1GFH49-0002o7-Ed for sipping@ietf.org; Mon, 21 Aug 2006 17:12:15 -0400 Received: from us.stoke.com (minsk.us.stoke.com [172.16.4.20]) by trio.stoke.com (8.13.3/8.13.3) with ESMTP id k7LLBuTS033371 for ; Mon, 21 Aug 2006 14:12:02 -0700 (PDT) (envelope-from yafan@stoke.com) X-MimeOLE: Produced By Microsoft Exchange V6.5.7226.0 Content-class: urn:content-classes:message MIME-Version: 1.0 Date: Mon, 21 Aug 2006 14:13:45 -0700 Message-ID: <23EECEC9B06584478B1C9E38C253D35F74ACF2@minsk.us.stoke.com> X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: Fixed Mobile Convergence (draft-yafan-fmc-mancho-00.txt) Thread-Index: AcbFZq+8XXo8hqTMRregC7/hP3Ig9g== From: "Yafan An" To: X-Virus-Scanned: ClamAV 0.88/1702/Mon Aug 21 08:23:21 2006 on trio.stoke.com X-Virus-Status: Clean X-Spam-Score: -2.6 (--) X-Scan-Signature: 0bb031f3a6fb29f760794ac9bf1997ae Subject: [Sipping] Fixed Mobile Convergence (draft-yafan-fmc-mancho-00.txt) X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Content-Type: multipart/mixed; boundary="===============0146759010==" Errors-To: sipping-bounces@ietf.org This is a multi-part message in MIME format. --===============0146759010== Content-class: urn:content-classes:message Content-Type: multipart/alternative; boundary="----_=_NextPart_001_01C6C566.B00DC507" This is a multi-part message in MIME format. ------_=_NextPart_001_01C6C566.B00DC507 Content-Type: text/plain; charset="US-ASCII" Content-Transfer-Encoding: quoted-printable Dear Sipping Enthusiasts: If you have not noticed the ietf drafts, I'd like to invite you to read and comment on the submitted draft. You can send your comments to me or the sipping list. Thank you. Yafan Title : Mobile Assisted Handover across VoIP and Cellular Domains Using SIP as Access Call Control Author(s) : Y. An Filename : draft-yafan-fmc-mancho-00.txt Pages : 61 Date : 2006-7-31 =09 This document describes the use of SIP between the dual mode handset (DMH) and the mobility SIP proxy to enable network controlled handovers across voice over IP (VoIP) and cellular networks. The mobility SIP proxy has a peering relationship with the cellular network. Mobile assisted and network controlled handovers require the exchange of cellular radio parameters between the DMH and the mobility SIP proxy. The set of cellular related parameters is collectively called Session Handover Protocol (SHP). The SHP is a residual protocol of SIP, where all of its messages are transported as MIME attachment to SIP messages. This document describes the requirements, various SIP message flows for cross-domain roaming and bidirectional handovers between VoIP and cellular domains, and detailed specification of the residual SHP protocol. A URL for this Internet-Draft is: http://www.ietf.org/internet-drafts/draft-yafan-fmc-mancho-00.txt YAFAN AN - CHIEF ARCHITCET=20 T: 408.855.2957 M: 510.386.2647 F: 408.855.2978 =20 SERVICES WITHOUT BOUNDARIES =09 ------_=_NextPart_001_01C6C566.B00DC507 Content-Type: text/html; charset="US-ASCII" Content-Transfer-Encoding: quoted-printable Fixed Mobile Convergence (draft-yafan-fmc-mancho-00.txt)

Dear = Sipping = Enthusiasts:

If = you have not noticed the ietf drafts, Id like to invite you to = read and comment on the submitted draft. You can send your comments = to me or the sipping list.

Thank = you.

Yafan


        Title   =         : Mobile Assisted Handover across VoIP and Cellular = Domains Using SIP as Access Call Control

        Author(s)       : Y. = An

        Filename        = : draft-yafan-fmc-mancho-00.txt

        Pages   =         : 61

        Date    =         : 2006-7-31

       

This document describes the use of SIP between the dual mode = handset

(DMH) and the mobility SIP proxy to enable network = controlled

handovers across voice over IP (VoIP) and cellular networks.  = The

mobility SIP proxy has a peering relationship with the = cellular

network.  Mobile assisted and network controlled handovers = require

the exchange of cellular radio parameters between the DMH and = the

mobility SIP proxy.  The set of cellular related parameters = is

collectively called Session Handover Protocol (SHP).  The SHP = is a

residual protocol of SIP, where all of its messages are = transported

as MIME attachment to SIP messages.

This document describes the requirements, various SIP message = flows

for cross-domain roaming and bidirectional handovers between VoIP = and

cellular domains, and detailed specification of the residual = SHP

protocol.


A URL for this Internet-Draft is:

http://www.ietf.org/internet-drafts/draft-yafan-fmc-mancho-00.txt


        YAFAN AN CHIEF ARCHITCET
T: 408.855.2957  M: 510.386.2647  F: = 408.855.2978  
SERVICES = WITHOUT BOUNDARIES    

------_=_NextPart_001_01C6C566.B00DC507-- --===============0146759010== Content-Type: text/plain; charset="us-ascii" MIME-Version: 1.0 Content-Transfer-Encoding: 7bit Content-Disposition: inline _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP --===============0146759010==-- From sipping-bounces@ietf.org Mon Aug 21 21:15:04 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GFKpu-0002kw-AV; Mon, 21 Aug 2006 21:13:46 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GFKps-0002kq-H0 for sipping@ietf.org; Mon, 21 Aug 2006 21:13:44 -0400 Received: from exprod6og53.obsmtp.com ([64.18.1.187]) by ietf-mx.ietf.org with smtp (Exim 4.43) id 1GFKpq-0005FY-Ld for sipping@ietf.org; Mon, 21 Aug 2006 21:13:44 -0400 Received: from source ([192.150.11.134]) by exprod6ob53.postini.com ([64.18.5.12]) with SMTP; Mon, 21 Aug 2006 18:13:41 PDT Received: from inner-relay-3.eur.adobe.com (inner-relay-3.adobe.com [192.150.20.198] (may be forged)) by outbound-smtp-1.corp.adobe.com (8.12.10/8.12.10) with ESMTP id k7M1BbBl007575; Mon, 21 Aug 2006 18:11:39 -0700 (PDT) Received: from fe1.corp.adobe.com (fe1.corp.adobe.com [10.8.192.70]) by inner-relay-3.eur.adobe.com (8.12.10/8.12.9) with ESMTP id k7M1BtRE013556; Mon, 21 Aug 2006 18:13:33 -0700 (PDT) Received: from namail5.corp.adobe.com ([10.8.192.88]) by fe1.corp.adobe.com with Microsoft SMTPSVC(6.0.3790.1830); Mon, 21 Aug 2006 18:13:32 -0700 X-MimeOLE: Produced By Microsoft Exchange V6.5 Content-class: urn:content-classes:message MIME-Version: 1.0 Subject: RE: [Sipping] Fixed Mobile Convergence (draft-yafan-fmc-mancho-00.txt) Date: Mon, 21 Aug 2006 18:13:31 -0700 Message-ID: <24CCCC428EFEA2469BF046DB3C7A8D22113AC5@namail5.corp.adobe.com> In-Reply-To: <23EECEC9B06584478B1C9E38C253D35F74ACF2@minsk.us.stoke.com> X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: [Sipping] Fixed Mobile Convergence (draft-yafan-fmc-mancho-00.txt) Thread-Index: AcbFZq+8XXo8hqTMRregC7/hP3Ig9gAHPdRg From: "Henry Sinnreich" To: "Yafan An" , X-OriginalArrivalTime: 22 Aug 2006 01:13:32.0498 (UTC) FILETIME=[2F36C320:01C6C588] X-Spam-Score: 0.0 (/) X-Scan-Signature: c40ff0c70161549dbde2f89c7946385a Cc: X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Content-Type: multipart/mixed; boundary="===============0120305040==" Errors-To: sipping-bounces@ietf.org This is a multi-part message in MIME format. --===============0120305040== Content-class: urn:content-classes:message Content-Type: multipart/alternative; boundary="----_=_NextPart_001_01C6C588.2EA8B596" This is a multi-part message in MIME format. ------_=_NextPart_001_01C6C588.2EA8B596 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: quoted-printable Yafan, =20 I am not sure if it is right to have only the limiting assumptions in the I-D: =20 - The 'network' is in control of the handover decision and not the user, - The end user has no control over choosing the network or service provider, - Other networks such as Wi-Fi and WiMAX are not considered where the end user may want to switch to. =20 Last but not least, the Internet point of view is of interest in the IETF. IMHO, 3G is only one of the access options to the Internet ("the Internet is THE service") and the user must have choices and control to exercise them. =20 Thanks, Henry =20 ________________________________ From: Yafan An [mailto:yafan@stoke.com]=20 Sent: Monday, August 21, 2006 4:14 PM To: sipping@ietf.org Subject: [Sipping] Fixed Mobile Convergence (draft-yafan-fmc-mancho-00.txt) =20 Dear Sipping Enthusiasts: If you have not noticed the ietf drafts, I'd like to invite you to read and comment on the submitted draft. You can send your comments to me or the sipping list. Thank you. Yafan =20 Title : Mobile Assisted Handover across VoIP and Cellular Domains Using SIP as Access Call Control Author(s) : Y. An Filename : draft-yafan-fmc-mancho-00.txt Pages : 61 Date : 2006-7-31 =20 This document describes the use of SIP between the dual mode handset (DMH) and the mobility SIP proxy to enable network controlled handovers across voice over IP (VoIP) and cellular networks. The mobility SIP proxy has a peering relationship with the cellular network. Mobile assisted and network controlled handovers require the exchange of cellular radio parameters between the DMH and the mobility SIP proxy. The set of cellular related parameters is collectively called Session Handover Protocol (SHP). The SHP is a residual protocol of SIP, where all of its messages are transported as MIME attachment to SIP messages. This document describes the requirements, various SIP message flows for cross-domain roaming and bidirectional handovers between VoIP and cellular domains, and detailed specification of the residual SHP protocol. =20 A URL for this Internet-Draft is: http://www.ietf.org/internet-drafts/draft-yafan-fmc-mancho-00.txt =20 =20 YAFAN AN - CHIEF ARCHITCET T: 408.855.2957 M: 510.386.2647 F: 408.855.2978 =20 SERVICES WITHOUT BOUNDARIES =20 ------_=_NextPart_001_01C6C588.2EA8B596 Content-Type: text/html; charset="us-ascii" Content-Transfer-Encoding: quoted-printable Fixed Mobile Convergence (draft-yafan-fmc-mancho-00.txt)

Yafan,

 

I am not sure if it is right to = have only the limiting assumptions in the I-D:

 

- The ‘network’ is in = control of the handover decision and not the user,

- The end user has no control over choosing the network or service provider,

- Other networks such as Wi-Fi and = WiMAX are not considered where the end user may want to switch = to.

 

Last but not least, the Internet = point of view is of interest in the IETF. IMHO, 3G is only one of the access = options to the Internet (“the Internet is THE service”) and the user = must have choices and control to exercise them.

 

Thanks, = Henry

 


From: Yafan = An [mailto:yafan@stoke.com]
Sent: Monday, August 21, = 2006 4:14 PM
To: sipping@ietf.org
Subject: [Sipping] Fixed = Mobile Convergence (draft-yafan-fmc-mancho-00.txt)

 

Dear Sipping Enthusiasts:

If you have not noticed the ietf drafts, I’d like to invite you to read = and comment on the submitted draft. You can send your comments = to me or the sipping list.

Thank you.

Yafan

 

        Title           : Mobile = Assisted Handover across VoIP and Cellular Domains Using SIP as Access Call = Control

        Author(s)       = : Y. An

        Filename        : draft-yafan-fmc-mancho-00.txt

        Pages           : = 61

        Date            : = 2006-7-31

       

This document describes the use of SIP between the dual = mode handset

(DMH) and the mobility SIP proxy to enable network = controlled

handovers across voice over IP (VoIP) and cellular networks.  The

mobility SIP proxy has a peering relationship with the = cellular

network.  Mobile assisted and network controlled handovers = require

the exchange of cellular radio parameters between the DMH = and the

mobility SIP proxy.  The set of cellular related = parameters is

collectively called Session Handover Protocol = (SHP).  The SHP is a

residual protocol of SIP, where all of its messages = are transported

as MIME attachment to SIP = messages.

This document describes the requirements, various SIP = message flows

for cross-domain roaming and bidirectional handovers = between VoIP and

cellular domains, and detailed specification of the = residual SHP

protocol.

 

A URL for this Internet-Draft = is:

http://www.ietf.org/internet-drafts/draft-yafan-fmc-mancho-00.txt

 

      &n= bsp; YAFAN AN – CHIEF ARCHITCET
T: 408.855.2957  M: 510.386.2647  F: 408.855.2978  
SERVICES WITHOUT = BOUNDARIES    

------_=_NextPart_001_01C6C588.2EA8B596-- --===============0120305040== Content-Type: text/plain; charset="us-ascii" MIME-Version: 1.0 Content-Transfer-Encoding: 7bit Content-Disposition: inline _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP --===============0120305040==-- From sipping-bounces@ietf.org Tue Aug 22 01:26:15 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GFOkd-0003fs-9n; Tue, 22 Aug 2006 01:24:35 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GFOka-0003eG-N8 for sipping@ietf.org; Tue, 22 Aug 2006 01:24:32 -0400 Received: from nat.stoke.com ([209.78.40.100] helo=trio.stoke.com) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GFOWR-0000k6-E0 for sipping@ietf.org; Tue, 22 Aug 2006 01:09:57 -0400 Received: from us.stoke.com (minsk.us.stoke.com [172.16.4.20]) by trio.stoke.com (8.13.3/8.13.3) with ESMTP id k7M59ktl040612 for ; Mon, 21 Aug 2006 22:09:46 -0700 (PDT) (envelope-from yafan@stoke.com) X-MimeOLE: Produced By Microsoft Exchange V6.5.7226.0 Content-class: urn:content-classes:message MIME-Version: 1.0 Subject: RE: [Sipping] Fixed Mobile Convergence (draft-yafan-fmc-mancho-00.txt) Date: Mon, 21 Aug 2006 22:11:34 -0700 Message-ID: <23EECEC9B06584478B1C9E38C253D35F74AD2B@minsk.us.stoke.com> X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: [Sipping] Fixed Mobile Convergence (draft-yafan-fmc-mancho-00.txt) Thread-Index: AcbFZq+8XXo8hqTMRregC7/hP3Ig9gAHPdRgAAjOopA= From: "Yafan An" To: "Henry Sinnreich" , X-Virus-Scanned: ClamAV 0.88/1705/Mon Aug 21 18:18:23 2006 on trio.stoke.com X-Virus-Status: Clean X-Spam-Score: 0.0 (/) X-Scan-Signature: 8949cc4fd406a34204d26327803246d1 Cc: X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Content-Type: multipart/mixed; boundary="===============1286489110==" Errors-To: sipping-bounces@ietf.org This is a multi-part message in MIME format. --===============1286489110== Content-class: urn:content-classes:message Content-Type: multipart/alternative; boundary="----_=_NextPart_001_01C6C5A9.70C2F01D" This is a multi-part message in MIME format. ------_=_NextPart_001_01C6C5A9.70C2F01D Content-Type: text/plain; charset="US-ASCII" Content-Transfer-Encoding: quoted-printable Henry, =20 The term "network-controlled" handovers refers to the fact that a network element is in control of the execution of the handover procedures. This is in contrast to other schemes where, e.g. the handset is in control of the handover procedures. =20 User selection of which access network to use for access can well be exercised by a policy-based (or other means) on the handset or on a network element. Such control may provision that an on-going session can be handed over from WLAN to/from cellular by a network element according to certain threshold settings. =20 These two types of "controls" do not necessarily conflict with each other. =20 How the control is exercised may be determined by many factors. In GSM, handover is network-controlled (and mobile assisted). =20 Thanks, Yafan =20 ________________________________ From: Henry Sinnreich [mailto:hsinnrei@adobe.com]=20 Sent: Monday, August 21, 2006 6:14 PM To: Yafan An; sipping@ietf.org Subject: RE: [Sipping] Fixed Mobile Convergence (draft-yafan-fmc-mancho-00.txt) =20 Yafan, =20 I am not sure if it is right to have only the limiting assumptions in the I-D: =20 - The 'network' is in control of the handover decision and not the user, - The end user has no control over choosing the network or service provider, - Other networks such as Wi-Fi and WiMAX are not considered where the end user may want to switch to. =20 Last but not least, the Internet point of view is of interest in the IETF. IMHO, 3G is only one of the access options to the Internet ("the Internet is THE service") and the user must have choices and control to exercise them. =20 Thanks, Henry =20 ________________________________ From: Yafan An [mailto:yafan@stoke.com]=20 Sent: Monday, August 21, 2006 4:14 PM To: sipping@ietf.org Subject: [Sipping] Fixed Mobile Convergence (draft-yafan-fmc-mancho-00.txt) =20 Dear Sipping Enthusiasts: If you have not noticed the ietf drafts, I'd like to invite you to read and comment on the submitted draft. You can send your comments to me or the sipping list. Thank you. Yafan =20 Title : Mobile Assisted Handover across VoIP and Cellular Domains Using SIP as Access Call Control Author(s) : Y. An Filename : draft-yafan-fmc-mancho-00.txt Pages : 61 Date : 2006-7-31 =20 This document describes the use of SIP between the dual mode handset (DMH) and the mobility SIP proxy to enable network controlled handovers across voice over IP (VoIP) and cellular networks. The mobility SIP proxy has a peering relationship with the cellular network. Mobile assisted and network controlled handovers require the exchange of cellular radio parameters between the DMH and the mobility SIP proxy. The set of cellular related parameters is collectively called Session Handover Protocol (SHP). The SHP is a residual protocol of SIP, where all of its messages are transported as MIME attachment to SIP messages. This document describes the requirements, various SIP message flows for cross-domain roaming and bidirectional handovers between VoIP and cellular domains, and detailed specification of the residual SHP protocol. =20 A URL for this Internet-Draft is: http://www.ietf.org/internet-drafts/draft-yafan-fmc-mancho-00.txt =20 =20 YAFAN AN - CHIEF ARCHITCET T: 408.855.2957 M: 510.386.2647 F: 408.855.2978 =20 SERVICES WITHOUT BOUNDARIES =20 ------_=_NextPart_001_01C6C5A9.70C2F01D Content-Type: text/html; charset="US-ASCII" Content-Transfer-Encoding: quoted-printable Fixed Mobile Convergence (draft-yafan-fmc-mancho-00.txt)

Henry,

 

The term = “network-controlled” handovers refers to the fact that a network element is in control of the execution of the handover procedures. This is in contrast to other = schemes where, e.g. the handset is in control of the handover = procedures.

 

User selection of which access = network to use for access can well be exercised by a policy-based (or other means) = on the handset or on a network element. Such control may provision that an = on-going session can be handed over from WLAN to/from cellular by a network = element according to certain threshold settings.

 

These two types of = “controls” do not necessarily conflict with each = other.

 

How the control is exercised may be determined by many factors. In GSM, handover is network-controlled (and = mobile assisted).

 

Thanks,

=

Yafan

 


From: Henry = Sinnreich [mailto:hsinnrei@adobe.com]
Sent: Monday, August 21, = 2006 6:14 PM
To: Yafan An; sipping@ietf.org
Subject: RE: [Sipping] = Fixed Mobile Convergence = (draft-yafan-fmc-mancho-00.txt)

 

Yafan,

 

I am not sure if it is right to = have only the limiting assumptions in the I-D:

 

- The ‘network’ is in = control of the handover decision and not the user,

- The end user has no control over choosing the network or service provider,

- Other networks such as Wi-Fi and = WiMAX are not considered where the end user may want to switch = to.

 

Last but not least, the Internet = point of view is of interest in the IETF. IMHO, 3G is only one of the access = options to the Internet (“the Internet is THE service”) and the user = must have choices and control to exercise them.

 

Thanks, = Henry

 


From: = Yafan An [mailto:yafan@stoke.com]
Sent: Monday, August 21, = 2006 4:14 PM
To: sipping@ietf.org
Subject: [Sipping] Fixed = Mobile Convergence (draft-yafan-fmc-mancho-00.txt)

 

Dear Sipping Enthusiasts:

If you have not noticed the ietf drafts, I’d like to invite you to = read and comment on the submitted draft. You can send your comments = to me or the sipping list.

Thank you.

Yafan

 

        Title           : Mobile Assisted = Handover across VoIP and Cellular Domains Using SIP as Access Call = Control

        Author(s)       = : Y. An

        Filename        : draft-yafan-fmc-mancho-00.txt

        Pages           : = 61

        Date            : = 2006-7-31

       

This document describes the use of SIP between the dual = mode handset

(DMH) and the mobility SIP proxy to enable network = controlled

handovers across voice over IP (VoIP) and cellular networks.  The

mobility SIP proxy has a peering relationship with the = cellular

network.  Mobile assisted and network controlled handovers = require

the exchange of cellular radio parameters between the DMH = and the

mobility SIP proxy.  The set of cellular related = parameters is

collectively called Session Handover Protocol = (SHP).  The SHP is a

residual protocol of SIP, where all of its messages = are transported

as MIME attachment to SIP = messages.

This document describes the requirements, various SIP = message flows

for cross-domain roaming and bidirectional handovers = between VoIP and

cellular domains, and detailed specification of the = residual SHP

protocol.

 

A URL for this Internet-Draft = is:

http://www.ietf.org/internet-drafts/draft-yafan-fmc-mancho-00.txt

 

      &n= bsp; YAFAN AN – CHIEF ARCHITCET
T: 408.855.2957  M: 510.386.2647  F: 408.855.2978  
SERVICES WITHOUT = BOUNDARIES    

------_=_NextPart_001_01C6C5A9.70C2F01D-- --===============1286489110== Content-Type: text/plain; charset="us-ascii" MIME-Version: 1.0 Content-Transfer-Encoding: 7bit Content-Disposition: inline _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP --===============1286489110==-- From sipping-bounces@ietf.org Tue Aug 22 03:23:39 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GFQb1-0006ET-2q; Tue, 22 Aug 2006 03:22:47 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GFQaz-0006EO-VU for sipping@ietf.org; Tue, 22 Aug 2006 03:22:45 -0400 Received: from 66.254.241.83.in-addr.dgcsystems.net ([83.241.254.66] helo=smtp.dgcsystems.net) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GFQau-0004e2-LJ for sipping@ietf.org; Tue, 22 Aug 2006 03:22:45 -0400 Received: from Misan ([217.13.240.136]) by smtp.dgcsystems.net with Microsoft SMTPSVC(6.0.3790.1830); Tue, 22 Aug 2006 09:22:33 +0200 From: =?iso-8859-1?Q?Gunnar_Hellstr=F6m?= To: "Frank Shearar" , Subject: RE: [Sipping] Minor grammatical nits in draft-ietf-sipping-toip-06 Date: Tue, 22 Aug 2006 09:22:17 +0200 Message-ID: MIME-Version: 1.0 Content-Type: text/plain; charset="iso-8859-1" Content-Transfer-Encoding: 8bit X-Priority: 3 (Normal) X-MSMail-Priority: Normal X-Mailer: Microsoft Outlook IMO, Build 9.0.6604 (9.0.2911.0) X-MIMEOLE: Produced By Microsoft MimeOLE V6.00.2900.2962 In-Reply-To: <006701c6c297$02b8e600$0600000a@wsfrank> Importance: Normal X-OriginalArrivalTime: 22 Aug 2006 07:22:33.0698 (UTC) FILETIME=[BC65C420:01C6C5BB] X-Spam-Score: 0.0 (/) X-Scan-Signature: 50a516d93fd399dc60588708fd9a3002 Cc: X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Frank, I agree with your two comments and change proposals. Gunnar ---------------------------------------------------------------------------- - Gunnar Hellström, Omnitor gunnar.hellstrom@omnitor.se -----Original Message----- From: Frank Shearar [mailto:frank.shearar@angband.za.org] Sent: Friday, August 18, 2006 9:22 AM To: sipping@ietf.org Subject: [Sipping] Minor grammatical nits in draft-ietf-sipping-toip-06 Section 5.2.3.2: They are a) "No-gain" codec solution, b) the Cellular Text Telephony Modem (CTM) solution [8] both collectively called "Baudot mode" solution in the USA. should read "They are the a) ...., and b) ..." Section 6.2.5.1: This means that the other user may be used to formal turn taking during the call. should maybe read This means that the textphone user may be used to formal turn taking during the call. Looks good! frank _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Tue Aug 22 08:22:16 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GFVFw-0002mI-Jl; Tue, 22 Aug 2006 08:21:20 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GFVFu-0002mB-G0 for sipping@ietf.org; Tue, 22 Aug 2006 08:21:18 -0400 Received: from nf-out-0910.google.com ([64.233.182.189]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GFVFt-0007bu-5C for sipping@ietf.org; Tue, 22 Aug 2006 08:21:18 -0400 Received: by nf-out-0910.google.com with SMTP id n15so41341nfc for ; Tue, 22 Aug 2006 05:21:16 -0700 (PDT) DomainKey-Signature: a=rsa-sha1; q=dns; c=nofws; s=beta; d=gmail.com; h=received:message-id:date:from:to:subject:cc:in-reply-to:mime-version:content-type:content-transfer-encoding:content-disposition:references; b=g79nD7g+M99AmD9O5Ah/Tv+anOOiN0CEVQWWxrL1avBr6iehXV4EdGfBfa35pkFdwBuSmTnUxauHdvl3vsh8sB3dvdBQGODco/zhwEDBwWQDTJDlw13WiFubOks18ygv/9qEcPRDMkPuYXrakd2SKkJ5ine+6b3nGsDU96L4xtM= Received: by 10.49.43.2 with SMTP id v2mr307361nfj; Tue, 22 Aug 2006 05:21:16 -0700 (PDT) Received: by 10.49.14.3 with HTTP; Tue, 22 Aug 2006 05:21:15 -0700 (PDT) Message-ID: Date: Tue, 22 Aug 2006 08:21:15 -0400 From: "Arjun Roychowdhury" To: "Yafan An" Subject: Re: [Sipping] Fixed Mobile Convergence (draft-yafan-fmc-mancho-00.txt) In-Reply-To: <23EECEC9B06584478B1C9E38C253D35F74ACF2@minsk.us.stoke.com> MIME-Version: 1.0 Content-Type: text/plain; charset=WINDOWS-1252; format=flowed Content-Transfer-Encoding: quoted-printable Content-Disposition: inline References: <23EECEC9B06584478B1C9E38C253D35F74ACF2@minsk.us.stoke.com> X-Spam-Score: 0.0 (/) X-Scan-Signature: cf3becbbd6d1a45acbe2ffd4ab88bdc2 Cc: sipping@ietf.org X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Yafan, Has the referred to document 'yafan-fmc-arch' been discussed in lists before ? this I-D though interesting seems to be a direct outcome of a particular configuration /product deployment and assumption of nodes such as MP-CSCF. Finally, this draft makes no mention of 3GPP/2 VCC and why a different network reference model is required. regds arjun On 8/21/06, Yafan An wrote: > > > > Dear Sipping Enthusiasts: > > If you have not noticed the ietf drafts, I'd like to invite you to read a= nd > comment on the submitted draft. You can send your comments to me or the > sipping list. > > Thank you. > > Yafan > > > Title : Mobile Assisted Handover across VoIP and Cellul= ar > Domains Using SIP as Access Call Control > > Author(s) : Y. An > > Filename : draft-yafan-fmc-mancho-00.txt > > Pages : 61 > > Date : 2006-7-31 > > > > This document describes the use of SIP between the dual mode handset > > (DMH) and the mobility SIP proxy to enable network controlled > > handovers across voice over IP (VoIP) and cellular networks. The > > mobility SIP proxy has a peering relationship with the cellular > > network. Mobile assisted and network controlled handovers require > > the exchange of cellular radio parameters between the DMH and the > > mobility SIP proxy. The set of cellular related parameters is > > collectively called Session Handover Protocol (SHP). The SHP is a > > residual protocol of SIP, where all of its messages are transported > > as MIME attachment to SIP messages. > > This document describes the requirements, various SIP message flows > > for cross-domain roaming and bidirectional handovers between VoIP and > > cellular domains, and detailed specification of the residual SHP > > protocol. > > > A URL for this Internet-Draft is: > > http://www.ietf.org/internet-drafts/draft-yafan-fmc-mancho-00.txt > > > > > YAFAN AN =96 CHIEF ARCHITCET > T: 408.855.2957 M: 510.386.2647 F: 408.855.2978 > SERVICES WITHOUT BOUNDARIES > > > > _______________________________________________ > Sipping mailing list > https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP > Use sip-implementors@cs.columbia.edu for questions on > current sip > Use sip@ietf.org for new developments of core SIP > > --=20 Arjun Roychowdhury _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Tue Aug 22 10:02:48 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GFWp1-0000QX-MK; Tue, 22 Aug 2006 10:01:39 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GFWp0-0000QR-MC for sipping@ietf.org; Tue, 22 Aug 2006 10:01:38 -0400 Received: from exprod6og56.obsmtp.com ([64.18.1.208]) by ietf-mx.ietf.org with smtp (Exim 4.43) id 1GFWov-0001YV-ML for sipping@ietf.org; Tue, 22 Aug 2006 10:01:38 -0400 Received: from source ([192.150.20.142]) by exprod6ob56.postini.com ([64.18.5.12]) with SMTP; Tue, 22 Aug 2006 07:01:30 PDT Received: from inner-relay-3.eur.adobe.com (inner-relay-3b [10.128.4.236]) by outbound-smtp-2.corp.adobe.com (8.12.10/8.12.10) with ESMTP id k7ME1euv015435; Tue, 22 Aug 2006 07:01:41 -0700 (PDT) Received: from fe2.corp.adobe.com (fe2.corp.adobe.com [10.8.192.72]) by inner-relay-3.eur.adobe.com (8.12.10/8.12.9) with ESMTP id k7ME1MQw022220; Tue, 22 Aug 2006 07:01:23 -0700 (PDT) Received: from namail5.corp.adobe.com ([10.8.192.88]) by fe2.corp.adobe.com with Microsoft SMTPSVC(6.0.3790.1830); Tue, 22 Aug 2006 07:01:22 -0700 X-MimeOLE: Produced By Microsoft Exchange V6.5 Content-class: urn:content-classes:message MIME-Version: 1.0 Subject: RE: [Sipping] Fixed Mobile Convergence (draft-yafan-fmc-mancho-00.txt) Date: Tue, 22 Aug 2006 06:59:11 -0700 Message-ID: <24CCCC428EFEA2469BF046DB3C7A8D22113AF1@namail5.corp.adobe.com> In-Reply-To: <23EECEC9B06584478B1C9E38C253D35F74AD2B@minsk.us.stoke.com> X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: [Sipping] Fixed Mobile Convergence (draft-yafan-fmc-mancho-00.txt) Thread-Index: AcbFZq+8XXo8hqTMRregC7/hP3Ig9gAHPdRgAAjOopAAEr0ZIA== From: "Henry Sinnreich" To: "Yafan An" , X-OriginalArrivalTime: 22 Aug 2006 14:01:22.0564 (UTC) FILETIME=[731CF440:01C6C5F3] X-Spam-Score: 0.0 (/) X-Scan-Signature: 9d5866e4c615ceea0db8f42c46495d22 Cc: X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Content-Type: multipart/mixed; boundary="===============2123537449==" Errors-To: sipping-bounces@ietf.org This is a multi-part message in MIME format. --===============2123537449== Content-class: urn:content-classes:message Content-Type: multipart/alternative; boundary="----_=_NextPart_001_01C6C5F3.727BB066" This is a multi-part message in MIME format. ------_=_NextPart_001_01C6C5F3.727BB066 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: quoted-printable Yafan, =20 Your I-D is good engineering work and the explanations below are accurate.=20 =20 It would help to clarify in the Introduction what options there are, which of the architectural choices are made in this I-D (as explained below) and for good balance to include some references for the other options. =20 Multimodal mobile devices may cover 2G, 3G, and 4G networks. The user who pays for both the device and the services will naturally go where they are given a choice. Under such an assumption, handover control from the device is too big an issue to be omitted. =20 Thanks for your understanding, =20 Henry =20 ________________________________ From: Yafan An [mailto:yafan@stoke.com]=20 Sent: Tuesday, August 22, 2006 12:12 AM To: Henry Sinnreich; sipping@ietf.org Subject: RE: [Sipping] Fixed Mobile Convergence (draft-yafan-fmc-mancho-00.txt) =20 Henry, =20 The term "network-controlled" handovers refers to the fact that a network element is in control of the execution of the handover procedures. This is in contrast to other schemes where, e.g. the handset is in control of the handover procedures. =20 User selection of which access network to use for access can well be exercised by a policy-based (or other means) on the handset or on a network element. Such control may provision that an on-going session can be handed over from WLAN to/from cellular by a network element according to certain threshold settings. =20 These two types of "controls" do not necessarily conflict with each other. =20 How the control is exercised may be determined by many factors. In GSM, handover is network-controlled (and mobile assisted). =20 Thanks, Yafan =20 ________________________________ From: Henry Sinnreich [mailto:hsinnrei@adobe.com]=20 Sent: Monday, August 21, 2006 6:14 PM To: Yafan An; sipping@ietf.org Subject: RE: [Sipping] Fixed Mobile Convergence (draft-yafan-fmc-mancho-00.txt) =20 Yafan, =20 I am not sure if it is right to have only the limiting assumptions in the I-D: =20 - The 'network' is in control of the handover decision and not the user, - The end user has no control over choosing the network or service provider, - Other networks such as Wi-Fi and WiMAX are not considered where the end user may want to switch to. =20 Last but not least, the Internet point of view is of interest in the IETF. IMHO, 3G is only one of the access options to the Internet ("the Internet is THE service") and the user must have choices and control to exercise them. =20 Thanks, Henry =20 ________________________________ From: Yafan An [mailto:yafan@stoke.com]=20 Sent: Monday, August 21, 2006 4:14 PM To: sipping@ietf.org Subject: [Sipping] Fixed Mobile Convergence (draft-yafan-fmc-mancho-00.txt) =20 Dear Sipping Enthusiasts: If you have not noticed the ietf drafts, I'd like to invite you to read and comment on the submitted draft. You can send your comments to me or the sipping list. Thank you. Yafan =20 Title : Mobile Assisted Handover across VoIP and Cellular Domains Using SIP as Access Call Control Author(s) : Y. An Filename : draft-yafan-fmc-mancho-00.txt Pages : 61 Date : 2006-7-31 =20 This document describes the use of SIP between the dual mode handset (DMH) and the mobility SIP proxy to enable network controlled handovers across voice over IP (VoIP) and cellular networks. The mobility SIP proxy has a peering relationship with the cellular network. Mobile assisted and network controlled handovers require the exchange of cellular radio parameters between the DMH and the mobility SIP proxy. The set of cellular related parameters is collectively called Session Handover Protocol (SHP). The SHP is a residual protocol of SIP, where all of its messages are transported as MIME attachment to SIP messages. This document describes the requirements, various SIP message flows for cross-domain roaming and bidirectional handovers between VoIP and cellular domains, and detailed specification of the residual SHP protocol. =20 A URL for this Internet-Draft is: http://www.ietf.org/internet-drafts/draft-yafan-fmc-mancho-00.txt =20 =20 YAFAN AN - CHIEF ARCHITCET T: 408.855.2957 M: 510.386.2647 F: 408.855.2978 =20 SERVICES WITHOUT BOUNDARIES =20 ------_=_NextPart_001_01C6C5F3.727BB066 Content-Type: text/html; charset="us-ascii" Content-Transfer-Encoding: quoted-printable Fixed Mobile Convergence (draft-yafan-fmc-mancho-00.txt)

Yafan,

 

Your I-D is good engineering work = and the explanations below are accurate.

 

It would help to clarify in the Introduction what options there are, which of the architectural choices = are made in this I-D (as explained below) and for good balance to include = some references for the other options.

 

Multimodal mobile devices may cover = 2G, 3G, and 4G networks. The user who pays for both the device and the = services will naturally go where they are given a choice. Under such an = assumption, handover control from the device is too big an issue to be = omitted.

 

Thanks for your = understanding,

 

Henry

 


From: Yafan = An [mailto:yafan@stoke.com]
Sent: Tuesday, August 22, = 2006 12:12 AM
To: Henry Sinnreich; sipping@ietf.org
Subject: RE: [Sipping] = Fixed Mobile Convergence = (draft-yafan-fmc-mancho-00.txt)

 

Henry,

 

The term = “network-controlled” handovers refers to the fact that a network element is in control of the = execution of the handover procedures. This is in contrast to other schemes where, = e.g. the handset is in control of the handover = procedures.

 

User selection of which access = network to use for access can well be exercised by a policy-based (or other means) = on the handset or on a network element. Such control may provision that an = on-going session can be handed over from WLAN to/from cellular by a network = element according to certain threshold settings.

 

These two types of = “controls” do not necessarily conflict with each = other.

 

How the control is exercised may be determined by many factors. In GSM, handover is network-controlled (and = mobile assisted).

 

Thanks,

=

Yafan

 


From: Henry = Sinnreich [mailto:hsinnrei@adobe.com]
Sent: Monday, August 21, = 2006 6:14 PM
To: Yafan An; sipping@ietf.org
Subject: RE: [Sipping] = Fixed Mobile Convergence = (draft-yafan-fmc-mancho-00.txt)

 

Yafan,

 

I am not sure if it is right to = have only the limiting assumptions in the I-D:

 

- The ‘network’ is in = control of the handover decision and not the user,

- The end user has no control over choosing the network or service provider,

- Other networks such as Wi-Fi and = WiMAX are not considered where the end user may want to switch = to.

 

Last but not least, the Internet = point of view is of interest in the IETF. IMHO, 3G is only one of the access = options to the Internet (“the Internet is THE service”) and the user = must have choices and control to exercise them.

 

Thanks, = Henry

 


From: = Yafan An [mailto:yafan@stoke.com]
Sent: Monday, August 21, = 2006 4:14 PM
To: sipping@ietf.org
Subject: [Sipping] Fixed = Mobile Convergence (draft-yafan-fmc-mancho-00.txt)

 

Dear Sipping Enthusiasts:

If you have not noticed the ietf drafts, I’d like to invite you to = read and comment on the submitted draft. You can send your comments = to me or the sipping list.

Thank you.

Yafan

 

        Title           : Mobile Assisted = Handover across VoIP and Cellular Domains Using SIP as Access Call = Control

        Author(s)       = : Y. An

        Filename        : draft-yafan-fmc-mancho-00.txt

        Pages           : = 61

        Date            : = 2006-7-31

       

This document describes the use of SIP between the dual = mode handset

(DMH) and the mobility SIP proxy to enable network = controlled

handovers across voice over IP (VoIP) and cellular networks.  The

mobility SIP proxy has a peering relationship with the = cellular

network.  Mobile assisted and network controlled handovers = require

the exchange of cellular radio parameters between the DMH = and the

mobility SIP proxy.  The set of cellular related = parameters is

collectively called Session Handover Protocol = (SHP).  The SHP is a

residual protocol of SIP, where all of its messages = are transported

as MIME attachment to SIP = messages.

This document describes the requirements, various SIP = message flows

for cross-domain roaming and bidirectional handovers = between VoIP and

cellular domains, and detailed specification of the = residual SHP

protocol.

 

A URL for this Internet-Draft = is:

http://www.ietf.org/internet-drafts/draft-yafan-fmc-mancho-00.txt

 

      &n= bsp; YAFAN AN – CHIEF ARCHITCET
T: 408.855.2957  M: 510.386.2647  F: 408.855.2978  
SERVICES WITHOUT = BOUNDARIES    

------_=_NextPart_001_01C6C5F3.727BB066-- --===============2123537449== Content-Type: text/plain; charset="us-ascii" MIME-Version: 1.0 Content-Transfer-Encoding: 7bit Content-Disposition: inline _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP --===============2123537449==-- From sipping-bounces@ietf.org Tue Aug 22 12:13:05 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GFYrE-0001Qb-D6; Tue, 22 Aug 2006 12:12:04 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GFYrD-0001Pd-2q for sipping@ietf.org; Tue, 22 Aug 2006 12:12:03 -0400 Received: from stsc1260-eth-s1-s1p1-vip.va.neustar.com ([156.154.16.129] helo=chiedprmail1.ietf.org) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GFX2l-0003dA-8T for sipping@ietf.org; Tue, 22 Aug 2006 10:15:51 -0400 Received: from nf-out-0910.google.com ([64.233.182.185]) by chiedprmail1.ietf.org with esmtp (Exim 4.43) id 1GFWyn-0000RV-6z for sipping@ietf.org; Tue, 22 Aug 2006 10:11:47 -0400 Received: by nf-out-0910.google.com with SMTP id n15so68551nfc for ; Tue, 22 Aug 2006 07:11:41 -0700 (PDT) DomainKey-Signature: a=rsa-sha1; q=dns; c=nofws; s=beta; d=gmail.com; h=received:message-id:date:from:to:subject:cc:in-reply-to:mime-version:content-type:content-transfer-encoding:content-disposition:references; b=EEEGyQrCvJcACxdkVUd4qgkXuA0JNOJKyvuWNLQEyQHBE310Y5iw8wpvXP6D8IEYxfK0xojPEI2/diRyF89boqZyHr41lVd718tOOpC8aCt6nhRFn2p8bqQU+xjuyiDdmZ1zGesQZaMDF2HZLwNUGvYV6ZH5jqiWERlRHmNhdQw= Received: by 10.49.75.2 with SMTP id c2mr487915nfl; Tue, 22 Aug 2006 07:11:41 -0700 (PDT) Received: by 10.49.14.3 with HTTP; Tue, 22 Aug 2006 07:11:40 -0700 (PDT) Message-ID: Date: Tue, 22 Aug 2006 10:11:40 -0400 From: "Arjun Roychowdhury" To: "Yafan An" Subject: Re: [Sipping] Fixed Mobile Convergence (draft-yafan-fmc-arch-00.txt) In-Reply-To: <23EECEC9B06584478B1C9E38C253D35F74ACF4@minsk.us.stoke.com> MIME-Version: 1.0 Content-Type: text/plain; charset=WINDOWS-1252; format=flowed Content-Transfer-Encoding: quoted-printable Content-Disposition: inline References: <23EECEC9B06584478B1C9E38C253D35F74ACF4@minsk.us.stoke.com> X-Spam-Score: -2.5 (--) X-Scan-Signature: 8fbbaa16f9fd29df280814cb95ae2290 Cc: sipping@ietf.org X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Yafan, Just noticed this post. Comments while reading through the draft: General_1: * This is not really an over-arching FMC draft. It does not factor in GANs such as WiMAX / WLAN/DSL since the network reference model depicts MSC/VLR etc. which may not exist depending on the access network being connected to. Similarly, the identification tokens seem to be specific to the celluar wor= ld. Similary, the document contains many references to IMSI/MSISDN which again assumes interworking of an existing cellular network to a VoIP network. If we assume a PC soft-client on a desktop over WLAN needs to interwork, what will its IMSI be ? This comment comes due to the scope specified in your Abstract - I think the rest of the document has a much more restricted scope than the abstract quotes General_2: The benefits of this architecture over proposed interworking architectures by 3G is unclear to me. Can you please state the advantages ? I noticed that you also propose network initiated handover - this , afaik, is work under investigation as well with VCC and MiH (802.21) regds arjun * I have a fundamental problem with the MP-CSCF. It does not exist today and does not fit neatly into the defined 3G architecture. Most vendors I know have ready P-CSCFs and S-CSCFs. Pushing the anchoring function to the AS node (as VCC does) fits well into this model - mobility anchoring can be 'added' as an application node. Putting an MP-CSCF upfront means it needs to support all the published interfaces of the P-CSCF and more. * 3.2: PSI is referred to as the 'Public Service Identifier' in the 3G/IMS world - you use it to refer to 'pkt switched i/d'. (also CSI is referred to as the much debated 'Communication Service Identifier' ). Consider using common terminologies - PS/CS * 3.2.1: I am confused why you call the 'S-CSCF' a soft-switch (3.2.1) . This seems to indicate a product configuration. From a standardization perspective, an S-CSCF does not need to have traditional soft-switching capabiltiies nor class-5 features as you mention. The S-CSCF is supposed to execute the iFC and forward to the appropriate AS/MGCF for execution of relevant services. * 6.2: I believe P-Access-Network-Info and P-Visited-Network-ID could be used instead of overloading existing SIP headers. On 8/21/06, Yafan An wrote: > > > > Dear Sipping Enthusiasts: > > If you have not noticed the ietf drafts, I'd like to invite you to read a= nd > comment on the submitted draft. You can send your comments to me or the > sipping list. > > Thank you. > > Yafan > > > Title : An Architecture Framework for Fixed Mobile > Convergence Using SIP as Access Call Control Protocol > > Author(s) : Y. An > > Filename : draft-yafan-fmc-arch-00.txt > > Pages : 32 > > Date : 2006-7-28 > > > > This document describes an architecture framework for achieving > > converged mobile services across alternative radio access networks > > (GAN, or Generic Access Network), such as WLAN, WiMax, and the > > cellular network, while utilizing SIP for media session control > > between the dual mode handset (DMH) and a network SIP proxy. The key > > benefits of this architecture include (1) fast and flexible service > > deployment in the IP domain, and (2) allow seamless voice call > > continuity services across IP and circuit-switched cellular domains. > > > A URL for this Internet-Draft is: > > http://www.ietf.org/internet-drafts/draft-yafan-fmc-arch-00.txt > > > > > YAFAN AN =96 CHIEF ARCHITCET > T: 408.855.2957 M: 510.386.2647 F: 408.855.2978 > SERVICES WITHOUT BOUNDARIES > > > > _______________________________________________ > Sipping mailing list > https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP > Use sip-implementors@cs.columbia.edu for questions on > current sip > Use sip@ietf.org for new developments of core SIP > > --=20 Arjun Roychowdhury _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Tue Aug 22 12:52:50 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GFZUC-0000BW-IZ; Tue, 22 Aug 2006 12:52:20 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GFZUB-0000Az-8e for sipping@ietf.org; Tue, 22 Aug 2006 12:52:19 -0400 Received: from nat.stoke.com ([209.78.40.100] helo=trio.stoke.com) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GFZU9-0002Dz-RG for sipping@ietf.org; Tue, 22 Aug 2006 12:52:19 -0400 Received: from us.stoke.com (minsk.us.stoke.com [172.16.4.20]) by trio.stoke.com (8.13.3/8.13.3) with ESMTP id k7MGqDhN050195 for ; Tue, 22 Aug 2006 09:52:13 -0700 (PDT) (envelope-from yafan@stoke.com) X-MimeOLE: Produced By Microsoft Exchange V6.5.7226.0 Content-class: urn:content-classes:message MIME-Version: 1.0 Content-Type: text/plain; charset="US-ASCII" Content-Transfer-Encoding: quoted-printable Subject: RE: [Sipping] Fixed Mobile Convergence (draft-yafan-fmc-mancho-00.txt) Date: Tue, 22 Aug 2006 09:54:03 -0700 Message-ID: <23EECEC9B06584478B1C9E38C253D35F74AD8E@minsk.us.stoke.com> X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: [Sipping] Fixed Mobile Convergence (draft-yafan-fmc-mancho-00.txt) Thread-Index: AcbF5brcPnU/AgZRQJW9kuwVpjqkmQAJMfzg From: "Yafan An" To: "Arjun Roychowdhury" X-Virus-Scanned: ClamAV 0.88/1708/Tue Aug 22 05:43:00 2006 on trio.stoke.com X-Virus-Status: Clean X-Spam-Score: 0.0 (/) X-Scan-Signature: 825e642946eda55cd9bc654a36dab8c2 Cc: sipping@ietf.org X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Arjun, The 'yafan-fmc-arch' draft is submitted to this list for discussion at the same time. I agree that the 3GPP VCC approach should be referenced in text, it will be added. Yafan -----Original Message----- From: Arjun Roychowdhury [mailto:arjunrc@gmail.com]=20 Sent: Tuesday, August 22, 2006 5:21 AM To: Yafan An Cc: sipping@ietf.org Subject: Re: [Sipping] Fixed Mobile Convergence (draft-yafan-fmc-mancho-00.txt) Yafan, Has the referred to document 'yafan-fmc-arch' been discussed in lists before ? this I-D though interesting seems to be a direct outcome of a particular configuration /product deployment and assumption of nodes such as MP-CSCF. Finally, this draft makes no mention of 3GPP/2 VCC and why a different network reference model is required. regds arjun On 8/21/06, Yafan An wrote: > > > > Dear Sipping Enthusiasts: > > If you have not noticed the ietf drafts, I'd like to invite you to read and > comment on the submitted draft. You can send your comments to me or the > sipping list. > > Thank you. > > Yafan > > > Title : Mobile Assisted Handover across VoIP and Cellular > Domains Using SIP as Access Call Control > > Author(s) : Y. An > > Filename : draft-yafan-fmc-mancho-00.txt > > Pages : 61 > > Date : 2006-7-31 > > > > This document describes the use of SIP between the dual mode handset > > (DMH) and the mobility SIP proxy to enable network controlled > > handovers across voice over IP (VoIP) and cellular networks. The > > mobility SIP proxy has a peering relationship with the cellular > > network. Mobile assisted and network controlled handovers require > > the exchange of cellular radio parameters between the DMH and the > > mobility SIP proxy. The set of cellular related parameters is > > collectively called Session Handover Protocol (SHP). The SHP is a > > residual protocol of SIP, where all of its messages are transported > > as MIME attachment to SIP messages. > > This document describes the requirements, various SIP message flows > > for cross-domain roaming and bidirectional handovers between VoIP and > > cellular domains, and detailed specification of the residual SHP > > protocol. > > > A URL for this Internet-Draft is: > > http://www.ietf.org/internet-drafts/draft-yafan-fmc-mancho-00.txt > > > > > YAFAN AN - CHIEF ARCHITCET > T: 408.855.2957 M: 510.386.2647 F: 408.855.2978 > SERVICES WITHOUT BOUNDARIES > > > > _______________________________________________ > Sipping mailing list > https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP > Use sip-implementors@cs.columbia.edu for questions on > current sip > Use sip@ietf.org for new developments of core SIP > > --=20 Arjun Roychowdhury _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Tue Aug 22 15:27:55 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GFbuA-0002na-4N; Tue, 22 Aug 2006 15:27:18 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GFbu8-0002nS-Lg for sipping@ietf.org; Tue, 22 Aug 2006 15:27:16 -0400 Received: from cs.columbia.edu ([128.59.16.20]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GFbu5-0002UF-E6 for sipping@ietf.org; Tue, 22 Aug 2006 15:27:16 -0400 Received: from lion.cs.columbia.edu (IDENT:s37E2EtbRV3yU4/qC6yug1ajgi4/Dv9A@lion.cs.columbia.edu [128.59.16.120]) by cs.columbia.edu (8.12.10/8.12.10) with ESMTP id k7MJRAGb026707 (version=TLSv1/SSLv3 cipher=EDH-RSA-DES-CBC3-SHA bits=168 verify=NOT) for ; Tue, 22 Aug 2006 15:27:10 -0400 (EDT) Received: from [128.59.16.206] (chairpc.win.cs.columbia.edu [128.59.16.206]) (authenticated bits=0) by lion.cs.columbia.edu (8.12.9/8.12.9) with ESMTP id k7MJR9g7009136 (version=TLSv1/SSLv3 cipher=RC4-MD5 bits=128 verify=NOT) for ; Tue, 22 Aug 2006 15:27:10 -0400 Message-ID: <44EB5A88.1090606@cs.columbia.edu> Date: Tue, 22 Aug 2006 15:27:04 -0400 From: Henning Schulzrinne Organization: Columbia University User-Agent: Thunderbird 1.5.0.5 (Windows/20060719) MIME-Version: 1.0 To: sipping Content-Type: text/plain; charset=ISO-8859-1; format=flowed Content-Transfer-Encoding: 7bit X-PerlMx-Spam: Gauge=IIIIIII, Probability=7%, X-Seen-By filter1.cs.columbia.edu X-Spam-Score: 0.0 (/) X-Scan-Signature: cf4fa59384e76e63313391b70cd0dd25 Subject: [Sipping] Comment on draft-vandyke-mscml-09 X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org I have two comments on the draft: (1) As a high-level comment, Section 3 does not answer why, instead of INFO, a new method wouldn't have been more appropriate. (2) We've been using MSCML in a local application, for a PSAP mixer. For that application, one needs a slightly different mute function, namely something that blocks *outgoing* audio. I don't know the precise motivation, but it might involve privacy concerns for the call taker and the bridged first responders. One can create custom mixes, but that is rather cumbersome for a variety of reasons, such as dealing with new arrivals. It would seem more appropriate to have a comprehensive configure_leg type that allows to disable listening, not just talking, i.e., has four status (neither, talking only, listening only, both). Henning _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Tue Aug 22 15:46:44 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GFcCl-0002HM-Tp; Tue, 22 Aug 2006 15:46:31 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GFcCk-0002HG-Bg for sipping@ietf.org; Tue, 22 Aug 2006 15:46:30 -0400 Received: from stsc1260-eth-s1-s1p1-vip.va.neustar.com ([156.154.16.129] helo=chiedprmail1.ietf.org) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GFbxZ-00033M-GE for sipping@ietf.org; Tue, 22 Aug 2006 15:30:49 -0400 Received: from nat.stoke.com ([209.78.40.100] helo=trio.stoke.com) by chiedprmail1.ietf.org with esmtp (Exim 4.43) id 1GFbnz-000890-LM for sipping@ietf.org; Tue, 22 Aug 2006 15:20:58 -0400 Received: from us.stoke.com (minsk.us.stoke.com [172.16.4.20]) by trio.stoke.com (8.13.3/8.13.3) with ESMTP id k7MJKoLi054412 for ; Tue, 22 Aug 2006 12:20:50 -0700 (PDT) (envelope-from yafan@stoke.com) X-MimeOLE: Produced By Microsoft Exchange V6.5.7226.0 Content-class: urn:content-classes:message MIME-Version: 1.0 Content-Type: text/plain; charset="US-ASCII" Content-Transfer-Encoding: quoted-printable Subject: RE: [Sipping] Fixed Mobile Convergence (draft-yafan-fmc-arch-00.txt) Date: Tue, 22 Aug 2006 12:22:40 -0700 Message-ID: <23EECEC9B06584478B1C9E38C253D35F74ADC0@minsk.us.stoke.com> X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: [Sipping] Fixed Mobile Convergence (draft-yafan-fmc-arch-00.txt) Thread-Index: AcbF9Si0+D72oOnnR5qvbFYRmiNbCQAFm6IA From: "Yafan An" To: "Arjun Roychowdhury" X-Virus-Scanned: ClamAV 0.88/1708/Tue Aug 22 05:43:00 2006 on trio.stoke.com X-Virus-Status: Clean X-Spam-Score: -2.6 (--) X-Scan-Signature: 6ba8aaf827dcb437101951262f69b3de Cc: sipping@ietf.org X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org -----Original Message----- From: Arjun Roychowdhury [mailto:arjunrc@gmail.com]=20 Sent: Tuesday, August 22, 2006 7:12 AM To: Yafan An Cc: sipping@ietf.org Subject: Re: [Sipping] Fixed Mobile Convergence (draft-yafan-fmc-arch-00.txt) Yafan, Just noticed this post. Comments while reading through the draft: General_1: * This is not really an over-arching FMC draft. It does not factor in GANs such as WiMAX / WLAN/DSL since the network reference model depicts MSC/VLR etc. which may not exist depending on the access network being connected to. [Yafan An] It should have an "IP access" cloud which represents WLAN/WiMax/Broadband etc. This draft addresses some convergence issues between these IP access networks and the cellular network. Similarly, the identification tokens seem to be specific to the celluar world. Similary, the document contains many references to IMSI/MSISDN which again assumes interworking of an existing cellular network to a VoIP network. If we assume a PC soft-client on a desktop over WLAN needs to interwork, what will its IMSI be ? This comment comes due to the scope specified in your Abstract - I think the rest of the document has a much more restricted scope than the abstract quotes [Yafan An] For Dual Mode Handset (DMH), we should assume that an IMSI is always available for cellular access. A PC soft client that is not a dual mode handset, is outside the scope of this draft. There are other works which addresses transfers between SIP UAs, which I am currently evaluating. General_2: The benefits of this architecture over proposed interworking architectures by 3G is unclear to me. Can you please state the advantages ? I noticed that you also propose network initiated handover - this , afaik, is work under investigation as well with VCC and MiH (802.21) [Yafan An] I agree, this will be address in future updates. regds arjun * I have a fundamental problem with the MP-CSCF. It does not exist today and does not fit neatly into the defined 3G architecture. Most vendors I know have ready P-CSCFs and S-CSCFs. Pushing the anchoring function to the AS node (as VCC does) fits well into this model - mobility anchoring can be 'added' as an application node. Putting an MP-CSCF upfront means it needs to support all the published interfaces of the P-CSCF and more. [Yafan An] MP-CSCF is a specialized P-CSCF. You have a good argument here. I am considering separating the mobility piece out and name it separately. * 3.2: PSI is referred to as the 'Public Service Identifier' in the 3G/IMS world - you use it to refer to 'pkt switched i/d'. (also CSI is referred to as the much debated 'Communication Service Identifier' ). Consider using common terminologies - PS/CS [Yafan An] Point taken. * 3.2.1: I am confused why you call the 'S-CSCF' a soft-switch (3.2.1) . This seems to indicate a product configuration. From a standardization perspective, an S-CSCF does not need to have traditional soft-switching capabiltiies nor class-5 features as you mention. The S-CSCF is supposed to execute the iFC and forward to the appropriate AS/MGCF for execution of relevant services. [Yafan An] Point taken. Will add AS into the picture and clarify related text. * 6.2: I believe P-Access-Network-Info and P-Visited-Network-ID could be used instead of overloading existing SIP headers. [Yafan An] P-Visited-Network-ID was considered. However, it was defined in RFC3455 for conveying ID to the home network. The coding of it may not be as "free style" as the "Organization" field for display purposes. On 8/21/06, Yafan An wrote: > > > > Dear Sipping Enthusiasts: > > If you have not noticed the ietf drafts, I'd like to invite you to read and > comment on the submitted draft. You can send your comments to me or the > sipping list. > > Thank you. > > Yafan > > > Title : An Architecture Framework for Fixed Mobile > Convergence Using SIP as Access Call Control Protocol > > Author(s) : Y. An > > Filename : draft-yafan-fmc-arch-00.txt > > Pages : 32 > > Date : 2006-7-28 > > > > This document describes an architecture framework for achieving > > converged mobile services across alternative radio access networks > > (GAN, or Generic Access Network), such as WLAN, WiMax, and the > > cellular network, while utilizing SIP for media session control > > between the dual mode handset (DMH) and a network SIP proxy. The key > > benefits of this architecture include (1) fast and flexible service > > deployment in the IP domain, and (2) allow seamless voice call > > continuity services across IP and circuit-switched cellular domains. > > > A URL for this Internet-Draft is: > > http://www.ietf.org/internet-drafts/draft-yafan-fmc-arch-00.txt > > > > > YAFAN AN - CHIEF ARCHITCET > T: 408.855.2957 M: 510.386.2647 F: 408.855.2978 > SERVICES WITHOUT BOUNDARIES > > > > _______________________________________________ > Sipping mailing list > https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP > Use sip-implementors@cs.columbia.edu for questions on > current sip > Use sip@ietf.org for new developments of core SIP > > --=20 Arjun Roychowdhury _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Tue Aug 22 15:50:06 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GFcFx-0003sq-Ls; Tue, 22 Aug 2006 15:49:49 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GFcFv-0003sk-64 for sipping@ietf.org; Tue, 22 Aug 2006 15:49:47 -0400 Received: from stsc1260-eth-s1-s1p1-vip.va.neustar.com ([156.154.16.129] helo=chiedprmail1.ietf.org) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GFavo-0002cg-0l for sipping@ietf.org; Tue, 22 Aug 2006 14:24:56 -0400 Received: from host10.216.41.24.conversent.net ([216.41.24.10] helo=acmepacket.com) by chiedprmail1.ietf.org with esmtp (Exim 4.43) id 1GFaZV-0006Ap-QI for sipping@ietf.org; Tue, 22 Aug 2006 14:01:54 -0400 Received: from BPenfield [10.0.200.6] by acmepacket.com with ESMTP (SMTPD-9.03) id A6520238; Tue, 22 Aug 2006 14:00:50 -0400 Message-ID: <001b01c6c614$ec201950$800101df@acmepacket.com> From: "Bob Penfield" To: "SIPPING" Date: Tue, 22 Aug 2006 14:00:48 -0400 MIME-Version: 1.0 Content-Type: text/plain; format=flowed; charset="iso-8859-1"; reply-type=original Content-Transfer-Encoding: 7bit X-Priority: 3 X-MSMail-Priority: Normal X-Mailer: Microsoft Outlook Express 6.00.2900.2869 X-MimeOLE: Produced By Microsoft MimeOLE V6.00.2900.2962 X-Spam-Score: -1.9 (-) X-Scan-Signature: 9af087f15dbdd4c64ae6bbcdbc5b1d44 Cc: volkerh@bell-labs.com, Gonzalo Camarillo , Mary Barnes Subject: [Sipping] Review of draft-ietf-sipping-session-policy-framework-01 X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Draft: draft-ietf-sipping-session-policy-framework-01 Reviewer: Bob Penfield Review Date: August 22, 2006 Review Deadline: August 26, 2006 Status: Initial review Summary: This draft is on the right track but has open issues, described in the review. [ISSUE-1] Dependence on Config Framework documents On Page 6, Section 3.2: It says: A UA compliant to this specification MUST support the User Agent Profile Data Set for Media Policy [3] and the Schema for SIP User Agent Profile Data Sets [8]. This MUST makes [8] (draft-petrie-sipping-profile-datasets-03) a normative reference. That document is not (yet?) a working group document and has also expired. What is the plan for [8]? Also, I could not find a definition for "the MIME type of the Schema for SIP User Agent Profile Data Sets" in the other documents ([4] or [8]). [ISSUE-2] Normative text on disclosing answer to policy server On Page 14, Section 4.4.1. It says: A UA SHOULD generally disclose the offer and the answer to the policy server. However, the policy server may indicate on the policy channel (after receiving the offer) that the disclosure of the answer is not needed for this session. In this case, the UAC MAY skip the disclosure of the answer for this particular session. Why is the last sentence a MAY? If the policy server indicated it did not need/want to see the answer, the UA SHOULD NOT disclose the answer. Suggest rewording as follows: A UA SHOULD disclose the offer to the policy server. A UA SHOULD disclose the answer to the policy server, unless the policy server has indicated on the policy channel (when processing the offer) that the disclosure of the answer is not needed for this session. When disclosing the answer to the policy servers, the UA MUST contact the same policy servers it has contacted for the offer. Also, on Page 15, Section 4.4.3. It says: If the UAS receives an ACK containing an answer, it SHOULD contact the policy servers again with the answer. In this case, it MUST contact the same policy servers it has contacted for the offer. However, the policy server may have indicated in response to the offer that the disclosure of the answer is not needed for this session. In this case, the UAS MAY skip the disclosure of the answer for this particular session. Suggest: If the UAS receives an ACK containing an answer, it SHOULD contact the policy servers again with the answer, unless the policy server indicated in response to the offer that the disclosure of the answer is not needed for this session. When disclosing the answer to the policy servers, the UAS MUST contact the same policy servers it has contacted for the offer. [ISSUE-3] UA Behavior for PRACK needed In Section 4.4.1 (UAC Behavior) there should be procedures for an answer in a PRACK for a 1xx reliable provisional response which contained the offer. These would be similar to the procedures for ACK. This also applies to Section 4.4.3 (UAS Behavior). I wonder if it might be better to say "a request containing a session description", rather that specifying INVITE, UPDATE, ACK and PRACK as the only methods the procedures apply to (although ACK, PRACK, and INVITE responses still need special consideration). [ISSUE-4] Caching requirements for UAS Section 4.4.2 talks about UACs caching/storing the policy server URI. Shouldn't the UAS also cache/store the policy server URIs it learns from the Policy-Contact header(s) in an INVITE or UPDATE request for use in re-INVITE/UPDATE toward the calling party? [ISSUE-5] Header Definition and Syntax Since the policy server is always contacted using SUBSCRIBE/NOTIFY, the policy server URI should be a SIP or SIPS URI rather than an absoluteURI. Also, the header syntax should account for possible future header field parameters. [ISSUE-6] Security Considerations The Security Considerations section is a good start, but it needs work. It needs to be a bit more organized in identifying the specific treats using the terminology in RFC 3552. Also, normative requirements language needs to be used in the recommendations to mitigate the identified threats. [Editorial] Sections 4.4.1 thru 4.4.3 (UA Behavior) If caching applies to the UAS too, you may want to have a "User Agent Behavior" section that describes the general UA behavior and then UAC and UAS behavior sub-sections. Also, there is normative text in the UAC section that applies to UAs in general that should be in the general UA behavior section. [Editorial] Section 4.4.2: Caching of Policy Server URIs This section talks about "caching" policy server URIs and "storing" them in the session state. Readers might confuse these two. Suggest adding text which clarifies that "caching" means remembering the URIs across multiple sessions and that the URIs are "stored" in session state even when the "non-cacheable" parameter is present. [Editorial] On Page 4, Section 1, it says: Since these policies are not based on a specific session description, they can be created independent of an attempt to set up a session and only need to be conveyed to the user agent once (e.g. at the time the device is powered on). However, the UA is also informed when the policy changes. Suggest: ... and only need to be conveyed to the user agent when it initializes (e.g. at the time the device is powered on) and when the policies are changed. [Editorial] On Page 5, Section 3.1, it says: The local network may, for example, have policies that support the access network infrastructure (e.g. in a wireless network where bandwidth is scarce, a provider may restrict the bandwidth available to an individual user) Suggest the following text might read better: The local network may have policies that support the access network infrastructure. For example, in a wireless network where bandwidth is scarce, a provider may restrict the bandwidth available to an individual user. [Editorial] On Page 5, Section 3.1, it says: Thus, in most cases, a UA will receive session-independent policies from one or at most two policy servers. Suggest removal of "at most" as this implies that there will never be more than two policy servers. [Editorial] On Page 14, Section 4.4.2, there is a typo (s/is/it/) Change: The UAC SHOULD store the list of policy server URIs is has contacted To: The UAC SHOULD store the list of policy server URIs it has contacted [Editorial] On Page 18, Section 4.5, it says: If this update causes a change in the session description of a session, the UA may need to generate a re-INVITE or UPDATE message to re-negotiate the modified session description with its peer UA. You should probably mention that the re-INVITE/UPDATE message is generated in accordance with section 4.2 (i.e. it will contain a Policy-Id header with the policy server URI that sent the updated policy). cheers, (-:bob Robert F. Penfield Chief Software Architect Acme Packet, Inc. 71 Third Avenue Burlington, MA 01803 bpenfield@acmepacket.com _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Wed Aug 23 08:03:35 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GFrQA-0007CA-P3; Wed, 23 Aug 2006 08:01:22 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GFrQ8-0007Bz-Pr for sipping@ietf.org; Wed, 23 Aug 2006 08:01:20 -0400 Received: from sj-iport-6.cisco.com ([171.71.176.117]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GFrQ7-0003g4-6T for sipping@ietf.org; Wed, 23 Aug 2006 08:01:20 -0400 Received: from sj-dkim-3.cisco.com ([171.71.179.195]) by sj-iport-6.cisco.com with ESMTP; 23 Aug 2006 05:01:18 -0700 Received: from india-core-1.cisco.com (india-core-1.cisco.com [64.104.129.221]) by sj-dkim-3.cisco.com (8.12.11.20060308/8.12.11) with ESMTP id k7NC1GP5008908 for ; Wed, 23 Aug 2006 05:01:17 -0700 Received: from xbh-blr-412.apac.cisco.com (xbh-blr-412.cisco.com [64.104.140.149]) by india-core-1.cisco.com (8.12.10/8.12.6) with ESMTP id k7NC1FQJ014712 for ; Wed, 23 Aug 2006 12:01:16 GMT Received: from xmb-blr-418.apac.cisco.com ([64.104.140.147]) by xbh-blr-412.apac.cisco.com with Microsoft SMTPSVC(6.0.3790.1830); Wed, 23 Aug 2006 17:31:15 +0530 X-MimeOLE: Produced By Microsoft Exchange V6.5 Content-class: urn:content-classes:message MIME-Version: 1.0 Subject: RE: [Sipping] SIP Status code for communicating about UA inminimalstate-Post active dialog Date: Wed, 23 Aug 2006 17:31:14 +0530 Message-ID: X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: [Sipping] SIP Status code for communicating about UA inminimalstate-Post active dialog Thread-Index: AcZhKBgBoAnnLx0FTUeVxh4cXup2QwAzRQFQAAwsTWAZIW6zYA== From: "Rajeev Narang \(ranarang\)" To: X-OriginalArrivalTime: 23 Aug 2006 12:01:15.0473 (UTC) FILETIME=[D5C3E410:01C6C6AB] DKIM-Signature: a=rsa-sha1; q=dns; l=39660; t=1156334478; x=1157198478; c=relaxed/simple; s=sjdkim3002; h=Content-Type:From:Subject:Content-Transfer-Encoding:MIME-Version; d=cisco.com; i=ranarang@cisco.com; z=From:=22Rajeev=20Narang=20\(ranarang\)=22=20 |Subject:RE=3A=20[Sipping]=20SIP=20Status=20code=20for=20communicating=20about=20 UA=20inminimalstate-Post=20active=20dialog; X=v=3Dcisco.com=3B=20h=3DwosFHfHqJRP3GYnGYhPCosn9l0o=3D; b=VwIGBrypYITVmLFr77MtLhO0I90jcm6AvdQEw3+U3h7psZvqAs5mDSEMNXDgdMYaCA6pyB3J hmvMWIFk9WrPhpNew8wQt98IWo+MDVGtkKcB4JmfpBasyKsGMk4VuaKU; Authentication-Results: sj-dkim-3.cisco.com; header.From=ranarang@cisco.com; dkim=pass ( 59 extraneous bytes; sig from cisco.com verified; ); X-Spam-Score: 0.1 (/) X-Scan-Signature: f4cf803cc0263974f8c67805a39d1b18 X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Content-Type: multipart/mixed; boundary="===============1825973264==" Errors-To: sipping-bounces@ietf.org This is a multi-part message in MIME format. --===============1825973264== Content-class: urn:content-classes:message Content-Type: multipart/alternative; boundary="----_=_NextPart_001_01C6C6AB.D5AC6138" This is a multi-part message in MIME format. ------_=_NextPart_001_01C6C6AB.D5AC6138 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: quoted-printable The draft for the discussed issue is available for review in the link below. =20 http://www.ietf.org/internet-drafts/draft-ranarang-sipping-middialog-sip -status-00.txt =20 Regards, Rajeev ________________________________ From: Michael Hammer (mhammer)=20 Sent: Monday, April 17, 2006 7:38 PM To: Rajeev Narang (ranarang); Mahesh Govind Cc: sipping@ietf.org Subject: RE: [Sipping] SIP Status code for communicating about UA inminimalstate-Post active dialog The existence of proxies depends on which may have record-routed. If so, then they would likely not be forking the call again. But, in any case a 4xx or 450 would cause the dialog to end. But, that is the intent is it not? =20 =20 My understanding is that the intent is for the corresponding media not to also be ended. Perhaps this would also require some type of caller preferences and some type of new proxy directive. But, since media is "supposed to be" end-to-end the main issue is whether the two media endpoints understand the meaning of this form of "call, but not media" release. =20 I'll need to read the proposed draft to see how this all works. =20 Mike =20 =20 ________________________________ From: Rajeev Narang (ranarang)=20 Sent: Monday, April 17, 2006 4:06 AM To: Mahesh Govind Cc: sipping@ietf.org Subject: RE: [Sipping] SIP Status code for communicating about UA inminimalstate-Post active dialog =09 =09 Hi Mahesh, =20 It seems the 8.1.3.2 section is for the out of dialog behavior for initial call setup. =20 In our case mainly the usage for 450 response code will be mid-call (post established dialog) when the link from b2bua to remote party gets out of order. I think if b2bua doesn't understand 450 and treats it as 400, probably for mid-call 400 handling will be b2bua specific (may not tear down the call). =09 The mid-call transactions will probably be directly between UA's, so probably Proxy may not need to mandatory address the forwarding of the 450 status code. =09 So I think semi failure response (4xx) may be fine. =20 Thanks much, Rajeev =09 ________________________________ From: Mahesh Govind [mailto:vu3mmg@gmail.com]=20 Sent: Sunday, April 16, 2006 1:03 PM To: Rajeev Narang (ranarang) Cc: sipping@ietf.org Subject: Re: [Sipping] SIP Status code for communicating about UA in minimalstate-Post active dialog =09 =09 HI Rajeev, =20 I have a small query regarding to the response class , whether , we need to convey a semi successful response or a semi failure response state .ie Whether we need to send a 2xx class or 4xx class. If B2BUA1 receives a 450 and it doesn't understands 450 , will it consider the 450 as a 4xx and consider it as a failure ?(8.1.3.2 Unrecognized Responses of 3261) .Instead if it gets a 250 and even if it don't understands a 250 , it will consider it as a success case and may not tear the call..=20 Also I think if some proxy is in between it will take the 2xx response end to end ? With thanks and regards Mahesh =20 =20 =20 On 4/15/06, Rajeev Narang (ranarang) wrote: Mahesh, =20 I looked into the draft provided by you. It seems useful for the scenarios where GW/b2bua level "state" information is to be notified. And it will need the framework to be implemented and both way subscription/notification will be needed for every call. I think it will be useful if specific service state information is needed per call.=20 =20 My requirement is very basic for the sip dialogs and that situation could encounter in any b2bua's interworking with the phone. It is mainly for addressing situations where any dialog from b2bua to the phone could encounter the network loss. And assumption is media is direct between phones or through media agents. The b2bua to phone network loss must not impact the established media for currently active calls.=20 Basically even if one side of b2bua losses network to the phone, it doesn't mean that media/session between the phones (which could be direct on a different network) is also lost, so we won't want to terminate the call to the other side of b2bua.=20 =20 Effectively to support this, it needs to pass the appropriate mid dialog SIP Status code on the sip dialog to the other end of b2bua.=20 =20 We could send the BYE to remote UA with appropriate SIP Status info about this situation (basically n/w out of order with the phone).=20 So, it is upto the remote UA, whether to tear down the call or take it as a partial network issue and assume that media between the phones could still be active.=20 So the remote UA will wait for hangup from the phone on its side.=20 =20 Main idea is not to restore the call, but to preserve the established media until graceful hangup. =20 The current plan as per discussion with Mike, is to propose a draft with a new SIP Status code "450 Call in Minimal State" for this situation, which will reflect the status of the whole b2bua.=20 =20 Thanks much, Rajeev ________________________________ From: Mahesh Govind [mailto:vu3mmg@gmail.com]=20 Sent: Saturday, April 15, 2006 11:16 AM =09 To: sipping@ietf.org Subject: [Sipping] SIP Status code for communicating about UA in minimalstate-Post active dialog =09 =20 =09 sorry, the previous copy went to sip list accidentally . regds Mahesh =20 =20 =09 HI, We had earlier written a draft about service state of sip entities .Basically it is an event package. =09 (http://www.ietf.org/internet-drafts/draft-mahesh-sipping-svc-state-noti fy-00.txt ) The draft is not telling exactly about the requirement in the thread , but some thing similiar to it . If you have time please go through it .Currently we have defined only one state , Out of service, but we see that we can define other intermediate states (dormant ) and another final state inservice .also and this can be extended very easily. =09 =20 (PS :In the examples couple of bugs are there about the usage and implict subscription) =20 With thanks and regards =09 =09 Mahesh =09 =20 =20 =20 =20 On 4/14/06, Ranga wrote:=20 Correct me if I am wrong.=20 =09 As per the protocol, SIP responses are sent in response to the request but not generated. =09 In this perticular case, you could implement this on top of NOTIFY ( implicit SUBCRIPTION ).=20 =20 =09 Ranga =09 =09 =09 On 4/12/06, Michael Hammer (mhammer) < mhammer@cisco.com > wrote: =09 =09 My concern is that the SIP status code is status about the dialog transaction, which is not experiencing any difficulty. What you are reporting status on is something outside that dialog, which seems inappropriate.=20 =20 Mike =20 ________________________________ From: Rajeev Narang (ranarang)=20 Sent: Wednesday, April 12, 2006 10:14 AM To: Michael Hammer (mhammer); ' sipping@ietf.org ' Subject: RE: [Sipping] SIP Status code for communicating about UA in minimalstate- Post active dialog =09 =20 Right, these are B2BUAs. =20 Actually we need to keep B2BUA1 in knowledge of this specific state at B2BUA2 (as mentioned in scenario below). In case network link from B2BUA1 to B2BUA2 is also lost, we need to do some action at B2BUA1. And action at UA1 will be dependent on whether B2BUA2 was fully active or got into some minimal state due to its network issues with phone2.=20 So was thinking if we can use some standard SIP Status code from B2BUA2 to B2BUA1. =20 Thanks, Rajeev ________________________________ From: Michael Hammer (mhammer)=20 Sent: Wednesday, April 12, 2006 7:10 PM To: Rajeev Narang (ranarang); sipping@ietf.org Subject: RE: [Sipping] SIP Status code for communicating about UA in minimalstate- Post active dialog =09 =20 You are using B2BUAs are you not?=20 =20 They should have the intelligence to not involve dialog1 when something bad happens to the dialog with phone2. They are separate dialogs, are they not? I assume you are intending to restore the dialog with phone 2 at some point.=20 =20 I don't see the need to do anything new with dialog1. =20 Mike =20 ________________________________ From: Rajeev Narang (ranarang)=20 Sent: Wednesday, April 12, 2006 2:44 AM To: sipping@ietf.org Subject: [Sipping] SIP Status code for communicating about UA in minimalstate- Post active dialog =09 =20 Hi, =20 Say we have the following scenario:- =20 phone1-----B2BUA1----dialog1 (active) ----B2BUA2----phone2 =20 There is a session timers active on the dialog1. Say UA1 is the refresher. =20 An event occurs which results in B2BUA2 loosing network with phone2.=20 For our requirement we still want to allow dialog1 to be active with no interupption to the periodic session refresh. The B2BUA2 must keep on responding to the session refresh.=20 =20 The requirement is that we want to communicate to B2BUA1 about this recent situation with B2BUA2 in response to the session refresh. The B2BUA1 must be allowed to take some specific action based on the state of the B2BUA2. Also Proxy server between the UA's mustn't tear down the call.=20 =20 Basically the response code should be such that the session refresh procedure must go "on" but remote (refresher) must come to know about this situation. =20 From the available SIP status codes the following are choices:- 1) 200 OK:- Using this status code would indicate a normal situation, so doesn't seem appropriate. 2) 488 Not Acceptable Here:- Not sure if this will make sense for mid call. 3) 480 Temporarily Unavailable:- Not sure if this will make sense for mid call. =20 Any suggestions on the SIP Status code so that we can handle it in a standard way. =20 How about creating a new SIP Status code. Say:- "450 Call in Minimal/Preserved State" =20 Thanks much, Rajeev =09 _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping =20 This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip=20 Use sip@ietf.org for new developments of core SIP=20 =09 =09 =20 _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping =20 This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP=20 =09 =09 ------_=_NextPart_001_01C6C6AB.D5AC6138 Content-Type: text/html; charset="us-ascii" Content-Transfer-Encoding: quoted-printable
The draft for the discussed issue is = available for=20 review in the link below.
 
http://www.ietf.org/internet-drafts/draft-ranarang= -sipping-middialog-sip-status-00.txt
 
Regards,
Rajeev


From: Michael Hammer (mhammer) =
Sent:=20 Monday, April 17, 2006 7:38 PM
To: Rajeev Narang (ranarang); = Mahesh=20 Govind
Cc: sipping@ietf.org
Subject: RE: [Sipping] = SIP=20 Status code for communicating about UA inminimalstate-Post active=20 dialog

The existence of proxies depends on which may have=20 record-routed.  If so, then they would likely not be forking the = call=20 again.  But, in any case a 4xx or 450 would cause the dialog to = end. =20 But, that is the intent is it not? 
 
My understanding is that the intent is for the = corresponding media=20 not to also be ended.  Perhaps this would also require some type of = caller=20 preferences and some type of new proxy directive.  But, since media = is=20 "supposed to be" end-to-end the main issue is whether the two media = endpoints=20 understand the meaning of this form of "call, but not media"=20 release.
 
I'll need=20 to read the proposed draft to see how this all = works.
 
Mike
 
 


From: Rajeev Narang (ranarang)=20
Sent: Monday, April 17, 2006 4:06 AM
To: Mahesh=20 Govind
Cc: sipping@ietf.org
Subject: RE: [Sipping] = SIP=20 Status code for communicating about UA inminimalstate-Post active=20 dialog

Hi Mahesh,
 
It seems the 8.1.3.2=20 section is for the out of = dialog=20 behavior for initial call setup.
 
In our case=20 mainly the usage for 450 response code will be mid-call (post established = dialog)=20 when the link from b2bua to = remote=20 party gets out of = order.
I think=20 if b2bua doesn't understand 450 and treats it as 400, probably for = mid-call=20 400 handling will be b2bua specific = (may not=20 tear down the call).

The mid-call transactions will = probably be=20 directly between UA's, so probably Proxy may not need to mandatory = address=20 the forwarding of the 450 = status=20 code.
So I think semi failure response (4xx) may be = fine.
 
Thanks much,
Rajeev


From: Mahesh Govind = [mailto:vu3mmg@gmail.com]=20
Sent: Sunday, April 16, 2006 1:03 PM
To: Rajeev = Narang=20 (ranarang)
Cc: sipping@ietf.org
Subject: Re: = [Sipping] SIP=20 Status code for communicating about UA in minimalstate-Post active=20 dialog

HI Rajeev,
 
I have a small query regarding to the response class  , = whether , we=20 need to convey a  semi successful  response or a semi = failure=20 response state .ie Whether we need to send a 2xx class or  4xx=20 class.
If B2BUA1 receives a 450  and it doesn't understands 450 , = will it=20 consider the 450 as a 4xx and consider it as a failure ?(8.1.3.2 Unrecognized Responses of 3261) = .Instead if=20 it gets a 250 and even if  it don't understands a 250 , it will = consider=20 it as a success case and may not tear the call..
Also I think if some proxy is in between it will take the 2xx = response=20 end to end ?
With thanks and regards
Mahesh
 
 


 
On 4/15/06, Rajeev=20 Narang (ranarang) <ranarang@cisco.com> = wrote:=20
Mahesh,
 
I looked=20 into the draft provided by you. It seems useful for the scenarios = where=20 GW/b2bua level "state" information is to be notified. And it will = need the=20 framework to be implemented and both way subscription/notification = will be=20 needed for every call. I think it will be useful if specific service = state=20 information is needed per call.
 
My=20 requirement is very basic for the sip dialogs and that = situation could=20 encounter in any b2bua's interworking with the phone. It is mainly = for=20 addressing situations where any dialog from b2bua to the phone could = encounter the network loss. And assumption is media is direct = between phones=20 or through media agents. The b2bua to phone network loss must not = impact the=20 established media for currently active calls.
Basically = even if one=20 side of b2bua losses network to the phone, it doesn't mean that=20 media/session between the phones (which could be direct on a = different=20 network) is also lost, so we won't want to terminate the call = to the=20 other side of b2bua.
 
Effectively to support this, it needs to pass the = appropriate mid=20 dialog SIP Status code on the sip dialog to the other end of=20 b2bua.
 
We could=20 send the BYE to remote UA with appropriate SIP Status info about = this=20 situation (basically n/w out of order with the phone). =
So, it=20 is upto the remote UA, whether to tear down the call or take it as a = partial=20 network issue and assume that media between the phones could still = be=20 active.
So the=20 remote UA will wait for hangup from the phone on its side.=20
 
Main=20 idea is not to restore the call, but to preserve the established = media until=20 graceful hangup.
 
The current plan as per discussion with Mike, is to propose = a draft=20 with a new SIP Status code "450 Call in Minimal State" for this = situation,=20 which will reflect the status of the whole b2bua.=20
 
Thanks=20 much,
Rajeev

From: Mahesh Govind=20 [mailto:vu3mmg@gmail.com]=20
Sent: Saturday, April 15, 2006 11:16 AM

To: sipping@ietf.org
Subject: [Sipping] = SIP Status=20 code for communicating about UA in minimalstate-Post active=20 dialog

 
sorry, the previous copy went to sip list = accidentally .
regds
Mahesh
 

 

HI,
We had earlier written a draft about service state of sip = entities=20 .Basically it is an event package.
The draft is not telling  exactly about the requirement in = the=20 thread  , but some thing similiar to it . If you have time = please go=20 through it .Currently we have defined only one state = ,
Out of service, but we see that we can define  other = intermediate=20 states (dormant ) and another final state inservice .also and  = this can=20 be extended very easily.
 
(PS :In the examples couple of bugs are there about the usage = and=20 implict subscription)

 
With thanks and regards
Mahesh
 
 
 

 
On 4/14/06, Ranga=20 <rangarao.v@gmail.com> wrote:
Correct me if I am wrong.

As = per the=20 protocol, SIP responses are sent in response to the request but = not=20 generated.

In this perticular case, you could implement = this on top=20 of NOTIFY ( implicit SUBCRIPTION ).
 

Ranga


On = 4/12/06, Michael Hammer (mhammer) < = mhammer@cisco.com>=20 wrote:
My=20 concern is that the SIP status code is status about the dialog=20 transaction, which is not experiencing any difficulty.  = What you=20 are reporting status on is something outside that dialog, which = seems=20 inappropriate.
 
Mike
 


From: Rajeev Narang = (ranarang)=20
Sent: Wednesday, April 12, 2006 10:14 = AM
To:=20 Michael Hammer (mhammer); '=20 sipping@ietf.org'
Subject: RE: [Sipping] SIP = Status code=20 for communicating about UA in minimalstate- Post active=20 dialog

 
Right, these are=20 B2BUAs.
 
Actually we need=20 to keep B2BUA1 in knowledge of this specific state at B2BUA2 = (as=20 mentioned in scenario below). In case network link from B2BUA1 = to=20 B2BUA2 is also lost, we need to do some action at B2BUA1. And = action=20 at UA1 will be dependent on whether B2BUA2 was fully active or = got=20 into some minimal state due to its network issues with phone2. =
So was thinking=20 if we can use some standard SIP Status code from B2BUA2 to=20 B2BUA1.
 
Thanks,
Rajeev


From: Michael Hammer = (mhammer)=20
Sent: Wednesday, April 12, 2006 7:10 = PM
To:=20 Rajeev Narang (ranarang); sipping@ietf.org
Subject: RE: = [Sipping]=20 SIP Status code for communicating about UA in minimalstate- = Post=20 active dialog

 
You are=20 using B2BUAs are you not?
 
They=20 should have the intelligence to not involve dialog1 when = something bad=20 happens to the dialog with phone2.  They = are separate=20 dialogs, are they not?  I assume you are intending to = restore the=20 dialog with phone 2 at some point.
 
I don't=20 see the need to do anything new with = dialog1.
 
Mike
 


From: Rajeev Narang = (ranarang)=20
Sent: Wednesday, April 12, 2006 2:44 = AM
To: sipping@ietf.org
Subject: = [Sipping] SIP=20 Status code for communicating about UA in minimalstate- Post = active=20 dialog

 
Hi,
 
Say we have the = following=20 scenario:-
 
 phone1-----B2BUA1----dialog1 = (active)=20 ----B2BUA2----phone2
 
There is a session = timers active=20 on the dialog1. Say UA1 is the = refresher.
 
An event occurs which = results in=20 B2BUA2 loosing network with phone2.
For our requirement = we still want=20 to allow dialog1 to be active with no interupption to the = periodic=20 session refresh. The B2BUA2 must keep on responding to the = session=20 refresh.
 
The = requirement=20 is that we want to communicate to B2BUA1 about this recent = situation=20 with B2BUA2 in response to the session refresh. The B2BUA1 = must be=20 allowed to take some specific action based on the state of = the=20 B2BUA2. Also Proxy server between the UA's mustn't tear down = the=20 call.
 
Basically the = response code=20 should be such that the session refresh procedure must = go "on"=20 but remote (refresher) must come to know about = this=20 situation.
 
From the available = SIP status=20 codes the following are choices:-
1) 200 OK:- Using = this status=20 code would indicate a normal situation, so doesn't seem=20 appropriate.
2) 488 Not Acceptable = Here:-=20 Not sure if = this will=20 make sense for mid call.
3) 480 Temporarily = Unavailable:-=20 Not sure if this will make sense for mid = call.
 
Any suggestions on = the SIP Status=20 code so that we can handle it in a standard = way.
 
How about creating a = new SIP=20 Status code.
Say:-=20 "450 Call in Minimal/Preserved=20 State"
 
Thanks = much,
Rajeev

_______________________________________________
Sipping= =20 mailing list  =20 https://www1.ietf.org/mailman/listinfo/sipping
This list = is for=20 NEW development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions = on=20 current sip
Use sip@ietf.org for = new=20 developments of core SIP

=

 

______________________= _________________________
Sipping=20 mailing list  =20 https://www1.ietf.org/mailman/listinfo/sipping
This list is = for NEW=20 development of the application of SIP
Use sip-implementors@cs.columbia.edu for questions = on=20 current sip
Use sip@ietf.org for = new=20 developments of core SIP=20 =



------_=_NextPart_001_01C6C6AB.D5AC6138-- --===============1825973264== Content-Type: text/plain; charset="us-ascii" MIME-Version: 1.0 Content-Transfer-Encoding: 7bit Content-Disposition: inline _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP --===============1825973264==-- From sipping-bounces@ietf.org Thu Aug 24 05:44:50 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GGBjy-00020b-0e; Thu, 24 Aug 2006 05:43:10 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GGBjw-00020V-Or for sipping@ietf.org; Thu, 24 Aug 2006 05:43:08 -0400 Received: from tcmail23.telekom.de ([217.6.95.237]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GGBjv-00076U-9Z for sipping@ietf.org; Thu, 24 Aug 2006 05:43:08 -0400 Received: from g9jbr.mgb01.telekom.de (g9jbr.mgb01.telekom.de [164.20.31.6]) by tcmail21.dmz.telekom.de with ESMTP; Thu, 24 Aug 2006 11:42:59 +0200 Received: by G9JBR.mgb01.telekom.de with Internet Mail Service (5.5.2653.19) id ; Thu, 24 Aug 2006 11:42:59 +0200 Message-Id: From: "Huelsemann, Martin" To: sipping@ietf.org, mary.barnes@nortel.com, Gonzalo.Camarillo@ericsson.com Date: Thu, 24 Aug 2006 11:42:46 +0200 MIME-Version: 1.0 X-Mailer: Internet Mail Service (5.5.2653.19) X-Spam-Score: 0.1 (/) X-Scan-Signature: 9a2be21919e71dc6faef12b370c4ecf5 Cc: Oscar.Novo@ericsson.com Subject: [Sipping] draft-poetzl-sipping-call-completion-01 X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Content-Type: multipart/mixed; boundary="===============0344458906==" Errors-To: sipping-bounces@ietf.org This message is in MIME format. Since your mail reader does not understand this format, some or all of this message may not be legible. --===============0344458906== Content-Type: multipart/alternative; boundary="----_=_NextPart_001_01C6C761.A6E2CB82" This message is in MIME format. Since your mail reader does not understand this format, some or all of this message may not be legible. ------_=_NextPart_001_01C6C761.A6E2CB82 Content-Type: text/plain; charset="iso-8859-1" Hi all, a revised version of the Call Completion draft is available at http://www.ietf.org/internet-drafts/draft-poetzl-sipping-call-completion-01.txt . The intention of the draft is to propose solutions for the requirements for the ETSI TISPAN NGN services Completion of Call to Busy Subscribers and Completion of Call on No reply. The main change to the previous version is, that there is only one call completion event package now, which can be used for CCBS and CCNR. The indication for CCBS is 'Allow-events: call-completion' in a 486 Busy response, 'Allow-events: call-completion' in a 180 Ringing response is the indication for CCNR. Then I have added some more explanatory text and signalling flows, I hope that makes the requirements and solutions described in this draft clearer now. As you renember from the IETF 66 discussion, the main topic of discussion was, is it appropriate to use SUBSCRIBE/NOTIFY, besides subscribing to a call completion event itself, also for managing the call completion queue, for example for suspend and resume procedures. Questions and comments are very welcome. Best regards, Martin ------_=_NextPart_001_01C6C761.A6E2CB82 Content-Type: text/html; charset="iso-8859-1" Content-Transfer-Encoding: quoted-printable draft-poetzl-sipping-call-completion-01

Hi all,

a revised version of the Call = Completion draft is available at

http://www.ietf.org/internet-drafts/draft-poetzl-sipping-= call-completion-01.txt.

The intention of the draft is to = propose solutions for the requirements for the ETSI TISPAN NGN services = Completion of Call to Busy Subscribers and Completion of Call on No = reply.

The main change to the previous = version is, that there is only one call completion event package now, = which can be used for CCBS and CCNR. The indication for CCBS is = 'Allow-events: call-completion' in a 486 Busy response, 'Allow-events: = call-completion' in a 180 Ringing response is the indication for = CCNR.

Then I have added some more = explanatory text and signalling flows, I hope that makes the = requirements and solutions described in this draft clearer = now.



As you renember from the IETF 66 = discussion, the main topic of discussion was, is it appropriate to use = SUBSCRIBE/NOTIFY, besides subscribing to a call completion event = itself, also for managing the call completion queue, for example for = suspend and resume procedures.



Questions and comments are very = welcome.


Best regards, Martin



------_=_NextPart_001_01C6C761.A6E2CB82-- --===============0344458906== Content-Type: text/plain; charset="us-ascii" MIME-Version: 1.0 Content-Transfer-Encoding: 7bit Content-Disposition: inline _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP --===============0344458906==-- From sipping-bounces@ietf.org Thu Aug 24 14:56:24 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GGKMN-0002SF-Ej; Thu, 24 Aug 2006 14:55:23 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GGKML-0002S9-TP for sipping@ietf.org; Thu, 24 Aug 2006 14:55:21 -0400 Received: from sj-iport-6.cisco.com ([171.71.176.117]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GGKMK-0005Pf-KI for sipping@ietf.org; Thu, 24 Aug 2006 14:55:21 -0400 Received: from sj-dkim-2.cisco.com ([171.71.179.186]) by sj-iport-6.cisco.com with ESMTP; 24 Aug 2006 11:55:20 -0700 Received: from sj-core-5.cisco.com (sj-core-5.cisco.com [171.71.177.238]) by sj-dkim-2.cisco.com (8.12.11.20060308/8.12.11) with ESMTP id k7OItJSR013889 for ; Thu, 24 Aug 2006 11:55:19 -0700 Received: from xbh-sjc-231.amer.cisco.com (xbh-sjc-231.cisco.com [128.107.191.100]) by sj-core-5.cisco.com (8.12.10/8.12.6) with ESMTP id k7OItJ1E023936 for ; Thu, 24 Aug 2006 11:55:19 -0700 (PDT) Received: from xfe-sjc-211.amer.cisco.com ([171.70.151.174]) by xbh-sjc-231.amer.cisco.com with Microsoft SMTPSVC(6.0.3790.211); Thu, 24 Aug 2006 11:55:19 -0700 Received: from jmpolk-wxp.cisco.com ([10.21.121.92]) by xfe-sjc-211.amer.cisco.com with Microsoft SMTPSVC(6.0.3790.1830); Thu, 24 Aug 2006 11:55:18 -0700 Message-Id: <4.3.2.7.2.20060824135239.04016830@email.cisco.com> X-Sender: jmpolk@email.cisco.com X-Mailer: QUALCOMM Windows Eudora Version 4.3.2 Date: Thu, 24 Aug 2006 13:55:16 -0500 To: sipping@ietf.org From: "James M. Polk" Mime-Version: 1.0 Content-Type: text/plain; charset="us-ascii"; format=flowed X-OriginalArrivalTime: 24 Aug 2006 18:55:19.0136 (UTC) FILETIME=[D827C600:01C6C7AE] DKIM-Signature: a=rsa-sha1; q=dns; l=216; t=1156445719; x=1157309719; c=relaxed/simple; s=sjdkim2002; h=Content-Type:From:Subject:Content-Transfer-Encoding:MIME-Version; d=cisco.com; i=jmpolk@cisco.com; z=From:=22James=20M.=20Polk=22=20 |Subject:Q=20about=20draft-poetzl-sipping-call-completion-01=20; X=v=3Dcisco.com=3B=20h=3DYQfiGqhxGGwXzBJtiMQ/RxCZk5E=3D; b=rbpgAFP5NaL+Wr8Ge7cKjoI1fpmkFxM16/HRBC+d6L7qWxy/U/5OyOa9An493raFKwRq8K8F 9ritLZo4Po0KO0H8ay3SMAMrUExTz8OLoBGVripUlSiU0p/h1abd6A3E; Authentication-Results: sj-dkim-2.cisco.com; header.From=jmpolk@cisco.com; dkim=pass ( sig from cisco.com verified; ); X-Spam-Score: 0.0 (/) X-Scan-Signature: 6d62ab47271805379d7172ee693a45db Subject: [Sipping] Q about draft-poetzl-sipping-call-completion-01 X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Isn't this call-completion performing the same function of how we now subscribe to to someone's (rather their UA's) presence state, to be notified when the UA becomes available? If so, why is this necessary? _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Fri Aug 25 08:53:18 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GGbAS-0002QU-9F; Fri, 25 Aug 2006 08:52:12 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GGbAQ-0002QI-92 for sipping@ietf.org; Fri, 25 Aug 2006 08:52:10 -0400 Received: from mail1.telekom.de ([62.225.183.202]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GGbAO-0006se-RM for sipping@ietf.org; Fri, 25 Aug 2006 08:52:10 -0400 Received: from g9jbr.mgb01.telekom.de by G8SBV.dmz.telekom.de with ESMTP; Fri, 25 Aug 2006 14:52:05 +0200 Received: by G9JBR.mgb01.telekom.de with Internet Mail Service (5.5.2653.19) id ; Fri, 25 Aug 2006 14:52:05 +0200 Message-Id: From: "Huelsemann, Martin" To: jmpolk@cisco.com, sipping@ietf.org Subject: AW: [Sipping] Q about draft-poetzl-sipping-call-completion-01 Date: Fri, 25 Aug 2006 14:52:01 +0200 MIME-Version: 1.0 X-Mailer: Internet Mail Service (5.5.2653.19) Content-Type: text/plain; charset="iso-8859-1" Content-Transfer-Encoding: quoted-printable X-Spam-Score: 0.0 (/) X-Scan-Signature: 0ddefe323dd869ab027dbfff7eff0465 Cc: X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Hi James, all, although you could in principle use presence information to inform a = user about a the call-completion condition, basically the meaning of = the two events is a littlebit different. With presence information the = remote user can be informed that you are ready to accept Instant = Messages, or multimedia communication sessions in general, so that you = are basically available, but it doesn't mean that a communication = session establishment will be successfull, because there might be other = call-completion requests active. And as the presence server wouldn't know if the presence subscription = is really for receiving presence information or more for receiving = call-completion information, it wouldn't know whether to inform the = subscribing user in every case of a change in the presence state or, if = there are other subscribers, only to inform him when his position in = the call-completion queue allows it. So, to make it short, presence and call-completion events can provide = different information, and to have the possibility to receive this = information independent from each other, the two packages are needed. Hope this clarifies. Best regards, Martin -----Urspr=FCngliche Nachricht----- Von: James M. Polk [mailto:jmpolk@cisco.com]=20 Gesendet: Donnerstag, 24. August 2006 20:55 An: sipping@ietf.org Betreff: [Sipping] Q about draft-poetzl-sipping-call-completion-01=20 Isn't this call-completion performing the same function of how we now=20 subscribe to to someone's (rather their UA's) presence state, to be=20 notified when the UA becomes available? If so, why is this necessary? _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Fri Aug 25 13:08:05 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GGf9N-0006Uk-01; Fri, 25 Aug 2006 13:07:21 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GGf9L-0006Ue-1F for sipping@ietf.org; Fri, 25 Aug 2006 13:07:19 -0400 Received: from zcars04e.nortel.com ([47.129.242.56]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GGf9J-0001FO-LJ for sipping@ietf.org; Fri, 25 Aug 2006 13:07:19 -0400 Received: from zrc2hxm1.corp.nortel.com (zrc2hxm1.corp.nortel.com [47.103.123.72]) by zcars04e.nortel.com (Switch-2.2.0/Switch-2.2.0) with ESMTP id k7PH14I08213; Fri, 25 Aug 2006 13:01:05 -0400 (EDT) X-MimeOLE: Produced By Microsoft Exchange V6.5 Content-class: urn:content-classes:message MIME-Version: 1.0 Content-Type: text/plain; charset="US-ASCII" Content-Transfer-Encoding: quoted-printable Date: Fri, 25 Aug 2006 12:07:09 -0500 Message-ID: X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: Updated SIPPING WG document review status Thread-Index: AcbIaOX0edomSWyETpiSR1hWmIjlqQ== From: "Mary Barnes" To: X-Spam-Score: 0.0 (/) X-Scan-Signature: 2086112c730e13d5955355df27e3074b Cc: gonzalo.camarillo@ericsson.com Subject: [Sipping] Updated SIPPING WG document review status X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Hi all,=20 I've updated the spreadsheets with the new reviewers and proposed review dates, along with adding annotation reflecting the SIPPING WG drafts that are being moved to the SIP WG.=20 http://www.softarmor.com/sipping/process/review_procedure.html (root directory for reviews) http://www.softarmor.com/sipping/process/wg-review/sipping-rt.html (sorted by Review due date) http://www.softarmor.com/sipping/process/wg-review/sipping-rt-rev-due.ht ml (sorted by revision due date) http://www.softarmor.com/sipping/process/wg-review/sipping-rt-wglc.html (sorted by WGLC dates) http://www.softarmor.com/sipping/process/wg-review/sipping-rt-editor.htm l (sorted by document editor) I've summarized some miscellaneous changes/additions to the formatting at the end of the email for folks that are interested. To summarize, the following documents are being moved to the SIP WG due to protocol impacts: draft-ietf-sipping-consent-framework draft-ietf-sipping-session-policy-framework draft-ietf-sipping-multiple-refer=20 The highlights of the documents currently under Initial Review within SIPPING with a review deadline of 25 August 2006 (yes, that's today!) are: draft-ietf-sipping-session-policy-framework-01.txt (revised doc will be=20 submitted to SIP WG for Last call) draft-ietf-sipping-policy-package-01.txt (initial WG review) draft-ietf-sipping-media-policy-dataset-01.txt (initial WG review) With the following documents to be reviewed very soon:=20 - draft-ietf-sipping-capacity-attribute-00.txt (WGLC 18 Aug - 8 Sept)=20 Note: We still need at least 1 more reviewer for this one!) - draft-camarillo-sipping-consent-format-01.txt (WGLC 28 Aug- 18 Sept) - draft-camarillo-sipping-pending-additions-00.txt (WGLC 28 Aug- 18 Sept) Note: the above two docs will go through WGLC at the same time as WGLC of=20 consent-framework in SIP WG.=20 - draft-ietf-sipping-dialogusage-03.txt (WGLC 11 Sept - 2 Oct) Per the above, there are a few items that are already out of date. In addition, there is an error in the spreadsheet with regards to the draft-ietf-sipping-xcap-config. That document won't be moving to SIP, but there will likely be SIP impacts from that document (related to a new response code to be discussed on the SIP WG mailing list). =20 So, these spreadsheets will be very fluid and the goal is to update regularly (at least every 2 weeks). There are already some reviews for the policy documents that need to be uploaded.=20 The approach taken was to have no more than 2 sets of documents being reviewed at the same time. Ideally, we should have periods where there is only one document under review, but that wasn't possible and still get the documents that we need to get reviewed done within a reasonable time period before the IETF-67 meeting deadlines. Also, any of the documents that aren't available in time for their review slot will have to be rescheduled. There are a few WG documents that are ready for an initial review, that are lower priority that could be slipped into those review slots.=20 For folks that aren't official reviewers, your comments are always welcome and of course, the schedule doesn't preclude discussion or review of other drafts at anytime.=20 And, finally, feedback on the proposed milestone dates for WGLC and IESG would be appreciated as those need to be rolled up into the official WG milestones. =20 Regards, Mary ------------------------------------------------------------------------ ---------Miscellaneous notes on the formatting of the information: 1) The most useful basic sorting of the information seemed to be based "Review due", "Status" and "IESG Target".=20 2) Re-arranged the columns to make it more manageable for updating dates. 3) Added a column for primary "Editor". This lets us know who we need to ping about dates, etc. for documents. Thus, we've added a new sorting which is by "Editor". 4) As listed, there is a separate file which sorts by the Revision due date that is likely useful for the authors/editors in tracking their deliverables. =20 5) For documents that we're still working on, I've kept the old entry for the document and rather than pointing to the ID tracker (for working group documents only) the link now points to the old version maintained on the tools page. Also, just a note on the link into the ID tracker for pre-IESG documents - I've got a bug in the link that for some reason, you'll need to re-enter SIPPING as the WG and redisplay to get the ID tracker entry. =20 6) I've updated the date for the "Review due", with a new category indicating it's been completed: "Review Completed" or "Re-review" for the documents whose newer versions will be re-reviewed. I debated whether to keep the old date, but you can find the original date in the template of the review (i.e., just click on one of the Reviewer names). If folks feel this isn't sufficient, I could annotate the field with a date as I did for some of the official WG milestones. Note that this "Review Completed" also implies another version of the document is due, wherease the documents with a "Completed" will not have another WG review (unless substantive changes result from the IESG review). _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Fri Aug 25 21:06:25 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GGmc9-0004MH-HG; Fri, 25 Aug 2006 21:05:33 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GGmc8-0004Jo-Eu for sipping@ietf.org; Fri, 25 Aug 2006 21:05:32 -0400 Received: from zcars04f.nortel.com ([47.129.242.57]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GGmc6-0002DT-2g for sipping@ietf.org; Fri, 25 Aug 2006 21:05:32 -0400 Received: from zrc2hxm1.corp.nortel.com (zrc2hxm1.corp.nortel.com [47.103.123.72]) by zcars04f.nortel.com (Switch-2.2.6/Switch-2.2.0) with ESMTP id k7Q15QT08209; Fri, 25 Aug 2006 21:05:26 -0400 (EDT) X-MimeOLE: Produced By Microsoft Exchange V6.5 Content-class: urn:content-classes:message MIME-Version: 1.0 Content-Type: text/plain; charset="US-ASCII" Content-Transfer-Encoding: quoted-printable Date: Fri, 25 Aug 2006 20:05:10 -0500 Message-ID: X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: Review of draft-ietf-sipping-session-policy-framework-01 Thread-Index: AcbIq61+JCB+x41cRs+x9SRFLVkHrw== From: "Mary Barnes" To: X-Spam-Score: 1.1 (+) X-Scan-Signature: b132cb3ed2d4be2017585bf6859e1ede Cc: sipping@ietf.org, gonzalo.camarillo@ericsson.com Subject: [Sipping] Review of draft-ietf-sipping-session-policy-framework-01 X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Draft: draft-ietf-sipping-session-policy-framework-01 Reviewer: Mary Barnes Review Date: 25 Aug 06 Review Deadline: 25 Aug 06 Status: Initial review=20 Summary: This draft is almost ready for publication, but needs some clarifications and has nits that should be fixed before publication. The requested clarifications are listed first: Section 4.4.2, 1st paragraph, last sentence. It mentions that "The UAC SHOULD contact the cached local policy server URI when creating a new INVITE or UPDATE request, before they are sent." I'm wondering if this shouldn't be a MUST or the conditions and consequences for when it would be okay to not contact the cached local policy should be stated.=20 Section 4.4.4, 3rd paragraph, last sentence. Can that SHOULD be changed to a MUST? Otherwise, the consequences of not removing that policy server URI should be described. Editorial nits: --------------- General: Add figure #s and appropriate figure references for the flows. Abstract: - I would suggest combining the 2nd and 3rd sentences into a more definitive statement about what the functionality in this document. OLD: However, there is currently no standard mechanism by which a proxy can define=20 or influence policies on sessions such as the codecs or media types to be used. This document specifies a framework for SIP session policies that provides this capability to proxies.=20 NEW: This document specifies a framework for SIP session policies that provides a standard mechanism by which a proxy can define=20 or influence policies on sessions, such as the codecs or media types to be used.=20 - It might also be useful to highlight the new protocol mechanisms in the abstract.=20 Section 1: - 6th paragraph, 1st sentence: I would suggest adding "all" to qualify "the SIP sessions": OLD: Session-independent policies on the other hand are policies that are created independent of a session and generally apply to the SIP sessions set up by a user agent. NEW:=20 Session-independent policies on the other hand are policies that are created independent of a session and generally apply to all the SIP sessions set up by a user agent. Section 3:=20 - 1st sentence: I would suggest adding "all" to qualify "the sessions": OLD: Session-independent policies are policies that are created independent of a session and generally apply to the sessions a user agent is setting up.=20 NEW: Session-independent policies are policies that are created independent of a session and generally apply to all the sessions a user agent is setting up.=20 - 2nd sentence: I would suggest changing "the sessions" to "any sessions": OLD: They typically remain stable for a longer period of time and apply=20 to the sessions set up while they are valid. NEW:=20 They typically remain stable for a longer period of time and apply=20 to any sessions set up while they are valid. Section 3.2: - 2nd paragraph, 3rd sentence: the use of the semi-colon doesn't make sense. Should that be replaced with "and"? =20 Section 4.2: - 2nd paragraph, 3rd sentence: remove the text in parenthesis (2 occurences) and insert commas: OLD:=20 It enables the use of separate encryption mechanisms on the signaling path (to secure the communication between endpoints) and on the policy channel (to secure the communication between endpoint and policy server). NEW: It enables the use of separate encryption mechanisms on the signaling path to secure the communication between endpoints, and on the policy channel to secure the communication between endpoint and policy server. =20 Section 4.4.3: - 1st paragraph, 1st sentence: change lowercase "may" to uppercase "MAY" Section 5, Security Considerations: - 4th paragraph, 1st sentence. Suggest to change the lower case should to SHOULD.=20 - 5th paragraph, 2nd sentence, insert "a" in the sentence. =20 OLD: An attacker can use this mechanism to refer a UA to compromised policy server. NEW: An attacker can use this mechanism to refer a UA to a compromised policy=20 server. =20 _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Fri Aug 25 21:21:22 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GGmr6-0004Yy-Gp; Fri, 25 Aug 2006 21:21:00 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GGmr5-0004Yg-9D for sipping@ietf.org; Fri, 25 Aug 2006 21:20:59 -0400 Received: from zcars04f.nortel.com ([47.129.242.57]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GGmr2-0003Ko-RR for sipping@ietf.org; Fri, 25 Aug 2006 21:20:59 -0400 Received: from zrc2hxm1.corp.nortel.com (zrc2hxm1.corp.nortel.com [47.103.123.72]) by zcars04f.nortel.com (Switch-2.2.6/Switch-2.2.0) with ESMTP id k7Q1KqT09032; Fri, 25 Aug 2006 21:20:52 -0400 (EDT) X-MimeOLE: Produced By Microsoft Exchange V6.5 Content-class: urn:content-classes:message MIME-Version: 1.0 Date: Fri, 25 Aug 2006 20:20:36 -0500 Message-ID: X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: Review of draft-ietf-sipping-media-policy-dataset-01 Thread-Index: AcbIrdTwfaKYuPLVQtmBbVvnL/zf7w== From: "Mary Barnes" To: X-Spam-Score: 0.1 (/) X-Scan-Signature: 93df555cbdbcdae9621e5b95d44b301e Cc: sipping@ietf.org, gonzalo.camarillo@ericsson.com Subject: [Sipping] Review of draft-ietf-sipping-media-policy-dataset-01 X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Content-Type: multipart/mixed; boundary="===============0437795865==" Errors-To: sipping-bounces@ietf.org This is a multi-part message in MIME format. --===============0437795865== Content-class: urn:content-classes:message Content-Type: multipart/alternative; boundary="----_=_NextPart_001_01C6C8AD.D5775793" This is a multi-part message in MIME format. ------_=_NextPart_001_01C6C8AD.D5775793 Content-Type: text/plain; charset="US-ASCII" Content-Transfer-Encoding: quoted-printable Draft: draft-ietf-sipping-media-policy-dataset-01 Reviewer: Mary Barnes Review Date: 25 Aug 2006 Review Deadline: 25 Aug 06 Status: Initial review=20 Summary: This draft is on the right track for publication, but has nits and open issues that should be fixed before publication. Caveat: I'm not an XML guru and I'm assuming that the XML will be validated by a tool and reviewed by XML experts.=20 Section 1: - last pargraph, 2nd sentence: insert "be" before "merged" OLD: These documents need to merged into a single document the user agent can work with.=20 NEW: These documents need to be merged into a single document the user agent can work with.=20 Section 3: - OPEN ISSUE: has a typo - not sure why the word "might" is in that sentence. I'm not enough of an XML guru to know the pros/cons of reusing the MIME type from the SIP User Agent Profile Dataset. But, it seems that since you're building on that profile that it would be reasonable to just reuse that same MIME type. Are there any XML gurus that know why that's not a good idea?=20 Section 5.1: - 1st paragraph, 2nd sentence: remove "to" and capitalize "must". OLD: The stream to which a property applies to must be identifiable=20 through a label [6]. NEW: The stream to which a property applies MUST be identifiable=20 through a label [6]. - 3rd paragraph, last sentence: suggest to change the statement to be more definitive (I can't see that this is an example).=20 OLD: These labels are, for example, part of the session description. NEW: These labels are MUST be part of the session description. Section 11, references: - [13] should reference the BEHAVE WG TURN document: http://tools.ietf.org/wg/behave/draft-ietf-behave-turn/draft-ietf-behave -turn-01.txt ------_=_NextPart_001_01C6C8AD.D5775793 Content-Type: text/html; charset="US-ASCII" Content-Transfer-Encoding: quoted-printable Review of draft-ietf-sipping-media-policy-dataset-01

Draft:  = draft-ietf-sipping-media-policy-dataset-01
Reviewer: Mary Barnes
Review Date: 25 Aug 2006
Review Deadline: 25 Aug = 06
Status: Initial review

Summary: This draft is on the = right track for publication, but has nits and open issues that should be = fixed before publication.

Caveat: I'm not an XML guru and = I'm assuming that the XML will be validated by a tool and reviewed by = XML experts.

Section 1:
- last pargraph, 2nd sentence: = insert "be" before "merged"
OLD:
   These documents = need to merged into a single document the
   user agent can work = with.
NEW:
   These documents = need to be merged into a single document the
   user agent can work = with.

Section 3:
- OPEN ISSUE: has a typo - not = sure why the word "might" is in that sentence. I'm not enough = of an XML guru to know the pros/cons of reusing the MIME type from the = SIP User Agent Profile Dataset.  But, it seems that since you're = building on that profile that it would be reasonable to just reuse that = same MIME type. Are there any XML gurus that know why that's not a good = idea?

Section 5.1:
- 1st paragraph, 2nd sentence: = remove "to" and capitalize "must".
OLD:
   The stream to which = a property applies to must be identifiable
   through a label = [6].
NEW:
   The stream to which = a property applies MUST be identifiable
   through a label = [6].

- 3rd paragraph, last sentence: = suggest to change the statement to be more definitive (I can't see that = this is an example).

OLD:
   These labels are, = for example, part of the session description.
NEW:
   These labels are = MUST be part of the session description.

Section 11, references:
- [13] should reference the = BEHAVE WG TURN document:
http://tools.ietf.org/wg/behave/draft-ietf-behave-turn/draft-ietf-be= have-turn-01.txt






------_=_NextPart_001_01C6C8AD.D5775793-- --===============0437795865== Content-Type: text/plain; charset="us-ascii" MIME-Version: 1.0 Content-Transfer-Encoding: 7bit Content-Disposition: inline _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP --===============0437795865==-- From sipping-bounces@ietf.org Fri Aug 25 22:04:04 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GGnVO-0000lk-JW; Fri, 25 Aug 2006 22:02:38 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GGnVN-0000lX-9C for sipping@ietf.org; Fri, 25 Aug 2006 22:02:37 -0400 Received: from zrtps0kp.nortel.com ([47.140.192.56]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GGnVK-0000dV-7c for sipping@ietf.org; Fri, 25 Aug 2006 22:02:37 -0400 Received: from zrc2hxm1.corp.nortel.com (zrc2hxm1.corp.nortel.com [47.103.123.72]) by zrtps0kp.nortel.com (Switch-2.2.6/Switch-2.2.0) with ESMTP id k7Q22PZ03184; Fri, 25 Aug 2006 22:02:25 -0400 (EDT) X-MimeOLE: Produced By Microsoft Exchange V6.5 Content-class: urn:content-classes:message MIME-Version: 1.0 Content-Type: text/plain; charset="US-ASCII" Content-Transfer-Encoding: quoted-printable Date: Fri, 25 Aug 2006 21:02:25 -0500 Message-ID: X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: Review of draft-ietf-sipping-policy-package-01 Thread-Index: AcbIs6zMSwiqHk6RSAydr0yUxWeQzw== From: "Mary Barnes" To: X-Spam-Score: 0.0 (/) X-Scan-Signature: b2809b6f39decc6de467dcf252f42af1 Cc: sipping@ietf.org, gonzalo.camarillo@ericsson.com Subject: [Sipping] Review of draft-ietf-sipping-policy-package-01 X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Draft: draft-ietf-sipping-policy-package-01 Reviewer: Mary Barnes Review Date: 25 Aug 2006 Review Deadline: 25 Aug 2006 Status: Initial review=20 Summary: This draft is on the right track but has open issues, described in the review. Section 1: - 2nd paragraph, last sentence: suggest to qualify "the SIP sessions" with "all" OLD: Session-independent policies on the other hand are policies that are=20 created independent of a session and generally apply to the SIP sessions=20 set up by a user agent (see [6]). NEW: Session-independent policies on the other hand are policies that are=20 created independent of a session and generally apply to all the SIP sessions=20 set up by a user agent (see [6]). Section 3.3:=20 - 1st paragraph, 3rd & 4th sentences. I found this a bit to confusing to read. Suggest to precede the last sentence with "However," to link that to the previous sentence.=20 OLD: In this event package, the Request-URI, the event package name and event parameters are not sufficient to determine the resource a subscription is for. With the session description in the SUBSCRIBE body, the notifier can generate the requested policy decision and create policy events for this resource. NEW: In this event package, the Request-URI, the event package name and event parameters are not sufficient to determine the resource a subscription is for. However, with the session description in the SUBSCRIBE body, the notifier can generate the requested policy decision and create policy events for this resource. - OPEN ISSUE: I like the idea of using XML (versus a copy of the SDP) and would be curious if anyone has any objections?=20 Section 3.4: - 1st paragraph, 1st sentence: suggest to remove the parenthetical statement and clarify and add the condition(s) under which a subscription MAY be terminated early or add an example. The current wording of "of course" leaves alot for the reader.=20 Section 3.5: - 4th paragraph, last two sentences. I'm having difficulty seeing how these two sentences relate. It first mentions that the notifier can't count on a user understanding other format extensions, which is reasonable. But, I can't see how that last statement on rationale explaining that a modified format is often returns justifies why the notifier can't count on a user understanding other format extensions. It would seem to me that the last sentence would fit much better if it were moved after the first sentence. So, I'd propose a rewrite of that paragraph something like the following(keeping the first sentence the same), although, I'm not certain it entirely captures the points you wanted to make either: OLD: If the notifier uses the same format in NOTIFY bodies that was used by the subscriber in the SUBSCRIBE body (e.g. "application/ session-policy+xml"), the notifier can expect that the subscriber supports all format extensions that were used in the SUBSCRIBE body. However, the notifier cannot assume that the subscriber supports other extensions beyond that. If the notifier uses other format extensions, it cannot count on the fact that they will be understood by the subscriber. The rationale behind this is that the notifier will often return a modified version of the document that was submitted by the subscriber. NEW: If the notifier uses the same format in NOTIFY bodies that was used by the subscriber in the SUBSCRIBE body (e.g. "application/ session-policy+xml"), the notifier can expect that the subscriber supports all format extensions that were used in the SUBSCRIBE body. The notifier SHOULD NOT assume that the subscriber supports other extensions beyond that. The notifier MAY return a modified=20 version of the document that was submitted by the subscriber.=20 However, if the notifier uses other format extensions, it SHOULD NOT=20 expect that they will be understood by the subscriber. =20 Section 3.6: - OPEN ISSUE: This is a really good and important question. My initial reaction is yes, we should be able to define at least some basic default examples to justify and validate why this functionaltiy is even useful or needed. There used to be a scenarios document and I'm wondering if it wouldn't be useful to revamp that. Overall, I found this document to be extremely abstract and feel that there's a need for some more concrete examples in this document in general. =20 Section 3.7:=20 - 3rd paragraph, 1st sentence: "Responding timely..." -> "A timely response..." - 3rd paragraph, last sentence: break out that parenthetical statement into its own sentence.=20 Section 3.8: - 1st paragraph, last sentence: Here's where a more concrete example would be really useful.=20 - 3rd paragraph, 3rd sentence: How would the determination be made as to whether a notifier would "expect that the session can be admitted at a later point in time"? =20 Section 3.9:=20 - 3rd paragraph, 2nd sentence: Could that SHOULD be a MUST? If not, there should be a statement about the consequences of the subscriber not applying the new policy decision to this session, or least an example of when a subscriber might not want to.=20 - 3rd paragraph, 3rd sentence: It's not clear to me why a subscriber would ever need to terminate a session based upon an updated policy. Could you add an example here? =20 Section 3.13: - 1st paragraph, last sentence: insert "a" between "on" and "transport". - Example Flow. It would be really informative to show the detailed XML in terms of understanding the changes that might happen between the SUBSCRIBE and NOTIFY. And also, example for the local and remote. I know it's tedious, but despite not being normative, examples tend to be really useful for developers to really "get it". You could borrow the example that is given in section 7 of the media policy dataset document and build from there. =20 =20 _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Sat Aug 26 11:36:07 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GH09s-0003bI-Ll; Sat, 26 Aug 2006 11:33:16 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GH09q-0003QI-CX for sipping@ietf.org; Sat, 26 Aug 2006 11:33:14 -0400 Received: from host10.216.41.24.conversent.net ([216.41.24.10] helo=acmepacket.com) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GH09p-0000K1-15 for sipping@ietf.org; Sat, 26 Aug 2006 11:33:14 -0400 Received: from BPenfield [10.0.200.6] by acmepacket.com with ESMTP (SMTPD-9.03) id A8BD0170; Sat, 26 Aug 2006 11:29:01 -0400 Message-ID: <006201c6c924$5b5635e0$800101df@acmepacket.com> From: "Bob Penfield" To: "SIPPING" Date: Sat, 26 Aug 2006 11:29:01 -0400 MIME-Version: 1.0 Content-Type: text/plain; format=flowed; charset="iso-8859-1"; reply-type=original Content-Transfer-Encoding: 7bit X-Priority: 3 X-MSMail-Priority: Normal X-Mailer: Microsoft Outlook Express 6.00.2900.2869 X-MimeOLE: Produced By Microsoft MimeOLE V6.00.2900.2962 X-Spam-Score: 0.1 (/) X-Scan-Signature: 7da5a831c477fb6ef97f379a05fb683c Cc: volkerh@bell-labs.com, Mary Barnes , Gonzalo Camarillo Subject: [Sipping] Review of draft-ietf-sipping-policy-package-01 X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Draft: draft-ietf-sipping-policy-package-01 Reviewer: Bob Penfield Review Date: August 26, 2006 Review Deadline: August 25, 2006 Status: Initial review Summary: This draft is on the right track but has open issues, described in the review. [Technical] Correlation of policy decision and INVITE request These are really a general comments on the whole set of session policy documents. The session-specific policy documents talk about including only session description information. However, I believe that additional information from the SIP signaling needs to be included in order for this to work. The existing framework and event package may support this, but there is no discussion of how it would work. There are a couple of reasons for this: 1. In order to verify that the policy server was properly contacted and that the policy server did actually authorize the session, the proxy and policy server would need to share data in order to correlate the INVITE with the policy decision. Thus the policy server would need to receive the dialog identifying information, or the policy decision would need to include some sort of token that was included in the INVITE to correlate the request with policy decision. 2. When a media intermediary needs to be included, the routing decision made by the proxy could affect which media intermediary is chosen. This would require that the request target be disclosed to the policy server, or that the proxy include some sort of token or cookie in a (non-cacheable) policy-server URI that it passes back to the UA. [Technical] In Section 3.3, it says: A SUBSCRIBE for the session-specific policy package SHOULD contain a body that describes a SIP session. Should this be a MUST? Under what circumstances would it be valid to not include a body in the SUBSCRIBE? [Technical] Open Issue in Section 3.3 on page 5 Is this still an open issue? I think using XML is the right choice. Much of the text in the open issue should be retained since it provides the motivation for choosing XML over SDP. [Technical] On page 5, section 3.3, it says: When the remote description becomes available, the subscriber SHOULD refresh the subscription by sending another SUBSCRIBE request, which then contains the local and the remote session description. The subscriber MAY skip sending the remote session description to the notifier if it has received a NOTIFY with the "local-only" parameter. A notifier will typically include this parameter in a NOTIFY when it has received the local session description and does not need to see the remote session description. Suggest the following wording since the remote session description should not be included if the notifier indicated that it is not needed: When the remote description becomes available, the subscriber SHOULD refresh the subscription by sending another SUBSCRIBE request, which then contains the local and the remote session description, unless the subscriber received a NOTIFY with the "local-only" parameter indicating that it does not need to see the remote session description. Alternatively, change "MAY skip sending" to "SHOULD NOT send" [Technical] On page 9, section 3.9, it says: However, the subscriber MAY still keep up the subscription to session-specific policies for this session since the policy decision may change and the session may be admitted at a later time. Why would the subscription kept alive if the session cannon be initiated? Does this mean that the subscriber might change the session description and try again? Wouldn't that be done more or less immediately? Shouldn't the subscriber terminate the subscription if it aborts the session initiation? [Technical] On page 9, section 3.9, it says: If the subscriber receives a NOTIFY that contains the "local-only" event parameter, it MAY stop inserting the remote session description in SUBSCRIBE requests within this subscription. It MAY skip refreshing the subscription in order to convey information about the remote session description to the notifier. Being consistent with my prior comments, the subscriber SHOULD NOT include the remote description when the notifier has indicated that it is not needed. Suggest changing to: If the subscriber receives a NOTIFY that contains the "local-only" event parameter, it SHOULD NOT include the remote session description in subsequent SUBSCRIBE requests within this subscription. [Editorial] Introduction The Intro seems a bit wordy and repetitive. For example, there are at least three places where it talks about a user agent disclosing the session information to the policy server. I think it needs to be more concise. [Editorial] on page 8, in section 3.7, it says: However, subscriptions may also be established by applications and automata (e.g. a conference server). I don't think that "and automata" adds anything to this sentence. How is it different than "applications"? [Editorial] on page 8, in section 3.7, it says: Responding timely to a SUBSCRIBE request is crucial for this event package. Suggest "Responding in a timely manner to ..." or "Timely responses to ..." cheers, (-:bob Robert F. Penfield Chief Software Architect Acme Packet, Inc. 71 Third Avenue Burlington, MA 01803 bpenfield@acmepacket.com _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Mon Aug 28 10:29:07 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GHi52-00014O-Bo; Mon, 28 Aug 2006 10:27:12 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GHi50-000146-Ss for sipping@ietf.org; Mon, 28 Aug 2006 10:27:10 -0400 Received: from stsc1260-eth-s1-s1p1-vip.va.neustar.com ([156.154.16.129] helo=chiedprmail1.ietf.org) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GHi50-0000DR-RB for sipping@ietf.org; Mon, 28 Aug 2006 10:27:10 -0400 Received: from m5-143.126.com ([202.108.5.143]) by chiedprmail1.ietf.org with smtp (Exim 4.43) id 1GHi4m-0005Jf-AW for sipping@ietf.org; Mon, 28 Aug 2006 10:26:59 -0400 Received: from jiyang (unknown [221.218.212.65]) by smtp6 (Coremail) with SMTP id wKjSjzuATBMp_fJEdY3vAQ==.8240S2; Mon, 28 Aug 2006 22:26:49 +0800 (CST) Date: Mon, 28 Aug 2006 22:27:24 +0800 From: "jiyangbupt" To: "sipping" Message-ID: <200608282227236871618@126.com> X-mailer: Foxmail 6, 0, 101, 11 [cn] Mime-Version: 1.0 X-Spam-Score: 2.0 (++) X-Scan-Signature: 52e1467c2184c31006318542db5614d5 Subject: [Sipping] post the sipping mailist X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Content-Type: multipart/mixed; boundary="===============0857388315==" Errors-To: sipping-bounces@ietf.org This is a multi-part message in MIME format. --===============0857388315== Content-Type: multipart/alternative; boundary="=====003_Dragon760478716528_=====" This is a multi-part message in MIME format. --=====003_Dragon760478716528_===== Content-Type: text/plain; charset="gb2312" Content-Transfer-Encoding: 7bit Best regards JI Yang 2006-08-28 --=====003_Dragon760478716528_===== Content-Type: text/html; charset="gb2312" Content-Transfer-Encoding: 7bit
 
 
 
Best regards
JI Yang
2006-08-28
--=====003_Dragon760478716528_=====-- --===============0857388315== Content-Type: text/plain; charset="us-ascii" MIME-Version: 1.0 Content-Transfer-Encoding: 7bit Content-Disposition: inline _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP --===============0857388315==-- From sipping-bounces@ietf.org Mon Aug 28 16:42:06 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GHnuu-0002gj-K8; Mon, 28 Aug 2006 16:41:08 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GHnut-0002ge-Dg for sipping@ietf.org; Mon, 28 Aug 2006 16:41:07 -0400 Received: from [2001:5c0:8fff:fffe::4c49] (helo=nostrum.com) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GHnuq-0006tb-1B for sipping@ietf.org; Mon, 28 Aug 2006 16:41:07 -0400 Received: from [172.17.2.252] (vicuna.estacado.net [72.1.129.69]) (authenticated bits=0) by nostrum.com (8.13.6/8.13.6) with ESMTP id k7SKejLu042609 (version=TLSv1/SSLv3 cipher=RC4-SHA bits=128 verify=NO); Mon, 28 Aug 2006 15:40:48 -0500 (CDT) (envelope-from rjsparks@nostrum.com) In-Reply-To: <44E5BCD9.9050600@cisco.com> References: <70798CF421F00F4DA018059E5B7EEB8C7A5FD8@sonusinmail01.sonusnet.com> <44E5BCD9.9050600@cisco.com> Mime-Version: 1.0 (Apple Message framework v752.2) Content-Type: text/plain; charset=UTF-8; delsp=yes; format=flowed Message-Id: <045D8825-41BE-4BDB-B6BA-7158BB9E7E92@nostrum.com> Content-Transfer-Encoding: quoted-printable From: Robert Sparks Subject: Re: [Sipping] Re: [Sip] comments on draft-ietf-sipping-dialogusage-02.txt Date: Mon, 28 Aug 2006 15:40:49 -0500 To: Paul Kyzivat X-Mailer: Apple Mail (2.752.2) Received-SPF: pass (nostrum.com: 72.1.129.69 is authenticated by a trusted mechanism) X-Spam-Score: 0.0 (/) X-Scan-Signature: 71f780ffdd80c541d3e75aa5f2710d3d Cc: sipping@ietf.org X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Sorry for the lag on replying to this thread. To Nataraju's primary concern (as I understand it) : We don't really =20= have such a thing as response authentication in SIP (see the many long discussions during =20 work on Identity). The concern you express about a MitM injecting a response that tears =20 things down is a known risk and work on mitigating that risk does not belong in this =20 document. I _can_ call it out in the security considerations section to remind implementer's that it is =20 there and that they need to think about the implications. Paul's response is pointing to a different place in the handshake - =20 that something shouldn't legitimately issue a 404/410/etc. when a 401 would be more =20 appropriate. I'm going through the document today to prep a version for last call early next month. =20 I'll look for a place to call this out, but off the top of my head, I don't see anywhere that it =20 would naturally fit. RjS On Aug 18, 2006, at 8:12 AM, Paul Kyzivat wrote: > > > B, Nataraju wrote: >> I was thinking making it explicit would be better. At least shall =20 >> we mention this one as normative information? Otherwise many more =20 >> others might think the same way (which I thought earlier) and =20 >> develop the buggy software... > > Well, I *thought* it was clear in 3261. But I went back to check, =20 > and it is far from clear. > > Section 8.2 says: > > UASs SHOULD process the requests in the order of the steps that > follow in this section (that is, starting with authentication, then > inspecting the method, the header fields, and so on throughout the > remainder of this section). > > which certainly implies that authentication should be done first. =20 > Unfortunately, the subsections that follow this intro don't mention =20= > authentication at all. :-( > > And, this section pertains to requests outside a dialog, so it =20 > doesn't really apply anyway. > > Section 12, which defines behavior for dialog establishment and =20 > requests within dialogs doesn't mention authentication or =20 > authorization at all. > > So I guess it isn't at all clear that that if 401 is applicable to =20 > a request then it should be issued in preference to 404, 410, or =20 > 416, and that this should be true inside a dialog as well as outside. > > It would be good to get some input from others on this. > > Paul > >> Thanks, >> Nataraju A B >>> -----Original Message----- >>> From: Paul Kyzivat [mailto:pkyzivat@cisco.com] >>> Sent: Thursday, August 17, 2006 6:29 PM >>> To: B, Nataraju >>> Cc: sipping@ietf.org >>> Subject: Re: [Sip] comments on draft-ietf-sipping-dialogusage-02.txt >>> >>> >>> >>> B, Nataraju wrote: >>>> Paul, >>>> >>>> What I meant to mention here was that, in the scenarios which =20 >>>> might lead >>> to these responses must be authenticated before we call for this >>> Reponses... >>>> I mean if we receive a NOTIFY(un-authenticated) 404 should not = be >>> sent out which will terminate the dialog/usage. >>>> Authenticate the NTFY request first then only it could = be >>> rejected with 404 or any other appropriate replies which would =20 >>> result in >>> dialog termination. >>> >>> If I understand you, this is just standard procedure, applicable >>> regardless of dialog usages. Authentication is supposed to be done >>> first, so no extra information is given to unauthenticated callers. >>> >>> Paul >>> >>>> Thanks, >>>> Nataraju A B >>>> >>>>> -----Original Message----- >>>>> From: Paul Kyzivat [mailto:pkyzivat@cisco.com] >>>>> Sent: Monday, August 14, 2006 7:34 PM >>>>> To: B, Nataraju >>>>> Cc: sip@ietf.org >>>>> Subject: Re: [Sip] comments on draft-ietf-sipping-=20 >>>>> dialogusage-02.txt >>>>> >>>>> >>>>> >>>>> B, Nataraju wrote: >>>>> >>>>>> 1. Sec 4 describes what the effects on dialog are and its =20 >>>>>> usages when >>>>>> particular responses are received for NOTIFY message, the =20 >>>>>> responses >>> 404, >>>>>> 410, 416, lead to dialog and its usage termination. >>>>>> >>>>>> a. I feel we should have mentioned ONLY the =20 >>>>>> authenticated NOTIFY >>>>>> requests will have this effect on the dialogs/usages. =20 >>>>>> Otherwise any >>>>>> middle men can trace the message and send the NOTIFY messages =20 >>>>>> which >>>>>> terminate the dialogs, which is undesirable =E2=98=B9. If the = entity =20 >>>>>> sending >>>>>> NOTIFY really intend to do that it can resend NOTIFY with >>> authorization >>>>>> credentials which will help to terminate the associated dialogs. >>>>> I agree we wouldn't want to enable a MITM attack here. >>>>> >>>>> But I think that is already covered. If the request isn't properly >>>>> authenticated, then the response will be 401, not 404, 410, or =20 >>>>> 416. >>>>> >>>>> Paul >>>> >>>> >>>> -------------------------------------------------------------------=20= >>>> ----- >>>> >>>> Subject: >>>> [Sip] comments on draft-ietf-sipping-dialogusage-02.txt >>>> From: >>>> "B, Nataraju" >>>> Date: >>>> Mon, 14 Aug 2006 12:04:19 +0530 >>>> To: >>>> >>>> >>>> To: >>>> >>>> >>>> >>>> Hi Sparks, >>>> >>>> >>>> >>>> I had gone through the document, here are few comments. >>>> >>>> >>>> >>>> Note: I assume the behavior mentioned in sec 4 applies both at =20 >>>> client >>> receiving these responses (Notifier) and server generating these =20 >>> replies >>> (Subscriber). If this was meant to be only @ sender of NOTIFY =20 >>> shall we >>> need to mention the similar set of actions @ Subscriber side too? >>>> >>>> >>>> 1. Sec 4 describes what the effects on dialog are and its usages =20= >>>> when >>> particular responses are received for NOTIFY message, the =20 >>> responses 404, >>> 410, 416, lead to dialog and its usage termination. >>>> a. I feel we should have mentioned ONLY the authenticated =20 >>>> NOTIFY >>> requests will have this effect on the dialogs/usages. Otherwise =20 >>> any middle >>> men can trace the message and send the NOTIFY messages which =20 >>> terminate the >>> dialogs, which is undesirable =E2=98=B9. If the entity sending = NOTIFY =20 >>> really >>> intend to do that it can resend NOTIFY with authorization =20 >>> credentials >>> which will help to terminate the associated dialogs. >>>> >>>> >>>> b. Other alternative would be that let those responses =20 >>>> have no >>> effect on the dialog/usages, eventually the subscription duration =20= >>> would >>> take care of dialog termination or the actual entity would =20 >>> terminate or >>> extend the subscription. Any way the request from actual notifier =20= >>> would >>> have the proper URIs or credentials which will never lead to these >>> responses. >>>> >>>> >>>> The same logic would need to extend for replies 481, 483, 484, =20 >>>> 489 and >>> 604 also. >>>> >>>> >>>> 2. Sec 4 of the draft >>>> >>>> >>>> >>>> <<< 423 Interval Too Brief: This response won't happen in our >>> example >>>> scenario, but if it came in response to a reSUBSCRIBE, the >>>> >>>> subscribe usage is not destroyed (or otherwise affected). No >>>> >>>> other usages of the dialog are affected. >>> >>>> >>>> >>>> >>>> In the above Para the meaning of last part of 1st sentence (I >>> mean, the subscribe usage is not destroyed (or otherwise =20 >>> affected)) and >>> 2nd sentence is not clear, both mention the same meaning? Can we =20 >>> make it >>> explicit about what we really mean here ? >>>> >>>> >>>> Thanks, >>>> >>>> Nataraju A B >>>> >>>> Sonus Networks India Pvt Ltd >>>> >>>> >>>> >>>> -------------------------------------------------------------------=20= >>>> ----- >>>> >>>> _______________________________________________ >>>> Sip mailing list https://www1.ietf.org/mailman/listinfo/sip >>>> This list is for NEW development of the core SIP Protocol >>>> Use sip-implementors@cs.columbia.edu for questions on current sip >>>> Use sipping@ietf.org for new developments on the application of sip > > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP > Use sip-implementors@cs.columbia.edu for questions on current sip > Use sip@ietf.org for new developments of core SIP _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Mon Aug 28 18:34:16 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GHpfW-0007HV-Pn; Mon, 28 Aug 2006 18:33:22 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GHpfV-0007HP-ML for sipping@ietf.org; Mon, 28 Aug 2006 18:33:21 -0400 Received: from stsc1260-eth-s1-s1p1-vip.va.neustar.com ([156.154.16.129] helo=chiedprmail1.ietf.org) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GHoLI-0003PR-RA for sipping@ietf.org; Mon, 28 Aug 2006 17:08:24 -0400 Received: from shaman.nostrum.com ([72.232.15.10] helo=nostrum.com) by chiedprmail1.ietf.org with esmtp (Exim 4.43) id 1GHo4C-0000iM-MC for sipping@ietf.org; Mon, 28 Aug 2006 16:50:47 -0400 Received: from [172.17.2.252] (vicuna.estacado.net [72.1.129.69]) (authenticated bits=0) by nostrum.com (8.13.6/8.13.6) with ESMTP id k7SKof7x043019 (version=TLSv1/SSLv3 cipher=RC4-SHA bits=128 verify=NO); Mon, 28 Aug 2006 15:50:42 -0500 (CDT) (envelope-from rjsparks@nostrum.com) In-Reply-To: <70798CF421F00F4DA018059E5B7EEB8C7A5BDA@sonusinmail01.sonusnet.com> References: <70798CF421F00F4DA018059E5B7EEB8C7A5BDA@sonusinmail01.sonusnet.com> Mime-Version: 1.0 (Apple Message framework v752.2) Message-Id: From: Robert Sparks Date: Mon, 28 Aug 2006 15:50:45 -0500 To: "B, Nataraju" X-Mailer: Apple Mail (2.752.2) Received-SPF: pass (nostrum.com: 72.1.129.69 is authenticated by a trusted mechanism) X-Spam-Score: -2.2 (--) X-Scan-Signature: 8de5f93cb2b4e3bee75302e9eacc33db Cc: sipping Subject: [Sipping] Re: [Sip] comments on draft-ietf-sipping-dialogusage-02.txt X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Content-Type: multipart/mixed; boundary="===============1459303676==" Errors-To: sipping-bounces@ietf.org --===============1459303676== Content-Type: multipart/alternative; boundary=Apple-Mail-9-249902429 --Apple-Mail-9-249902429 Content-Transfer-Encoding: 7bit Content-Type: text/plain; charset=US-ASCII; delsp=yes; format=flowed On Aug 14, 2006, at 1:34 AM, B, Nataraju wrote (to the SIP list, but this response is sent to SIPPING) > Hi Sparks, > 2. Sec 4 of the draft > > > > <<< 423 Interval Too Brief: This response won't happen in our > example > > scenario, but if it came in response to a reSUBSCRIBE, the > > subscribe usage is not destroyed (or otherwise affected). No > > other usages of the dialog are affected. >>> > > > > In the above Para the meaning of last part of 1st sentence (I > mean, the subscribe usage is not destroyed (or otherwise affected)) > and 2nd sentence is not clear, both mention the same meaning? Can > we make it explicit about what we really mean here ? > > This pattern of text occurs frequently in the document. What caused you to single this one instance out? On review, I think the text _DOES_ say what it really means, but if it's confused you, it's probably going to confuse others. Can you suggest an alternative? RjS --Apple-Mail-9-249902429 Content-Transfer-Encoding: quoted-printable Content-Type: text/html; charset=ISO-8859-1
On Aug 14, 2006, = at 1:34 AM, B, Nataraju wrote (to the SIP list, but this response is = sent to SIPPING)

Hi = Sparks,

<snip><= BR>

2. Sec 4 of the = draft

=A0=A0 =

=A0=A0=A0 = <<<=A0=A0 423 Interval Too Brief: This response won't happen in = our example

=A0=A0=A0= =A0=A0 scenario, but if it came in response to a reSUBSCRIBE, = the

=A0=A0=A0=A0=A0 = subscribe usage is not destroyed (or otherwise affected).=A0 = No

=A0=A0=A0=A0=A0 = other usages of the dialog are affected. = >>>

=A0

=A0=A0=A0= =A0=A0 In the above Para the meaning = of last part of 1st sentence (I mean, the subscribe usage is not = destroyed (or otherwise affected)) and 2nd sentence is not clear, both = mention the same meaning? Can we make it explicit about what we really = mean here ?

=A0

This pattern of text occurs frequently in the document. What=A0 caused = you to single this one instance out?

On review, I think the text = _DOES_ say what it really means, but if it's confused you, it's probably = going to confuse
others. Can you suggest an = alternative?

RjS
= --Apple-Mail-9-249902429-- --===============1459303676== Content-Type: text/plain; charset="us-ascii" MIME-Version: 1.0 Content-Transfer-Encoding: 7bit Content-Disposition: inline _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP --===============1459303676==-- From sipping-bounces@ietf.org Tue Aug 29 02:10:34 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GHwlj-00007s-Pr; Tue, 29 Aug 2006 02:08:15 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GHwlh-00007m-LF for sipping@ietf.org; Tue, 29 Aug 2006 02:08:13 -0400 Received: from stsc1260-eth-s1-s1p1-vip.va.neustar.com ([156.154.16.129] helo=chiedprmail1.ietf.org) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GHwlh-0004O5-JR for sipping@ietf.org; Tue, 29 Aug 2006 02:08:13 -0400 Received: from sonussf1.sonusnet.com ([208.45.178.26]) by chiedprmail1.ietf.org with esmtp (Exim 4.43) id 1GHwfn-0006ga-CD for sipping@ietf.org; Tue, 29 Aug 2006 02:02:09 -0400 Received: from sonusmail06.sonusnet.com (sonusmail06.sonusnet.com [10.128.32.156]) by sonussf1.sonusnet.com (8.13.3/8.13.3) with ESMTP id k7T625gt030447; Tue, 29 Aug 2006 02:02:05 -0400 Received: from SONUSINMAIL01.sonusnet.com ([10.128.254.7]) by sonusmail06.sonusnet.com with Microsoft SMTPSVC(6.0.3790.1830); Tue, 29 Aug 2006 02:02:05 -0400 content-class: urn:content-classes:message Subject: RE: [Sipping] Re: [Sip] comments on draft-ietf-sipping-dialogusage-02.txt MIME-Version: 1.0 Date: Tue, 29 Aug 2006 11:32:00 +0530 X-MimeOLE: Produced By Microsoft Exchange V6.0.6603.0 Message-ID: <70798CF421F00F4DA018059E5B7EEB8C84BC05@sonusinmail01.sonusnet.com> X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: [Sipping] Re: [Sip] comments on draft-ietf-sipping-dialogusage-02.txt Thread-Index: AcbK4kzxcn4VIB6wRu2FEVGOCK535gARv/WA From: "B, Nataraju" To: "Robert Sparks" , "Paul Kyzivat" X-OriginalArrivalTime: 29 Aug 2006 06:02:05.0218 (UTC) FILETIME=[A74A7020:01C6CB30] X-Spam-Score: -1.8 (-) X-Scan-Signature: 7e439b86d3292ef5adf93b694a43a576 Cc: sipping@ietf.org X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Content-Type: multipart/mixed; boundary="===============1969089031==" Errors-To: sipping-bounces@ietf.org --===============1969089031== content-class: urn:content-classes:message Content-Type: text/plain; charset="utf-8" Content-Transfer-Encoding: base64 SGkgU3BhcmtzLCANCg0KTXkgb3JpZ2luYWwgaW1wb3J0YW50IGNvbW1lbnQgd2FzIHRoYXQsIGFu IHVuLWF1dGhlbnRpY2F0ZWQgTk9USUZZIHNob3VsZCBub3QgdGVybWluYXRlIGFueSBkaWFsb2cu 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(helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GI1tW-0003Zx-6X; Tue, 29 Aug 2006 07:36:38 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GI1tU-0003Zs-GP for sipping@ietf.org; Tue, 29 Aug 2006 07:36:36 -0400 Received: from smail.alcatel.fr ([62.23.212.165]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GI1tN-00022C-WA for sipping@ietf.org; Tue, 29 Aug 2006 07:36:36 -0400 Received: from bemail02.netfr.alcatel.fr (bemail02.netfr.alcatel.fr [155.132.251.36]) by smail.alcatel.fr (8.13.4/8.13.4/ICT) with ESMTP id k7TBaTC2004240 for ; Tue, 29 Aug 2006 13:36:29 +0200 Received: from [138.203.251.40] ([138.203.251.40]) by bemail02.netfr.alcatel.fr (Lotus Domino Release 6.5.4FP2HF315) with ESMTP id 2006082913362716-8454 ; Tue, 29 Aug 2006 13:36:27 +0200 Message-ID: <44F426B6.60605@alcatel.be> Date: Tue, 29 Aug 2006 13:36:22 +0200 From: olivier.ro.rousseau@alcatel.be Organization: Alcatel User-Agent: Mozilla/5.0 (Windows; U; Windows NT 5.1; en-US; rv:1.7.10) Gecko/20050716 X-Accept-Language: en-us, en MIME-Version: 1.0 To: sipping@ietf.org X-MIMETrack: Itemize by SMTP Server on BEMAIL02/BE/ALCATEL(Release 6.5.4FP2HF315 | February 22, 2006) at 08/29/2006 13:36:27, Serialize by Router on BEMAIL02/BE/ALCATEL(Release 6.5.4FP2HF315 | February 22, 2006) at 08/29/2006 13:36:28, Serialize complete at 08/29/2006 13:36:28 Content-Type: multipart/mixed; boundary="------------090608050600000301060902" X-Scanned-By: MIMEDefang 2.51 on 155.132.180.81 X-Spam-Score: 0.2 (/) X-Scan-Signature: 5d7a7e767f20255fce80fa0b77fb2433 Subject: [Sipping] [Fwd: Re: [Sip] question about RFC4240 (NETANN)] X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org This is a multi-part message in MIME format. --------------090608050600000301060902 Content-Transfer-Encoding: 7bit Content-Type: text/plain; charset=us-ascii; format=flowed --------------090608050600000301060902 Content-Transfer-Encoding: 7bit Content-Type: message/rfc822; name="Re: [Sip] question about RFC4240 (NETANN)" Content-Disposition: inline; filename="Re: [Sip] question about RFC4240 (NETANN)" Content-Transfer-Encoding: quoted-printable Received: from smail.alcatel.fr ([155.132.180.81]) by bemail02.netfr.alcatel.fr (Lotus Domino Release 6.5.4FP2HF315) with ESMTP id 2006082912214773-7277 ; Tue, 29 Aug 2006 12:21:47 +0200 Received: from mailgate.alcatel.fr (mailgate.alcatel.fr [155.132.180.93]) by smail.alcatel.fr (8.13.4/8.13.4/ICT) with ESMTP id k7TALmHW022731 (version=TLSv1/SSLv3 cipher=DHE-RSA-AES256-SHA bits=256 verify=NOT) for ; Tue, 29 Aug 2006 12:21:49 +0200 Received: from ihemail2.lucent.com (ihemail2.lucent.com [192.11.222.163]) by mailgate.alcatel.fr (ALCANET) with ESMTP id k7TALj81028224 for ; Tue, 29 Aug 2006 12:21:46 +0200 Received: from ilexp02.ndc.lucent.com (h135-3-39-2.lucent.com [135.3.39.2]) by ihemail2.lucent.com (8.13.6/IER-o) with ESMTP id k7TALj68019976 for ; Tue, 29 Aug 2006 05:21:45 -0500 (CDT) Received: from DEEXP01.de.lucent.com ([135.248.187.65]) by ilexp02.ndc.lucent.com with Microsoft SMTPSVC(6.0.3790.0); Tue, 29 Aug 2006 05:21:45 -0500 Received: from DEEXC1U01.de.lucent.com ([135.248.187.20]) by DEEXP01.de.lucent.com with Microsoft SMTPSVC(6.0.3790.0); Tue, 29 Aug 2006 12:21:43 +0200 Subject: RE: [Sip] question about RFC4240 (NETANN) Date: Tue, 29 Aug 2006 12:21:42 +0200 Message-ID: <5D1A7985295922448D5550C94DE2918034DD5C@DEEXC1U01.de.lucent.com> From: "Drage, Keith \(Keith\)" To: olivier.ro.rousseau@alcatel.be Content-class: urn:content-classes:message MIME-Version: 1.0 X-MimeOLE: Produced By Microsoft Exchange V6.5.7226.0 X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: [Sip] question about RFC4240 (NETANN) Thread-Index: AcbIPQmsI7z4JpONQz+QBog9l7NkmgDF9CbQ X-OriginalArrivalTime: 29 Aug 2006 10:21:43.0504 (UTC) FILETIME=[ECAC3D00:01C6CB54] X-Scanned-By: '' on 155.132.180.93 X-MIMETrack: Itemize by SMTP Server on BEMAIL02/BE/ALCATEL(Release 6.5.4FP2HF315 | February 22, 2006) at 08/29/2006 12:21:47, Serialize by Router on BEMAIL02/BE/ALCATEL(Release 6.5.4FP2HF315 | February 22, 2006) at 08/29/2006 12:21:49, Serialize complete at 08/29/2006 12:21:49 Content-Type: text/plain; charset="us-ascii" Note that this document was produced by the SIPPING group rather than the SIP group. Regards Keith=20 > -----Original Message----- > From: olivier.ro.rousseau@alcatel.be=20 > [mailto:olivier.ro.rousseau@alcatel.be]=20 > Sent: 25 August 2006 12:49 > To: Sip@ietf.org > Subject: [Sip] question about RFC4240 (NETANN) >=20 > Hello, >=20 > I have a problem with the RFC 4240. > The language parameter is defined in the page 10. > locale-param =3D "locale=3D" token > ; per RFC 3066, usually > ; ISO639-1_ISO3166-1 > ; e.g., en, en_US, en_UK, etc. > and is explained in page 7 > locale > Specifies the language and optionally country variant of the > announcement sequence named in the "play=3D" parameter. RFC = 3066 > [9] specifies the locale tag. The locale tag is=20 > usually a two- or > three-letter code per ISO 639-1 [11]. The country=20 > variant is also > often a two-letter code per ISO 3166-1 [12]. These elements are > concatenated with a single under bar (%x5F) character, such as > "en_CA". >=20 > However whe I look in the RFC3066 I found > Language-Tag =3D Primary-subtag *( "-" Subtag ) >=20 > Primary-subtag =3D 1*8ALPHA >=20 > Subtag =3D 1*8(ALPHA / DIGIT) >=20 > In the RFC3261, the Accept-Langugae is defined as=20 > Accept-Language =3D "Accept-Language" HCOLON > [ language *(COMMA language) ] > language =3D language-range *(SEMI accept-param) > language-range =3D ( ( 1*8ALPHA *( "-" 1*8ALPHA ) ) / "*" ) >=20 > You can see that one RFC says that a "_" must be used between=20 > the two subtags while the two others says that it must be a "-". > And worse the RFC4240 refers to RFC3066 but uses a different=20 > syntax in its BNF and in its text ( >=20 > Could someone clarify what I should use? >=20 > Thanks and regards, > Olivier >=20 >=20 >=20 > _______________________________________________ > Sip mailing list https://www1.ietf.org/mailman/listinfo/sip > This list is for NEW development of the core SIP Protocol Use=20 > sip-implementors@cs.columbia.edu for questions on current sip=20 > Use sipping@ietf.org for new developments on the application of sip >=20 --------------090608050600000301060902 Content-Type: text/plain; charset="us-ascii" MIME-Version: 1.0 Content-Transfer-Encoding: 7bit Content-Disposition: inline _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP --------------090608050600000301060902-- From sipping-bounces@ietf.org Tue Aug 29 09:08:05 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GI3Iw-0005bZ-Ax; Tue, 29 Aug 2006 09:06:58 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GI3Iu-0005bU-Jx for sipping@ietf.org; Tue, 29 Aug 2006 09:06:56 -0400 Received: from rtp-iport-2.cisco.com ([64.102.122.149]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GI3It-0002HW-AN for sipping@ietf.org; Tue, 29 Aug 2006 09:06:56 -0400 Received: from rtp-dkim-1.cisco.com ([64.102.121.158]) by rtp-iport-2.cisco.com with ESMTP; 29 Aug 2006 09:06:54 -0400 X-IronPort-AV: i="4.08,180,1154923200"; d="scan'208"; a="99451241:sNHT32272684" Received: from rtp-core-2.cisco.com (rtp-core-2.cisco.com [64.102.124.13]) by rtp-dkim-1.cisco.com (8.12.11.20060308/8.12.11) with ESMTP id k7TD6sd7027466; Tue, 29 Aug 2006 09:06:54 -0400 Received: from xbh-rtp-201.amer.cisco.com (xbh-rtp-201.cisco.com [64.102.31.12]) by rtp-core-2.cisco.com (8.12.10/8.12.6) with ESMTP id k7TD6suI027998; Tue, 29 Aug 2006 09:06:54 -0400 (EDT) Received: from xfe-rtp-201.amer.cisco.com ([64.102.31.38]) by xbh-rtp-201.amer.cisco.com with Microsoft SMTPSVC(6.0.3790.1830); Tue, 29 Aug 2006 09:06:54 -0400 Received: from [161.44.79.176] ([161.44.79.176]) by xfe-rtp-201.amer.cisco.com with Microsoft SMTPSVC(6.0.3790.1830); Tue, 29 Aug 2006 09:06:53 -0400 Message-ID: <44F43BED.8040507@cisco.com> Date: Tue, 29 Aug 2006 09:06:53 -0400 From: Paul Kyzivat User-Agent: Thunderbird 1.5.0.5 (Windows/20060719) MIME-Version: 1.0 To: Robert Sparks Subject: Re: [Sipping] Re: [Sip] comments on draft-ietf-sipping-dialogusage-02.txt References: <70798CF421F00F4DA018059E5B7EEB8C7A5BDA@sonusinmail01.sonusnet.com> In-Reply-To: Content-Type: text/plain; charset=ISO-8859-1; format=flowed Content-Transfer-Encoding: 7bit X-OriginalArrivalTime: 29 Aug 2006 13:06:53.0729 (UTC) FILETIME=[FFA09110:01C6CB6B] DKIM-Signature: a=rsa-sha1; q=dns; l=1546; t=1156856814; x=1157720814; c=relaxed/simple; s=rtpdkim1001; h=Content-Type:From:Subject:Content-Transfer-Encoding:MIME-Version; d=cisco.com; i=pkyzivat@cisco.com; z=From:Paul=20Kyzivat=20 |Subject:Re=3A=20[Sipping]=20Re=3A=20[Sip]=20comments=20on=09draft-ietf-sipping-d ialogusage-02.txt |To:Robert=20Sparks=20; X=v=3Dcisco.com=3B=20h=3DVEfiHuaUKIzVmtBfLv9CWsOrmpY=3D; b=dyDkyqQlhUDW4M/yW123r4vatsnVBr52ydlcXQ4qnHuL5h7oXx8joD60586kRqr1lMr7FESC vjYHwYhMcHWuHzW+x54a4IZhLLGUPv1FXYu3PULQlcDOYDjSm+4QEDVZ; Authentication-Results: rtp-dkim-1.cisco.com; header.From=pkyzivat@cisco.com; dkim=pass ( sig from cisco.com verified; ); X-Spam-Score: 0.0 (/) X-Scan-Signature: 4d87d2aa806f79fed918a62e834505ca Cc: sipping X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org This seems perfectly clear to me. Paul Robert Sparks wrote: > > On Aug 14, 2006, at 1:34 AM, B, Nataraju wrote (to the SIP list, but > this response is sent to SIPPING) > >> Hi Sparks, >> > >> >> 2. Sec 4 of the draft >> >> >> >> <<< 423 Interval Too Brief: This response won't happen in our >> example >> >> scenario, but if it came in response to a reSUBSCRIBE, the >> >> subscribe usage is not destroyed (or otherwise affected). No >> >> other usages of the dialog are affected. >>> >> >> >> >> In the above Para the meaning of last part of 1st sentence (I >> mean, the subscribe usage is not destroyed (or otherwise affected)) >> and 2nd sentence is not clear, both mention the same meaning? Can we >> make it explicit about what we really mean here ? >> >> >> > This pattern of text occurs frequently in the document. What caused you > to single this one instance out? > > On review, I think the text _DOES_ say what it really means, but if it's > confused you, it's probably going to confuse > others. Can you suggest an alternative? > > RjS > > > ------------------------------------------------------------------------ > > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP > Use sip-implementors@cs.columbia.edu for questions on current sip > Use sip@ietf.org for new developments of core SIP _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Tue Aug 29 11:02:08 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GI54A-0001sb-Jb; Tue, 29 Aug 2006 10:59:50 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GI548-0001sW-RU for sipping@ietf.org; Tue, 29 Aug 2006 10:59:48 -0400 Received: from [2001:5c0:8fff:fffe::4c3d] (helo=vicuna.estacado.net) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GI546-0007E0-0d for sipping@ietf.org; Tue, 29 Aug 2006 10:59:48 -0400 Received: from [172.17.2.251] ([172.17.2.251]) (authenticated bits=0) by vicuna.estacado.net (8.13.6/8.13.4) with ESMTP id k7TExOQJ059418 (version=TLSv1/SSLv3 cipher=DHE-RSA-AES256-SHA bits=256 verify=NO); Tue, 29 Aug 2006 09:59:25 -0500 (CDT) (envelope-from adam@estacado.net) Message-ID: <44F45637.7070500@estacado.net> Date: Tue, 29 Aug 2006 09:59:03 -0500 From: Adam Roach User-Agent: Thunderbird 1.5.0.5 (Windows/20060719) MIME-Version: 1.0 To: "Huelsemann, Martin" References: In-Reply-To: Content-Type: text/plain; charset=ISO-8859-1; format=flowed Content-Transfer-Encoding: 7bit X-Spam-Score: -2.8 (--) X-Scan-Signature: 3fbd9b434023f8abfcb1532abaec7a21 Cc: drage@LUCENT.COM, SIPPING WG Subject: [Sipping] Re: WG: draft-poetzl-sipping-call-completion-01 X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org I have a handful of comments on the current call completion document as it relates to RFC 3265. 1. I have some heartburn with the behavior described for the event package. The SUBSCRIBE/NOTIFY mechanism is intended to *subscribe* to remote state, not *manipulate* it. Parameters like "queue-operation" are obviously intended to act as an RPC mechanism. It is important that SUBSCRIBE requests remain idempotent: sending the same SUBSCRIBE request over and over should not cause users to be inserted into a queue multiple times. As this document defines the mechanism, this is not the case. 2. I find the choice of event header field parameters to return resource state to be both baroque and in violation of the description of NOTIFY bodies in section 4.4.5 of RFC 3265. For example, "queue-state" and "denial-reason" communicate resource state, and should be sent in the NOTIFY body (not Event header field parameters). 3. Disallowing forked requests for the event package seems an eccentric choice to me. Since there are necessarily no state agents in the network for this event package, it is EXTREMELY likely that subscribe requests will end up getting forked to multiple terminals. The prospect that CCBS/CCNR can apply to only one terminal seems really counterintuitive. I can't imagine how this will ever seem like the right behavior to users. For example, if I have two phones (one at my office in Dallas, where I spend most of my time, and one at my office in Denver, where I spend three weeks a year), and you call my number, but give up and activate the CCNR service: does it make sense that you will establish a CCNR relationship with my the phone in Colorado (which won't be used for another month or so), and REJECT the relationship with my phone in Dallas (which I'll probably use within an hour or two)? I also have some minor issues with the "P-Service-Indication" header. In this case, an RFC 3087-based solution will suit your purposes just fine (that is, the called party generates a GRUU with a GRID that it recognizes as a CCBS/CCNR service request, and provides it to the caller. When the caller uses this GRUU, the called party will recognize the GRID as being related to the CCBS/CCNR service, and know that the request is a CCBS/CCNR request.) You'll need to use a GRUU anyway, since the service as you've defined it is very specific to the called device, not to the user's AOR in general. I'll outline a solution that I think is much closer to the intention of RFC 3265 and other SIP documents: * Called parties, when returning any provisional response (180, 181, 182, 183, any others), or any other indication that the called party is currently unavailable (486, 480, 600) can indicate support for the CCBS extension using the "Supported" header field (NOT the "Allow-Events" header field). They also include a new header field which contains a GRUU that routes back to the device; this GRUU contains a GRID that the device recognizes as being related to the CCBS/CCNR service. * Called parties can react to such provisional or busy responses by sending a message that is an explicit request to insert themselves in the queue. These requests are sent to the service request GRUU. I'm not yet certain specifically what this looks like (that is, what method to use), but the request to insert the caller into the queue (and to pause and unpause the position in the queue, as is described in section 5 of the current call completion document) needs to be EXPLICIT and consistent with the meaning of the method in which it is carried -- not IMPLICIT as a side effect of requesting a subscription. This likely means that you'll either want a new method or you'll want to use INVITE to set up a queue management session. * Given the above, there are several ways to complete the service. One such option would be: The caller now subscribes to the state of the queue, and receives periodic notifications about the position and disposition of his request in the queue. The NOTIFYs would look something like this: NOTIFY sip:adam@172.17.1.47;grid=58sjgf SIP/2.0 From: ;tag=4598d-fg8 To: ;tag=f85lkfi-28s Call-ID: 30awvs09xmvf90325ax0932kgz0gnj20 CSeq: 58721 NOTIFY Via: SIP/2.0/UDP proxy.estacado.net:5060;branch=z9hg4bkg09832gnjz095 Via: SIP/2.0/UDP 172.17.16.52:5060;branch=z9hg4bks582fd0gk109d Contact: Event: cc-queuestate Content-Type: application/queue-state+xml (the exact syntax and contents of this message aren't terribly important for the purpose of understanding what I'm proposing, and probably need more than a little tweaking). When the user's request becomes active, the body of the notify will indicate "ready" instead of "queued" for the request. At this point, the caller can then use the GRUU that it received before to initiate the call. (The called party can reject the call if the GRID in the INVITE doesn't match the GRID for the call marked as "ready" in the queue). Finally, I would suggest that you consider the ability to deploy a system in which some non-terminal server handles queue management instead of the endpoints -- doing so would allow the "Dallas/Denver" example I provide above to work in a more sane fashion. The non-terminal server could either track dialog state by being call stateful, or it could subscribe to the dialog state for each of the user's registered devices. /a Huelsemann, Martin wrote: > > Hi all, > > a revised version of the Call Completion draft is available at > > _http://www.ietf.org/internet-drafts/draft-poetzl-sipping-call-completion-01.txt_. > > > The intention of the draft is to propose solutions for the > requirements for the ETSI TISPAN NGN services Completion of Call to > Busy Subscribers and Completion of Call on No reply. > > The main change to the previous version is, that there is only one > call completion event package now, which can be used for CCBS and > CCNR. The indication for CCBS is 'Allow-events: call-completion' in a > 486 Busy response, 'Allow-events: call-completion' in a 180 Ringing > response is the indication for CCNR. > > Then I have added some more explanatory text and signalling flows, I > hope that makes the requirements and solutions described in this draft > clearer now. > > > > As you renember from the IETF 66 discussion, the main topic of > discussion was, is it appropriate to use SUBSCRIBE/NOTIFY, besides > subscribing to a call completion event itself, also for managing the > call completion queue, for example for suspend and resume procedures. > > > > Questions and comments are very welcome. > > > Best regards, Martin > > > _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Tue Aug 29 11:10:38 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GI5Ec-0006nt-3p; Tue, 29 Aug 2006 11:10:38 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GI5EZ-0006nN-Op; Tue, 29 Aug 2006 11:10:35 -0400 Received: from [2001:5c0:8fff:fffe::4c49] (helo=nostrum.com) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GI5ET-0008TX-Kz; Tue, 29 Aug 2006 11:10:35 -0400 Received: from [172.17.2.252] (vicuna.estacado.net [72.1.129.69]) (authenticated bits=0) by nostrum.com (8.13.6/8.13.6) with ESMTP id k7TFAJ5F092087 (version=TLSv1/SSLv3 cipher=RC4-SHA bits=128 verify=NO); Tue, 29 Aug 2006 10:10:19 -0500 (CDT) (envelope-from rjsparks@nostrum.com) Mime-Version: 1.0 (Apple Message framework v752.2) Content-Transfer-Encoding: 7bit Message-Id: <7FEE86DE-A536-4B2A-90D2-7E429B9D2913@nostrum.com> Content-Type: text/plain; charset=US-ASCII; delsp=yes; format=flowed To: IETF SIP List , sipping , simple@ietf.org, sip-implementors@cs.columbia.edu From: Robert Sparks Date: Tue, 29 Aug 2006 10:10:23 -0500 X-Mailer: Apple Mail (2.752.2) Received-SPF: pass (nostrum.com: 72.1.129.69 is authenticated by a trusted mechanism) X-Spam-Score: 0.0 (/) X-Scan-Signature: de4f315c9369b71d7dd5909b42224370 Cc: Subject: [Sipping] SIPit 19 registration X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org A couple of reminders: SIPit19 registration is open. The event will be Oct 16-20 at the University of New Hampshire InterOperability Laboratory. While the registration deadline is still a few weeks away (9/29), we are already over 70% full. If you are planning to attend and have not yet registered, you should do so soon. Also, please make your hotel reservations early - this is prime leaf- peeping season in New Hampshire and last minute hotels will be very hard to find. See www.sipit.net for more information. RjS _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Tue Aug 29 16:23:21 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GIA5u-0008QG-Fb; Tue, 29 Aug 2006 16:21:58 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GIA5t-0008Q6-Ei for sipping@ietf.org; Tue, 29 Aug 2006 16:21:57 -0400 Received: from agnada.com ([69.36.182.244]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GIA5r-00011F-3G for sipping@ietf.org; Tue, 29 Aug 2006 16:21:57 -0400 Received: from [192.168.0.101] (S010600095b9792b5.vc.shawcable.net [24.85.243.115]) (authenticated) by agnada.com (8.11.6/8.11.6) with ESMTP id k7TKLX227607 for ; Tue, 29 Aug 2006 14:21:34 -0600 Message-ID: <44F4A1CB.2020705@ntt-at.com> Date: Tue, 29 Aug 2006 13:21:31 -0700 From: Shida Schubert User-Agent: Thunderbird 1.5.0.5 (Windows/20060719) MIME-Version: 1.0 To: sipping@ietf.org Content-Type: text/plain; charset=ISO-2022-JP Content-Transfer-Encoding: 7bit X-Spam-Score: 1.1 (+) X-Scan-Signature: a8a20a483a84f747e56475e290ee868e Subject: [Sipping] Review of draft-ietf-sipping-session-policy-framework-01 X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list Reply-To: shida@ntt-at.com List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Draft: draft-ietf-sipping-session-policy-framework-01 Reviewer: Shida Schubert Review Date: 28 Aug 06 Review Deadline: 25 Aug 06 Status: Initial review Summary: This draft is almost ready for publication, but needs some clarifications and has nits that should be fixed before publication. The requested clarifications are listed: Section 1. - 4th paragraph, 3rd sentence mentions a user-interaction for providing consent on the policy provided by the policy server but there is no mentioning of this in section 4.x. Did you mean user or UA here? I think this needs to be clarified. Section 4.4.1 Last sentence in the 4th paragraph, defines a normative behavior for UAC where UAC SHOULD NOT change the order of policy servers Section 4.4.2 - 2nd paragraph, last sentence: I think when non-cachable is appended after the URI, recipient(UAC) is only supposed to remove the URI from the cache that has the same value as the one received with the "non-cacheable". If that's true, the last sentence should say the following. OLD: It SHOULD remove the current URI from the cache when receiving a "non-cacheable" URI. NEW: It SHOULD not store the URI in the cache when receiving a "non-cacheable" URI and SHOULD remove the URI if it was already cached. Section 4.4.2 - 3rd paragraph and 4th paragraph contradicts one another. 4th paragraph mentions that UAC SHOULD store the list of policy server's URI it contacted but 3rd paragraph states that UAC SHOULD only cache URI for the local policy server. I think some coherence is needed between the paragraph. Section 4.4.4 - There is no normative behavior for Policy server when UAS is inquiring for the policy. I think if there is nothing for policy server to do, then that should be stated as well. Editorial nits: -------------------------- Section 1: - 6th paragraph, last sentence: I would suggest the following changes to clarify that session-independent policy does change. OLD: Since these policies are not based on a specific session description, they can be created independent of an attempt to set up a session and only need to be conveyed to the user agent once (e.g. at the time the device is powered on). NEW: Since these policies are not based on a specific session description, they can be created independent of an attempt to set up a session and only need to be conveyed to the user agent once (e.g. at the time the device is powered on) until relevant policies change Section 4.2: - 1st paragraph, 4th sentence to the end of 2nd paragraph explains the benefit of separate channel over piggy back model, but do we still need this in the draft? I personally think this can be removed. May be you can note the decision that was made in the change-log. Section 4.4.1 - 2nd paragrah, last sentence: I would suggest the following changes to clarify that request reflecting the policies are to be sent to the same proxy where original request was sent. Also behavior of UAC when no policies were provided should be mentioned as well. OLD: The UAC SHOULD apply the policies received and resend the updated request. NEW: The UAC SHOULD apply the policies received and resend the updated request to the same proxy it sent the original request. If no policies were received it SHOULD resend the request without changing its session parameters to the same proxy it sent the original request. Regards Shida Schubert _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Tue Aug 29 19:13:13 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GICk6-0004h0-RO; Tue, 29 Aug 2006 19:11:38 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GICk5-0004gv-8j for sipping@ietf.org; Tue, 29 Aug 2006 19:11:37 -0400 Received: from agnada.com ([69.36.182.244]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GICk3-0008DH-Vr for sipping@ietf.org; Tue, 29 Aug 2006 19:11:37 -0400 Received: from [192.168.0.101] (S010600095b9792b5.vc.shawcable.net [24.85.243.115]) (authenticated) by agnada.com (8.11.6/8.11.6) with ESMTP id k7TNBZm12950; Tue, 29 Aug 2006 17:11:35 -0600 Message-ID: <44F4C9A8.6060900@ntt-at.com> Date: Tue, 29 Aug 2006 16:11:36 -0700 From: Shida Schubert User-Agent: Thunderbird 1.5.0.5 (Windows/20060719) MIME-Version: 1.0 To: sipping@ietf.org, volkerh@bell-labs.com Content-Type: text/plain; charset=ISO-2022-JP Content-Transfer-Encoding: 7bit X-Spam-Score: 0.0 (/) X-Scan-Signature: e1e48a527f609d1be2bc8d8a70eb76cb Cc: Subject: [Sipping] Review of draft-ietf-sipping-policy-package-01 X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list Reply-To: shida@ntt-at.com List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Draft: draft-ietf-sipping-policy-package-01 Reviewer: Shida Schubert Review Date: 28 Aug 2006 Review Deadline: 25 Aug 2006 Status: Initial review Summary: This draft is on the right track but has open issues, described in the review. I only have some technical concerns: Section 3.6/3.9 May be it belongs in the framework document but there is no normative behavior in neither of documents for UAC or UAS when policy server either returns an error upon Subscription or when Notification does not arrive within reasonable time frame. Section 3.7 -1st paragraph, 2nd sentence: Because remote policy server may be contacted, the techniques listed in the draft might not suffice the authentication/authorization. Mentioning of P-A-ID/AIB/SIP-Identity as a potential way to authenticate a client(subscriber) may be good. Section 4 All the security consideration focuses on UA but with policy server possibly providing policy information of the domain to an outside entity, there is a security consideration surrounding the domain's security as well. If no proper security is set for providing policy information to an entity outside of the domain, malicious user can gain access on potential attacking point such as IP/port to use etc. Currently all you need is policy server's URI to gain access to this, as there is no correlation between the request that caused 488 and the SUBSCRIBE request. Regards Shida _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Wed Aug 30 06:25:03 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GINDV-0004mD-0E; Wed, 30 Aug 2006 06:22:41 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GINDT-0004m8-LE for sipping@ietf.org; Wed, 30 Aug 2006 06:22:39 -0400 Received: from mxs2.siemens.at ([194.138.12.133] helo=atvies1zrx.siemens.at) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GINDK-0001Vu-33 for sipping@ietf.org; Wed, 30 Aug 2006 06:22:39 -0400 Received: from vies1kbx.sie.siemens.at ([158.226.129.82]) by atvies1zrx.siemens.at with ESMTP id k7UANCu2011633 for ; Wed, 30 Aug 2006 12:23:13 +0200 Received: from localhost ([158.226.129.98]) by vies1kbx.sie.siemens.at (8.12.11.20060308/8.12.1) with SMTP id k7UAMK8B009582 for ; Wed, 30 Aug 2006 12:22:20 +0200 Received: from prga004a.ww300.siemens.net ([163.242.71.105]) by nets138a.ww300.siemens.net with Microsoft SMTPSVC(6.0.3790.1830); Wed, 30 Aug 2006 12:22:09 +0200 Content-class: urn:content-classes:message MIME-Version: 1.0 Subject: [sipping] Removing media stream from all legs and from own leg in tightly coupled SIP conferencing X-MimeOLE: Produced By Microsoft Exchange V6.5 Date: Wed, 30 Aug 2006 12:22:07 +0200 Message-ID: <3E4278088AD82C48B4663DDFE762CEF301A1BE7D@prga004a.ww300.siemens.net> X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: [sipping] Removing media stream from all legs and from own leg in tightly coupled SIP conferencing Thread-Index: AcbMHiWVQUEUSpC0TV2W6iZhby5fSQ== From: "Sedlacek, Ivo" To: X-OriginalArrivalTime: 30 Aug 2006 10:22:09.0094 (UTC) FILETIME=[2656B660:01C6CC1E] X-Spam-Score: 0.1 (/) X-Scan-Signature: d16ce744298aacf98517bc7c108bd198 X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Content-Type: multipart/mixed; boundary="===============1707038236==" Errors-To: sipping-bounces@ietf.org This is a multi-part message in MIME format. --===============1707038236== Content-class: urn:content-classes:message Content-Type: multipart/alternative; boundary="----_=_NextPart_001_01C6CC1E.26068ABE" This is a multi-part message in MIME format. ------_=_NextPart_001_01C6CC1E.26068ABE Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: quoted-printable Hello, =20 RFC4353 defines the tightly coupled SIP conferencing. =20 One of the operations defined there is 5.6. Adding and Removing Media. =20 It says: =20 ------------------------------------------------------------------------ ------------------------ 5.6. Adding and Removing Media =20 Each conference is composed of a particular set of media that the focus is managing. For example, a conference might contain a video stream and an audio stream. The set of media streams that constitute the conference can be changed by participants. When the set of media in the conference change, the focus will need to generate a re-INVITE to each participant in order to add or remove the media stream to each participant. When a media stream is being added, a participant can reject the offered media stream, in which case it will not receive or contribute to that stream. Rejection of a stream by a participant does not imply that the stream is no longer part of the conference, only that the participant is not involved in it. =20 A SIP re-INVITE can be used by a participant to add or remove a media stream. This is accomplished using the standard offer/answer techniques for adding media streams to a session [13]. This will trigger the focus to generate its own re-INVITEs. ------------------------------------------------------------------------ ------------------------ =20 Given the description above, it is possible that a participant decides to use only subset of the offered media streams when the re-INVITE is delivered to the participant. =20 Since the participant may decide to use a subset of the offered media streams when re-INVITE is delivered to the participant, the participant should also have the same option when the participant initiates re-INVITE. =20 Thus when removing the media stream in re-INVITE, the participant has two possibilities how to remove media stream: 1) either participant wants to remove the media stream from own leg (between participant and focus) 2) or participant wants to remove the media stream from all the legs (completely from the conference) =20 How should these two possibilities be distinguished in SIP or SDP? =20 Kind regards =20 Ivo Sedlacek Siemens ANF DATA MCS CD3 phone: +420 54310 6532 mailto: ivo.sedlacek@siemens.com =20 Mailcode: C0gmHhFp=20 =20 =20 ------_=_NextPart_001_01C6CC1E.26068ABE Content-Type: text/html; charset="us-ascii" Content-Transfer-Encoding: quoted-printable
Hello,
 
RFC4353 defines the = tightly=20 coupled SIP conferencing.
 
One of the operations defined = there is=20 5.6. Adding and Removing Media.
 
It=20 says:
 
------------------------------------------------------= ------------------------------------------
5.6. =20 Adding and Removing Media
 
   Each conference is = composed=20 of a particular set of media that the
   focus is = managing. =20 For example, a conference might contain a video
   stream = and an=20 audio stream.  The set of media streams that = constitute
   the=20 conference can be changed by participants.  When the set of=20 media
   in the conference change, the focus will need to = generate=20 a re-INVITE
   to each participant in order to add or = remove the=20 media stream to
   each participant.  When a media = stream is=20 being added, a participant
   can reject the offered media = stream,=20 in which case it will not
   receive or contribute to that=20 stream.  Rejection of a stream by a
   participant = does not=20 imply that the stream is no longer part of the
   = conference, only=20 that the participant is not involved in it.
 
   A = SIP=20 re-INVITE can be used by a participant to add or remove a = media
  =20 stream.  This is accomplished using the standard=20 offer/answer
   techniques for adding media streams to a = session=20 [13].  This will
   trigger the focus to generate its = own=20 re-INVITEs.
----------------------------------------------------------= --------------------------------------
 
Given=20 the description above, it is possible that a participant decides to use = only=20 subset of the offered media streams when the re-INVITE is delivered to = the=20 participant.
 
Since the participant may decide to use a = subset of=20 the offered media streams when re-INVITE is delivered to the = participant, the=20 participant should also have the same option when the participant = initiates=20 re-INVITE.
 
Thus when removing the media stream in = re-INVITE, the=20 participant has two possibilities how to remove media stream:
1) = either=20 participant wants to remove the media stream from own leg (between = participant=20 and focus)
2) or participant wants to remove the media stream from = all the=20 legs (completely from the conference)
 
How should these two=20 possibilities be distinguished in SIP or SDP?
 
Kind=20 regards
 
Ivo Sedlacek
Siemens
ANF DATA MCS = CD3
phone: +420 54310 6532
mailto: ivo.sedlacek@siemens.com
 
Mailcode: C0gmHhFp
 
 
------_=_NextPart_001_01C6CC1E.26068ABE-- --===============1707038236== Content-Type: text/plain; charset="us-ascii" MIME-Version: 1.0 Content-Transfer-Encoding: 7bit Content-Disposition: inline _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP --===============1707038236==-- From sipping-bounces@ietf.org Wed Aug 30 08:06:09 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GIOoY-0008LN-Df; Wed, 30 Aug 2006 08:05:02 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GIOoX-0008LF-0k for sipping@ietf.org; Wed, 30 Aug 2006 08:05:01 -0400 Received: from sj-iport-5.cisco.com ([171.68.10.87]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GIOoV-0004k3-Hl for sipping@ietf.org; Wed, 30 Aug 2006 08:05:00 -0400 Received: from sj-dkim-8.cisco.com ([171.68.10.93]) by sj-iport-5.cisco.com with ESMTP; 30 Aug 2006 05:04:59 -0700 X-IronPort-AV: i="4.08,187,1154934000"; d="scan'208"; a="315799875:sNHT33978416" Received: from sj-core-3.cisco.com (sj-core-3.cisco.com [171.68.223.137]) by sj-dkim-8.cisco.com (8.12.11.20060308/8.12.11) with ESMTP id k7UC4wk6006507; Wed, 30 Aug 2006 05:04:58 -0700 Received: from xbh-rtp-201.amer.cisco.com (xbh-rtp-201.cisco.com [64.102.31.12]) by sj-core-3.cisco.com (8.12.10/8.12.6) with ESMTP id k7UC4ww7010632; Wed, 30 Aug 2006 05:04:58 -0700 (PDT) Received: from xfe-rtp-201.amer.cisco.com ([64.102.31.38]) by xbh-rtp-201.amer.cisco.com with Microsoft SMTPSVC(6.0.3790.1830); Wed, 30 Aug 2006 08:04:58 -0400 Received: from [10.86.243.47] ([10.86.243.47]) by xfe-rtp-201.amer.cisco.com with Microsoft SMTPSVC(6.0.3790.1830); Wed, 30 Aug 2006 08:04:57 -0400 Message-ID: <44F57EE8.2080708@cisco.com> Date: Wed, 30 Aug 2006 08:04:56 -0400 From: Paul Kyzivat User-Agent: Thunderbird 1.5.0.5 (Windows/20060719) MIME-Version: 1.0 To: "Sedlacek, Ivo" Subject: Re: [sipping] Removing media stream from all legs and from own leg in tightly coupled SIP conferencing References: <3E4278088AD82C48B4663DDFE762CEF301A1BE7D@prga004a.ww300.siemens.net> In-Reply-To: <3E4278088AD82C48B4663DDFE762CEF301A1BE7D@prga004a.ww300.siemens.net> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Content-Transfer-Encoding: 7bit X-OriginalArrivalTime: 30 Aug 2006 12:04:57.0869 (UTC) FILETIME=[833713D0:01C6CC2C] DKIM-Signature: a=rsa-sha1; q=dns; l=3261; t=1156939498; x=1157803498; c=relaxed/relaxed; s=sjdkim8002; h=Content-Type:From:Subject:Content-Transfer-Encoding:MIME-Version; d=cisco.com; i=pkyzivat@cisco.com; z=From:Paul=20Kyzivat=20 |Subject:Re=3A=20[sipping]=20Removing=20media=20stream=20from=20all=20legs=20and= 20from=20own=20leg=0A=20in=09tightly=20coupled=20SIP=20conferencing; X=v=3Dcisco.com=3B=20h=3Dai4H5jUPNgRjHXYlDYmg9gqOjdo=3D; b=lEmrxpkTFqqNPx1MWWi4zKvwUgUGt2oLWyPhvzIPuyt1dPbpIyEVNrndmcKf8X8T4mRgie5j qohaclXHSHZp+PZ4xa96rcaipKn4sU9KyHSDTelMNP+HwIXnnuGmLyX/; Authentication-Results: sj-dkim-8.cisco.com; header.From=pkyzivat@cisco.com; dkim=pass ( sig from cisco.com verified; ); X-Spam-Score: 0.0 (/) X-Scan-Signature: 6cca30437e2d04f45110f2ff8dc1b1d5 Cc: sipping@ietf.org X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Ivo, The focus controls the media it offers to each participant. Using sip signaling a participant can only control the media it uses. Should it so choose, the focus could follow the lead of "special" participant about what to offer to the others. But that owuld be a policy of the focus. Otherwise you would need to use some other mechanism to control the behavior of the focus. Paul Sedlacek, Ivo wrote: > Hello, > > RFC4353 defines the tightly coupled SIP conferencing. > > One of the operations defined there is 5.6. Adding and Removing Media. > > It says: > > ------------------------------------------------------------------------------------------------ > 5.6. Adding and Removing Media > > Each conference is composed of a particular set of media that the > focus is managing. For example, a conference might contain a video > stream and an audio stream. The set of media streams that constitute > the conference can be changed by participants. When the set of media > in the conference change, the focus will need to generate a re-INVITE > to each participant in order to add or remove the media stream to > each participant. When a media stream is being added, a participant > can reject the offered media stream, in which case it will not > receive or contribute to that stream. Rejection of a stream by a > participant does not imply that the stream is no longer part of the > conference, only that the participant is not involved in it. > > A SIP re-INVITE can be used by a participant to add or remove a media > stream. This is accomplished using the standard offer/answer > techniques for adding media streams to a session [13]. This will > trigger the focus to generate its own re-INVITEs. > ------------------------------------------------------------------------------------------------ > > Given the description above, it is possible that a participant decides > to use only subset of the offered media streams when the re-INVITE is > delivered to the participant. > > Since the participant may decide to use a subset of the offered media > streams when re-INVITE is delivered to the participant, the participant > should also have the same option when the participant initiates re-INVITE. > > Thus when removing the media stream in re-INVITE, the participant has > two possibilities how to remove media stream: > 1) either participant wants to remove the media stream from own leg > (between participant and focus) > 2) or participant wants to remove the media stream from all the legs > (completely from the conference) > > How should these two possibilities be distinguished in SIP or SDP? > > Kind regards > > Ivo Sedlacek > Siemens > ANF DATA MCS CD3 > phone: +420 54310 6532 > mailto: ivo.sedlacek@siemens.com > > Mailcode: C0gmHhFp > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP > Use sip-implementors@cs.columbia.edu for questions on current sip > Use sip@ietf.org for new developments of core SIP _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Wed Aug 30 11:06:50 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GIRd9-0001Px-J0; Wed, 30 Aug 2006 11:05:27 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GIRd6-0001Ps-Th for sipping@ietf.org; Wed, 30 Aug 2006 11:05:24 -0400 Received: from mailn.telecomitalia.it ([217.169.121.54]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GIRcz-0006Rk-VX for sipping@ietf.org; Wed, 30 Aug 2006 11:05:24 -0400 thread-index: AcbMRbOi/Wz12td2SJeYVne5M2VY1w== Received: from ptpxch007rm001.idc.cww.telecomitalia.it ([156.54.205.169]) by mailn.telecomitalia.it with Microsoft SMTPSVC(6.0.3790.1830); Wed, 30 Aug 2006 17:05:07 +0200 Received: from ptpxch006rm001.idc.cww.telecomitalia.it ([156.54.205.168]) by ptpxch007rm001.idc.cww.telecomitalia.it with Microsoft SMTPSVC(6.0.3790.1830); Wed, 30 Aug 2006 17:05:07 +0200 Received: from [163.162.233.32] ([163.162.233.32]) by ptpxch006rm001.idc.cww.telecomitalia.it over TLS secured channel with Microsoft SMTPSVC(6.0.3790.1830); Wed, 30 Aug 2006 17:05:07 +0200 Message-ID: <44F5A93A.7020200@telecomitalia.it> Content-Class: urn:content-classes:message Importance: normal Priority: normal X-MimeOLE: Produced By Microsoft MimeOLE V6.00.3790.2757 Date: Wed, 30 Aug 2006 17:05:30 +0200 From: "Silvia Tessa" User-Agent: Debian Thunderbird 1.0.7 (X11/20051017) X-Accept-Language: en-us, en MIME-Version: 1.0 To: "Adam Roach" , "SIPPING WG" Subject: Re: [Sipping] Re: WG: draft-poetzl-sipping-call-completion-01 References: <44F45637.7070500@estacado.net> In-Reply-To: <44F45637.7070500@estacado.net> Content-Type: text/plain; format=flowed; charset="iso-8859-1" Content-Transfer-Encoding: quoted-printable X-OriginalArrivalTime: 30 Aug 2006 15:05:07.0154 (UTC) FILETIME=[AE0D1320:01C6CC45] X-Spam-Score: 0.0 (/) X-Scan-Signature: 33cc095b503da4365ce57c727e553cf1 Cc: drage@LUCENT.COM X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Hi all, comments inline. Adam Roach wrote: > I have a handful of comments on the current call completion document > as it relates to RFC 3265. > > 1. I have some heartburn with the behavior described for the event > package. The SUBSCRIBE/NOTIFY mechanism is intended to = *subscribe* > to remote state, not *manipulate* it. Parameters like > "queue-operation" are obviously intended to act as an RPC > mechanism. It is important that SUBSCRIBE requests remain > idempotent: sending the same SUBSCRIBE request over and over > should not cause users to be inserted into a queue multiple = times. > As this document defines the mechanism, this is not the case. It is not clear to me which exact scenarios you have in mind: if the=20 SUBSCRIBE is not sent on an existing dialog, it does create a new=20 subscription, no matter which event package we are talking about:=20 sending the same SUBSCRIBE request over and over on different dialogs=20 tipically cause users to be notified of status changes multiple times. If the SUBSCRIBE is sent on the same dialog but with different=20 queue-operation it can simply change the internal subscription status=20 (suspended or not): sending over and over the same SUBSCRIBE with a=20 different Expires Header value, causes the SUBSCRIPTION to continuously=20 change its duration attribute, sending it with a different=20 queue-operation (suspend/resume) makes it change its suspension = attribute. If the SUBSCRIBE is sent on the same dialog with the same=20 queue-operation value, no action is taken. > > 2. I find the choice of event header field parameters to return > resource state to be both baroque and in violation of the > description of NOTIFY bodies in section 4.4.5 of RFC 3265. For > example, "queue-state" and "denial-reason" communicate resource > state, and should be sent in the NOTIFY body (not Event header > field parameters). It can be baroque, but not in violation, as far as I understand.... > > 3. Disallowing forked requests for the event package seems an > eccentric choice to me. Since there are necessarily no state > agents in the network for this event package, it is EXTREMELY > likely that subscribe requests will end up getting forked to > multiple terminals. The prospect that CCBS/CCNR can apply to = only > one terminal seems really counterintuitive. I can't imagine how > this will ever seem like the right behavior to users. For = example, > if I have two phones (one at my office in Dallas, where I spend > most of my time, and one at my office in Denver, where I spend > three weeks a year), and you call my number, but give up and > activate the CCNR service: does it make sense that you will > establish a CCNR relationship with my the phone in Colorado = (which > won't be used for another month or so), and REJECT the > relationship with my phone in Dallas (which I'll probably use > within an hour or two)? You are raising a good issue! Actually, this is not solvable by solving=20 the forking or not-forking matter, but introducing a centralised entity, = holding all the queue information. That's actually the idea behind the=20 draft: the call completion event server is a centralized entity=20 monitoring (via dialog event package or just logging users=20 communication, that's out of scope) the user call status. Being=20 centralized, forking could be useless. > > I also have some minor issues with the "P-Service-Indication" header. > In this case, an RFC 3087-based solution will suit your purposes just > fine (that is, the called party generates a GRUU with a GRID that it > recognizes as a CCBS/CCNR service request, and provides it to the > caller. When the caller uses this GRUU, the called party will > recognize the GRID as being related to the CCBS/CCNR service, and = know > that the request is a CCBS/CCNR request.) You'll need to use a GRUU > anyway, since the service as you've defined it is very specific to = the > called device, not to the user's AOR in general. In the above example, the entity generating the GRID is the UAS, but in=20 reality, it would be generated by a centralized entity, that not=20 necessarily would be involved in the new incoming call. In that scenario = the univocally generated grid would not be understood. Therefore, unfortunately, this mechanism is not sufficient. > > I'll outline a solution that I think is much closer to the intention > of RFC 3265 and other SIP documents: > * Called parties, when returning any provisional response (180, = 181, > 182, 183, any others), or any other indication that the called > party is currently unavailable (486, 480, 600) can indicate > support for the CCBS extension using the "Supported" header = field > (NOT the "Allow-Events" header field). They also include a new > header field which contains a GRUU that routes back to the = device; > this GRUU contains a GRID that the device recognizes as being > related to the CCBS/CCNR service. =20 let me understand, are you suggesting the use of the support header, in=20 order not to use the SUBSCRIBE method afterwords ? Regarding the new header, you would see it's necessary any time an error = response (4xx) is generated since the Contact is not allowed, or were=20 you thinking about other gaps ? > > * Called parties can react to such provisional or busy responses = by > sending a message that is an explicit request to insert = themselves > in the queue. These requests are sent to the service request = GRUU. > I'm not yet certain specifically what this looks like (that is, > what method to use), but the request to insert the caller into = the > queue (and to pause and unpause the position in the queue, as is > described in section 5 of the current call completion document) > needs to be EXPLICIT and consistent with the meaning of the = method > in which it is carried -- not IMPLICIT as a side effect of > requesting a subscription. This likely means that you'll either > want a new method or you'll want to use INVITE to set up a queue > management session. Actually, we are not handling implicit or explicit information, but just = defining an event package with different notification policies. > * Given the above, there are several ways to complete the service. > One such option would be: > > [snip] > Finally, I would suggest that you consider the ability to deploy a > system in which some non-terminal server handles queue management > instead of the endpoints -- doing so would allow the "Dallas/Denver" > example I provide above to work in a more sane fashion. The > non-terminal server could either track dialog state by being call > stateful, or it could subscribe to the dialog state for each of the > user's registered devices. The Call-Completion Event Server was meant to be a non-terminal server handling queues, that could, as an implementation matter, base itself upon dialog event package to retrieve the dialog information: what you are asking for, is just the same server that use MYNEWMETHOD instead of SUBSCRIBE, following the same apporch as REFER did ? Best Regards Silvia -------------------------------------------------------------------- CONFIDENTIALITY NOTICE This message and its attachments are addressed solely to the persons = above and may contain confidential information. If you have received the = message in error, be informed that any use of the content hereof is = prohibited. Please return it immediately to the sender and delete the = message. Should you have any questions, please contact us by replying to = webmaster@telecomitalia.it. Thank you www.telecomitalia.it -------------------------------------------------------------------- =20 _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Wed Aug 30 11:27:38 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GIRyN-0002e2-5o; Wed, 30 Aug 2006 11:27:23 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GIRxs-00021j-6a for sipping@ietf.org; Wed, 30 Aug 2006 11:26:52 -0400 Received: from rtp-iport-1.cisco.com ([64.102.122.148]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GIRnO-0000p5-3m for sipping@ietf.org; Wed, 30 Aug 2006 11:16:03 -0400 Received: from rtp-dkim-1.cisco.com ([64.102.121.158]) by rtp-iport-1.cisco.com with ESMTP; 30 Aug 2006 08:16:02 -0700 X-IronPort-AV: i="4.08,189,1154934000"; d="scan'208"; a="38996335:sNHT33933786" Received: from rtp-core-2.cisco.com (rtp-core-2.cisco.com [64.102.124.13]) by rtp-dkim-1.cisco.com (8.12.11.20060308/8.12.11) with ESMTP id k7UFG19H019202; Wed, 30 Aug 2006 11:16:01 -0400 Received: from xbh-rtp-211.amer.cisco.com (xbh-rtp-211.cisco.com [64.102.31.102]) by rtp-core-2.cisco.com (8.12.10/8.12.6) with ESMTP id k7UFFwuK007256; Wed, 30 Aug 2006 11:16:01 -0400 (EDT) Received: from xfe-rtp-202.amer.cisco.com ([64.102.31.21]) by xbh-rtp-211.amer.cisco.com with Microsoft SMTPSVC(6.0.3790.1830); Wed, 30 Aug 2006 11:15:59 -0400 Received: from [161.44.79.176] ([161.44.79.176]) by xfe-rtp-202.amer.cisco.com with Microsoft SMTPSVC(6.0.3790.1830); Wed, 30 Aug 2006 11:15:59 -0400 Message-ID: <44F5ABAF.7010407@cisco.com> Date: Wed, 30 Aug 2006 11:15:59 -0400 From: Paul Kyzivat User-Agent: Thunderbird 1.5.0.5 (Windows/20060719) MIME-Version: 1.0 To: "Sedlacek, Ivo" Subject: Re: [sipping] Removing media stream from all legs and from own leg in tightly coupled SIP conferencing References: <3E4278088AD82C48B4663DDFE762CEF301A1C153@prga004a.ww300.siemens.net> In-Reply-To: <3E4278088AD82C48B4663DDFE762CEF301A1C153@prga004a.ww300.siemens.net> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Content-Transfer-Encoding: 7bit X-OriginalArrivalTime: 30 Aug 2006 15:15:59.0506 (UTC) FILETIME=[32E21720:01C6CC47] DKIM-Signature: a=rsa-sha1; q=dns; l=4946; t=1156950961; x=1157814961; c=relaxed/simple; s=rtpdkim1001; h=Content-Type:From:Subject:Content-Transfer-Encoding:MIME-Version; d=cisco.com; i=pkyzivat@cisco.com; z=From:Paul=20Kyzivat=20 |Subject:Re=3A=20[sipping]=20Removing=20media=20stream=20from=20all=20legs=20and= 20from=20own=20leg=0A=20in=09tightly=20coupled=20SIP=20conferencing |To:=22Sedlacek,=20Ivo=22=20; X=v=3Dcisco.com=3B=20h=3Dai4H5jUPNgRjHXYlDYmg9gqOjdo=3D; b=lQ5NOH6P8HESNl6aoa9QyrsXNBWvuRdRFz2ia8Biz9Z5+AcNLE2z+0gUrmzhr9tKQ9vrcPPp eLzt7Yt29PcIHqAFB4Hgxq3CKQjnoE6lftZiO3gKXO2SGeKpw1SNEtXz; Authentication-Results: rtp-dkim-1.cisco.com; header.From=pkyzivat@cisco.com; dkim=pass ( sig from cisco.com verified; ); X-Spam-Score: 0.0 (/) X-Scan-Signature: 37af5f8fbf6f013c5b771388e24b09e7 Cc: sipping@ietf.org X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Sedlacek, Ivo wrote: > Hello Paul, > > thanks for your mail. > > > The focus controls the media it offers to each participant. Using sip > > signaling a participant can only control the media it uses. > > Yes. The participant directly works only with the media streams it uses. > But it can also cause focus to do some actions > > In RFC4353 there is stated: I think you read too much into what 4353 is saying. Yes it is true that a participant in a conference could send a reinvite to offer a stream it is not currently using, and that other participants also are not adding. The focus could accept that or reject it. If it accepts it, it could offer the same to other participants and bridge the new media - if it wants. There is nothing to force it to do so, and no guarantee that it is capable of doing so. Similarly, if all participants are using two media streams, say voice and video, and one participant removes the video, the focus *could* remove it from all participants. But that might be quite impolite to the other participants who might want to continue using it. It *could* follow the lead of a "special" participant in this regard. For instance if there is a moderator it might not allow any stream unless the moderator accepts that stream. But in general what can be done in this way is limited, because the offer/answer protocol doesn't provide a suitable way to control all the things that a focus might need to do. I suggest you take your question to the xcon wg. I have stopped following their work, but I expect they will have a solution for what you want to do. Thanks, Paul > ------------------------------------------------------------------------------------------------ > 5.6. Adding and Removing Media > ... > The set of media streams that constitute > the conference can be changed by participants. When the set of media > in the conference change, the focus will need to generate a re-INVITE > to each participant in order to add or remove the media stream to > each participant. > ... > ------------------------------------------------------------------------------------------------ > > So there is already a method how a participant offers new media stream > to other participants through usage of the focus. > > For media stream removal, this RFC4353 is not so clear - it is stated > that rejection of a media stream by a participant is valid only for the > participant. Other participants can continue using it. > > ------------------------------------------------------------------------------------------------ > 5.6. Adding and Removing Media > ... > Rejection of a stream by a > participant does not imply that the stream is no longer part of the > conference, only that the participant is not involved in it. > ------------------------------------------------------------------------------------------------ > > > Should it so > > choose, the focus could follow the lead of "special" > > participant about > > what to offer to the others. But that owuld be a policy of the focus. > > Well, such policy is certainly possible. > > Unfortunately, this is quite restrictive. I believe it would work only > in case where there is only one "special participant" and this one uses > all the media streams used in the conference. > > Let's consider the following use case: > - a conference is set up with 3 participants A, B, C. After set up A > uses media streams X, Y, Z; B uses media streams X and Y; C uses media > streams X and Z. B is the "special participant". > - now B sends re-INVITE and wants to add media stream W (otherwise the > conference should stay as it is). So B includes in SDP offer X with > port!=0; Y with port!=0, Z with port=0 and W with port!=0. > - since the focus follows the lead of B, the focus not only offers the > media streams X, Y, W to A and C, but it also offers removal of media > stream Z from A and C (which was not the intention of B). > > Another use case is where there are multiple "special participants", > which use different (partly overlapping) sets of media streams. Which of > them will the focus follow? > > > Otherwise you would need to use some other mechanism to control the > > behavior of the focus. > > There may be situations when the participant is allow to do both > following actions: > 1. remove media stream from own leg > 2. remove media stream from all the legs > > 1. can be achieved by rejecting the media stream as specified in RFC4353. > Is there any IETF RFC or draft which would allow participant to do 2. > (in situations when the participant is allowed to do both 1. or 2.)? > > Kind regards > > Ivo Sedlacek > Siemens > ANF DATA MCS CD3 > phone: +420 54310 6532 > mailto: ivo.sedlacek@siemens.com > > Mailcode: C0gmHhFp _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Wed Aug 30 12:20:32 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GISmQ-0000Rh-FV; Wed, 30 Aug 2006 12:19:06 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GISmP-0000RN-9c for sipping@ietf.org; Wed, 30 Aug 2006 12:19:05 -0400 Received: from [2001:5c0:8fff:fffe::4c3d] (helo=vicuna.estacado.net) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GISmI-0003cA-3m for sipping@ietf.org; Wed, 30 Aug 2006 12:19:05 -0400 Received: from [172.17.2.251] ([172.17.2.251]) (authenticated bits=0) by vicuna.estacado.net (8.13.6/8.13.4) with ESMTP id k7UGIb8A003617 (version=TLSv1/SSLv3 cipher=DHE-RSA-AES256-SHA bits=256 verify=NO); Wed, 30 Aug 2006 11:18:37 -0500 (CDT) (envelope-from adam@estacado.net) Message-ID: <44F5BA46.5060109@estacado.net> Date: Wed, 30 Aug 2006 11:18:14 -0500 From: Adam Roach User-Agent: Thunderbird 1.5.0.5 (Windows/20060719) MIME-Version: 1.0 To: Silvia Tessa Subject: Re: [Sipping] Re: WG: draft-poetzl-sipping-call-completion-01 References: <44F45637.7070500@estacado.net> <44F5A93A.7020200@telecomitalia.it> In-Reply-To: <44F5A93A.7020200@telecomitalia.it> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Content-Transfer-Encoding: 7bit X-Spam-Score: -2.8 (--) X-Scan-Signature: 3002fc2e661cd7f114cb6bae92fe88f1 Cc: drage@LUCENT.COM, SIPPING WG X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Silvia Tessa wrote: > Hi all, > comments inline. > > Adam Roach wrote: > > > I have a handful of comments on the current call completion document > > as it relates to RFC 3265. > > > > 1. I have some heartburn with the behavior described for the event > > package. The SUBSCRIBE/NOTIFY mechanism is intended to *subscribe* > > to remote state, not *manipulate* it. Parameters like > > "queue-operation" are obviously intended to act as an RPC > > mechanism. It is important that SUBSCRIBE requests remain > > idempotent: sending the same SUBSCRIBE request over and over > > should not cause users to be inserted into a queue multiple times. > > As this document defines the mechanism, this is not the case. > > It is not clear to me which exact scenarios you have in mind: if the > SUBSCRIBE is not sent on an existing dialog, it does create a new > subscription, no matter which event package we are talking about: > sending the same SUBSCRIBE request over and over on different dialogs > tipically cause users to be notified of status changes multiple times. > > If the SUBSCRIBE is sent on the same dialog but with different > queue-operation it can simply change the internal subscription status > (suspended or not): sending over and over the same SUBSCRIBE with a > different Expires Header value, causes the SUBSCRIPTION to > continuously change its duration attribute, sending it with a > different queue-operation (suspend/resume) makes it change its > suspension attribute. > > If the SUBSCRIBE is sent on the same dialog with the same > queue-operation value, no action is taken. It's good that you view these operations as idempotent (the document does not make this clear, and the implication is otherwise) -- however, the issue remains that you are creating and manipulating NON-SUBSCRIPTION state with SUBSCRIBE messages. That's not what SUBSCRIBE is for, and you'll almost certainly run into implementation issues going down this path. > > > > 2. I find the choice of event header field parameters to return > > resource state to be both baroque and in violation of the > > description of NOTIFY bodies in section 4.4.5 of RFC 3265. For > > example, "queue-state" and "denial-reason" communicate resource > > state, and should be sent in the NOTIFY body (not Event header > > field parameters). > > It can be baroque, but not in violation, as far as I understand.... "The NOTIFY body is used to report state on the resource being monitored." I'm not sure how this could be any clearer. > You are raising a good issue! Actually, this is not solvable by > solving the forking or not-forking matter, but introducing a > centralised entity, holding all the queue information. That's actually > the idea behind the draft: the call completion event server is a > centralized entity monitoring (via dialog event package or just > logging users communication, that's out of scope) the user call status. If such is the case, then the document has far more problems than I originally determined. Are you expecting this centralized entity to add "Allow-Events" header fields to the 486 and 1xx responses generated by terminals? Section 3.3.7 is abundantly clear on this topic: "Note that "Allow-Events" headers MUST NOT be inserted by proxies." The signaling flows in section 7 of the current draft certainly imply that the entity processing the INVITE requests is the same entity processing the SUBSCRIBE requests -- and this would be the terminal, not some server in the middle of the network. If the intention is that the mechanism is handled on some centralized server, then the document needs some *serious* rework to make that fact clear. Until the model is clearer, there's no use discussing the remainder of my comments. /a _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Wed Aug 30 13:52:49 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GIUEH-0006M2-4s; Wed, 30 Aug 2006 13:51:57 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GIUEE-0006L0-V1 for sipping@ietf.org; Wed, 30 Aug 2006 13:51:54 -0400 Received: from stsc1260-eth-s1-s1p1-vip.va.neustar.com ([156.154.16.129] helo=chiedprmail1.ietf.org) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GIRbt-0006Cr-FN for sipping@ietf.org; Wed, 30 Aug 2006 11:04:09 -0400 Received: from mxs1.siemens.at ([194.138.12.131] helo=atvies1zqx.siemens.at) by chiedprmail1.ietf.org with esmtp (Exim 4.43) id 1GIRVO-00016G-4f for sipping@ietf.org; Wed, 30 Aug 2006 10:57:32 -0400 Received: from vies1kbx.sie.siemens.at ([158.226.129.82]) by atvies1zqx.siemens.at with ESMTP id k7UEvobB023906; Wed, 30 Aug 2006 16:57:50 +0200 Received: from localhost ([158.226.129.98]) by vies1kbx.sie.siemens.at (8.12.11.20060308/8.12.1) with SMTP id k7UEvFeo002519; Wed, 30 Aug 2006 16:57:15 +0200 Received: from prga004a.ww300.siemens.net ([163.242.71.105]) by nets139a.ww300.siemens.net with Microsoft SMTPSVC(6.0.3790.1830); Wed, 30 Aug 2006 16:51:25 +0200 X-MimeOLE: Produced By Microsoft Exchange V6.5 Content-class: urn:content-classes:message MIME-Version: 1.0 Subject: RE: [sipping] Removing media stream from all legs and from own leg in tightly coupled SIP conferencing Date: Wed, 30 Aug 2006 16:51:23 +0200 Message-ID: <3E4278088AD82C48B4663DDFE762CEF301A1C153@prga004a.ww300.siemens.net> In-Reply-To: <44F57EE8.2080708@cisco.com> X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: [sipping] Removing media stream from all legs and from own leg in tightly coupled SIP conferencing Thread-Index: AcbMLPW2IpNQ9x88S52g6V3zOXNM3QAEI8SA From: "Sedlacek, Ivo" To: "Paul Kyzivat" X-OriginalArrivalTime: 30 Aug 2006 14:51:25.0124 (UTC) FILETIME=[C4154040:01C6CC43] X-Spam-Score: -0.8 (/) X-Scan-Signature: 88b11fc64c1bfdb4425294ef5374ca07 Cc: sipping@ietf.org X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Content-Type: multipart/mixed; boundary="===============1758598430==" Errors-To: sipping-bounces@ietf.org This is a multi-part message in MIME format. --===============1758598430== Content-class: urn:content-classes:message Content-Type: multipart/alternative; boundary="----_=_NextPart_001_01C6CC43.C398C339" This is a multi-part message in MIME format. ------_=_NextPart_001_01C6CC43.C398C339 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: quoted-printable Hello Paul, thanks for your mail. > The focus controls the media it offers to each participant. Using sip > signaling a participant can only control the media it uses. Yes. The participant directly works only with the media streams it uses. But it can also cause focus to do some actions In RFC4353 there is stated: ------------------------------------------------------------------------ ------------------------ 5.6. Adding and Removing Media ... The set of media streams that constitute the conference can be changed by participants. When the set of media in the conference change, the focus will need to generate a re-INVITE to each participant in order to add or remove the media stream to each participant. ... ------------------------------------------------------------------------ ------------------------ So there is already a method how a participant offers new media stream to other participants through usage of the focus. For media stream removal, this RFC4353 is not so clear - it is stated that rejection of a media stream by a participant is valid only for the participant. Other participants can continue using it. ------------------------------------------------------------------------ ------------------------ 5.6. Adding and Removing Media ... Rejection of a stream by a participant does not imply that the stream is no longer part of the conference, only that the participant is not involved in it. ------------------------------------------------------------------------ ------------------------ > Should it so > choose, the focus could follow the lead of "special" > participant about > what to offer to the others. But that owuld be a policy of the focus. Well, such policy is certainly possible. Unfortunately, this is quite restrictive. I believe it would work only in case where there is only one "special participant" and this one uses all the media streams used in the conference. Let's consider the following use case: - a conference is set up with 3 participants A, B, C. After set up A uses media streams X, Y, Z; B uses media streams X and Y; C uses media streams X and Z. B is the "special participant". - now B sends re-INVITE and wants to add media stream W (otherwise the conference should stay as it is). So B includes in SDP offer X with port!=3D0; Y with port!=3D0, Z with port=3D0 and W with port!=3D0. - since the focus follows the lead of B, the focus not only offers the media streams X, Y, W to A and C, but it also offers removal of media stream Z from A and C (which was not the intention of B). Another use case is where there are multiple "special participants", which use different (partly overlapping) sets of media streams. Which of them will the focus follow? > Otherwise you would need to use some other mechanism to control the > behavior of the focus. There may be situations when the participant is allow to do both following actions: 1. remove media stream from own leg 2. remove media stream from all the legs 1. can be achieved by rejecting the media stream as specified in RFC4353. Is there any IETF RFC or draft which would allow participant to do 2. (in situations when the participant is allowed to do both 1. or 2.)? Kind regards Ivo Sedlacek Siemens ANF DATA MCS CD3 phone: +420 54310 6532 mailto: ivo.sedlacek@siemens.com Mailcode: C0gmHhFp=20 ------_=_NextPart_001_01C6CC43.C398C339 Content-Type: text/html; charset="us-ascii" Content-Transfer-Encoding: quoted-printable
Hello Paul,

thanks for your=20 mail.

> The focus controls the media it offers to each = participant.=20 Using sip
> signaling a participant can only control the media it=20 uses.

Yes. The participant directly works only with the media = streams it=20 uses. But it can also cause focus to do some actions

In RFC4353 = there is=20 stated:

----------------------------------------------------------= --------------------------------------
5.6. =20 Adding and Removing Media
   ...
   The set of = media=20 streams that constitute
   the conference can be changed by = participants.  When the set of media
   in the = conference=20 change, the focus will need to generate a re-INVITE
   to = each=20 participant in order to add or remove the media stream = to
   each=20 participant.
  =20 ...
------------------------------------------------------------------= ------------------------------

So=20 there is already a method how a participant offers new media stream to = other=20 participants through usage of the focus.

For media stream = removal, this=20 RFC4353 is not so clear - it is stated that rejection of a media stream = by a=20 participant is valid only for the participant. Other participants can = continue=20 using=20 it.

--------------------------------------------------------------= ----------------------------------
5.6. =20 Adding and Removing Media
   ...
   Rejection = of a=20 stream by a
   participant does not imply that the stream = is no=20 longer part of the
   conference, only that the participant = is not=20 involved in=20 it.
------------------------------------------------------------------= ------------------------------

>=20 Should it so
> choose, the focus could follow the lead of=20 "special"
> participant about
> what to offer to the others. = But=20 that owuld be a policy of the focus.

Well, such policy is = certainly=20 possible.

Unfortunately, this is quite restrictive. I believe it = would=20 work only in case where there is only one "special participant" and this = one=20 uses all the media streams used in the conference.

Let's consider = the=20 following use case:
- a conference is set up with 3 participants A, = B, C.=20 After set up A uses media streams X, Y, Z; B uses media streams X and Y; = C uses=20 media streams X and Z. B is the "special participant".
- now B sends=20 re-INVITE and wants to add media stream W (otherwise the conference = should stay=20 as it is). So B includes in SDP offer X with port!=3D0; Y with = port!=3D0, Z with=20 port=3D0 and W with port!=3D0.
- since the focus follows the lead of = B, the focus=20 not only offers the media streams X, Y, W to A and C, but it also offers = removal=20 of media stream Z from A and C (which was not the intention of=20 B).

Another use case is where there are multiple "special = participants",=20 which use different (partly overlapping) sets of media streams. Which of = them=20 will the focus follow?

> Otherwise you would need to use some = other=20 mechanism to control the
> behavior of the focus.

There may = be=20 situations when the participant is allow to do both following = actions:
1.=20 remove media stream from own leg
2. remove media stream from all the=20 legs

1. can be achieved by rejecting the media stream as = specified in=20 RFC4353.
Is there any IETF RFC or draft which would allow participant = to do=20 2. (in situations when the participant is allowed to do both 1. or=20 2.)?

Kind regards

Ivo Sedlacek
Siemens
ANF DATA MCS=20 CD3
phone: +420 54310 6532
mailto:=20 ivo.sedlacek@siemens.com

Mailcode: C0gmHhFp =
------_=_NextPart_001_01C6CC43.C398C339-- --===============1758598430== Content-Type: text/plain; charset="us-ascii" MIME-Version: 1.0 Content-Transfer-Encoding: 7bit Content-Disposition: inline _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP --===============1758598430==-- From sipping-bounces@ietf.org Wed Aug 30 14:25:44 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GIUjw-00020S-05; Wed, 30 Aug 2006 14:24:40 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GIUjv-00020N-03 for sipping@ietf.org; Wed, 30 Aug 2006 14:24:39 -0400 Received: from rtp-iport-2.cisco.com ([64.102.122.149]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GIUjs-0005ci-Ax for sipping@ietf.org; Wed, 30 Aug 2006 14:24:38 -0400 Received: from rtp-dkim-1.cisco.com ([64.102.121.158]) by rtp-iport-2.cisco.com with ESMTP; 30 Aug 2006 14:24:36 -0400 X-IronPort-AV: i="4.08,189,1154923200"; d="scan'208"; a="99813275:sNHT37240536" Received: from rtp-core-2.cisco.com (rtp-core-2.cisco.com [64.102.124.13]) by rtp-dkim-1.cisco.com (8.12.11.20060308/8.12.11) with ESMTP id k7UIOZOX024865; Wed, 30 Aug 2006 14:24:35 -0400 Received: from xbh-rtp-211.amer.cisco.com (xbh-rtp-211.cisco.com [64.102.31.102]) by rtp-core-2.cisco.com (8.12.10/8.12.6) with ESMTP id k7UIOZuI017394; Wed, 30 Aug 2006 14:24:35 -0400 (EDT) Received: from xmb-rtp-20b.amer.cisco.com ([64.102.31.53]) by xbh-rtp-211.amer.cisco.com with Microsoft SMTPSVC(6.0.3790.1830); Wed, 30 Aug 2006 14:24:35 -0400 X-MimeOLE: Produced By Microsoft Exchange V6.5 Content-class: urn:content-classes:message MIME-Version: 1.0 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: quoted-printable Subject: RE: [Sipping] Re: WG: draft-poetzl-sipping-call-completion-01 Date: Wed, 30 Aug 2006 14:24:33 -0400 Message-ID: <072C5B76F7CEAB488172C6F64B30B5E301DDAD9D@xmb-rtp-20b.amer.cisco.com> X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: [Sipping] Re: WG: draft-poetzl-sipping-call-completion-01 Thread-Index: AcbMULoNv0rejOFwRCOdr0CcG8MV2QADW1VQ From: "Michael Hammer \(mhammer\)" To: "Adam Roach" , "Silvia Tessa" X-OriginalArrivalTime: 30 Aug 2006 18:24:35.0157 (UTC) FILETIME=[8B896850:01C6CC61] DKIM-Signature: a=rsa-sha1; q=dns; l=6199; t=1156962275; x=1157826275; c=relaxed/simple; s=rtpdkim1001; h=Content-Type:From:Subject:Content-Transfer-Encoding:MIME-Version; d=cisco.com; i=mhammer@cisco.com; z=From:=22Michael=20Hammer=20\(mhammer\)=22=20 |Subject:RE=3A=20[Sipping]=20Re=3A=20WG=3A=20draft-poetzl-sipping-call-completion -01 |To:=22Adam=20Roach=22=20, =0A=20=20=20=20=20=20=20=20=22S ilvia=20Tessa=22=20; X=v=3Dcisco.com=3B=20h=3D+ycxeVUwtDJXz/t5FCDDSUi4q/k=3D; b=jT/CoXi1sKi2jgtvdPC0lzuIDXudhROLeTAoMCUcRAycq3yJPg96W1H6bb5FQkW6rQAazsSO +KhwGtXIb2EDeOKfe6+UlnSS8wUfWwyyRE6zxp7VcrjPOhs5fjk4IPdk; Authentication-Results: rtp-dkim-1.cisco.com; header.From=mhammer@cisco.com; dkim=pass ( sig from cisco.com verified; ); X-Spam-Score: 0.0 (/) X-Scan-Signature: 03169bfe4792634a390035a01a6c6d2f Cc: drage@LUCENT.COM, SIPPING WG X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Adam, Tessa, Joachim, If the queue were positioned as an independent presentity with multiple watchers that could include both the calling and called parties;=20 that due to its association with an E.164 number allows an SP to provision it; that allows the called party to SUBSCRIBE as watcher of the status and accept only calls from the head of queue while it contains active callers; that allows the called party to call the active party at the head of the queue in CCBS or CCNA situations; (simulation) **OR** that allows the calling parties to SUBSCRIBE as watchers, get NOTIFIES of their status (head of queue is ephemeral - call now or get dropped), which triggers them to call; (emulation) that allows callers to PUBLISH queue status (based on XML package definition) of: - active: allows moving up in queue as head identities are removed,=20 - suspended: allows moving up, but not to head of queue, - removed: drops from queue altogether; (Note the additional rejection reasons could be added to the responses to PUBLISH.) that does not allow callers to see who else is in the queue (filtered NOTIFYs); that allows the CCBS/CCNA PUBLISH requests to be charged on a per use or subscription basis; would that address the modeling concern? Mike > -----Original Message----- > From: Adam Roach [mailto:adam@estacado.net]=20 > Sent: Wednesday, August 30, 2006 12:18 PM > To: Silvia Tessa > Cc: drage@LUCENT.COM; SIPPING WG > Subject: Re: [Sipping] Re: WG: draft-poetzl-sipping-call-completion-01 >=20 > Silvia Tessa wrote: > > Hi all, > > comments inline. > > > > Adam Roach wrote: > > > > > I have a handful of comments on the current call=20 > completion document=20 > > > as it relates to RFC 3265. > > > > > > 1. I have some heartburn with the behavior described=20 > for the event > > > package. The SUBSCRIBE/NOTIFY mechanism is intended=20 > to *subscribe* > > > to remote state, not *manipulate* it. Parameters like > > > "queue-operation" are obviously intended to act as an RPC > > > mechanism. It is important that SUBSCRIBE requests remain > > > idempotent: sending the same SUBSCRIBE request over and over > > > should not cause users to be inserted into a queue=20 > multiple times. > > > As this document defines the mechanism, this is not the case. > > > > It is not clear to me which exact scenarios you have in=20 > mind: if the=20 > > SUBSCRIBE is not sent on an existing dialog, it does create a new=20 > > subscription, no matter which event package we are talking about: > > sending the same SUBSCRIBE request over and over on=20 > different dialogs=20 > > tipically cause users to be notified of status changes=20 > multiple times. > > > > If the SUBSCRIBE is sent on the same dialog but with different=20 > > queue-operation it can simply change the internal=20 > subscription status=20 > > (suspended or not): sending over and over the same SUBSCRIBE with a=20 > > different Expires Header value, causes the SUBSCRIPTION to=20 > > continuously change its duration attribute, sending it with a=20 > > different queue-operation (suspend/resume) makes it change its=20 > > suspension attribute. > > > > If the SUBSCRIBE is sent on the same dialog with the same=20 > > queue-operation value, no action is taken. >=20 > It's good that you view these operations as idempotent (the=20 > document does not make this clear, and the implication is=20 > otherwise) -- however, the issue remains that you are=20 > creating and manipulating NON-SUBSCRIPTION state with=20 > SUBSCRIBE messages. That's not what SUBSCRIBE is for, and=20 > you'll almost certainly run into implementation issues going=20 > down this path. >=20 > > > > > > 2. I find the choice of event header field parameters to return > > > resource state to be both baroque and in violation of the > > > description of NOTIFY bodies in section 4.4.5 of RFC=20 > 3265. For > > > example, "queue-state" and "denial-reason"=20 > communicate resource > > > state, and should be sent in the NOTIFY body (not=20 > Event header > > > field parameters). > > > > It can be baroque, but not in violation, as far as I understand.... >=20 > "The NOTIFY body is used to report state on the resource being > monitored." >=20 >=20 > I'm not sure how this could be any clearer. >=20 > > You are raising a good issue! Actually, this is not solvable by=20 > > solving the forking or not-forking matter, but introducing a=20 > > centralised entity, holding all the queue information.=20 > That's actually=20 > > the idea behind the draft: the call completion event server is a=20 > > centralized entity monitoring (via dialog event package or just=20 > > logging users communication, that's out of scope) the user=20 > call status. >=20 > If such is the case, then the document has far more problems=20 > than I originally determined. Are you expecting this=20 > centralized entity to add "Allow-Events" header fields to the=20 > 486 and 1xx responses generated by terminals? Section 3.3.7=20 > is abundantly clear on this topic: >=20 > "Note that "Allow-Events" headers MUST NOT be inserted by=20 > proxies." >=20 > =20 >=20 > The signaling flows in section 7 of the current draft=20 > certainly imply that the entity processing the INVITE=20 > requests is the same entity processing the SUBSCRIBE requests=20 > -- and this would be the terminal, not some server in the=20 > middle of the network. If the intention is that the mechanism=20 > is handled on some centralized server, then the document=20 > needs some *serious* rework to make that fact clear. >=20 > Until the model is clearer, there's no use discussing the=20 > remainder of my comments. >=20 > /a >=20 > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP=20 > Use sip-implementors@cs.columbia.edu for questions on current=20 > sip Use sip@ietf.org for new developments of core SIP >=20 _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Wed Aug 30 15:51:06 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GIW4n-0004ze-9a; Wed, 30 Aug 2006 15:50:17 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GIW4Y-0004RT-Ho; Wed, 30 Aug 2006 15:50:02 -0400 Received: from ns1.neustar.com ([2001:503:c779:1a::9c9a:108a]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GIW4Y-0007jA-8S; Wed, 30 Aug 2006 15:50:02 -0400 Received: from stiedprstage1.ietf.org (stiedprstage1.va.neustar.com [10.31.47.10]) by ns1.neustar.com (Postfix) with ESMTP id 2E6C526E5E; Wed, 30 Aug 2006 19:50:02 +0000 (GMT) Received: from ietf by stiedprstage1.ietf.org with local (Exim 4.43) id 1GIW4Y-0003hG-1z; Wed, 30 Aug 2006 15:50:02 -0400 Content-Type: Multipart/Mixed; Boundary="NextPart" Mime-Version: 1.0 To: i-d-announce@ietf.org From: Internet-Drafts@ietf.org Message-Id: Date: Wed, 30 Aug 2006 15:50:02 -0400 X-Spam-Score: -2.5 (--) X-Scan-Signature: c3a18ef96977fc9bcc21a621cbf1174b Cc: sipping@ietf.org Subject: [Sipping] I-D ACTION:draft-ietf-sipping-toip-07.txt X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org --NextPart A New Internet-Draft is available from the on-line Internet-Drafts directories. This draft is a work item of the Session Initiation Proposal Investigation Working Group of the IETF. Title : Framework for real-time text over IP using the Session Initiation Protocol (SIP) Author(s) : A. Van Wijk, G. Gybels Filename : draft-ietf-sipping-toip-07.txt Pages : 27 Date : 2006-8-30 This document lists the essential requirements for real-time Text- over-IP (ToIP) and defines a framework for implementation of all required functions based on the Session Initiation Protocol (SIP) and the Real-Time Transport Protocol (RTP). This includes interworking between Text-over-IP and existing text telephony on the PSTN and other networks. A URL for this Internet-Draft is: http://www.ietf.org/internet-drafts/draft-ietf-sipping-toip-07.txt To remove yourself from the I-D Announcement list, send a message to i-d-announce-request@ietf.org with the word unsubscribe in the body of the message. You can also visit https://www1.ietf.org/mailman/listinfo/I-D-announce to change your subscription settings. Internet-Drafts are also available by anonymous FTP. Login with the username "anonymous" and a password of your e-mail address. After logging in, type "cd internet-drafts" and then "get draft-ietf-sipping-toip-07.txt". A list of Internet-Drafts directories can be found in http://www.ietf.org/shadow.html or ftp://ftp.ietf.org/ietf/1shadow-sites.txt Internet-Drafts can also be obtained by e-mail. Send a message to: mailserv@ietf.org. In the body type: "FILE /internet-drafts/draft-ietf-sipping-toip-07.txt". NOTE: The mail server at ietf.org can return the document in MIME-encoded form by using the "mpack" utility. To use this feature, insert the command "ENCODING mime" before the "FILE" command. To decode the response(s), you will need "munpack" or a MIME-compliant mail reader. Different MIME-compliant mail readers exhibit different behavior, especially when dealing with "multipart" MIME messages (i.e. documents which have been split up into multiple messages), so check your local documentation on how to manipulate these messages. Below is the data which will enable a MIME compliant mail reader implementation to automatically retrieve the ASCII version of the Internet-Draft. --NextPart Content-Type: Multipart/Alternative; Boundary="OtherAccess" --OtherAccess Content-Type: Message/External-body; access-type="mail-server"; server="mailserv@ietf.org" Content-Type: text/plain Content-ID: <2006-8-30142453.I-D@ietf.org> ENCODING mime FILE /internet-drafts/draft-ietf-sipping-toip-07.txt --OtherAccess Content-Type: Message/External-body; name="draft-ietf-sipping-toip-07.txt"; site="ftp.ietf.org"; access-type="anon-ftp"; directory="internet-drafts" Content-Type: text/plain Content-ID: <2006-8-30142453.I-D@ietf.org> --OtherAccess-- --NextPart Content-Type: text/plain; charset="us-ascii" MIME-Version: 1.0 Content-Transfer-Encoding: 7bit Content-Disposition: inline _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP --NextPart-- From sipping-bounces@ietf.org Wed Aug 30 18:26:46 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GIYVZ-0002Q1-T2; Wed, 30 Aug 2006 18:26:05 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GIYVV-0002PP-UO; Wed, 30 Aug 2006 18:26:01 -0400 Received: from stsc1260-eth-s1-s1p1-vip.va.neustar.com ([156.154.16.129] helo=chiedprmail1.ietf.org) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GIWaj-0005FI-Vt; Wed, 30 Aug 2006 16:23:18 -0400 Received: from ns3.neustar.com ([156.154.24.138]) by chiedprmail1.ietf.org with esmtp (Exim 4.43) id 1GIWMA-0002Rr-HG; Wed, 30 Aug 2006 16:08:16 -0400 Received: from stiedprstage1.ietf.org (stiedprstage1.va.neustar.com [10.31.47.10]) by ns3.neustar.com (Postfix) with ESMTP id 1B33317615; Wed, 30 Aug 2006 19:50:02 +0000 (GMT) Received: from ietf by stiedprstage1.ietf.org with local (Exim 4.43) id 1GIW4X-0003gX-Li; Wed, 30 Aug 2006 15:50:01 -0400 Content-Type: Multipart/Mixed; Boundary="NextPart" Mime-Version: 1.0 To: i-d-announce@ietf.org From: Internet-Drafts@ietf.org Message-Id: Date: Wed, 30 Aug 2006 15:50:01 -0400 X-Spam-Score: -5.9 (-----) X-Scan-Signature: b5d20af10c334b36874c0264b10f59f1 Cc: sipping@ietf.org Subject: [Sipping] I-D ACTION:draft-ietf-sipping-dialogusage-03.txt X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org --NextPart A New Internet-Draft is available from the on-line Internet-Drafts directories. This draft is a work item of the Session Initiation Proposal Investigation Working Group of the IETF. Title : Multiple Dialog Usages in the Session Initiation Protocol Author(s) : R. Sparks Filename : draft-ietf-sipping-dialogusage-03.txt Pages : 28 Date : 2006-8-30 Several methods in the Session Initiation Protocol can create an association between endpoints known as a dialog. Some of these methods can also create a different, but related, association within an existing dialog. These multiple associations, or dialog usages, require carefully coordinated processing as they have independent life-cycles, but share common dialog state. This memo argues that multiple dialog usages should be avoided. It discusses alternatives to their use and clarifies essential behavior for elements that cannot currently avoid them. This is an informative document and makes no normative statements of any kind. A URL for this Internet-Draft is: http://www.ietf.org/internet-drafts/draft-ietf-sipping-dialogusage-03.txt To remove yourself from the I-D Announcement list, send a message to i-d-announce-request@ietf.org with the word unsubscribe in the body of the message. You can also visit https://www1.ietf.org/mailman/listinfo/I-D-announce to change your subscription settings. Internet-Drafts are also available by anonymous FTP. Login with the username "anonymous" and a password of your e-mail address. After logging in, type "cd internet-drafts" and then "get draft-ietf-sipping-dialogusage-03.txt". A list of Internet-Drafts directories can be found in http://www.ietf.org/shadow.html or ftp://ftp.ietf.org/ietf/1shadow-sites.txt Internet-Drafts can also be obtained by e-mail. Send a message to: mailserv@ietf.org. In the body type: "FILE /internet-drafts/draft-ietf-sipping-dialogusage-03.txt". NOTE: The mail server at ietf.org can return the document in MIME-encoded form by using the "mpack" utility. To use this feature, insert the command "ENCODING mime" before the "FILE" command. To decode the response(s), you will need "munpack" or a MIME-compliant mail reader. Different MIME-compliant mail readers exhibit different behavior, especially when dealing with "multipart" MIME messages (i.e. documents which have been split up into multiple messages), so check your local documentation on how to manipulate these messages. Below is the data which will enable a MIME compliant mail reader implementation to automatically retrieve the ASCII version of the Internet-Draft. --NextPart Content-Type: Multipart/Alternative; Boundary="OtherAccess" --OtherAccess Content-Type: Message/External-body; access-type="mail-server"; server="mailserv@ietf.org" Content-Type: text/plain Content-ID: <2006-8-30131454.I-D@ietf.org> ENCODING mime FILE /internet-drafts/draft-ietf-sipping-dialogusage-03.txt --OtherAccess Content-Type: Message/External-body; name="draft-ietf-sipping-dialogusage-03.txt"; site="ftp.ietf.org"; access-type="anon-ftp"; directory="internet-drafts" Content-Type: text/plain Content-ID: <2006-8-30131454.I-D@ietf.org> --OtherAccess-- --NextPart Content-Type: text/plain; charset="us-ascii" MIME-Version: 1.0 Content-Transfer-Encoding: 7bit Content-Disposition: inline _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP --NextPart-- From sipping-bounces@ietf.org Wed Aug 30 21:56:39 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GIbmi-0004MU-9n; Wed, 30 Aug 2006 21:56:00 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GIbmh-0004MO-GG for sipping@ietf.org; Wed, 30 Aug 2006 21:55:59 -0400 Received: from stsc1260-eth-s1-s1p1-vip.va.neustar.com ([156.154.16.129] helo=chiedprmail1.ietf.org) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GIXxL-0001kb-B3 for sipping@ietf.org; Wed, 30 Aug 2006 17:50:43 -0400 Received: from mclmx2.mail.saic.com ([149.8.64.32]) by chiedprmail1.ietf.org with esmtp (Exim 4.43) id 1GIXvb-00064c-B4 for sipping@ietf.org; Wed, 30 Aug 2006 17:48:57 -0400 Received: from 0015-its-ieg01.mail.saic.com ([149.8.64.21] [149.8.64.21]) by mclmx2.mail.saic.com for sipping@ietf.org; Wed, 30 Aug 2006 17:48:47 -0400 Received: from mcl-its-exig01.mail.saic.com ([149.8.64.12]) by 0015-its-ieg01.mail.saic.com (SMSSMTP 4.0.5.66) with SMTP id M2006083017484701043 for ; Wed, 30 Aug 2006 17:48:47 -0400 Received: by mcl-its-exig01.mail.saic.com with Internet Mail Service (5.5.2657.72) id ; Wed, 30 Aug 2006 17:48:47 -0400 Message-Id: <62D92A9A02BCC845B202323D49A48426E1CE2E@0591-ITS-EXMP02.us.saic.com> From: "Roy, Radhika R." To: sipping@ietf.org Subject: RE: [Sipping] I-D ACTION:draft-ietf-sipping-toip-07.txt Date: Wed, 30 Aug 2006 17:46:48 -0400 MIME-Version: 1.0 X-Mailer: Internet Mail Service (5.5.2657.72) content-class: urn:content-classes:message x-mimeole: Produced By Microsoft Exchange V6.5 x-originalarrivaltime: 30 Aug 2006 21:48:43.0401 (UTC) FILETIME=[100F1F90:01C6CC7E] Content-Type: text/plain; charset="iso-8859-1" X-Spam-Score: -2.3 (--) X-Scan-Signature: bdc523f9a54890b8a30dd6fd53d5d024 X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Mary: Did WG appointed 2 editors for this? Did WG remove others as co-authors? BR\RRR _____ From: sipping-bounces@ietf.org on behalf of Internet-Drafts@ietf.org Sent: Wed 8/30/2006 3:50 PM To: i-d-announce@ietf.org Cc: sipping@ietf.org Subject: [Sipping] I-D ACTION:draft-ietf-sipping-toip-07.txt A New Internet-Draft is available from the on-line Internet-Drafts directories. This draft is a work item of the Session Initiation Proposal Investigation Working Group of the IETF. Title : Framework for real-time text over IP using the Session Initiation Protocol (SIP) Author(s) : A. Van Wijk, G. Gybels Filename : draft-ietf-sipping-toip-07.txt Pages : 27 Date : 2006-8-30 This document lists the essential requirements for real-time Text- over-IP (ToIP) and defines a framework for implementation of all required functions based on the Session Initiation Protocol (SIP) and the Real-Time Transport Protocol (RTP). This includes interworking between Text-over-IP and existing text telephony on the PSTN and other networks. A URL for this Internet-Draft is: http://www.ietf.org/internet-drafts/draft-ietf-sipping-toip-07.txt To remove yourself from the I-D Announcement list, send a message to i-d-announce-request@ietf.org with the word unsubscribe in the body of the message. You can also visit https://www1.ietf.org/mailman/listinfo/I-D-announce to change your subscription settings. Internet-Drafts are also available by anonymous FTP. Login with the username "anonymous" and a password of your e-mail address. After logging in, type "cd internet-drafts" and then "get draft-ietf-sipping-toip-07.txt". A list of Internet-Drafts directories can be found in http://www.ietf.org/shadow.html or ftp://ftp.ietf.org/ietf/1shadow-sites.txt Internet-Drafts can also be obtained by e-mail. Send a message to: mailserv@ietf.org. In the body type: "FILE /internet-drafts/draft-ietf-sipping-toip-07.txt". NOTE: The mail server at ietf.org can return the document in MIME-encoded form by using the "mpack" utility. To use this feature, insert the command "ENCODING mime" before the "FILE" command. To decode the response(s), you will need "munpack" or a MIME-compliant mail reader. Different MIME-compliant mail readers exhibit different behavior, especially when dealing with "multipart" MIME messages (i.e. documents which have been split up into multiple messages), so check your local documentation on how to manipulate these messages. Below is the data which will enable a MIME compliant mail reader implementation to automatically retrieve the ASCII version of the Internet-Draft. _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Thu Aug 31 01:46:15 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GIfNM-0001BV-3W; Thu, 31 Aug 2006 01:46:04 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GIfNK-0001BQ-EG for sipping@ietf.org; Thu, 31 Aug 2006 01:46:02 -0400 Received: from fw.polycom.co.il ([212.179.41.2] helo=isrexch01.israel.polycom.com) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GIfNJ-0001VS-EM for sipping@ietf.org; Thu, 31 Aug 2006 01:46:02 -0400 X-MimeOLE: Produced By Microsoft Exchange V6.5 Content-class: urn:content-classes:message MIME-Version: 1.0 Subject: RE: [sipping] Removing media stream from all legs and from own legin tightly coupled SIP conferencing Date: Thu, 31 Aug 2006 08:45:54 +0300 Message-ID: <144ED8561CE90C41A3E5908EDECE315C03D3E907@IsrExch01.israel.polycom.com> In-Reply-To: <3E4278088AD82C48B4663DDFE762CEF301A1C153@prga004a.ww300.siemens.net> X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: [sipping] Removing media stream from all legs and from own legin tightly coupled SIP conferencing Thread-Index: AcbMLPW2IpNQ9x88S52g6V3zOXNM3QAEI8SAACCdnjA= From: "Even, Roni" To: "Sedlacek, Ivo" , "Paul Kyzivat" X-Spam-Score: 0.1 (/) X-Scan-Signature: 22e211536bda974b2dcf811522dc525d Cc: sipping@ietf.org X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Content-Type: multipart/mixed; boundary="===============0929007354==" Errors-To: sipping-bounces@ietf.org This is a multi-part message in MIME format. --===============0929007354== Content-class: urn:content-classes:message Content-Type: multipart/alternative; boundary="----_=_NextPart_001_01C6CCC0.BB29FC8C" This is a multi-part message in MIME format. ------_=_NextPart_001_01C6CCC0.BB29FC8C Content-Type: text/plain; charset="US-ASCII" Content-Transfer-Encoding: quoted-printable Hi, RFC 4353 introduction has the following text "This framework presents a general architectural model for these conferences and presents terminology used to discuss such conferences. It also discusses the ways in which SIP itself is involved in conferencing. The aim of the framework is to meet the general requirements for conferencing that are outlined in [3]. This specification alludes to non-SIP-specific mechanisms for achieving several conferencing functions. Those mechanisms are outside the scope of this specification." =20 The questions you are raising have to do with conference functions that are discussed in XCON working group and are addressed in the drafts prepared there. =20 Having said that I would agree with Paul that you can define a specific policy to your conference server that will define its behavior based on users input including having only the common media, adding media if one of the participants adds it. Having more complicated policies will require some sort of protocol and authorization to make changes on the conference media which again in the scope of XCON group. =20 Thanks Roni Even=20 =20 =20 ________________________________ From: Sedlacek, Ivo [mailto:ivo.sedlacek@siemens.com]=20 Sent: Wednesday, August 30, 2006 5:51 PM To: Paul Kyzivat Cc: sipping@ietf.org Subject: RE: [sipping] Removing media stream from all legs and from own legin tightly coupled SIP conferencing =20 Hello Paul, thanks for your mail. > The focus controls the media it offers to each participant. Using sip > signaling a participant can only control the media it uses. Yes. The participant directly works only with the media streams it uses. But it can also cause focus to do some actions In RFC4353 there is stated: ------------------------------------------------------------------------ ------------------------ 5.6. Adding and Removing Media ... The set of media streams that constitute the conference can be changed by participants. When the set of media in the conference change, the focus will need to generate a re-INVITE to each participant in order to add or remove the media stream to each participant. ... ------------------------------------------------------------------------ ------------------------ So there is already a method how a participant offers new media stream to other participants through usage of the focus. For media stream removal, this RFC4353 is not so clear - it is stated that rejection of a media stream by a participant is valid only for the participant. Other participants can continue using it. ------------------------------------------------------------------------ ------------------------ 5.6. Adding and Removing Media ... Rejection of a stream by a participant does not imply that the stream is no longer part of the conference, only that the participant is not involved in it. ------------------------------------------------------------------------ ------------------------ > Should it so > choose, the focus could follow the lead of "special" > participant about > what to offer to the others. But that owuld be a policy of the focus. Well, such policy is certainly possible. Unfortunately, this is quite restrictive. I believe it would work only in case where there is only one "special participant" and this one uses all the media streams used in the conference. Let's consider the following use case: - a conference is set up with 3 participants A, B, C. After set up A uses media streams X, Y, Z; B uses media streams X and Y; C uses media streams X and Z. B is the "special participant". - now B sends re-INVITE and wants to add media stream W (otherwise the conference should stay as it is). So B includes in SDP offer X with port!=3D0; Y with port!=3D0, Z with port=3D0 and W with port!=3D0. - since the focus follows the lead of B, the focus not only offers the media streams X, Y, W to A and C, but it also offers removal of media stream Z from A and C (which was not the intention of B). Another use case is where there are multiple "special participants", which use different (partly overlapping) sets of media streams. Which of them will the focus follow? > Otherwise you would need to use some other mechanism to control the > behavior of the focus. There may be situations when the participant is allow to do both following actions: 1. remove media stream from own leg 2. remove media stream from all the legs 1. can be achieved by rejecting the media stream as specified in RFC4353. Is there any IETF RFC or draft which would allow participant to do 2. (in situations when the participant is allowed to do both 1. or 2.)? Kind regards Ivo Sedlacek Siemens ANF DATA MCS CD3 phone: +420 54310 6532 mailto: ivo.sedlacek@siemens.com Mailcode: C0gmHhFp=20 ------_=_NextPart_001_01C6CCC0.BB29FC8C Content-Type: text/html; charset="US-ASCII" Content-Transfer-Encoding: quoted-printable

Hi,

RFC 4353 introduction has the = following text

"This framework =
presents a general architectural model for
   these conferences and presents =
terminology used to discuss =
such
   conferences.  It also =
discusses the ways in which SIP itself =
is
   involved in conferencing.  =
The aim of the framework is to meet =
the
   general requirements for =
conferencing that are outlined in [3].  =
This
   specification alludes to =
non-SIP-specific mechanisms for =
achieving
   several conferencing =
functions.  Those mechanisms are outside =
the
   scope of this =
specification."
 
The =
questions you are raising have to do with conference functions that are =
discussed in XCON working group and are addressed in the drafts prepared =
there.
 
Having =
said that I would agree with Paul that you can define a specific policy =
to your conference server that will define its behavior based on users =
input including having only the common media, adding media if one of the =
participants adds it. Having more complicated policies will require some =
sort of protocol and authorization to make changes on the conference =
media which again in the scope of XCON =
group.
 
Thanks
Roni Even =

 

 


From: = Sedlacek, Ivo [mailto:ivo.sedlacek@siemens.com]
Sent: Wednesday, August = 30, 2006 5:51 PM
To: Paul Kyzivat
Cc: sipping@ietf.org
Subject: RE: [sipping] = Removing media stream from all legs and from own legin tightly coupled SIP = conferencing

 

Hello Paul,

thanks for your mail.

> The focus controls the media it offers to each participant. Using = sip
> signaling a participant can only control the media it uses.

Yes. The participant directly works only with the media streams it uses. = But it can also cause focus to do some actions

In RFC4353 there is stated:

-------------------------------------------------------------------------= -----------------------
5.6.  Adding and Removing Media
   ...
   The set of media streams that constitute
   the conference can be changed by participants.  When = the set of media
   in the conference change, the focus will need to generate a re-INVITE
   to each participant in order to add or remove the media = stream to
   each participant.
   ...
-------------------------------------------------------------------------= -----------------------

So there is already a method how a participant offers new media stream = to other participants through usage of the focus.

For media stream removal, this RFC4353 is not so clear - it is stated = that rejection of a media stream by a participant is valid only for the = participant. Other participants can continue using it.

-------------------------------------------------------------------------= -----------------------
5.6.  Adding and Removing Media
   ...
   Rejection of a stream by a
   participant does not imply that the stream is no longer = part of the
   conference, only that the participant is not involved in = it.
-------------------------------------------------------------------------= -----------------------

> Should it so
> choose, the focus could follow the lead of "special"
> participant about
> what to offer to the others. But that owuld be a policy of the = focus.

Well, such policy is certainly possible.

Unfortunately, this is quite restrictive. I believe it would work only = in case where there is only one "special participant" and this one = uses all the media streams used in the conference.

Let's consider the following use case:
- a conference is set up with 3 participants A, B, C. After set up A = uses media streams X, Y, Z; B uses media streams X and Y; C uses media streams X = and Z. B is the "special participant".
- now B sends re-INVITE and wants to add media stream W (otherwise the conference should stay as it is). So B includes in SDP offer X with = port!=3D0; Y with port!=3D0, Z with port=3D0 and W with port!=3D0.
- since the focus follows the lead of B, the focus not only offers the = media streams X, Y, W to A and C, but it also offers removal of media stream Z = from A and C (which was not the intention of B).

Another use case is where there are multiple "special = participants", which use different (partly overlapping) sets of media streams. Which of = them will the focus follow?

> Otherwise you would need to use some other mechanism to control = the
> behavior of the focus.

There may be situations when the participant is allow to do both = following actions:
1. remove media stream from own leg
2. remove media stream from all the legs

1. can be achieved by rejecting the media stream as specified in = RFC4353.
Is there any IETF RFC or draft which would allow participant to do 2. = (in situations when the participant is allowed to do both 1. or 2.)?

Kind regards

Ivo Sedlacek
Siemens
ANF DATA MCS CD3
phone: +420 54310 6532
mailto: ivo.sedlacek@siemens.com

Mailcode: C0gmHhFp

------_=_NextPart_001_01C6CCC0.BB29FC8C-- --===============0929007354== Content-Type: text/plain; charset="us-ascii" MIME-Version: 1.0 Content-Transfer-Encoding: 7bit Content-Disposition: inline _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP --===============0929007354==-- From sipping-bounces@ietf.org Thu Aug 31 02:56:46 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GIgTI-0000pB-RP; Thu, 31 Aug 2006 02:56:16 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GIgTH-0000o1-Fa; Thu, 31 Aug 2006 02:56:15 -0400 Received: from outbound.cantata.com ([208.236.123.102]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GIgTG-00073A-5v; Thu, 31 Aug 2006 02:56:15 -0400 Received: from ma02exchtmp01.Cantata.com ([10.128.18.41]) by outbound.Cantata.com with Microsoft SMTPSVC(6.0.3790.1830); Thu, 31 Aug 2006 02:56:18 -0400 Received: from 10.128.41.30 ([10.128.41.30]) by ma02exchtmp01.Cantata.com ([10.128.18.41]) with Microsoft Exchange Server HTTP-DAV ; Thu, 31 Aug 2006 06:56:16 +0000 User-Agent: Microsoft-Entourage/11.2.5.060620 Date: Thu, 31 Aug 2006 02:55:56 -0400 From: Eric Burger To: Sipping Message-ID: Thread-Topic: [Sip] question about RFC4240 (NETANN) Thread-Index: AcbMxxuVWjE78ji6EdutvgAWy5cvmwAA2Y5z In-Reply-To: Mime-version: 1.0 Content-type: text/plain; charset="US-ASCII" Content-transfer-encoding: 7bit X-OriginalArrivalTime: 31 Aug 2006 06:56:18.0029 (UTC) FILETIME=[8EF181D0:01C6CCCA] X-Spam-Score: 0.0 (/) X-Scan-Signature: 0fa76816851382eb71b0a882ccdc29ac Cc: SIP List Subject: [Sipping] Re: [Sip] question about RFC4240 (NETANN) X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Whoops - meant to post to SIPPING list. Please follow-up on SIPPING, not SIP. On 8/31/06 2:31 AM, "Eric Burger" wrote: > Oooh - good catch of what looks like an error. From a strict > constructionist point of view, the page 7 text is normative, so underbar > reigns. However, one can make a very convincing argument the reference to > RFC 3066 is normative, hence it should be a hyphen. > > AFAIK, all implementations to date of RFC 4240 use underbar. Is there > enough confusion out there to create an update to 4240? Note that 4240 is > not a PS, so this isn't a DS opportunity. > > > On 8/25/06 7:48 AM, "olivier.ro.rousseau@alcatel.be" > wrote: > >> Hello, >> >> I have a problem with the RFC 4240. >> The language parameter is defined in the page 10. >> locale-param = "locale=" token >> ; per RFC 3066, usually >> ; ISO639-1_ISO3166-1 >> ; e.g., en, en_US, en_UK, etc. >> and is explained in page 7 >> locale >> Specifies the language and optionally country variant of the >> announcement sequence named in the "play=" parameter. RFC 3066 >> [9] specifies the locale tag. The locale tag is usually a two- or >> three-letter code per ISO 639-1 [11]. The country variant is also >> often a two-letter code per ISO 3166-1 [12]. These elements are >> concatenated with a single under bar (%x5F) character, such as >> "en_CA". >> >> However whe I look in the RFC3066 I found >> Language-Tag = Primary-subtag *( "-" Subtag ) >> >> Primary-subtag = 1*8ALPHA >> >> Subtag = 1*8(ALPHA / DIGIT) >> >> In the RFC3261, the Accept-Langugae is defined as >> Accept-Language = "Accept-Language" HCOLON >> [ language *(COMMA language) ] >> language = language-range *(SEMI accept-param) >> language-range = ( ( 1*8ALPHA *( "-" 1*8ALPHA ) ) / "*" ) >> >> You can see that one RFC says that a "_" must be used between the two >> subtags while the two others says that it must be a "-". >> And worse the RFC4240 refers to RFC3066 but uses a different syntax in >> its BNF and in its text ( >> >> Could someone clarify what I should use? >> >> Thanks and regards, >> Olivier >> >> >> >> _______________________________________________ >> Sip mailing list https://www1.ietf.org/mailman/listinfo/sip >> This list is for NEW development of the core SIP Protocol >> Use sip-implementors@cs.columbia.edu for questions on current sip >> Use sipping@ietf.org for new developments on the application of sip >> > > > _______________________________________________ > Sip mailing list https://www1.ietf.org/mailman/listinfo/sip > This list is for NEW development of the core SIP Protocol > Use sip-implementors@cs.columbia.edu for questions on current sip > Use sipping@ietf.org for new developments on the application of sip > _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Thu Aug 31 06:52:30 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GIk7x-00072i-Em; Thu, 31 Aug 2006 06:50:29 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GIk7v-00072d-GJ for sipping@ietf.org; Thu, 31 Aug 2006 06:50:27 -0400 Received: from tcmail23.telekom.de ([217.6.95.237]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GIk7u-0003GJ-Uj for sipping@ietf.org; Thu, 31 Aug 2006 06:50:27 -0400 Received: from g9jbr.mgb01.telekom.de (g9jbr.mgb01.telekom.de [164.20.31.6]) by tcmail21.dmz.telekom.de with ESMTP; Thu, 31 Aug 2006 12:50:19 +0200 Received: by G9JBR.mgb01.telekom.de with Internet Mail Service (5.5.2653.19) id ; Thu, 31 Aug 2006 12:50:18 +0200 Message-Id: From: "Huelsemann, Martin" To: mhammer@cisco.com Subject: AW: [Sipping] Re: WG: draft-poetzl-sipping-call-completion-01 Date: Thu, 31 Aug 2006 12:50:07 +0200 MIME-Version: 1.0 X-Mailer: Internet Mail Service (5.5.2653.19) Content-Type: text/plain; charset="iso-8859-1" Content-Transfer-Encoding: quoted-printable X-Spam-Score: 0.0 (/) X-Scan-Signature: 93df555cbdbcdae9621e5b95d44b301e Cc: sipping@ietf.org X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Hi Mike, your idea seems to be good, the only thing we have to keep in mind is = the interworking to the PSTN, where the notification 'instance' (on a = MGC) has more the behaviour of a normal UA.=20 So a solution for call completion should work if the notifier is a UA = as well it should work with a centralized server. That we have to = check. Best regards, Martin -----Urspr=FCngliche Nachricht----- Von: Michael Hammer (mhammer) [mailto:mhammer@cisco.com]=20 Gesendet: Mittwoch, 30. August 2006 20:25 An: Adam Roach; Silvia Tessa Cc: drage@LUCENT.COM; SIPPING WG Betreff: RE: [Sipping] Re: WG: draft-poetzl-sipping-call-completion-01 Adam, Tessa, Joachim, If the queue were positioned as an independent presentity with multiple watchers that could include both the calling and called parties;=20 that due to its association with an E.164 number allows an SP to provision it; that allows the called party to SUBSCRIBE as watcher of the status and accept only calls from the head of queue while it contains active callers; that allows the called party to call the active party at the head of = the queue in CCBS or CCNA situations; (simulation) **OR** that allows the calling parties to SUBSCRIBE as watchers, get NOTIFIES of their status (head of queue is ephemeral - call now or get dropped), which triggers them to call; (emulation) that allows callers to PUBLISH queue status (based on XML package definition) of: - active: allows moving up in queue as head identities are removed,=20 - suspended: allows moving up, but not to head of queue, - removed: drops from queue altogether; (Note the additional rejection reasons could be added to the responses to PUBLISH.) that does not allow callers to see who else is in the queue (filtered NOTIFYs); that allows the CCBS/CCNA PUBLISH requests to be charged on a per use = or subscription basis; would that address the modeling concern? Mike > -----Original Message----- > From: Adam Roach [mailto:adam@estacado.net]=20 > Sent: Wednesday, August 30, 2006 12:18 PM > To: Silvia Tessa > Cc: drage@LUCENT.COM; SIPPING WG > Subject: Re: [Sipping] Re: WG: = draft-poetzl-sipping-call-completion-01 >=20 > Silvia Tessa wrote: > > Hi all, > > comments inline. > > > > Adam Roach wrote: > > > > > I have a handful of comments on the current call=20 > completion document=20 > > > as it relates to RFC 3265. > > > > > > 1. I have some heartburn with the behavior described=20 > for the event > > > package. The SUBSCRIBE/NOTIFY mechanism is intended=20 > to *subscribe* > > > to remote state, not *manipulate* it. Parameters like > > > "queue-operation" are obviously intended to act as an RPC > > > mechanism. It is important that SUBSCRIBE requests remain > > > idempotent: sending the same SUBSCRIBE request over and over > > > should not cause users to be inserted into a queue=20 > multiple times. > > > As this document defines the mechanism, this is not the = case. > > > > It is not clear to me which exact scenarios you have in=20 > mind: if the=20 > > SUBSCRIBE is not sent on an existing dialog, it does create a new=20 > > subscription, no matter which event package we are talking about: > > sending the same SUBSCRIBE request over and over on=20 > different dialogs=20 > > tipically cause users to be notified of status changes=20 > multiple times. > > > > If the SUBSCRIBE is sent on the same dialog but with different=20 > > queue-operation it can simply change the internal=20 > subscription status=20 > > (suspended or not): sending over and over the same SUBSCRIBE with a = > > different Expires Header value, causes the SUBSCRIPTION to=20 > > continuously change its duration attribute, sending it with a=20 > > different queue-operation (suspend/resume) makes it change its > > suspension attribute. > > > > If the SUBSCRIBE is sent on the same dialog with the same=20 > > queue-operation value, no action is taken. >=20 > It's good that you view these operations as idempotent (the=20 > document does not make this clear, and the implication is=20 > otherwise) -- however, the issue remains that you are=20 > creating and manipulating NON-SUBSCRIPTION state with=20 > SUBSCRIBE messages. That's not what SUBSCRIBE is for, and=20 > you'll almost certainly run into implementation issues going=20 > down this path. >=20 > > > > > > 2. I find the choice of event header field parameters to return > > > resource state to be both baroque and in violation of the > > > description of NOTIFY bodies in section 4.4.5 of RFC=20 > 3265. For > > > example, "queue-state" and "denial-reason"=20 > communicate resource > > > state, and should be sent in the NOTIFY body (not=20 > Event header > > > field parameters). > > > > It can be baroque, but not in violation, as far as I understand.... >=20 > "The NOTIFY body is used to report state on the resource being > monitored." >=20 >=20 > I'm not sure how this could be any clearer. >=20 > > You are raising a good issue! Actually, this is not solvable by=20 > > solving the forking or not-forking matter, but introducing a=20 > > centralised entity, holding all the queue information.=20 > That's actually=20 > > the idea behind the draft: the call completion event server is a=20 > > centralized entity monitoring (via dialog event package or just=20 > > logging users communication, that's out of scope) the user=20 > call status. >=20 > If such is the case, then the document has far more problems=20 > than I originally determined. Are you expecting this=20 > centralized entity to add "Allow-Events" header fields to the=20 > 486 and 1xx responses generated by terminals? Section 3.3.7=20 > is abundantly clear on this topic: >=20 > "Note that "Allow-Events" headers MUST NOT be inserted by=20 > proxies." >=20 > =20 >=20 > The signaling flows in section 7 of the current draft=20 > certainly imply that the entity processing the INVITE=20 > requests is the same entity processing the SUBSCRIBE requests=20 > -- and this would be the terminal, not some server in the=20 > middle of the network. If the intention is that the mechanism=20 > is handled on some centralized server, then the document=20 > needs some *serious* rework to make that fact clear. >=20 > Until the model is clearer, there's no use discussing the=20 > remainder of my comments. >=20 > /a >=20 > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP=20 > Use sip-implementors@cs.columbia.edu for questions on current=20 > sip Use sip@ietf.org for new developments of core SIP >=20 _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Thu Aug 31 09:51:53 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GImwY-00041T-SY; Thu, 31 Aug 2006 09:50:54 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GImwX-00041J-B6 for sipping@ietf.org; Thu, 31 Aug 2006 09:50:53 -0400 Received: from maill.telecomitalia.it ([217.169.121.52]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GImwR-0006iZ-0A for sipping@ietf.org; Thu, 31 Aug 2006 09:50:53 -0400 thread-index: AcbNBHH/I/A+H9lSSNSn6FrZsfqfuQ== Received: from ptpxch008rm001.idc.cww.telecomitalia.it ([156.54.205.170]) by maill.telecomitalia.it with Microsoft SMTPSVC(6.0.3790.1830); Thu, 31 Aug 2006 15:50:40 +0200 Received: from ptpxch006rm001.idc.cww.telecomitalia.it ([156.54.205.168]) by ptpxch008rm001.idc.cww.telecomitalia.it with Microsoft SMTPSVC(6.0.3790.1830); Thu, 31 Aug 2006 15:50:40 +0200 Received: from [163.162.3.5] ([163.162.3.5]) by ptpxch006rm001.idc.cww.telecomitalia.it with Microsoft SMTPSVC(6.0.3790.1830); Thu, 31 Aug 2006 15:50:39 +0200 Content-Class: urn:content-classes:message Importance: normal Priority: normal X-MimeOLE: Produced By Microsoft MimeOLE V6.00.3790.2757 Message-ID: <44F6E882.6080900@telecomitalia.it> Date: Thu, 31 Aug 2006 15:47:46 +0200 From: "Alberto Cuda" User-Agent: Thunderbird 1.5.0.5 (X11/20060812) MIME-Version: 1.0 To: "Adam Roach" Subject: Re: [Sipping] Re: WG: draft-poetzl-sipping-call-completion-01 References: <44F45637.7070500@estacado.net> <44F5A93A.7020200@telecomitalia.it> <44F5BA46.5060109@estacado.net> In-Reply-To: <44F5BA46.5060109@estacado.net> Content-Type: text/plain; format=flowed; charset="iso-8859-1" Content-Transfer-Encoding: quoted-printable X-OriginalArrivalTime: 31 Aug 2006 13:50:39.0474 (UTC) FILETIME=[7184F520:01C6CD04] X-Spam-Score: 0.0 (/) X-Scan-Signature: 67c1ea29f88502ef6a32ccec927970f0 Cc: drage@LUCENT.COM, SIPPING WG X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Adam Roach wrote: > Silvia Tessa wrote: >> >> If the SUBSCRIBE is sent on the same dialog with the same=20 >> queue-operation value, no action is taken. > > It's good that you view these operations as idempotent (the document=20 > does not make this clear, and the implication is otherwise)=20 Yes, probably some clarification on this should be added. > -- however, the issue remains that you are creating and manipulating=20 > NON-SUBSCRIPTION state with SUBSCRIBE messages. That's not what=20 > SUBSCRIBE is for, and you'll almost certainly run into implementation=20 > issues going down this path. From my perspective, we can split an event server in different = subsystems: + A database containing all the subscriptions (updated by SUBSCRIBE=20 requests); + A database containing all the state documents (updated by PUBLISH=20 requests); + A subsystem that looks up the subscription DB to send NOTIFY requests = when some change in the remote state happens. The third subsystem usually follows a straightforward approach: send a=20 NOTIFY request for each subscription, however what is needed here is to notify=20 one watcher at a time, and sometimes watchers even need to suspend their=20 subscriptions without loosing their place in this queue. Now, what is the best way for=20 watchers to perform these operations? Since their are operating on attributes of their=20 subscriptions, the SUBSCRIBE method looks like the most obvious tool to do this, at=20 least for us: do you think that the SUBSCRIBE body would be a better choice for = carrying the "suspended" information? >> > 2. I find the choice of event header field parameters to return >> > resource state to be both baroque and in violation of the >> > description of NOTIFY bodies in section 4.4.5 of RFC 3265. For >> > example, "queue-state" and "denial-reason" communicate = resource >> > state, and should be sent in the NOTIFY body (not Event header >> > field parameters). >> >> It can be baroque, but not in violation, as far as I understand.... > > "The NOTIFY body is used to report state on the resource being > monitored." > > I'm not sure how this could be any clearer. I agree the body is the best place where to put this information. > >> You are raising a good issue! Actually, this is not solvable by=20 >> solving the forking or not-forking matter, but introducing a=20 >> centralised entity, holding all the queue information. That's=20 >> actually the idea behind the draft: the call completion event server=20 >> is a centralized entity monitoring (via dialog event package or just=20 >> logging users communication, that's out of scope) the user call = status. > > If such is the case, then the document has far more problems than I=20 > originally determined. Are you expecting this centralized entity to=20 > add "Allow-Events" header fields to the 486 and 1xx responses=20 > generated by terminals? Section 3.3.7 is abundantly clear on this = topic: > > "Note that "Allow-Events" headers MUST NOT be inserted by proxies." In that case the centralized entity would not be a proxy, but a B2BUA. > > =20 > The signaling flows in section 7 of the current draft certainly imply=20 > that the entity processing the INVITE requests is the same entity=20 > processing the SUBSCRIBE requests -- and this would be the terminal,=20 > not some server in the middle of the network. If the intention is that = > the mechanism is handled on some centralized server, then the document = > needs some *serious* rework to make that fact clear. > > Until the model is clearer, there's no use discussing the remainder of = > my comments. Actually we did not make an explicit reference to the possible=20 architectural deployments to keep the approach as general as possible (the event server can be=20 either an UA or a centralized entity, acting as a B2BUA). The latter case can solve=20 the forking problem you mentioned: the B2BUA is involved in each call and keeps the queue so = it can both put the Allow-Events header, and monitor the state of the callee's=20 terminals. Cheers. Alberto. -------------------------------------------------------------------- CONFIDENTIALITY NOTICE This message and its attachments are addressed solely to the persons = above and may contain confidential information. If you have received the = message in error, be informed that any use of the content hereof is = prohibited. Please return it immediately to the sender and delete the = message. Should you have any questions, please contact us by replying to = webmaster@telecomitalia.it. Thank you www.telecomitalia.it -------------------------------------------------------------------- =20 _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Thu Aug 31 11:10:21 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GIo9d-0003aQ-Pg; Thu, 31 Aug 2006 11:08:29 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GIo9c-0003aC-81 for sipping@ietf.org; Thu, 31 Aug 2006 11:08:28 -0400 Received: from rtp-iport-2.cisco.com ([64.102.122.149]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GIo9a-0006pa-PP for sipping@ietf.org; Thu, 31 Aug 2006 11:08:28 -0400 Received: from rtp-dkim-1.cisco.com ([64.102.121.158]) by rtp-iport-2.cisco.com with ESMTP; 31 Aug 2006 11:08:17 -0400 X-IronPort-AV: i="4.08,195,1154923200"; d="scan'208"; a="100015666:sNHT24844756098" Received: from rtp-core-1.cisco.com (rtp-core-1.cisco.com [64.102.124.12]) by rtp-dkim-1.cisco.com (8.12.11.20060308/8.12.11) with ESMTP id k7VF8HrR022899; Thu, 31 Aug 2006 11:08:17 -0400 Received: from xbh-rtp-201.amer.cisco.com (xbh-rtp-201.cisco.com [64.102.31.12]) by rtp-core-1.cisco.com (8.12.10/8.12.6) with ESMTP id k7VF8HdM009122; Thu, 31 Aug 2006 11:08:17 -0400 (EDT) Received: from xmb-rtp-20b.amer.cisco.com ([64.102.31.53]) by xbh-rtp-201.amer.cisco.com with Microsoft SMTPSVC(6.0.3790.1830); Thu, 31 Aug 2006 11:08:17 -0400 X-MimeOLE: Produced By Microsoft Exchange V6.5 Content-class: urn:content-classes:message MIME-Version: 1.0 Content-Type: text/plain; charset="iso-8859-1" Content-Transfer-Encoding: quoted-printable Subject: RE: [Sipping] Re: WG: draft-poetzl-sipping-call-completion-01 Date: Thu, 31 Aug 2006 11:08:16 -0400 Message-ID: <072C5B76F7CEAB488172C6F64B30B5E301DDB00E@xmb-rtp-20b.amer.cisco.com> X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: [Sipping] Re: WG: draft-poetzl-sipping-call-completion-01 Thread-Index: AcbM61csnGXkeR8VT2aUwplLgAPzAgAImTTw From: "Michael Hammer \(mhammer\)" To: "Huelsemann, Martin" X-OriginalArrivalTime: 31 Aug 2006 15:08:17.0070 (UTC) FILETIME=[49A9B4E0:01C6CD0F] DKIM-Signature: a=rsa-sha1; q=dns; l=8745; t=1157036897; x=1157900897; c=relaxed/simple; s=rtpdkim1001; h=Content-Type:From:Subject:Content-Transfer-Encoding:MIME-Version; d=cisco.com; i=mhammer@cisco.com; z=From:=22Michael=20Hammer=20\(mhammer\)=22=20 |Subject:RE=3A=20[Sipping]=20Re=3A=20WG=3A=20draft-poetzl-sipping-call-completion -01 |To:=22Huelsemann,=20Martin=22=20; X=v=3Dcisco.com=3B=20h=3DAsKpkrRAn76+Gx1vNJF4kGRhLbo=3D; b=b72pv51E+NIv2RNiFUb0Gw2AHUdErVfupTt/TDCU0aZzAy7ZVlGh6A61CbMSfWnxsBbEEyMV wEaS3/sK5fZzs+G6770vYF0ztcwOPV3zOkytdrirEijkTMa9aWnxcnQT; Authentication-Results: rtp-dkim-1.cisco.com; header.From=mhammer@cisco.com; dkim=pass ( sig from cisco.com verified; ); X-Spam-Score: 0.0 (/) X-Scan-Signature: 0bb031f3a6fb29f760794ac9bf1997ae Cc: sipping@ietf.org X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Martin, If the queue management presentity is co-located on the PSTN GW with the = UA tied to the IWF to the PSTN, does that effectively achieve the same = effect? Also, the reason I used PUBLISH versus SUBSCRIBE to push data to the = presentity, is that I was thinking the queue was a composite of the = presence of the contributing callers. So, SUBSCRIBE is used just to = watch the resulting composite queue state. I had thought that SUBs = could send filter bodies, but was not sure if they can push data into = the presence as well. Perhaps others could comment on that. In the past (emulation), Notifies would be sent only when the head of = queue is reached, but it might be useful to see ones request progress up = the queue, in case one wants to cancel. Mike =20 > -----Original Message----- > From: Huelsemann, Martin [mailto:Martin.Huelsemann@t-com.net]=20 > Sent: Thursday, August 31, 2006 6:50 AM > To: Michael Hammer (mhammer) > Cc: sipping@ietf.org > Subject: AW: [Sipping] Re: WG: draft-poetzl-sipping-call-completion-01 >=20 >=20 > Hi Mike, >=20 > your idea seems to be good, the only thing we have to keep in=20 > mind is the interworking to the PSTN, where the notification=20 > 'instance' (on a MGC) has more the behaviour of a normal UA.=20 > So a solution for call completion should work if the notifier=20 > is a UA as well it should work with a centralized server.=20 > That we have to check. >=20 >=20 >=20 > Best regards, Martin >=20 >=20 >=20 >=20 > -----Urspr=FCngliche Nachricht----- > Von: Michael Hammer (mhammer) [mailto:mhammer@cisco.com] > Gesendet: Mittwoch, 30. August 2006 20:25 > An: Adam Roach; Silvia Tessa > Cc: drage@LUCENT.COM; SIPPING WG > Betreff: RE: [Sipping] Re: WG: draft-poetzl-sipping-call-completion-01 >=20 >=20 > Adam, Tessa, Joachim, >=20 > If the queue were positioned as an independent presentity=20 > with multiple > watchers that could include both the calling and called parties;=20 >=20 > that due to its association with an E.164 number allows an SP to > provision it; >=20 > that allows the called party to SUBSCRIBE as watcher of the status and > accept only calls from the head of queue while it contains active > callers; >=20 > that allows the called party to call the active party at the=20 > head of the > queue in CCBS or CCNA situations; (simulation) > **OR** > that allows the calling parties to SUBSCRIBE as watchers, get NOTIFIES > of their status (head of queue is ephemeral - call now or get=20 > dropped), > which triggers them to call; (emulation) >=20 > that allows callers to PUBLISH queue status (based on XML package > definition) of: > - active: allows moving up in queue as head identities are removed,=20 > - suspended: allows moving up, but not to head of queue, > - removed: drops from queue altogether; >=20 > (Note the additional rejection reasons could be added to the responses > to PUBLISH.) >=20 > that does not allow callers to see who else is in the queue (filtered > NOTIFYs); >=20 > that allows the CCBS/CCNA PUBLISH requests to be charged on a=20 > per use or > subscription basis; >=20 > would that address the modeling concern? >=20 > Mike >=20 >=20 > > -----Original Message----- > > From: Adam Roach [mailto:adam@estacado.net]=20 > > Sent: Wednesday, August 30, 2006 12:18 PM > > To: Silvia Tessa > > Cc: drage@LUCENT.COM; SIPPING WG > > Subject: Re: [Sipping] Re: WG:=20 > draft-poetzl-sipping-call-completion-01 > >=20 > > Silvia Tessa wrote: > > > Hi all, > > > comments inline. > > > > > > Adam Roach wrote: > > > > > > > I have a handful of comments on the current call=20 > > completion document=20 > > > > as it relates to RFC 3265. > > > > > > > > 1. I have some heartburn with the behavior described=20 > > for the event > > > > package. The SUBSCRIBE/NOTIFY mechanism is intended=20 > > to *subscribe* > > > > to remote state, not *manipulate* it. Parameters like > > > > "queue-operation" are obviously intended to act as an RPC > > > > mechanism. It is important that SUBSCRIBE requests remain > > > > idempotent: sending the same SUBSCRIBE request=20 > over and over > > > > should not cause users to be inserted into a queue=20 > > multiple times. > > > > As this document defines the mechanism, this is=20 > not the case. > > > > > > It is not clear to me which exact scenarios you have in=20 > > mind: if the=20 > > > SUBSCRIBE is not sent on an existing dialog, it does=20 > create a new=20 > > > subscription, no matter which event package we are talking about: > > > sending the same SUBSCRIBE request over and over on=20 > > different dialogs=20 > > > tipically cause users to be notified of status changes=20 > > multiple times. > > > > > > If the SUBSCRIBE is sent on the same dialog but with different=20 > > > queue-operation it can simply change the internal=20 > > subscription status=20 > > > (suspended or not): sending over and over the same=20 > SUBSCRIBE with a=20 > > > different Expires Header value, causes the SUBSCRIPTION to=20 > > > continuously change its duration attribute, sending it with a=20 > > > different queue-operation (suspend/resume) makes it change its=20 > > > suspension attribute. > > > > > > If the SUBSCRIBE is sent on the same dialog with the same=20 > > > queue-operation value, no action is taken. > >=20 > > It's good that you view these operations as idempotent (the=20 > > document does not make this clear, and the implication is=20 > > otherwise) -- however, the issue remains that you are=20 > > creating and manipulating NON-SUBSCRIPTION state with=20 > > SUBSCRIBE messages. That's not what SUBSCRIBE is for, and=20 > > you'll almost certainly run into implementation issues going=20 > > down this path. > >=20 > > > > > > > > 2. I find the choice of event header field parameters=20 > to return > > > > resource state to be both baroque and in violation of the > > > > description of NOTIFY bodies in section 4.4.5 of RFC=20 > > 3265. For > > > > example, "queue-state" and "denial-reason"=20 > > communicate resource > > > > state, and should be sent in the NOTIFY body (not=20 > > Event header > > > > field parameters). > > > > > > It can be baroque, but not in violation, as far as I=20 > understand.... > >=20 > > "The NOTIFY body is used to report state on the resource being > > monitored." > >=20 > >=20 > > I'm not sure how this could be any clearer. > >=20 > > > You are raising a good issue! Actually, this is not solvable by=20 > > > solving the forking or not-forking matter, but introducing a=20 > > > centralised entity, holding all the queue information.=20 > > That's actually=20 > > > the idea behind the draft: the call completion event server is a=20 > > > centralized entity monitoring (via dialog event package or just=20 > > > logging users communication, that's out of scope) the user=20 > > call status. > >=20 > > If such is the case, then the document has far more problems=20 > > than I originally determined. Are you expecting this=20 > > centralized entity to add "Allow-Events" header fields to the=20 > > 486 and 1xx responses generated by terminals? Section 3.3.7=20 > > is abundantly clear on this topic: > >=20 > > "Note that "Allow-Events" headers MUST NOT be inserted by=20 > > proxies." > >=20 > > =20 > >=20 > > The signaling flows in section 7 of the current draft=20 > > certainly imply that the entity processing the INVITE=20 > > requests is the same entity processing the SUBSCRIBE requests=20 > > -- and this would be the terminal, not some server in the=20 > > middle of the network. If the intention is that the mechanism=20 > > is handled on some centralized server, then the document=20 > > needs some *serious* rework to make that fact clear. > >=20 > > Until the model is clearer, there's no use discussing the=20 > > remainder of my comments. > >=20 > > /a > >=20 > > _______________________________________________ > > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > > This list is for NEW development of the application of SIP=20 > > Use sip-implementors@cs.columbia.edu for questions on current=20 > > sip Use sip@ietf.org for new developments of core SIP > >=20 >=20 > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP > Use sip-implementors@cs.columbia.edu for questions on current sip > Use sip@ietf.org for new developments of core SIP >=20 _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Thu Aug 31 18:17:03 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GIupk-0003sp-La; Thu, 31 Aug 2006 18:16:24 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GIupj-0003sk-Hh for sipping@ietf.org; Thu, 31 Aug 2006 18:16:23 -0400 Received: from sccrmhc13.comcast.net ([204.127.200.83]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GIupi-0006O3-9a for sipping@ietf.org; Thu, 31 Aug 2006 18:16:23 -0400 Received: from dragon.ariadne.com (dworley.hsd1.ma.comcast.net[24.34.79.42]) by comcast.net (sccrmhc13) with ESMTP id <20060831221619013005bvh7e>; Thu, 31 Aug 2006 22:16:19 +0000 Received: from dragon.ariadne.com (dragon.ariadne.com [127.0.0.1]) by dragon.ariadne.com (8.12.8/8.12.8) with ESMTP id k7VMGJWI009407 for ; Thu, 31 Aug 2006 18:16:19 -0400 Received: (from worley@localhost) by dragon.ariadne.com (8.12.8/8.12.8/Submit) id k7VMGJaa009403; Thu, 31 Aug 2006 18:16:19 -0400 Date: Thu, 31 Aug 2006 18:16:19 -0400 Message-Id: <200608312216.k7VMGJaa009403@dragon.ariadne.com> To: sipping@ietf.org From: Dale.Worley@comcast.net In-reply-to: <072C5B76F7CEAB488172C6F64B30B5E301DDB00E@xmb-rtp-20b.amer.cisco.com> (mhammer@cisco.com) Subject: Re: [Sipping] Re: WG: draft-poetzl-sipping-call-completion-01 References: <072C5B76F7CEAB488172C6F64B30B5E301DDB00E@xmb-rtp-20b.amer.cisco.com> X-Spam-Score: 0.2 (/) X-Scan-Signature: 82c9bddb247d9ba4471160a9a865a5f3 X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org I've only started looking at this document. It presents some interesting questions. Here are a number of observations. 1. Having a definitions section is useful, but the current one needs to be revised. The verbs "completed" and "executed" seem to be used to refer to the same thing, the establishemnt of a call by the CCBS/CCNR mechanism. The terms "call-completion user" and "call-completion target" are not defined. The term "destination B queue" is utterly mysterious, and is associated with the "retain option", which is left undescribed. 2. A caller's UA could perform most of the CCBS/CCNR operations by subscribing to the dialog event package of the callee. What is gained by having the service is the queuing of multiple callers in the case of high contention. But this fact is nowhere discussed, despite that it is the true value-add of this feature. 3. The philosophy of this document is to identify calls by caller and callee AOR. This is a mistake and will cause endless trouble at some point in the future (in various weird circumstances where multiple calls can pass between the same From AOR and To AOR). Calls must be identified by Call-Id/from-tag/to-tag to avoid mabiguity. 4. Supported, Allow-*, and Accept-* headers are intended for use by a UA to describe its capabilities, not properties of an individual call or request. So using Allow-Events to indicate the availability of CCBS works only if *all calls* to that UA support CCBS. 5. In regard to forking, all features should be carefully vetted for support of forking, because in *any* INVITE, the request may have been arbitrarily forked before it got to a proxy which understands the new feature. Additionally, the request-URI at the input to the proxy bears no discernable relationship to the From AOR that the caller used, and unless the UAS provides a proper contact address, the caller has no URI which is assured of reaching the UAS again. All of these militate against any design which assumes that the caller and callee are unique and are aware of each other's "real" addresses. 6. The use of SUBSCRIBE and NOTIFY is very non-standard. Certainly, putting status-reporting data in the Event header of a NOTIFY instead of the body of the message is very strange. And it's not a wise idea to have a SUBSCRIBE dialog *affect* anything other than the SUBSCRIBE dialog status in the UAS. People assume that subscription information can be cached, proxied, and redistributed without a problem. I would recommend using INFO or PUBLISH to carry what is essentially RPC operations, and use SUBSCRIBE only for retrieving status information. 7. On the other hand, what is wanted here is to implement "standing in a queue", and the server that implements the queue wants to know that the caller's UAs are continuing to pay attention to their position in the queue. In that regard, the SUBSCRIBE "soft state" mechanism implements a lot of what we want -- the UA can maintain its presence in the queue indefinitely, but it must periodically refresh its notification to the UAS that it is paying attention. Dale _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Thu Aug 31 18:50:43 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GIvMs-0003qI-9a; Thu, 31 Aug 2006 18:50:38 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GIvMo-0003kr-C5; Thu, 31 Aug 2006 18:50:34 -0400 Received: from nit.isi.edu ([128.9.160.116]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GIvMn-0003w6-0c; Thu, 31 Aug 2006 18:50:34 -0400 Received: from nit.isi.edu (loopback [127.0.0.1]) by nit.isi.edu (8.12.11.20060308/8.12.11) with ESMTP id k7VMoWIf020464; Thu, 31 Aug 2006 15:50:32 -0700 Received: (from apache@localhost) by nit.isi.edu (8.12.11.20060308/8.12.11/Submit) id k7VMoWKP020463; Thu, 31 Aug 2006 15:50:32 -0700 Date: Thu, 31 Aug 2006 15:50:32 -0700 Message-Id: <200608312250.k7VMoWKP020463@nit.isi.edu> To: ietf-announce@ietf.org, rfc-dist@rfc-editor.org From: rfc-editor@rfc-editor.org X-Spam-Score: -14.3 (--------------) X-Scan-Signature: fb6060cb60c0cea16e3f7219e40a0a81 Cc: sipping@ietf.org, rfc-editor@rfc-editor.org Subject: [Sipping] RFC 4484 on Trait-Based Authorization Requirements for the Session Initiation Protocol (SIP) X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org A new Request for Comments is now available in online RFC libraries. RFC 4484 Title: Trait-Based Authorization Requirements for the Session Initiation Protocol (SIP) Author: J. Peterson, J. Polk, D. Sicker, H. Tschofenig Status: Informational Date: August 2006 Mailbox: jon.peterson@neustar.biz, jmpolk@cisco.com, douglas.sicker@colorado.edu, Hannes.Tschofenig@siemens.com Pages: 15 Characters: 37169 Updates/Obsoletes/SeeAlso: None I-D Tag: draft-ietf-sipping-trait-authz-02.txt URL: http://www.rfc-editor.org/rfc/rfc4484.txt This document lays out a set of requirements related to trait-based authorization for the Session Initiation Protocol (SIP). While some authentication mechanisms are described in the base SIP specification, trait-based authorization provides information used to make policy decisions based on the attributes of a participant in a session. This approach provides a richer framework for authorization, as well as allows greater privacy for users of an identity system. This memo provides information for the Internet community. This document is a product of the Session Initiation Proposal Investigation Working Group of the IETF. INFORMATIONAL: This memo provides information for the Internet community. It does not specify an Internet standard of any kind. Distribution of this memo is unlimited. This announcement is sent to the IETF list and the RFC-DIST list. Requests to be added to or deleted from the IETF distribution list should be sent to IETF-REQUEST@IETF.ORG. Requests to be added to or deleted from the RFC-DIST distribution list should be sent to RFC-DIST-REQUEST@RFC-EDITOR.ORG. Details on obtaining RFCs via FTP or EMAIL may be obtained by sending an EMAIL message to rfc-info@RFC-EDITOR.ORG with the message body help: ways_to_get_rfcs. For example: To: rfc-info@RFC-EDITOR.ORG Subject: getting rfcs help: ways_to_get_rfcs Requests for special distribution should be addressed to either the author of the RFC in question, or to RFC-Manager@RFC-EDITOR.ORG. Unless specifically noted otherwise on the RFC itself, all RFCs are for unlimited distribution. Submissions for Requests for Comments should be sent to RFC-EDITOR@RFC-EDITOR.ORG. Please consult RFC 2223, Instructions to RFC Authors, for further information. Joyce K. Reynolds and Sandy Ginoza USC/Information Sciences Institute ... _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Thu Aug 31 18:50:56 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GIvNA-00042v-3i; Thu, 31 Aug 2006 18:50:56 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GIvN6-0003yM-0X; Thu, 31 Aug 2006 18:50:52 -0400 Received: from nit.isi.edu ([128.9.160.116]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GIvN4-0003wt-Ky; Thu, 31 Aug 2006 18:50:51 -0400 Received: from nit.isi.edu (loopback [127.0.0.1]) by nit.isi.edu (8.12.11.20060308/8.12.11) with ESMTP id k7VMooOH020474; Thu, 31 Aug 2006 15:50:50 -0700 Received: (from apache@localhost) by nit.isi.edu (8.12.11.20060308/8.12.11/Submit) id k7VMooSQ020473; Thu, 31 Aug 2006 15:50:50 -0700 Date: Thu, 31 Aug 2006 15:50:50 -0700 Message-Id: <200608312250.k7VMooSQ020473@nit.isi.edu> To: ietf-announce@ietf.org, rfc-dist@rfc-editor.org From: rfc-editor@rfc-editor.org X-Spam-Score: -14.8 (--------------) X-Scan-Signature: 00e94c813bef7832af255170dca19e36 Cc: sipping@ietf.org, rfc-editor@rfc-editor.org Subject: [Sipping] RFC 4575 on A Session Initiation Protocol (SIP) Event Package for Conference State X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org A new Request for Comments is now available in online RFC libraries. RFC 4575 Title: A Session Initiation Protocol (SIP) Event Package for Conference State Author: J. Rosenberg, H. Schulzrinne, O. Levin, Ed. Status: Standards Track Date: August 2006 Mailbox: jdrosen@cisco.com, schulzrinne@cs.columbia.edu, oritl@microsoft.com Pages: 48 Characters: 97484 Updates/Obsoletes/SeeAlso: None I-D Tag: draft-ietf-sipping-conference-package-12.txt URL: http://www.rfc-editor.org/rfc/rfc4575.txt This document defines a conference event package for tightly coupled conferences using the Session Initiation Protocol (SIP) events framework, along with a data format used in notifications for this package. The conference package allows users to subscribe to a conference Uniform Resource Identifier (URI). Notifications are sent about changes in the membership of this conference and optionally about changes in the state of additional conference components. [STANDARDS TRACK] This document is a product of the Session Initiation Proposal Investigation Working Group of the IETF This is now a Proposed Standard Protocol. STANDARDS TRACK: This document specifies an Internet standards track protocol for the Internet community,and requests discussion and suggestions for improvements.Please refer to the current edition of the Internet Official Protocol Standards (STD 1) for the standardization state and status of this protocol. Distribution of this memo is unlimited. This announcement is sent to the IETF list and the RFC-DIST list. Requests to be added to or deleted from the IETF distribution list should be sent to IETF-REQUEST@IETF.ORG. Requests to be added to or deleted from the RFC-DIST distribution list should be sent to RFC-DIST-REQUEST@RFC-EDITOR.ORG. Details on obtaining RFCs via FTP or EMAIL may be obtained by sending an EMAIL message to rfc-info@RFC-EDITOR.ORG with the message body help: ways_to_get_rfcs. For example: To: rfc-info@RFC-EDITOR.ORG Subject: getting rfcs help: ways_to_get_rfcs Requests for special distribution should be addressed to either the author of the RFC in question, or to RFC-Manager@RFC-EDITOR.ORG. Unless specifically noted otherwise on the RFC itself, all RFCs are for unlimited distribution. Submissions for Requests for Comments should be sent to RFC-EDITOR@RFC-EDITOR.ORG. Please consult RFC 2223, Instructions to RFC Authors, for further information. Joyce K. Reynolds and Sandy Ginoza USC/Information Sciences Institute ... _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Thu Aug 31 18:51:05 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GIvNJ-0004By-06; Thu, 31 Aug 2006 18:51:05 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GIvNF-00049h-KR; Thu, 31 Aug 2006 18:51:01 -0400 Received: from nit.isi.edu ([128.9.160.116]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GIvNE-0003xh-8g; Thu, 31 Aug 2006 18:51:01 -0400 Received: from nit.isi.edu (loopback [127.0.0.1]) by nit.isi.edu (8.12.11.20060308/8.12.11) with ESMTP id k7VMoxUo020479; Thu, 31 Aug 2006 15:50:59 -0700 Received: (from apache@localhost) by nit.isi.edu (8.12.11.20060308/8.12.11/Submit) id k7VMoxrW020478; Thu, 31 Aug 2006 15:50:59 -0700 Date: Thu, 31 Aug 2006 15:50:59 -0700 Message-Id: <200608312250.k7VMoxrW020478@nit.isi.edu> To: ietf-announce@ietf.org, rfc-dist@rfc-editor.org From: rfc-editor@rfc-editor.org X-Spam-Score: -14.8 (--------------) X-Scan-Signature: d8ae4fd88fcaf47c1a71c804d04f413d Cc: sipping@ietf.org, rfc-editor@rfc-editor.org Subject: [Sipping] BCP0119 RFC 4579 on Session Initiation Protocol (SIP) Call Control - Conferencing for User Agents X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org A new Request for Comments is now available in online RFC libraries. BCP0119 RFC 4579 Title: Session Initiation Protocol (SIP) Call Control - Conferencing for User Agents Author: A. Johnston, O. Levin Status: Best Current Practice Date: August 2006 Mailbox: alan@sipstation.com, oritl@microsoft.com Pages: 43 Characters: 96506 Updates: See-Also: BCP0119 I-D Tag: draft-ietf-sipping-cc-conferencing-07.txt URL: http://www.rfc-editor.org/rfc/rfc4579.txt This specification defines conferencing call control features for the Session Initiation Protocol (SIP). This document builds on the Conferencing Requirements and Framework documents to define how a tightly coupled SIP conference works. The approach is explored from the perspective of different user agent (UA) types: conference-unaware, conference-aware, and focus UAs. The use of Uniform Resource Identifiers (URIs) in conferencing, OPTIONS for capabilities discovery, and call control using REFER are covered in detail with example call flow diagrams. The usage of the isfocus feature tag is defined. This document specifies an Internet Best Current Practices for the Internet Community, and requests discussion and suggestions for improvements. This document is a product of the Session Initiation Proposal Investigation Working Group of the IETF. This document specifies an Internet Best Current Practices for the Internet Community, and requests discussion and suggestions for improvements. Distribution of this memo is unlimited. This announcement is sent to the IETF list and the RFC-DIST list. Requests to be added to or deleted from the IETF distribution list should be sent to IETF-REQUEST@IETF.ORG. Requests to be added to or deleted from the RFC-DIST distribution list should be sent to RFC-DIST-REQUEST@RFC-EDITOR.ORG. Details on obtaining RFCs via FTP or EMAIL may be obtained by sending an EMAIL message to rfc-info@RFC-EDITOR.ORG with the message body help: ways_to_get_rfcs. For example: To: rfc-info@RFC-EDITOR.ORG Subject: getting rfcs help: ways_to_get_rfcs Requests for special distribution should be addressed to either the author of the RFC in question, or to RFC-Manager@RFC-EDITOR.ORG. Unless specifically noted otherwise on the RFC itself, all RFCs are for unlimited distribution. Submissions for Requests for Comments should be sent to RFC-EDITOR@RFC-EDITOR.ORG. Please consult RFC 2223, Instructions to RFC Authors, for further information. Joyce K. Reynolds and Sandy Ginoza USC/Information Sciences Institute ... _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Thu Aug 31 22:14:03 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GIyWN-0002CF-Qm; Thu, 31 Aug 2006 22:12:39 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GIyWM-00029B-FV for sipping@ietf.org; Thu, 31 Aug 2006 22:12:38 -0400 Received: from alnrmhc11.comcast.net ([206.18.177.51]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GIyWL-0005Wg-8l for sipping@ietf.org; Thu, 31 Aug 2006 22:12:38 -0400 Received: from dragon.ariadne.com (dworley.hsd1.ma.comcast.net[24.34.79.42]) by comcast.net (alnrmhc11) with ESMTP id <20060901021236b1100emibae>; Fri, 1 Sep 2006 02:12:36 +0000 Received: from dragon.ariadne.com (dragon.ariadne.com [127.0.0.1]) by dragon.ariadne.com (8.12.8/8.12.8) with ESMTP id k812CZWI014814 for ; Thu, 31 Aug 2006 22:12:35 -0400 Received: (from worley@localhost) by dragon.ariadne.com (8.12.8/8.12.8/Submit) id k812CZOS014810; Thu, 31 Aug 2006 22:12:35 -0400 Date: Thu, 31 Aug 2006 22:12:35 -0400 Message-Id: <200609010212.k812CZOS014810@dragon.ariadne.com> To: sipping@ietf.org From: Dale.Worley@comcast.net In-reply-to: References: X-Spam-Score: 0.2 (/) X-Scan-Signature: cf4fa59384e76e63313391b70cd0dd25 Subject: [Sipping] Re: [Sip] question about RFC4240 (NETANN) X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org From: Eric Burger Oooh - good catch of what looks like an error. From a strict constructionist point of view, the page 7 text is normative, so underbar reigns. However, one can make a very convincing argument the reference to RFC 3066 is normative, hence it should be a hyphen. AFAIK, all implementations to date of RFC 4240 use underbar. Is there enough confusion out there to create an update to 4240? Note that 4240 is not a PS, so this isn't a DS opportunity. Ugh. Is there any chance we can change it to hyphen, to match HTTP/1.1 (RFC 2068, section 3.10)? I assume RFC 4240 was intended to match that. Dale _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From sipping-bounces@ietf.org Thu Aug 31 22:14:52 2006 Received: from [127.0.0.1] (helo=stiedprmman1.va.neustar.com) by megatron.ietf.org with esmtp (Exim 4.43) id 1GIyYW-0003wu-0U; Thu, 31 Aug 2006 22:14:52 -0400 Received: from [10.91.34.44] (helo=ietf-mx.ietf.org) by megatron.ietf.org with esmtp (Exim 4.43) id 1GIyYT-0003tg-St for sipping@ietf.org; Thu, 31 Aug 2006 22:14:49 -0400 Received: from zrtps0kn.nortel.com ([47.140.192.55]) by ietf-mx.ietf.org with esmtp (Exim 4.43) id 1GIyYR-00062C-H8 for sipping@ietf.org; Thu, 31 Aug 2006 22:14:49 -0400 Received: from zrc2hxm1.corp.nortel.com (zrc2hxm1.corp.nortel.com [47.103.123.72]) by zrtps0kn.nortel.com (Switch-2.2.6/Switch-2.2.0) with ESMTP id k812EFO16576; Thu, 31 Aug 2006 22:14:15 -0400 (EDT) X-MimeOLE: Produced By Microsoft Exchange V6.5 Content-class: urn:content-classes:message MIME-Version: 1.0 Content-Type: text/plain; charset="US-ASCII" Content-Transfer-Encoding: quoted-printable Subject: RE: [Sipping] I-D ACTION:draft-ietf-sipping-toip-07.txt Date: Thu, 31 Aug 2006 21:14:06 -0500 Message-ID: In-Reply-To: <62D92A9A02BCC845B202323D49A48426E1CE2E@0591-ITS-EXMP02.us.saic.com> X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: [Sipping] I-D ACTION:draft-ietf-sipping-toip-07.txt Thread-Index: AcbMoRKZO4ODtQ4MRHStkRsRAUaoqAAysd9Q From: "Mary Barnes" To: "Roy, Radhika R." , X-Spam-Score: 0.0 (/) X-Scan-Signature: 9a2be21919e71dc6faef12b370c4ecf5 Cc: X-BeenThere: sipping@ietf.org X-Mailman-Version: 2.1.5 Precedence: list List-Id: "SIPPING Working Group \(applications of SIP\)" List-Unsubscribe: , List-Post: List-Help: List-Subscribe: , Errors-To: sipping-bounces@ietf.org Radhika, Per the response on this related topic back in June, there are 2 editors for this document and the other authors are listed as contributors (per the RFC editor's desire to not have lots of authors on page 1): http://www1.ietf.org/mail-archive/web/sipping/current/msg11226.html Cheers, Mary -----Original Message----- From: Roy, Radhika R. [mailto:RADHIKA.R.ROY@saic.com]=20 Sent: Wednesday, August 30, 2006 4:47 PM To: sipping@ietf.org Subject: RE: [Sipping] I-D ACTION:draft-ietf-sipping-toip-07.txt Mary: =20 Did WG appointed 2 editors for this? =20 Did WG remove others as co-authors? =20 BR\RRR _____ =20 From: sipping-bounces@ietf.org on behalf of Internet-Drafts@ietf.org Sent: Wed 8/30/2006 3:50 PM To: i-d-announce@ietf.org Cc: sipping@ietf.org Subject: [Sipping] I-D ACTION:draft-ietf-sipping-toip-07.txt=20 A New Internet-Draft is available from the on-line Internet-Drafts=20 directories.=20 This draft is a work item of the Session Initiation Proposal Investigation Working Group of the IETF.=20 Title : Framework for real-time text over IP using the Session Initiation Protocol (SIP)=20 Author(s) : A. Van Wijk, G. Gybels=20 Filename : draft-ietf-sipping-toip-07.txt=20 Pages : 27=20 Date : 2006-8-30=20 =20 This document lists the essential requirements for real-time Text-=20 over-IP (ToIP) and defines a framework for implementation of all=20 required functions based on the Session Initiation Protocol (SIP) and=20 the Real-Time Transport Protocol (RTP). This includes interworking=20 between Text-over-IP and existing text telephony on the PSTN and other=20 networks.=20 A URL for this Internet-Draft is:=20 http://www.ietf.org/internet-drafts/draft-ietf-sipping-toip-07.txt =20 To remove yourself from the I-D Announcement list, send a message to=20 i-d-announce-request@ietf.org with the word unsubscribe in the body of=20 the message.=20 You can also visit https://www1.ietf.org/mailman/listinfo/I-D-announce =20 to change your subscription settings.=20 Internet-Drafts are also available by anonymous FTP. Login with the=20 username "anonymous" and a password of your e-mail address. After=20 logging in, type "cd internet-drafts" and then=20 "get draft-ietf-sipping-toip-07.txt".=20 A list of Internet-Drafts directories can be found in=20 http://www.ietf.org/shadow.html =20 or ftp://ftp.ietf.org/ietf/1shadow-sites.txt =20 Internet-Drafts can also be obtained by e-mail.=20 Send a message to:=20 mailserv@ietf.org.=20 In the body type:=20 "FILE /internet-drafts/draft-ietf-sipping-toip-07.txt".=20 =20 NOTE: The mail server at ietf.org can return the document in=20 MIME-encoded form by using the "mpack" utility. To use this=20 feature, insert the command "ENCODING mime" before the "FILE"=20 command. To decode the response(s), you will need "munpack" or=20 a MIME-compliant mail reader. Different MIME-compliant mail readers exhibit different behavior, especially when dealing with=20 "multipart" MIME messages (i.e. documents which have been split=20 up into multiple messages), so check your local documentation on how to manipulate these messages.=20 Below is the data which will enable a MIME compliant mail reader=20 implementation to automatically retrieve the ASCII version of the=20 Internet-Draft.=20 _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP