From bliss-bounces@ietf.org Tue Jul 15 05:29:12 2008 Return-Path: X-Original-To: bliss-archive@optimus.ietf.org Delivered-To: ietfarch-bliss-archive@core3.amsl.com Received: from [127.0.0.1] (localhost [127.0.0.1]) by core3.amsl.com (Postfix) with ESMTP id 03FEF3A6992; Tue, 15 Jul 2008 05:29:12 -0700 (PDT) X-Original-To: bliss@core3.amsl.com Delivered-To: bliss@core3.amsl.com Received: from localhost (localhost [127.0.0.1]) by core3.amsl.com (Postfix) with ESMTP id 8D01F3A698D; Tue, 15 Jul 2008 05:29:11 -0700 (PDT) X-Virus-Scanned: amavisd-new at amsl.com X-Spam-Flag: NO X-Spam-Score: -2.413 X-Spam-Level: X-Spam-Status: No, score=-2.413 tagged_above=-999 required=5 tests=[AWL=0.835, BAYES_00=-2.599, HELO_EQ_FR=0.35, HTML_MESSAGE=0.001, RCVD_IN_DNSWL_LOW=-1] Received: from mail.ietf.org ([64.170.98.32]) by localhost (core3.amsl.com [127.0.0.1]) (amavisd-new, port 10024) with ESMTP id bPwjcIcgXi5Y; Tue, 15 Jul 2008 05:29:10 -0700 (PDT) Received: from p-mail2.rd.francetelecom.com (p-mail2.rd.francetelecom.com [195.101.245.16]) by core3.amsl.com (Postfix) with ESMTP id 1F6E53A6992; Tue, 15 Jul 2008 05:29:09 -0700 (PDT) Received: from FTRDMEL2.rd.francetelecom.fr ([10.193.117.153]) by ftrdsmtp2.rd.francetelecom.fr with Microsoft SMTPSVC(6.0.3790.1830); Tue, 15 Jul 2008 14:29:34 +0200 X-MimeOLE: Produced By Microsoft Exchange V6.5 Content-class: urn:content-classes:message MIME-Version: 1.0 Date: Tue, 15 Jul 2008 14:29:33 +0200 Message-ID: <7DBAFEC6A76F3E42817DF1EBE64CB02605B8622A@ftrdmel2> X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: New draft submission Thread-Index: Acjmdm+3acX2c1qQSmqpbUpwLWo3KQ== From: To: , X-OriginalArrivalTime: 15 Jul 2008 12:29:34.0828 (UTC) FILETIME=[7083E2C0:01C8E676] Subject: [BLISS] New draft submission X-BeenThere: bliss@ietf.org X-Mailman-Version: 2.1.9 Precedence: list List-Id: "Basic Level of Interoperability for SIP Services \(BLISS\) BoF" List-Unsubscribe: , List-Archive: List-Post: List-Help: List-Subscribe: , Content-Type: multipart/mixed; boundary="===============1979612815==" Sender: bliss-bounces@ietf.org Errors-To: bliss-bounces@ietf.org This is a multi-part message in MIME format. --===============1979612815== Content-class: urn:content-classes:message Content-Type: multipart/alternative; boundary="----_=_NextPart_001_01C8E676.6FBCCC67" This is a multi-part message in MIME format. ------_=_NextPart_001_01C8E676.6FBCCC67 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: quoted-printable Hi, I've just submitted the following draft:=20 http://tools.ietf.org/html/draft-mohali-diversion-history-info-00 The motivation of the draft is to provide guidelines on mapping between the Diversion header and the History-Info header and possibly a Voicemail-uri.=20 We believe that having clearly defined mapping procedures between these two headers will greatly facilitate migration from the Diversion header to the History-Info header. Lack of clear mapping between both headers could result in significant interoperability issues and therefore delay widespread implementation of the History-Info header.=20 As you can see from the list of co-authors this issue is of interest to a number of people. We would like to get more feedback on the proposed solution and confirmation of interest from the IETF community.=20 Best Regards, Marianne Mohali ------_=_NextPart_001_01C8E676.6FBCCC67 Content-Type: text/html; charset="us-ascii" Content-Transfer-Encoding: quoted-printable New draft submission

Hi,

I've just = submitted the following draft:
http://tools.ietf.org/html/draft-mohali-diversion-history-info-00

The = motivation of the draft is to provide guidelines on mapping between the = Diversion header and the History-Info header and possibly a = Voicemail-uri.

We believe = that having clearly defined mapping procedures between these two headers = will greatly facilitate migration from the Diversion header to the = History-Info header.

Lack of = clear mapping between both headers could result in significant = interoperability issues and therefore delay widespread implementation of = the History-Info header.

As you can = see from the list of co-authors this issue is of interest to a number of = people. We would like to get more feedback on the proposed solution and = confirmation of interest from the IETF community.

Best = Regards,
Marianne = Mohali

------_=_NextPart_001_01C8E676.6FBCCC67-- --===============1979612815== Content-Type: text/plain; charset="us-ascii" MIME-Version: 1.0 Content-Transfer-Encoding: 7bit Content-Disposition: inline _______________________________________________ BLISS mailing list BLISS@ietf.org https://www.ietf.org/mailman/listinfo/bliss --===============1979612815==-- From bliss-bounces@ietf.org Wed Jul 16 18:38:30 2008 Return-Path: X-Original-To: bliss-archive@optimus.ietf.org Delivered-To: ietfarch-bliss-archive@core3.amsl.com Received: from [127.0.0.1] (localhost [127.0.0.1]) by core3.amsl.com (Postfix) with ESMTP id 31DBE3A6993; Wed, 16 Jul 2008 18:38:30 -0700 (PDT) X-Original-To: bliss@core3.amsl.com Delivered-To: bliss@core3.amsl.com Received: from localhost (localhost [127.0.0.1]) by core3.amsl.com (Postfix) with ESMTP id 533963A693B for ; Wed, 16 Jul 2008 18:38:29 -0700 (PDT) X-Virus-Scanned: amavisd-new at amsl.com X-Spam-Flag: NO X-Spam-Score: -2.599 X-Spam-Level: X-Spam-Status: No, score=-2.599 tagged_above=-999 required=5 tests=[BAYES_00=-2.599] Received: from mail.ietf.org ([64.170.98.32]) by localhost (core3.amsl.com [127.0.0.1]) (amavisd-new, port 10024) with ESMTP id fctRBsgGJk+v for ; Wed, 16 Jul 2008 18:38:28 -0700 (PDT) Received: from gateway07.websitewelcome.com (gateway07.websitewelcome.com [74.52.223.18]) by core3.amsl.com (Postfix) with SMTP id 7F1B53A696D for ; Wed, 16 Jul 2008 18:38:28 -0700 (PDT) Received: (qmail 31496 invoked from network); 17 Jul 2008 01:47:37 -0000 Received: from gator465.hostgator.com (69.56.174.130) by gateway07.websitewelcome.com with SMTP; 17 Jul 2008 01:47:37 -0000 Received: from udp157071uds.hawaiiantel.net ([72.253.0.217]:50440 helo=[192.168.1.45]) by gator465.hostgator.com with esmtpsa (TLSv1:AES128-SHA:128) (Exim 4.69) (envelope-from ) id 1KJISI-0005Ij-DY for bliss@ietf.org; Wed, 16 Jul 2008 20:38:52 -0500 Message-Id: <24C40B70-6D8D-4B69-8462-3F23E77ECAD7@ntt-at.com> From: Shida Schubert To: bliss@ietf.org Mime-Version: 1.0 (Apple Message framework v926) Date: Wed, 16 Jul 2008 15:38:49 -1000 X-Mailer: Apple Mail (2.926) X-AntiAbuse: This header was added to track abuse, please include it with any abuse report X-AntiAbuse: Primary Hostname - gator465.hostgator.com X-AntiAbuse: Original Domain - ietf.org X-AntiAbuse: Originator/Caller UID/GID - [47 12] / [47 12] X-AntiAbuse: Sender Address Domain - ntt-at.com X-Source: X-Source-Args: X-Source-Dir: Subject: [BLISS] Drat Agenda - 1st stab X-BeenThere: bliss@ietf.org X-Mailman-Version: 2.1.9 Precedence: list List-Id: "Basic Level of Interoperability for SIP Services \(BLISS\) BoF" List-Unsubscribe: , List-Archive: List-Post: List-Help: List-Subscribe: , Content-Transfer-Encoding: 7bit Content-Type: text/plain; charset="us-ascii"; Format="flowed" Sender: bliss-bounces@ietf.org Errors-To: bliss-bounces@ietf.org All; This is a first stab at the agenda for our meeting in Dublin. I will add the reading lists with links etc, once the agenda is fleshed out. I am still talking to few people about agenda time so I will update the agenda accordingly. 10m Agenda Bash Chair 20m Automatic Call Handling John 30m Call Completion Denis 40m Line Sharing (MLA) Alan 20m Any other business Regards Shida _______________________________________________ BLISS mailing list BLISS@ietf.org https://www.ietf.org/mailman/listinfo/bliss From bliss-bounces@ietf.org Fri Jul 18 09:19:27 2008 Return-Path: X-Original-To: bliss-archive@optimus.ietf.org Delivered-To: ietfarch-bliss-archive@core3.amsl.com Received: from [127.0.0.1] (localhost [127.0.0.1]) by core3.amsl.com (Postfix) with ESMTP id 287223A6A24; Fri, 18 Jul 2008 09:19:27 -0700 (PDT) X-Original-To: bliss@core3.amsl.com Delivered-To: bliss@core3.amsl.com Received: from localhost (localhost [127.0.0.1]) by core3.amsl.com (Postfix) with ESMTP id 117A23A6A24 for ; Fri, 18 Jul 2008 09:19:26 -0700 (PDT) X-Virus-Scanned: amavisd-new at amsl.com X-Spam-Flag: NO X-Spam-Score: -6.275 X-Spam-Level: X-Spam-Status: No, score=-6.275 tagged_above=-999 required=5 tests=[AWL=-0.277, BAYES_00=-2.599, J_CHICKENPOX_35=0.6, RCVD_IN_DNSWL_MED=-4, WHOIS_NETSOLPR=0.001] Received: from mail.ietf.org ([64.170.98.32]) by localhost (core3.amsl.com [127.0.0.1]) (amavisd-new, port 10024) with ESMTP id AE55tdOsg1vo for ; Fri, 18 Jul 2008 09:19:24 -0700 (PDT) Received: from sj-iport-1.cisco.com (sj-iport-1.cisco.com [171.71.176.70]) by core3.amsl.com (Postfix) with ESMTP id 9B0913A6839 for ; Fri, 18 Jul 2008 09:19:24 -0700 (PDT) X-IronPort-AV: E=Sophos;i="4.31,211,1215388800"; d="scan'208";a="54883704" Received: from sj-dkim-3.cisco.com ([171.71.179.195]) by sj-iport-1.cisco.com with ESMTP; 18 Jul 2008 16:19:58 +0000 Received: from sj-core-3.cisco.com (sj-core-3.cisco.com [171.68.223.137]) by sj-dkim-3.cisco.com (8.12.11/8.12.11) with ESMTP id m6IGJuau019639 for ; Fri, 18 Jul 2008 09:19:56 -0700 Received: from xbh-rtp-201.amer.cisco.com (xbh-rtp-201.cisco.com [64.102.31.12]) by sj-core-3.cisco.com (8.13.8/8.13.8) with ESMTP id m6IGJptN025732 for ; Fri, 18 Jul 2008 16:19:56 GMT 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, 18 Jul 2008 12:19:53 -0400 Received: from [10.82.248.28] ([10.82.248.28]) by xfe-rtp-201.amer.cisco.com with Microsoft SMTPSVC(6.0.3790.1830); Fri, 18 Jul 2008 12:19:53 -0400 Message-ID: <4880C2B1.2020203@cisco.com> Date: Fri, 18 Jul 2008 12:20:01 -0400 From: Jonathan Rosenberg User-Agent: Thunderbird 2.0.0.9 (Windows/20071031) MIME-Version: 1.0 To: bliss@ietf.org X-OriginalArrivalTime: 18 Jul 2008 16:19:53.0360 (UTC) FILETIME=[1C3DC900:01C8E8F2] DKIM-Signature: v=1; a=rsa-sha256; q=dns/txt; l=4241; t=1216397997; x=1217261997; c=relaxed/simple; s=sjdkim3002; h=Content-Type:From:Subject:Content-Transfer-Encoding:MIME-Version; d=cisco.com; i=jdrosen@cisco.com; z=From:=20Jonathan=20Rosenberg=20 |Subject:=20configuring=20the=20proxy=20for=20ACH |Sender:=20; bh=DvJEf8FzWjL9jNluo6KqmVCFrXVXTJPK1TzJLlI4vRc=; b=O/Ge8a6c3i+oplR5mt9HzlD43WW61cwRAMJv5xTxOf6fSW61qfRnHifM8e bZpaIz6Qrf8BseJA2Q6GtPkScLIYUE4nbkva1My000I+JiIhKnS53lRpzBa0 LfDkBDQCTF; Authentication-Results: sj-dkim-3; header.From=jdrosen@cisco.com; dkim=pass ( sig from cisco.com/sjdkim3002 verified; ); Subject: [BLISS] configuring the proxy for ACH X-BeenThere: bliss@ietf.org X-Mailman-Version: 2.1.9 Precedence: list List-Id: "Basic Level of Interoperability for SIP Services \(BLISS\) BoF" List-Unsubscribe: , List-Archive: List-Post: List-Help: List-Subscribe: , Content-Transfer-Encoding: 7bit Content-Type: text/plain; charset="us-ascii"; Format="flowed" Sender: bliss-bounces@ietf.org Errors-To: bliss-bounces@ietf.org Here are some thoughts and a proposal on configuring ACH at the proxy. Its really important to keep in mind that BLISS is about REAL interoperability. That means, figuring out what is REALLY causing the interop failures for these features (which the survey and follow on work have done a good job at doing), and proposing REAL recommendations to fix them. To me, REAL means, that this is something that we believe folks will actually do, and would actually solve the problem. Our metric of success is, if folks implement our specs, you can plug a phone from one vendor into a call agent from another and it works for that feature. I don't think BLISS should be an academic exercise. With that in mind, some thoughts: 1. for configuring the proxy with ACH rules, the ONLY things we need to consider are those functions which typically have a button or dedicated function on the phone/UA. If something is set up via a web page, we don't need to standardize interfaces for that. My sense is, most endpoints that have built-in keys or embedded code for ACH features are almost exclusively limited to call forwarding (with several variants) and DND. I don't think anyone is trying to do ToD-based forwarding rules from dedicated software in an endpoint. But if folks have examples, please bring them out. 2. XCAP *could* be a way to do this - in theory. But if you look at what folks have implemented, well xcap is not exactly a resounding success. If we mandate that all phones implement xcap to just set call forward, and that all call agents/servers/proxies implement it as well - do we think they actually would. Being honest here - I dont think so. 3. The problem we're trying to face here - data-oriented client-server transactions - this is actually something that the rest of the Internet world has been very successful at solving. At this point, I think it would be fruitless to mandate something that we have defined (i.e., xcap or a new sip mechanism), and instead, let us just ride the wave. From what I can tell, where most folks are going for this stuff is the most trivial solution possible - and thats REST. 4. And so, my proposal would be to define a trivial REST interface for call forwarding at least. SO something like: POST http://server.com/joe/profile/callforwarding?value=true and thats it. PUT vs. POST or whatever, we need to find what the best practices are in the "web2.0 world" (if you'll excuse the buzzword). We provide an interface for turning it on/off, indicating whether it goes to voicemail, allow a target for forwarding. 5. DND - for pre-call DND (where you hit a DND button and future calls never reach you). I think, in the modern world, DND is really presence, and you are seeing more and more presence deployments including a DND state. So my suggestion here would be to use presence for this - i.e., the user sets their presence status to "DND". We can debate whether this is a new state or is the state, but lets agree on something (most folks are using a separate state for this - and we don't have a value for it). 6. as a general rule, we should avoid as much as possible RELYING on a consistent mechanism for configuring the phone (i.e., config framework), for any of our features to work. Again, the REALITY is that endpoint configuration is wildly variable amongst implementations, much more variable than any other aspect of the protocol set on endpoint devices. I don't think that, having IETF recommend to implement the IETF config framework for some features to work, will change at all the likelihood of that happening in reality. Note that, I view configuring the phone as totally different from pushing data to the proxy (for which I am proposing a REST interface above). I suspect this will all be controversial, but,thats the fun part :) Fire away. -Jonathan R. -- Jonathan D. Rosenberg, Ph.D. 499 Thornall St. Cisco Fellow Edison, NJ 08837 Cisco, Voice Technology Group jdrosen@cisco.com http://www.jdrosen.net PHONE: (408) 902-3084 http://www.cisco.com _______________________________________________ BLISS mailing list BLISS@ietf.org https://www.ietf.org/mailman/listinfo/bliss From bliss-bounces@ietf.org Fri Jul 18 09:25:38 2008 Return-Path: X-Original-To: bliss-archive@optimus.ietf.org Delivered-To: ietfarch-bliss-archive@core3.amsl.com Received: from [127.0.0.1] (localhost [127.0.0.1]) by core3.amsl.com (Postfix) with ESMTP id 6B70A28C1BC; Fri, 18 Jul 2008 09:25:38 -0700 (PDT) X-Original-To: bliss@core3.amsl.com Delivered-To: bliss@core3.amsl.com Received: from localhost (localhost [127.0.0.1]) by core3.amsl.com (Postfix) with ESMTP id 03F4A28C1B9 for ; Fri, 18 Jul 2008 09:25:37 -0700 (PDT) X-Virus-Scanned: amavisd-new at amsl.com X-Spam-Flag: NO X-Spam-Score: -2.599 X-Spam-Level: X-Spam-Status: No, score=-2.599 tagged_above=-999 required=5 tests=[BAYES_00=-2.599] Received: from mail.ietf.org ([64.170.98.32]) by localhost (core3.amsl.com [127.0.0.1]) (amavisd-new, port 10024) with ESMTP id kD6EQTH8ilZl for ; Fri, 18 Jul 2008 09:25:36 -0700 (PDT) Received: from smtp.mitel.com (smtp.mitel.com [216.191.234.102]) by core3.amsl.com (Postfix) with ESMTP id 5635D28C1B1 for ; Fri, 18 Jul 2008 09:25:36 -0700 (PDT) Received: from localhost (smtp.mitel.com [127.0.0.1]) by smtp.mitel.com (Postfix) with ESMTP id D7F772C073 for ; Fri, 18 Jul 2008 12:26:09 -0400 (EDT) X-Virus-Scanned: by amavisd-new (virusonly) at mitel.com Received: from smtp.mitel.com ([127.0.0.1]) by localhost (smtp.mitel.com [127.0.0.1]) (amavisd-new, port 10024) with ESMTP id jgbxmVtYBj1Q for ; Fri, 18 Jul 2008 12:26:09 -0400 (EDT) Received: from ottlnmta.mitel.com (ottlnmta.mitel.com [10.44.16.150]) by smtp.mitel.com (Postfix) with ESMTP id B7ED72C060 for ; Fri, 18 Jul 2008 12:26:09 -0400 (EDT) From: Fred_Stacey@Mitel.com To: bliss@ietf.org Message-ID: Date: Fri, 18 Jul 2008 12:26:08 -0400 X-MIMETrack: Serialize by Router on OTTLNMTA/Mitel(Release 6.5.5FP2|October 23, 2006) at 07/18/2008 12:26:09 PM MIME-Version: 1.0 Subject: [BLISS] Fred Stacey is out of the office. X-BeenThere: bliss@ietf.org X-Mailman-Version: 2.1.9 Precedence: list List-Id: "Basic Level of Interoperability for SIP Services \(BLISS\) BoF" List-Unsubscribe: , List-Archive: List-Post: List-Help: List-Subscribe: , Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: 7bit Sender: bliss-bounces@ietf.org Errors-To: bliss-bounces@ietf.org I will be out of the office starting 07/16/2008 and will not return until 08/11/2008. _______________________________________________ BLISS mailing list BLISS@ietf.org https://www.ietf.org/mailman/listinfo/bliss From bliss-bounces@ietf.org Mon Jul 21 05:22:52 2008 Return-Path: X-Original-To: bliss-archive@optimus.ietf.org Delivered-To: ietfarch-bliss-archive@core3.amsl.com Received: from [127.0.0.1] (localhost [127.0.0.1]) by core3.amsl.com (Postfix) with ESMTP id AE3873A6905; Mon, 21 Jul 2008 05:22:52 -0700 (PDT) X-Original-To: bliss@core3.amsl.com Delivered-To: bliss@core3.amsl.com Received: from localhost (localhost [127.0.0.1]) by core3.amsl.com (Postfix) with ESMTP id D52C43A6943 for ; Mon, 21 Jul 2008 05:22:51 -0700 (PDT) X-Virus-Scanned: amavisd-new at amsl.com X-Spam-Flag: NO X-Spam-Score: -2.592 X-Spam-Level: X-Spam-Status: No, score=-2.592 tagged_above=-999 required=5 tests=[AWL=0.006, BAYES_00=-2.599, WHOIS_NETSOLPR=0.001] Received: from mail.ietf.org ([64.170.98.32]) by localhost (core3.amsl.com [127.0.0.1]) (amavisd-new, port 10024) with ESMTP id khSDg5v4OVvU for ; Mon, 21 Jul 2008 05:22:50 -0700 (PDT) Received: from mailgate.siemenscomms.co.uk (mailgate.siemenscomms.co.uk [195.171.110.225]) by core3.amsl.com (Postfix) with ESMTP id 2B35A3A68B3 for ; Mon, 21 Jul 2008 05:22:50 -0700 (PDT) Received: from GBNTHT12009MSX.gb002.siemens.net ([137.223.219.235]) by siemenscomms.co.uk (PMDF V6.3-x14 #31430) with ESMTP id <0K4C0049JVR4B4@siemenscomms.co.uk> for bliss@ietf.org; Mon, 21 Jul 2008 13:23:28 +0100 (BST) Date: Mon, 21 Jul 2008 13:23:27 +0100 From: "Elwell, John" In-reply-to: <4880C2B1.2020203@cisco.com> To: Jonathan Rosenberg , bliss@ietf.org Message-id: <0D5F89FAC29E2C41B98A6A762007F5D0E9C2AB@GBNTHT12009MSX.gb002.siemens.net> MIME-version: 1.0 X-MIMEOLE: Produced By Microsoft Exchange V6.5 Thread-Topic: [BLISS] configuring the proxy for ACH Thread-Index: Acjo81jfcc3Z5zhjTry6vI+lzgoFGQCKjXYQ Content-class: urn:content-classes:message X-MS-Has-Attach: X-MS-TNEF-Correlator: References: <4880C2B1.2020203@cisco.com> Subject: Re: [BLISS] configuring the proxy for ACH X-BeenThere: bliss@ietf.org X-Mailman-Version: 2.1.9 Precedence: list List-Id: "Basic Level of Interoperability for SIP Services \(BLISS\) BoF" List-Unsubscribe: , List-Archive: List-Post: List-Help: List-Subscribe: , Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: 7bit Sender: bliss-bounces@ietf.org Errors-To: bliss-bounces@ietf.org Jonathan, Thanks for these thoughts. See below. > -----Original Message----- > From: bliss-bounces@ietf.org [mailto:bliss-bounces@ietf.org] > On Behalf Of Jonathan Rosenberg > Sent: 18 July 2008 17:20 > To: bliss@ietf.org > Subject: [BLISS] configuring the proxy for ACH > > Here are some thoughts and a proposal on configuring ACH at the proxy. > > Its really important to keep in mind that BLISS is about REAL > interoperability. That means, figuring out what is REALLY causing the > interop failures for these features (which the survey and > follow on work > have done a good job at doing), and proposing REAL recommendations to > fix them. To me, REAL means, that this is something that we believe > folks will actually do, and would actually solve the problem. > Our metric > of success is, if folks implement our specs, you can plug a > phone from > one vendor into a call agent from another and it works for > that feature. > I don't think BLISS should be an academic exercise. > > With that in mind, some thoughts: > > 1. for configuring the proxy with ACH rules, the ONLY things > we need to > consider are those functions which typically have a button or > dedicated > function on the phone/UA. If something is set up via a web page, we > don't need to standardize interfaces for that. My sense is, most > endpoints that have built-in keys or embedded code for ACH > features are > almost exclusively limited to call forwarding (with several variants) > and DND. I don't think anyone is trying to do ToD-based > forwarding rules > from dedicated software in an endpoint. But if folks have examples, > please bring them out. [JRE] In principle, yes, as long as the mechanism is extensible to allow more complex rules to be configured in future. > > 2. XCAP *could* be a way to do this - in theory. But if you > look at what > folks have implemented, well xcap is not exactly a resounding > success. > If we mandate that all phones implement xcap to just set call > forward, > and that all call agents/servers/proxies implement it as well - do we > think they actually would. Being honest here - I dont think so. [JRE] Agreed. > > 3. The problem we're trying to face here - data-oriented > client-server > transactions - this is actually something that the rest of > the Internet > world has been very successful at solving. At this point, I think it > would be fruitless to mandate something that we have defined > (i.e., xcap > or a new sip mechanism), and instead, let us just ride the wave. From > what I can tell, where most folks are going for this stuff is > the most > trivial solution possible - and thats REST. > > 4. And so, my proposal would be to define a trivial REST > interface for > call forwarding at least. SO something like: > > POST http://server.com/joe/profile/callforwarding?value=true > > and thats it. PUT vs. POST or whatever, we need to find what the best > practices are in the "web2.0 world" (if you'll excuse the > buzzword). We > provide an interface for turning it on/off, indicating > whether it goes > to voicemail, allow a target for forwarding. [JRE] Sounds like a reasonable approach. What exactly would be standardised? Presumably everything after the server.com element, which would need to be configured in the UA. I think the call forwarding would at least need to be extended to cover some basic conditions, like busy, no reply, DND, and where to forward to (voicemail could be a default). > > 5. DND - for pre-call DND (where you hit a DND button and > future calls > never reach you). I think, in the modern world, DND is really > presence, > and you are seeing more and more presence deployments including a DND > state. So my suggestion here would be to use presence for > this - i.e., > the user sets their presence status to "DND". We can debate > whether this > is a new state or is the state, but lets agree on > something (most > folks are using a separate state for this - and we don't have a value > for it). [JRE] Are you arguing here for an enhancement to RPID (new state or substate of busy state)? Although DND might well be done via presence, this does not prevent a call arriving during the DND state, and we need some way to handle that call (i.e., indicate to the proxy that DND is in force, and configure the proxy to deal with calls in this state in a suitable way, such as forwarding to voicemail). > > 6. as a general rule, we should avoid as much as possible > RELYING on a > consistent mechanism for configuring the phone (i.e., config > framework), > for any of our features to work. Again, the REALITY is that endpoint > configuration is wildly variable amongst implementations, much more > variable than any other aspect of the protocol set on > endpoint devices. > I don't think that, having IETF recommend to implement the > IETF config > framework for some features to work, will change at all the > likelihood > of that happening in reality. [JRE] Agreed. The present ACH analysis draft only discusses the config framework in the context of configuring the UA whether to defer to the proxy, and even then it only uses the config framework as an example. > > Note that, I view configuring the phone as totally different from > pushing data to the proxy (for which I am proposing a REST interface > above). [JRE]Understood. John _______________________________________________ BLISS mailing list BLISS@ietf.org https://www.ietf.org/mailman/listinfo/bliss From bliss-bounces@ietf.org Mon Jul 28 06:46:01 2008 Return-Path: X-Original-To: bliss-archive@optimus.ietf.org Delivered-To: ietfarch-bliss-archive@core3.amsl.com Received: from [127.0.0.1] (localhost [127.0.0.1]) by core3.amsl.com (Postfix) with ESMTP id 528F83A6839; Mon, 28 Jul 2008 06:46:01 -0700 (PDT) X-Original-To: bliss@core3.amsl.com Delivered-To: bliss@core3.amsl.com Received: from localhost (localhost [127.0.0.1]) by core3.amsl.com (Postfix) with ESMTP id 6395B3A67A5 for ; Mon, 28 Jul 2008 06:45:59 -0700 (PDT) X-Virus-Scanned: amavisd-new at amsl.com X-Spam-Flag: NO X-Spam-Score: -6.599 X-Spam-Level: X-Spam-Status: No, score=-6.599 tagged_above=-999 required=5 tests=[BAYES_00=-2.599, RCVD_IN_DNSWL_MED=-4] Received: from mail.ietf.org ([64.170.98.32]) by localhost (core3.amsl.com [127.0.0.1]) (amavisd-new, port 10024) with ESMTP id Z1QIXZTIT35F for ; Mon, 28 Jul 2008 06:45:57 -0700 (PDT) Received: from ams-iport-1.cisco.com (ams-iport-1.cisco.com [144.254.224.140]) by core3.amsl.com (Postfix) with ESMTP id 6405F3A6808 for ; Mon, 28 Jul 2008 06:45:57 -0700 (PDT) X-IronPort-AV: E=Sophos;i="4.31,266,1215388800"; d="scan'208";a="15594232" Received: from ams-dkim-1.cisco.com ([144.254.224.138]) by ams-iport-1.cisco.com with ESMTP; 28 Jul 2008 13:45:38 +0000 Received: from ams-core-1.cisco.com (ams-core-1.cisco.com [144.254.224.150]) by ams-dkim-1.cisco.com (8.12.11/8.12.11) with ESMTP id m6SDjbMK023679 for ; Mon, 28 Jul 2008 15:45:37 +0200 Received: from xbh-ams-332.emea.cisco.com (xbh-ams-332.cisco.com [144.254.231.87]) by ams-core-1.cisco.com (8.13.8/8.13.8) with ESMTP id m6SDjbkh012043 for ; Mon, 28 Jul 2008 13:45:38 GMT Received: from xfe-ams-331.emea.cisco.com ([144.254.231.72]) by xbh-ams-332.emea.cisco.com with Microsoft SMTPSVC(6.0.3790.1830); Mon, 28 Jul 2008 15:45:37 +0200 Received: from [10.61.65.243] ([10.61.65.243]) by xfe-ams-331.emea.cisco.com with Microsoft SMTPSVC(6.0.3790.1830); Mon, 28 Jul 2008 15:45:37 +0200 Message-ID: <488DCD8A.2040202@cisco.com> Date: Mon, 28 Jul 2008 09:45:46 -0400 From: Jonathan Rosenberg User-Agent: Thunderbird 2.0.0.9 (Windows/20071031) MIME-Version: 1.0 To: bliss@ietf.org X-OriginalArrivalTime: 28 Jul 2008 13:45:37.0670 (UTC) FILETIME=[378D1260:01C8F0B8] DKIM-Signature: v=1; a=rsa-sha256; q=dns/txt; l=4368; t=1217252738; x=1218116738; c=relaxed/simple; s=amsdkim1002; h=Content-Type:From:Subject:Content-Transfer-Encoding:MIME-Version; d=cisco.com; i=jdrosen@cisco.com; z=From:=20Jonathan=20Rosenberg=20 |Subject:=20thoughts=20on=20draft-ietf-bliss-call-completio n |Sender:=20; bh=t2WuSe6p7Qa89Z/MlfK8KpWnBBRVbMIUX2RBkhZ4kmg=; b=hmBKS0uuSMt1IaiCzBbyPuWLMmKAR+DeS0JFOrY3LQNTa273qlMH1MZ2uh TpMWYBGpZ3Cetxm2iBPHAK5rzqK633BI5hWofzrz18y34y/dfufZrZL9ui4/ vei+LeDnU6; Authentication-Results: ams-dkim-1; header.From=jdrosen@cisco.com; dkim=pass ( sig from cisco.com/amsdkim1002 verified; ); Subject: [BLISS] thoughts on draft-ietf-bliss-call-completion X-BeenThere: bliss@ietf.org X-Mailman-Version: 2.1.9 Precedence: list List-Id: "Basic Level of Interoperability for SIP Services \(BLISS\) BoF" List-Unsubscribe: , List-Archive: List-Post: List-Help: List-Subscribe: , Content-Transfer-Encoding: 7bit Content-Type: text/plain; charset="us-ascii"; Format="flowed" Sender: bliss-bounces@ietf.org Errors-To: bliss-bounces@ietf.org A key part of the mechanism in here is that it focuses on the 'server to server' aspect of the problem. It defines only what would need to flow between domains or between organizations to accomplish this feature between organizations. I think that aspect of it needs to be emphasized. For example, some diagrams which show this, including different models for how the agents can be co-located. Very much related to that, ccbs has an effect of introducing work in one domain (the one of the callee) that primarily (if not exclusively) benefits the other (the one of the caller). This provides an unfortunate negative incentive for implementing this in a callee domain. I think we need to take care to minimize the amount of work required on the callee's side to support this. So for example, I am reluctant to introduce requirements to do things like suspend/resume, as these introduce additional work and state on behalf of the callee's agent. Also related, I think we need to consider the security implications of this. So, if a malicious caller calls a busy user many times, they can cause state to build up in the callee's agent. Indeed I think an implementation would almost be required to construct the call-info URI in a way that allowed it to contain all of the needed state, moving the burden to the caller. This isn't discussed at all, afaict, and its a critical design issue. I seem to recall that this was discussed previously. On the HERFP problem, I think the issue is really deeper than that. The issue is really targeting/re-forwarding. The question is, if A calls B and this is retargeted to C, does it make sense to call complete on B or on C? C kind of makes sense; but what if C is busy for a while and during that time, the forwarding rules change and calls to B no longer forward to C. Now, if A called B back they would reach B (who was the person they REALLY wanted to call anyway), but a call completion request would in fact go to C. This seems wrong to me. One might even argue that call completion and forwarding are incompatible. Frankly, if our initial solution basically didn't support it (no Call-Info passed back at all when a call is forwarded), I think that is arguably a feature. I suspect experts on feature interaction have looked at this one; would be curious on the best practices around this in the PSTN. I also think its a mistake to require 199 or 130 or any other 'HERFP' response - again, making this too complicated. Simplicity is key for this feature IMHO. I have a practical worry on using the Call-ID as a correlator for the subscription. The reality of deployments are many B2BUAs exist, and this is no longer a reliable e2e correlator for things. Anyway its not needed; the URI should be sufficient (and unlike call-id, is reliable). Also, I would NOT assume that SUBSCRIBE and INVITE to the same URI are routed identically; this is not even true in theory as proxies are allowed to do method-based routing. Indeed, typically SUBSCRIBE are routed to appropriate servers based on event packages. I think the callback INVITE needs to be to a different SIP URI. I suspect it will need different treatment on both sides; i.e., that call should not go to voicemail (I suspect). Using the philosophy embraced in: http://www.ietf.org/internet-drafts/draft-ietf-sipping-service-identification-02.txt a service URI is absolutely the right thing to differentiate here. I do not understand the usage or need for the 'm' and 'monitor' attributes. Monitor shows up as a URI param, in fact. Not at all sure why that is needed. What does a calling domain do when it wants to invoke this service and the terminating side doesn't support it. Do we have a recommended 'poor mans' version of this that requires no additional support? i.e., I am aware of implementations that actually periodically send INVITEs. Indeed, doing so may be incentive to properly deploy a sub/not solution to avoid such deluge... Thanks, Jonathan R. -- Jonathan D. Rosenberg, Ph.D. 499 Thornall St. Cisco Fellow Edison, NJ 08837 Cisco, Voice Technology Group jdrosen@cisco.com http://www.jdrosen.net PHONE: (408) 902-3084 http://www.cisco.com _______________________________________________ BLISS mailing list BLISS@ietf.org https://www.ietf.org/mailman/listinfo/bliss From bliss-bounces@ietf.org Mon Jul 28 09:32:06 2008 Return-Path: X-Original-To: bliss-archive@optimus.ietf.org Delivered-To: ietfarch-bliss-archive@core3.amsl.com Received: from [127.0.0.1] (localhost [127.0.0.1]) by core3.amsl.com (Postfix) with ESMTP id E14F528C1AA; Mon, 28 Jul 2008 09:32:06 -0700 (PDT) X-Original-To: bliss@core3.amsl.com Delivered-To: bliss@core3.amsl.com Received: from localhost (localhost [127.0.0.1]) by core3.amsl.com (Postfix) with ESMTP id EFDB628C10C for ; Mon, 28 Jul 2008 09:32:05 -0700 (PDT) X-Virus-Scanned: amavisd-new at amsl.com X-Spam-Flag: NO X-Spam-Score: -1.951 X-Spam-Level: X-Spam-Status: No, score=-1.951 tagged_above=-999 required=5 tests=[AWL=0.647, BAYES_00=-2.599, HTML_MESSAGE=0.001] Received: from mail.ietf.org ([64.170.98.32]) by localhost (core3.amsl.com [127.0.0.1]) (amavisd-new, port 10024) with ESMTP id BAdGSAN2OKoc for ; Mon, 28 Jul 2008 09:32:03 -0700 (PDT) Received: from mout.perfora.net (mout.perfora.net [74.208.4.194]) by core3.amsl.com (Postfix) with ESMTP id 3B4D33A6ABE for ; Mon, 28 Jul 2008 09:32:03 -0700 (PDT) Received: from s73602 ([130.129.23.81]) by mrelay.perfora.net (node=mrus0) with ESMTP (Nemesis) id 0MKp8S-1KNVdr0gBB-0005j1; Mon, 28 Jul 2008 12:32:13 -0400 Message-ID: <0f7901c8f0cf$9d57e9a0$850e10ac@china.huawei.com> From: "Spencer Dawkins" To: Date: Mon, 28 Jul 2008 11:33:01 -0500 MIME-Version: 1.0 X-Priority: 3 X-MSMail-Priority: Normal X-Mailer: Microsoft Outlook Express 6.00.2900.3138 X-MimeOLE: Produced By Microsoft MimeOLE V6.00.2900.3198 X-Provags-ID: V01U2FsdGVkX18QKMOWSfm+Cv2cwDA5lXLsafbtTu2Gkbs87HP xcSw1sd4baTO6sJIBC6iLZrFdIPAZemPPsiAspXRaOwDHzZlmB rNBT4auZhy5PHwyKhJ/KS/T5zCIDIdskC8KTcj/8OI= Subject: [BLISS] My notes from Bliss session X-BeenThere: bliss@ietf.org X-Mailman-Version: 2.1.9 Precedence: list List-Id: "Basic Level of Interoperability for SIP Services \(BLISS\) BoF" List-Unsubscribe: , List-Archive: List-Post: List-Help: List-Subscribe: , Content-Type: multipart/mixed; boundary="===============0802708359==" Sender: bliss-bounces@ietf.org Errors-To: bliss-bounces@ietf.org This is a multi-part message in MIME format. --===============0802708359== Content-Type: multipart/alternative; boundary="----=_NextPart_000_0F76_01C8F0A5.B13045A0" This is a multi-part message in MIME format. ------=_NextPart_000_0F76_01C8F0A5.B13045A0 Content-Type: text/plain; charset="iso-8859-1" Content-Transfer-Encoding: quoted-printable 1.1 BLISS 1.1.1 Agenda Bashing / Status Update Presented by: Chairs Jason unable to attend today. Design teams have been very active, but mostly NOT on the list. Call Park/Pickup - no teleconferences so no progress. 1.1.2 Automaric Call Handling Presented by: John Elwell Draft:draft-ietf-bliss-ach-analysis-02.txt Two topics at last meeting that we decided not to address at last = meeting. Focused on addressing conflict between proxy and UA (don't want = them to interfere with each other if they can both do ACH) - mostly = focused on making sure UA doesn't interfere with proxies. Proposing that dataset draft (in SIPPING, but it's not a working group = draft) is extended to include a parameter. Jonathan was very concerned that this is a huge hurdle - configuration = framework isn't widely deployed today - just to support one = configuration issue. Don't think people will do this. Francois - if this is a config option, just say that. If config = framework ever takes off, we can add the parameter then. Mary - why invent another way when we've got the config framework? Jonathan - benchmark is phone and PBX from different figures just work - = if you don't include a way to configure, is that helping = interoperability? Martin - these devices already have a configuration capability, and most = have ability to turn capabilities off and give control to the proxy. = Already exists and is being used. Shida - so we don't need to standardize anything here?=20 Jonathan - configure this property through whatever method your device = supports. Lots of discussions on design team about getting information from UA for = ACH at proxy. Down to busy/explicit rejection/silent rejection and = local/global scope. 487 code use is different from what's in 3261. No recommendation for = global silent rejection code value. 480 is suggested for DND in 3261, but this looks like overload with edge = proxy indicating temporary lack of resources. Is global silent rejection a RUCUS topic?=20 Francois - Is there a reason you want to reject globally silently? If = this is nothing but SPIT, leave it for RUCUS or some follow-on. Is anyone concerned about overloading 480?=20 Francois - don't have a concern about this, but if someone does, it's a = 3261 issue and don't think it's caused interop problems. Don't worry = about this unless we see a reason to worry. Don't alter meaning of 6xx response codes - if something global is = required, use 4xx/5xx. Configuring ACH at the proxy from UA - XCAP considered to be overkill, = Jonathan proposed REST - how to extend this, and where to standardize = it? Not a lot of list discussion on this topic. Any issue in pursuing REST? Francois - REST is a trivial solution - good, but don't think it's only = used by ACH - would be used by a number of drafts. Would need to set up a registry.=20 Is this limited in scope to BLISS? Jonathan - suggested this instead of XCAP because REST is deployed all = over the network - Web 2.0, just use the framework, who needs a = registry? Scott - support this in general, but don't generalize into a framework = until we're sure we have something that works. There are lots of PBX = features people want to support from phone-tops. Think we should focus = on example first and get it working - then generalize. Mary - go ahead and take a higher view, use this as a use case. Jonathan - "agile process" - have next rev of document, look at it and = decide. Don't think I know what a framework for this looks like. XCAP = disappears in a puff of smoke when you say "REST". One step at a time. Allen - seems reasonable for trivial things - won't you get into = situations where you need to read? That's actually GET. This is running on the Internet for way more = complicated stuff than this. Keith - haven't decided what parameter to set yet, so need to understand = scope first. Jonathan - the world has moved on here. We need to wake up and smell the = coffee. Concept is similar and doesn't have XCAP's problems. Keith - don't agree with the way forward - should be a different = document that makes a case for a different configuration framework. Don't want to specify exactly what ACH is - very = service-provider-specific, not what we are trying to do in the IETF. Christer - may not have common understanding of what ACH is. Martin - we know how to turn these parameters off and on - look at that, = start simple. Jonathan - just agree on parameters and interpretation by the process. = If we can't figure out the parameters for minimal ACH, discussion of = framework is pointless. 1.1.3 Call Completion Presented by: Denis Alexeitsev Draft:draft-ietf-bliss-call-completion-02.txt Suspend and resume procedures proposed to use PUBLISH. Proposing CC = event body.=20 How to correlate INVITE and SUBSCRIBE? Intermediate proxies might change = call-ID.=20 Adam - if you change the call-ID, you're not a proxy, and we shouldn't = bend over backwards to accommodate SBCs.=20 Jonathan - reality is that this is an unreliable header. Thought we had = figured out how to do this without adding state (which is necessary from = security perspective). Keith - what are you correlating? Initial INVITE and SUBSCRIBE. Jonathan - I thought we needed this because you can't ask to be queued = for call completion if you haven't called the person yet. Keith - that's in ISDN. Jonathan - security. if not, I can send arbitrary subscribes. Keith - this is basically setting up a call for when a user is available = Jonathan - you either have to keep state or use a URI as a token - this = is an authorization question. Keith - you don't need to create state. (some "yes"/"no" exchange happened about here). How to route the suspension PUBLISH to the appropriate monitor? Send = within the SUBSCRIBE dialog? Use a GRUU received in the NOTIFY body? Jonathan - defer this feature - consumes resources in called party = domain for benefit of calling party - already a disincentive for = deployment. Not familiar with PSTN side, but if this can be rejected on = PSTN side, should also reject it on SIP side. Don't think this is = mandatory - people poll for this today. Francois - getting complicated because we're trying to interwork with = ISDN, I think it's implemented the same way as ISDN, we're getting = complicated while spinning our wheels and making this less likely to be = deployed.=20 Adam - not disagreeing with Jonathan/Francois - but if we do go down = this path, GRUU is the way we're headed. I expect to press for = deprecating anything except GRUUs in Contact header when we rev 3261. Hums -=20 =B7 suspend/resume removed from the spec? =B7 unsubscribe and subscribe? This was silent. =B7 Event packages? This was silent Trying again =B7 Keep suspend/resume? Sounded pretty evenly divided (between = keep/don't keep) - will take to the list. Adam - can we do basic feature and do extensions later? Nail down = 99-percent use case. This is minor capability that gives us half the = complexity in the draft. John - people can enhance later, this is taking a lot of time to = produce, keep it as simple as we can, otherwise we'll never finish = anything. If you want the callback feature, you probably want this = finished quickly. 1.1.4 Line Sharing Presented by: Alan Johnston Draft:draft-johnston-bliss-mla-req-02.txt Three methods (INVITE overlap is new). Need to pick one and go forward. Most UAs don't have to implement any of the methods (don't have multiple = line appearances). Adam - no matter what method we pick, something in the middle has to = understand MLA. Implementation complexity is the same.=20 PUBLISH/NOTIFY has deployments, design team supports it. Floor control = is cleaner architecturally but has no supporters. INVITE stretches = overlap dialing but doesn't require protocol work. Jonathan - don't agree with 3265 violation. PUBLISH/NOTIFY is what = people are doing. Do what people are doing if you want interoperability. Francois - list so far is nobody likes BFCP, support for PUBLISH/NOTIFY, = was thinking INVITE was OK but don't like it now, looking at the call = flows. Does proxy decide a line has to be seized, or is it the UA? = Server would ask client to say "tell me when you're in 100 state". Not = clear what the intent was, just jumped into mechanisms. Regardless of = choice, tempted to say that INVITE is least attractive of three choices. Hum - strongly favored PUBLISH/NOTIFY mechanism. 1.1.5 History Info and Diversion Interworking Presented by: Xavier Marjou Draft:draft-mohali-diversion-history-info-00.txt Two solutions - Diversion header (proprietary but widely deployed) and = History-Info (RFC 4244, also adopted by other SDOs but not widely = implemented or deployed). Almost no equipment supports History-Info. Francois - History-Info isn't Diversion, we're sabotaging History-Info = by dumbing it down, so it won't work for anything except PSTN = interworking). My company is guilty, too. Adding a parameter (as = Jonathan suggests) is only way I see to do this that will work. Mary - don't disagree - 4458 isn't replacing History-Info. We started = out from Diversion header, we have something general, if operators want = this, let them pay vendors to develop - they aren't paying to develop = History-Info. Ben - we've decided not to go the Diversion route many times. Martin - there are many SIP phones that support Diversion and won't be = upgraded. We're going to be receiving this header for a long time. We = need a way to provide interworking - that will help carriers decide to = deploy History-Info. Steve - agree. Diversion is much wider than ISDN these days. Big bang = scenarios never work - you have to have a transition story. Jean-Francois - Diversion is there, widely deployed, in some cases it's = what people need. Unanimous feedback from several operators in cable = space was "no, Diversion is there and it's good enough". ??? - Proposal is not to standardize Diversion header, it's to = standardize a mapping mechanism. We have this problem today. We may = discuss, but at least the requirements are there. Robert - apologize for the poke, but if you ask vendors to implement a = non-standard, the pain should be yours. But let me ask - draft is = proposing bi-directional mapping. Why do we need = History-Info-to-Diversion mapping? Mary - we implemented both, why can't you carry both and switch over to = 4458 for interworking? In some cases, we do preserve both. Francois - not bidirectional, there are things you can't do in PSTN, so = you lose information (forking in SIP, etc.). History-Info tells you = what's in SIP, not what happens behind a gateway. If you merge them, = you're changing History-Info's goal and it becomes unreliable at = describing what happened to the request. If we're going to do this, why = do we need History-Info in the first place? Jean-Francois - but you have a good representation of what carriers are = saying in the draft. Problem is that operators need to deploy with = software that's been tested in their labs and is available, at some = point in time. We tried to get people to upgrade. It's not going to = work. Some operators are deploying new SIP services and will require = History-Info support, but they will have to deal with Diversion for a = long time. Christer - this reminds me of INFO discussion. customers didn't ask for = Diversion because it's better, they ask because that's what out there = today. Maybe mapping isn't right term, but we need interworking. We lose = information when we interconnect today. Martin - we have a fair amount of wireline carriers globally and in the = US on this draft. If you think every vendor's product is compliant to = these RFCs, you're nuts. There are TAs, SIP Phones, even IP-PBXes that = do Diversion - if we don't go this way, we'll just fix the problems on = an SBC. Mary - we should do an informational spec - not standards-track. Hadriel - if we do this mapping, we get SOME information, but if we = don't, we get NO information.=20 Jonathan - agree with many folks, including Martin(!). We can't just = change SIP when we have a new idea. Agnostic on details here, but = discussion about this topic is the right thing to do - not doing it is = like ignoring NATs. Not just a service-provider problem, it's also = deployed in enterprise networks. Jean-Francois - rough consensus and running code. Francois - this is surreal. We're lobotomizing History-Info to do this = mapping thing. There is a problem. Best way is to do this in parallel = (with a real History-Info definition). If group doesn't think that's a = good idea, we have to do transition in non-destructive way to show this = is a redirection in the classical sense. Keith - draft is about interop - if it's supposed to be migration, you = need to say so in the draft. Robert - no one has answered my question - asking again. Can see mapping = in one direction. Is there a need for a bi-directional mapping? Martin - IP-PBX call center, may have multiple redirections. Robert - that's scale =3D 1, not scale 100,000 and nothing is making = History-Info now - what's the chances it will start showing up? Hadriel - seeing History-Info in peering situations (but they may be = mapping it right back to Diversion). Eric - draft is expired and five years old.=20 Spencer - this is the kind of thing Informational publication is for. ------=_NextPart_000_0F76_01C8F0A5.B13045A0 Content-Type: text/html; charset="iso-8859-1" Content-Transfer-Encoding: quoted-printable

1.1     =20 BLISS

1.1.1           = ;=20 Agenda Bashing / Status = Update

Presented by: Chairs

Jason=20 unable to attend today=85

Design=20 teams have been very active, but mostly NOT on the list.

Call=20 Park/Pickup =96 no teleconferences so no progress.

1.1.2           = ;=20 Automaric Call = Handling

Presented by: John Elwell

Draft:draft-ietf-bliss-ach-analysis-02.txt

Two=20 topics at last meeting that we decided not to address at last meeting. = Focused=20 on addressing conflict between proxy and UA (don=92t want them to = interfere with=20 each other if they can both do ACH) =96 mostly focused on making sure UA = doesn=92t=20 interfere with proxies.

Proposing that dataset draft (in SIPPING, but it=92s not a = working group=20 draft) is extended to include a parameter.

Jonathan=20 was very concerned that this is a huge hurdle =96 configuration = framework isn=92t=20 widely deployed today =96 just to support one configuration issue. = Don=92t think=20 people will do this.

Francois=20 =96 if this is a config option, just say that. If config framework ever = takes off,=20 we can add the parameter then.

Mary =96=20 why invent another way when we=92ve got the config framework?

Jonathan=20 =96 benchmark is phone and PBX from different figures just work =96 if = you don=92t=20 include a way to configure, is that helping interoperability?

Martin =96=20 these devices already have a configuration capability, and most have = ability to=20 turn capabilities off and give control to the proxy. Already exists and = is being=20 used.

Shida =96=20 so we don=92t need to standardize anything here?

Jonathan=20 =96 configure this property through whatever method your device=20 supports.

Lots of=20 discussions on design team about getting information from UA for ACH at = proxy.=20 Down to busy/explicit rejection/silent rejection and local/global=20 scope.

487 code=20 use is different from what=92s in 3261. No recommendation for global = silent=20 rejection code value.

480 is=20 suggested for DND in 3261, but this looks like overload with edge proxy=20 indicating temporary lack of resources.

Is=20 global silent rejection a RUCUS topic?

Francois=20 - Is there a reason you want to reject globally silently?  If this is nothing but SPIT, = leave it=20 for RUCUS or some follow-on.

Is=20 anyone concerned about overloading 480?

Francois=20 =96 don=92t have a concern about this, but if someone does, it=92s a = 3261 issue and=20 don=92t think it=92s caused interop problems. Don=92t worry about this = unless we see a=20 reason to worry.

Don=92t=20 alter meaning of 6xx response codes =96 if something global is required, = use=20 4xx/5xx.

Configuring ACH at the proxy from UA =96 XCAP considered to be = overkill,=20 Jonathan proposed REST =96 how to extend this, and where to standardize=20 it?

Not a=20 lot of list discussion on this topic. Any issue in pursuing = REST?

Francois=20 =96 REST is a trivial solution =96 good, but don=92t think it=92s only = used by ACH =96=20 would be used by a number of drafts.

Would=20 need to set up a registry.

Is this=20 limited in scope to BLISS?

Jonathan=20 =96 suggested this instead of XCAP because REST is deployed all over the = network =96=20 Web 2.0, just use the framework, who needs a registry?

Scott =96=20 support this in general, but don=92t generalize into a framework until = we=92re sure=20 we have something that works. There are lots of PBX features people want = to=20 support from phone-tops. Think we should focus on example first and get = it=20 working =96 then generalize.

Mary =96=20 go ahead and take a higher view, use this as a use case.

Jonathan=20 =96 =93agile process=94 =96 have next rev of document, look at it and = decide. Don=92t=20 think I know what a framework for this looks like. XCAP disappears in a = puff of=20 smoke when you say =93REST=94. One step at a time.

Allen =96=20 seems reasonable for trivial things =96 won=92t you get into situations = where you=20 need to read?

That=92s=20 actually GET=85 This is running on the Internet for way more complicated = stuff=20 than this.

Keith =96=20 haven=92t decided what parameter to set yet, so need to understand scope = first.

Jonathan=20 =96 the world has moved on here. We need to wake up and smell the = coffee. Concept=20 is similar and doesn=92t have XCAP=92s problems.

Keith =96=20 don=92t agree with the way forward =96 should be a different document = that makes a=20 case for a different configuration framework.

Don=92t=20 want to specify exactly what ACH is =96 very service-provider-specific, = not what=20 we are trying to do in the IETF.

Christer=20 =96 may not have common understanding of what ACH is.

Martin =96=20 we know how to turn these parameters off and on =96 look at that, start=20 simple.

Jonathan=20 =96 just agree on parameters and interpretation by the process.  If we can=92t figure out the = parameters for=20 minimal ACH, discussion of framework is pointless.

1.1.3           = ;=20 Call = Completion

Presented by: Denis Alexeitsev

Draft:draft-ietf-bliss-call-completion-02.txt

Suspend=20 and resume procedures proposed to use PUBLISH.  Proposing CC event body. =

How to=20 correlate INVITE and SUBSCRIBE? Intermediate proxies might change = call-ID.=20

Adam =96=20 if you change the call-ID, you=92re not a proxy, and we shouldn=92t bend = over=20 backwards to accommodate SBCs.

Jonathan=20 =96 reality is that this is an unreliable header. Thought we had figured = out how=20 to do this without adding state (which is necessary from security=20 perspective).

Keith =96=20 what are you correlating? Initial INVITE and SUBSCRIBE.

Jonathan=20 =96 I thought we needed this because you can=92t ask to be queued for = call=20 completion if you haven=92t called the person yet.

Keith =96=20 that=92s in ISDN.

Jonathan=20 =96 security=85 if not, I can send arbitrary subscribes.

Keith =96=20 this is basically setting up a call for when a user is available =

Jonathan=20 =96 you either have to keep state or use a URI as a token =96 this is an = authorization question.

Keith =96=20 you don=92t need to create state.

(some=20 =93yes=94/=94no=94 exchange happened about here).

How to=20 route the suspension PUBLISH to the appropriate monitor? Send within the = SUBSCRIBE dialog? Use a GRUU received in the NOTIFY body?

Jonathan=20 =96 defer this feature =96 consumes resources in called party domain for = benefit of=20 calling party =96 already a disincentive for deployment. Not familiar = with PSTN=20 side, but if this can be rejected on PSTN side, should also reject it on = SIP=20 side. Don=92t think this is mandatory =96 people poll for this = today.

Francois=20 =96 getting complicated because we=92re trying to interwork with ISDN, I = think it=92s=20 implemented the same way as ISDN, we=92re getting complicated while = spinning our=20 wheels and making this less likely to be deployed.

Adam =96=20 not disagreeing with Jonathan/Francois =96 but if we do go down this = path, GRUU is=20 the way we=92re headed. I expect to press for deprecating anything = except GRUUs in=20 Contact header when we rev 3261.

Hums =96=20

=B7        =20 suspend/resume removed = from the=20 spec?

=B7        =20 unsubscribe and = subscribe? This was=20 silent.

=B7        =20 Event packages? This = was=20 silent

Trying=20 again

=B7        =20 Keep suspend/resume? = Sounded pretty=20 evenly divided (between keep/don=92t keep) =96 will take to the = list.

Adam =96=20 can we do basic feature and do extensions later? Nail down 99-percent = use case.=20 This is minor capability that gives us half the complexity in the=20 draft.

John =96=20 people can enhance later, this is taking a lot of time to produce, keep = it as=20 simple as we can, otherwise we=92ll never finish anything. If you want = the=20 callback feature, you probably want this finished quickly.

1.1.4           = ;=20 Line = Sharing

Presented by: Alan Johnston

Draft:draft-johnston-bliss-mla-req-02.txt

Three=20 methods (INVITE overlap is new). Need to pick one and go = forward.

Most UAs=20 don=92t have to implement any of the methods (don=92t have multiple line = appearances).

Adam =96=20 no matter what method we pick, something in the middle has to understand = MLA.=20 Implementation complexity is the same.

PUBLISH/NOTIFY has deployments, design team supports it. Floor = control is=20 cleaner architecturally but has no supporters. INVITE stretches overlap = dialing=20 but doesn=92t require protocol work.

Jonathan=20 =96 don=92t agree with 3265 violation. PUBLISH/NOTIFY is what people are = doing. Do=20 what people are doing if you want interoperability.

Francois=20 =96 list so far is nobody likes BFCP, support for PUBLISH/NOTIFY, was = thinking=20 INVITE was OK but don=92t like it now, looking at the call flows. Does = proxy=20 decide a line has to be seized, or is it the UA? Server would ask client = to say=20 =93tell me when you=92re in 100 state=94.  Not clear what the intent was, = just=20 jumped into mechanisms. Regardless of choice, tempted to say that INVITE = is=20 least attractive of three choices.

Hum =96=20 strongly favored PUBLISH/NOTIFY mechanism.

1.1.5           = ;=20 History Info and = Diversion=20 Interworking

Presented by: Xavier Marjou

Draft:draft-mohali-diversion-history-info-00.txt

Two=20 solutions =96 Diversion header (proprietary but widely deployed) and = History-Info=20 (RFC 4244, also adopted by other SDOs but not widely implemented or=20 deployed).

Almost=20 no equipment supports History-Info.

Francois=20 =96 History-Info isn=92t Diversion, we=92re sabotaging History-Info by = dumbing it=20 down, so it won=92t work for anything except PSTN interworking). My = company is=20 guilty, too. Adding a parameter (as Jonathan suggests) is only way I see = to do=20 this that will work.

Mary =96=20 don=92t disagree =96 4458 isn=92t replacing History-Info. We started out = from=20 Diversion header, we have something general, if operators want this, let = them=20 pay vendors to develop =96 they aren=92t paying to develop = History-Info.

Ben =96=20 we=92ve decided not to go the Diversion route many times.

Martin =96=20 there are many SIP phones that support Diversion and won=92t be = upgraded. We=92re=20 going to be receiving this header for a long time. We need a way to = provide=20 interworking =96 that will help carriers decide to deploy = History-Info.

Steve =96=20 agree. Diversion is much wider than ISDN these days. Big bang scenarios = never=20 work =96 you have to have a transition story.

Jean-Francois =96 Diversion is there, widely deployed, in some = cases it=92s=20 what people need. Unanimous feedback from several operators in cable = space was=20 =93no, Diversion is there and it=92s good enough=94.

??? =96=20 Proposal is not to standardize Diversion header, it=92s to standardize a = mapping=20 mechanism. We have this problem today. We may discuss, but at least the=20 requirements are there.

Robert =96=20 apologize for the poke, but if you ask vendors to implement a = non-standard, the=20 pain should be yours. But let me ask =96 draft is proposing = bi-directional=20 mapping. Why do we need History-Info-to-Diversion mapping?

Mary =96=20 we implemented both, why can=92t you carry both and switch over to 4458 = for=20 interworking?

In some=20 cases, we do preserve both.

Francois=20 =96 not bidirectional, there are things you can=92t do in PSTN, so you = lose=20 information (forking in SIP, etc.). History-Info tells you what=92s in = SIP, not=20 what happens behind a gateway. If you merge them, you=92re changing = History-Info=92s=20 goal and it becomes unreliable at describing what happened to the = request. If=20 we=92re going to do this, why do we need History-Info in the first=20 place?

Jean-Francois =96 but you have a good representation of what = carriers are=20 saying in the draft. Problem is that operators need to deploy with = software=20 that=92s been tested in their labs and is available, at some point in = time. We=20 tried to get people to upgrade. It=92s not going to work. Some operators = are=20 deploying new SIP services and will require History-Info support, but = they will=20 have to deal with Diversion for a long time.

Christer=20 =96 this reminds me of INFO discussion=85 customers didn=92t ask for = Diversion because=20 it=92s better, they ask because that=92s what out there today. Maybe = mapping isn=92t=20 right term, but we need interworking. We lose information when we = interconnect=20 today.

Martin =96=20 we have a fair amount of wireline carriers globally and in the = US on this draft. If you = think every=20 vendor=92s product is compliant to these RFCs, you=92re nuts. There are = TAs, SIP=20 Phones, even IP-PBXes that do Diversion =96 if we don=92t go this way, = we=92ll just=20 fix the problems on an SBC.

Mary =96=20 we should do an informational spec =96 not standards-track.

Hadriel=20 =96 if we do this mapping, we get SOME information, but if we don=92t, = we get NO=20 information.

Jonathan=20 =96 agree with many folks, including Martin(!). We can=92t just change = SIP when we=20 have a new idea. Agnostic on details here, but discussion about this = topic is=20 the right thing to do =96 not doing it is like ignoring NATs. Not just a = service-provider problem, it=92s also deployed in enterprise = networks.

Jean-Francois =96 rough consensus and running = code=85

Francois=20 =96 this is surreal. We=92re lobotomizing History-Info to do this = mapping thing.=20 There is a problem. Best way is to do this in parallel (with a real = History-Info=20 definition). If group doesn=92t think that=92s a good idea, we have to = do transition=20 in non-destructive way to show this is a redirection in the classical=20 sense.

Keith =96=20 draft is about interop =96 if it=92s supposed to be migration, you need = to say so in=20 the draft.

Robert =96=20 no one has answered my question =96 asking again. Can see mapping in one = direction. Is there a need for a bi-directional mapping?

Martin =96=20 IP-PBX call center, may have multiple redirections=85

Robert =96=20 that=92s scale =3D 1, not scale 100,000 and nothing is making = History-Info now =96=20 what=92s the chances it will start showing up?

Hadriel=20 =96 seeing History-Info in peering situations (but they may be mapping = it right=20 back to Diversion).

Eric =96=20 draft is expired and five years old.

Spencer=20 =96 this is the kind of thing Informational publication is=20 for=85

------=_NextPart_000_0F76_01C8F0A5.B13045A0-- --===============0802708359== Content-Type: text/plain; charset="us-ascii" MIME-Version: 1.0 Content-Transfer-Encoding: 7bit Content-Disposition: inline _______________________________________________ BLISS mailing list BLISS@ietf.org https://www.ietf.org/mailman/listinfo/bliss --===============0802708359==-- From bliss-bounces@ietf.org Mon Jul 28 10:51:03 2008 Return-Path: X-Original-To: bliss-archive@optimus.ietf.org Delivered-To: ietfarch-bliss-archive@core3.amsl.com Received: from [127.0.0.1] (localhost [127.0.0.1]) by core3.amsl.com (Postfix) with ESMTP id 03E8028C1D8; Mon, 28 Jul 2008 10:51:03 -0700 (PDT) X-Original-To: bliss@core3.amsl.com Delivered-To: bliss@core3.amsl.com Received: from localhost (localhost [127.0.0.1]) by core3.amsl.com (Postfix) with ESMTP id 2A7723A6829 for ; Mon, 28 Jul 2008 10:51:01 -0700 (PDT) X-Virus-Scanned: amavisd-new at amsl.com X-Spam-Flag: NO X-Spam-Score: -6.166 X-Spam-Level: X-Spam-Status: No, score=-6.166 tagged_above=-999 required=5 tests=[AWL=0.433, BAYES_00=-2.599, RCVD_IN_DNSWL_MED=-4] Received: from mail.ietf.org ([64.170.98.32]) by localhost (core3.amsl.com [127.0.0.1]) (amavisd-new, port 10024) with ESMTP id F8JZI5Q0xR-r for ; Mon, 28 Jul 2008 10:50:55 -0700 (PDT) Received: from rtp-iport-2.cisco.com (rtp-iport-2.cisco.com [64.102.122.149]) by core3.amsl.com (Postfix) with ESMTP id E3EDC3A6838 for ; Mon, 28 Jul 2008 10:50:54 -0700 (PDT) X-IronPort-AV: E=Sophos;i="4.31,266,1215388800"; d="scan'208";a="15758926" Received: from rtp-dkim-2.cisco.com ([64.102.121.159]) by rtp-iport-2.cisco.com with ESMTP; 28 Jul 2008 17:51:04 +0000 Received: from rtp-core-1.cisco.com (rtp-core-1.cisco.com [64.102.124.12]) by rtp-dkim-2.cisco.com (8.12.11/8.12.11) with ESMTP id m6SHp4H1003938 for ; Mon, 28 Jul 2008 13:51:04 -0400 Received: from xbh-rtp-201.amer.cisco.com (xbh-rtp-201.cisco.com [64.102.31.12]) by rtp-core-1.cisco.com (8.13.8/8.13.8) with ESMTP id m6SHp4L8004306 for ; Mon, 28 Jul 2008 17:51:04 GMT 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); Mon, 28 Jul 2008 13:51:04 -0400 Received: from [161.44.174.168] ([161.44.174.168]) by xfe-rtp-202.amer.cisco.com with Microsoft SMTPSVC(6.0.3790.1830); Mon, 28 Jul 2008 13:51:04 -0400 Message-ID: <488E073E.9040904@cisco.com> Date: Mon, 28 Jul 2008 13:51:58 -0400 From: Paul Kyzivat User-Agent: Thunderbird 2.0.0.16 (Windows/20080708) MIME-Version: 1.0 To: Jonathan Rosenberg References: <488DCD8A.2040202@cisco.com> In-Reply-To: <488DCD8A.2040202@cisco.com> X-OriginalArrivalTime: 28 Jul 2008 17:51:04.0165 (UTC) FILETIME=[8139E150:01C8F0DA] DKIM-Signature: v=1; a=rsa-sha256; q=dns/txt; l=8163; t=1217267464; x=1218131464; c=relaxed/simple; s=rtpdkim2001; h=Content-Type:From:Subject:Content-Transfer-Encoding:MIME-Version; d=cisco.com; i=pkyzivat@cisco.com; z=From:=20Paul=20Kyzivat=20 |Subject:=20Re=3A=20[BLISS]=20thoughts=20on=20draft-ietf-bl iss-call-completion |Sender:=20 |To:=20Jonathan=20Rosenberg=20; bh=B5gA8fdYJao6Pdvab1idreDdKgppktyeMJ2bMDKzduc=; b=GUyC5ZtK70eSujKt0iRSHXEskhxZ+OnlqpknOsK9Hv6rGeia6A3X1DO2EK EHn4Rj4O3D22qUBxltoWIfRsvknzRWQmTNdF9uGptOCuVr3q0U7QdpNn233M xKfJoZeyg4; Authentication-Results: rtp-dkim-2; header.From=pkyzivat@cisco.com; dkim=pass ( sig from cisco.com/rtpdkim2001 verified; ); Cc: bliss@ietf.org Subject: Re: [BLISS] thoughts on draft-ietf-bliss-call-completion X-BeenThere: bliss@ietf.org X-Mailman-Version: 2.1.9 Precedence: list List-Id: "Basic Level of Interoperability for SIP Services \(BLISS\) BoF" List-Unsubscribe: , List-Archive: List-Post: List-Help: List-Subscribe: , Content-Transfer-Encoding: 7bit Content-Type: text/plain; charset="us-ascii"; Format="flowed" Sender: bliss-bounces@ietf.org Errors-To: bliss-bounces@ietf.org Lots of comments below. Jonathan Rosenberg wrote: > A key part of the mechanism in here is that it focuses on the 'server to > server' aspect of the problem. It defines only what would need to flow > between domains or between organizations to accomplish this feature > between organizations. I think that aspect of it needs to be emphasized. > For example, some diagrams which show this, including different models > for how the agents can be co-located. > > Very much related to that, ccbs has an effect of introducing work in one > domain (the one of the callee) that primarily (if not exclusively) > benefits the other (the one of the caller). This provides an unfortunate > negative incentive for implementing this in a callee domain. I think we > need to take care to minimize the amount of work required on the > callee's side to support this. So for example, I am reluctant to > introduce requirements to do things like suspend/resume, as these > introduce additional work and state on behalf of the callee's agent. While I agree that it is important to consider what the incentives are for implementing this stuff, I don't think this should be viewed as primarily benefiting the caller at the expense of the callee. At least not if this is considered an *optional* feature at the callee. A callee will offer this capability if he considers it advantageous - namely if he considers the incoming calls to his benefit, and missing them to be a loss. If the callee doesn't view incoming calls that way he is free to omit the feature. To the caller who wants to get through, there are various potential solutions. One is to simply repeatedly reattempt the call. This can be done manually by the UAC user, but it can also be implemented in the UAC itself. Why isn't *that* a sufficient solution? Several reasons: 1) it increases signaling load on the network 2) there is no "fairness" - no guarantee that the first to try is the first to get through, or even that the caller the callee would most like to talk to gets through first. 3) the calling user has to stand by awaiting success. There is no opportunity to do other things and be alerted when the call can get through. Who benefits varies for these: 1) the network carriers benefit. This may be incentive enough. 2) using repeat dialing, the UA that can generate call attempts most frequently has the highest probablility of getting through quickly. The has the potential to aggrevate (1). (A clever UAC may even initiate additional INVITEs before the earlier one has had a response.) Its not clear if fairness for callers is of benefit to the callee or not - perhaps if some callers get mad for having had to wait a long time. To be effective, the queuing must prevent people from getting through without entering the queue so long as a queue exists. But that locks out callers that don't have support for the queuing mechanism. That makes rollout of the queuing support problematic. Additional benefit can accrue to the callee if he is given the opportunity to cherry pick the queue - choosing how it is ordered. 3) this clearly is benefit to the caller. The benefit to the callee is a happier caller. For instance, this might make long call wait times be more palatable. IMO the "caller" call completion service and the "callee" call completion service are logically separate. As a caller, I just want a way to complete my calls, and I want it regardless of what capabilities the callee does or doesn't have. So the UAC should have a bag of tricks that it can use to accomplish call completion. Repeat dialing and the ccna/ccbs queing are two tricks in the bag. There may be other possibilities. (E.g. probing with OPTIONS checking for busy.) The "callee" call completion service is probably the callee half of the queuing mechanism, offered as an option to callers that support it. It is there if and when it suits the callee to offer it. > Also related, I think we need to consider the security implications of > this. So, if a malicious caller calls a busy user many times, they can > cause state to build up in the callee's agent. Indeed I think an > implementation would almost be required to construct the call-info URI > in a way that allowed it to contain all of the needed state, moving the > burden to the caller. This isn't discussed at all, afaict, and its a > critical design issue. I seem to recall that this was discussed previously. Its my understanding that the server managing the queue is free to prune the queue as desired. For instance, it will almost certainly only keep a single entry from any particular caller. And it can just stop queuing when the queue gets as big as it desires to handle. > On the HERFP problem, I think the issue is really deeper than that. The > issue is really targeting/re-forwarding. The question is, if A calls B > and this is retargeted to C, does it make sense to call complete on B or > on C? C kind of makes sense; but what if C is busy for a while and > during that time, the forwarding rules change and calls to B no longer > forward to C. Now, if A called B back they would reach B (who was the > person they REALLY wanted to call anyway), but a call completion request > would in fact go to C. This seems wrong to me. One might even argue that > call completion and forwarding are incompatible. Frankly, if our initial > solution basically didn't support it (no Call-Info passed back at all > when a call is forwarded), I think that is arguably a feature. I suspect > experts on feature interaction have looked at this one; would be curious > on the best practices around this in the PSTN. Some discretion on the part of B and C could make this work better: If the forwarding is due to NA or BS, then it can be treated as forking, so that the CC requests go both places. Then whichever gets it first wins. Conversely, C can notice that the call was forwarded from B, and can make a policy decision not to support queuing of calls forwarded from B. > I also think its a mistake to require 199 or 130 or any other 'HERFP' > response - again, making this too complicated. Simplicity is key for > this feature IMHO. > > I have a practical worry on using the Call-ID as a correlator for the > subscription. The reality of deployments are many B2BUAs exist, and this > is no longer a reliable e2e correlator for things. Anyway its not > needed; the URI should be sufficient (and unlike call-id, is reliable). > Also, I would NOT assume that SUBSCRIBE and INVITE to the same URI are > routed identically; this is not even true in theory as proxies are > allowed to do method-based routing. Indeed, typically SUBSCRIBE are > routed to appropriate servers based on event packages. I think one must trust that proxies are configured to route SUBSCRIBE for *this* event package in a way that is conducive to its intended use. > I think the callback INVITE needs to be to a different SIP URI. I > suspect it will need different treatment on both sides; i.e., that call > should not go to voicemail (I suspect). Using the philosophy embraced in: > http://www.ietf.org/internet-drafts/draft-ietf-sipping-service-identification-02.txt > > > a service URI is absolutely the right thing to differentiate here. > > I do not understand the usage or need for the 'm' and 'monitor' > attributes. Monitor shows up as a URI param, in fact. Not at all sure > why that is needed. > > What does a calling domain do when it wants to invoke this service and > the terminating side doesn't support it. Do we have a recommended 'poor > mans' version of this that requires no additional support? i.e., I am > aware of implementations that actually periodically send INVITEs. > Indeed, doing so may be incentive to properly deploy a sub/not solution > to avoid such deluge... I discussed this above, ad nauseum. :-) Paul > Thanks, > Jonathan R. _______________________________________________ BLISS mailing list BLISS@ietf.org https://www.ietf.org/mailman/listinfo/bliss From bliss-bounces@ietf.org Tue Jul 29 02:26:17 2008 Return-Path: X-Original-To: bliss-archive@optimus.ietf.org Delivered-To: ietfarch-bliss-archive@core3.amsl.com Received: from [127.0.0.1] (localhost [127.0.0.1]) by core3.amsl.com (Postfix) with ESMTP id D2A753A69C4; Tue, 29 Jul 2008 02:26:17 -0700 (PDT) X-Original-To: bliss@core3.amsl.com Delivered-To: bliss@core3.amsl.com Received: from localhost (localhost [127.0.0.1]) by core3.amsl.com (Postfix) with ESMTP id 171E23A69C4 for ; Tue, 29 Jul 2008 02:26:17 -0700 (PDT) X-Virus-Scanned: amavisd-new at amsl.com X-Spam-Flag: NO X-Spam-Score: -2.599 X-Spam-Level: X-Spam-Status: No, score=-2.599 tagged_above=-999 required=5 tests=[BAYES_00=-2.599] Received: from mail.ietf.org ([64.170.98.32]) by localhost (core3.amsl.com [127.0.0.1]) (amavisd-new, port 10024) with ESMTP id Y5IIKblMUWf3 for ; Tue, 29 Jul 2008 02:26:15 -0700 (PDT) Received: from mailgate.siemenscomms.co.uk (mailgate.siemenscomms.co.uk [195.171.110.225]) by core3.amsl.com (Postfix) with ESMTP id A01A03A68E4 for ; Tue, 29 Jul 2008 02:26:15 -0700 (PDT) Received: from GBNTHT12009MSX.gb002.siemens.net ([137.223.219.235]) by siemenscomms.co.uk (PMDF V6.3-x14 #31430) with ESMTP id <0K4R0088XGVUQK@siemenscomms.co.uk> for bliss@ietf.org; Tue, 29 Jul 2008 10:26:24 +0100 (BST) Date: Tue, 29 Jul 2008 10:26:16 +0100 From: "Elwell, John" In-reply-to: <488DCD8A.2040202@cisco.com> To: Jonathan Rosenberg , bliss@ietf.org Message-id: <0D5F89FAC29E2C41B98A6A762007F5D0EF2F9C@GBNTHT12009MSX.gb002.siemens.net> MIME-version: 1.0 X-MIMEOLE: Produced By Microsoft Exchange V6.5 Thread-Topic: [BLISS] thoughts on draft-ietf-bliss-call-completion Thread-Index: AcjwuE2mA5Li3xbTSPeUhsV31GbukwAo1tvQ Content-class: urn:content-classes:message X-MS-Has-Attach: X-MS-TNEF-Correlator: References: <488DCD8A.2040202@cisco.com> Subject: Re: [BLISS] thoughts on draft-ietf-bliss-call-completion X-BeenThere: bliss@ietf.org X-Mailman-Version: 2.1.9 Precedence: list List-Id: "Basic Level of Interoperability for SIP Services \(BLISS\) BoF" List-Unsubscribe: , List-Archive: List-Post: List-Help: List-Subscribe: , Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: 7bit Sender: bliss-bounces@ietf.org Errors-To: bliss-bounces@ietf.org > -----Original Message----- > From: bliss-bounces@ietf.org [mailto:bliss-bounces@ietf.org] > On Behalf Of Jonathan Rosenberg > Sent: 28 July 2008 14:46 > To: bliss@ietf.org > Subject: [BLISS] thoughts on draft-ietf-bliss-call-completion > > A key part of the mechanism in here is that it focuses on the > 'server to > server' aspect of the problem. It defines only what would > need to flow > between domains or between organizations to accomplish this feature > between organizations. I think that aspect of it needs to be > emphasized. > For example, some diagrams which show this, including > different models > for how the agents can be co-located. > > Very much related to that, ccbs has an effect of introducing > work in one > domain (the one of the callee) that primarily (if not exclusively) > benefits the other (the one of the caller). This provides an > unfortunate > negative incentive for implementing this in a callee > domain. I think > we need to take care to minimize the amount of work required on the > callee's side to support this. So for example, I am reluctant to > introduce requirements to do things like suspend/resume, as these > introduce additional work and state on behalf of the callee's agent. > > Also related, I think we need to consider the security > implications of > this. So, if a malicious caller calls a busy user many times, > they can > cause state to build up in the callee's agent. Indeed I think an > implementation would almost be required to construct the > call-info URI > in a way that allowed it to contain all of the needed state, > moving the > burden to the caller. This isn't discussed at all, afaict, and its a > critical design issue. I seem to recall that this was > discussed previously. > > On the HERFP problem, I think the issue is really deeper than > that. The > issue is really targeting/re-forwarding. The question is, if > A calls B > and this is retargeted to C, does it make sense to call > complete on B or > on C? C kind of makes sense; but what if C is busy for a while and > during that time, the forwarding rules change and calls to B > no longer > forward to C. Now, if A called B back they would reach B (who was the > person they REALLY wanted to call anyway), but a call > completion request > would in fact go to C. This seems wrong to me. One might even > argue that > call completion and forwarding are incompatible. Frankly, if > our initial > solution basically didn't support it (no Call-Info passed back at all > when a call is forwarded), I think that is arguably a > feature. I suspect > experts on feature interaction have looked at this one; would > be curious > on the best practices around this in the PSTN. [JRE] In PSTN (or at least enterprise equivalents of PSTN) one practice is to make it depend on the conditions for call forwarding. If it is call forwarding because of busy or no reply, it makes sense for CCBS/CCNR to operate on B, rather than C, because there is just as much chance of B becoming available as C becoming available. And after all, B is really the person A wants to communicate with. For unconditional call forwarding (where B is out of the office for a day, say), it might make sense for CCBS/CCNR to operate on C. However, I don't think we want to impose rigid rules of this nature on our SIP solution. In some respects behaviour should be based on the policy of B, but I am not sure how to realise that. > > I also think its a mistake to require 199 or 130 or any other 'HERFP' > response - again, making this too complicated. Simplicity is key for > this feature IMHO. > > I have a practical worry on using the Call-ID as a correlator for the > subscription. The reality of deployments are many B2BUAs > exist, and this > is no longer a reliable e2e correlator for things. Anyway its not > needed; the URI should be sufficient (and unlike call-id, is > reliable). > Also, I would NOT assume that SUBSCRIBE and INVITE to the > same URI are > routed identically; this is not even true in theory as proxies are > allowed to do method-based routing. Indeed, typically SUBSCRIBE are > routed to appropriate servers based on event packages. [JRE] I agree. > > I think the callback INVITE needs to be to a different SIP URI. I > suspect it will need different treatment on both sides; i.e., > that call > should not go to voicemail (I suspect). Using the philosophy > embraced in: > http://www.ietf.org/internet-drafts/draft-ietf-sipping-service > -identification-02.txt > > a service URI is absolutely the right thing to differentiate here. [JRE] Yes, I have proposed in the past (on the BLISS-CC list, I think) that the NOTIFY from the monitor should provide the URI for the callback INVITE. I think people had concerns that this would be a problem for PSTN interworking, however. John > > I do not understand the usage or need for the 'm' and 'monitor' > attributes. Monitor shows up as a URI param, in fact. Not at all sure > why that is needed. > > What does a calling domain do when it wants to invoke this > service and > the terminating side doesn't support it. Do we have a > recommended 'poor > mans' version of this that requires no additional support? i.e., I am > aware of implementations that actually periodically send INVITEs. > Indeed, doing so may be incentive to properly deploy a > sub/not solution > to avoid such deluge... > > Thanks, > Jonathan R. > -- > Jonathan D. Rosenberg, Ph.D. 499 Thornall St. > Cisco Fellow Edison, NJ 08837 > Cisco, Voice Technology Group > jdrosen@cisco.com > http://www.jdrosen.net PHONE: (408) 902-3084 > http://www.cisco.com > _______________________________________________ > BLISS mailing list > BLISS@ietf.org > https://www.ietf.org/mailman/listinfo/bliss > _______________________________________________ BLISS mailing list BLISS@ietf.org https://www.ietf.org/mailman/listinfo/bliss From bliss-bounces@ietf.org Tue Jul 29 02:33:54 2008 Return-Path: X-Original-To: bliss-archive@optimus.ietf.org Delivered-To: ietfarch-bliss-archive@core3.amsl.com Received: from [127.0.0.1] (localhost [127.0.0.1]) by core3.amsl.com (Postfix) with ESMTP id 48FAF28C1F7; Tue, 29 Jul 2008 02:33:54 -0700 (PDT) X-Original-To: bliss@core3.amsl.com Delivered-To: bliss@core3.amsl.com Received: from localhost (localhost [127.0.0.1]) by core3.amsl.com (Postfix) with ESMTP id 9A91228C1E5 for ; Tue, 29 Jul 2008 02:33:52 -0700 (PDT) X-Virus-Scanned: amavisd-new at amsl.com X-Spam-Flag: NO X-Spam-Score: -2.599 X-Spam-Level: X-Spam-Status: No, score=-2.599 tagged_above=-999 required=5 tests=[BAYES_00=-2.599] Received: from mail.ietf.org ([64.170.98.32]) by localhost (core3.amsl.com [127.0.0.1]) (amavisd-new, port 10024) with ESMTP id YGXPLUi7Y1Cc for ; Tue, 29 Jul 2008 02:33:51 -0700 (PDT) Received: from mailgate.siemenscomms.co.uk (mailgate.siemenscomms.co.uk [195.171.110.225]) by core3.amsl.com (Postfix) with ESMTP id 5467C28C12F for ; Tue, 29 Jul 2008 02:33:51 -0700 (PDT) Received: from GBNTHT12009MSX.gb002.siemens.net ([137.223.219.235]) by siemenscomms.co.uk (PMDF V6.3-x14 #31430) with ESMTP id <0K4R009ASH8NC0@siemenscomms.co.uk> for bliss@ietf.org; Tue, 29 Jul 2008 10:34:01 +0100 (BST) Date: Tue, 29 Jul 2008 10:33:47 +0100 From: "Elwell, John" In-reply-to: <488E073E.9040904@cisco.com> To: Paul Kyzivat , Jonathan Rosenberg Message-id: <0D5F89FAC29E2C41B98A6A762007F5D0EF2FAD@GBNTHT12009MSX.gb002.siemens.net> MIME-version: 1.0 X-MIMEOLE: Produced By Microsoft Exchange V6.5 Thread-Topic: [BLISS] thoughts on draft-ietf-bliss-call-completion Thread-Index: Acjw2oyYNm1e9e6YT0alNwoVmh4scgAgvktw Content-class: urn:content-classes:message X-MS-Has-Attach: X-MS-TNEF-Correlator: References: <488DCD8A.2040202@cisco.com> <488E073E.9040904@cisco.com> Cc: bliss@ietf.org Subject: Re: [BLISS] thoughts on draft-ietf-bliss-call-completion X-BeenThere: bliss@ietf.org X-Mailman-Version: 2.1.9 Precedence: list List-Id: "Basic Level of Interoperability for SIP Services \(BLISS\) BoF" List-Unsubscribe: , List-Archive: List-Post: List-Help: List-Subscribe: , Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: 7bit Sender: bliss-bounces@ietf.org Errors-To: bliss-bounces@ietf.org > -----Original Message----- > From: bliss-bounces@ietf.org [mailto:bliss-bounces@ietf.org] > On Behalf Of Paul Kyzivat > Sent: 28 July 2008 18:52 > To: Jonathan Rosenberg > Cc: bliss@ietf.org > Subject: Re: [BLISS] thoughts on draft-ietf-bliss-call-completion > > Lots of comments below. > > Jonathan Rosenberg wrote: > > A key part of the mechanism in here is that it focuses on > the 'server to > > server' aspect of the problem. It defines only what would > need to flow > > between domains or between organizations to accomplish this feature > > between organizations. I think that aspect of it needs to > be emphasized. > > For example, some diagrams which show this, including > different models > > for how the agents can be co-located. > > > > Very much related to that, ccbs has an effect of > introducing work in one > > domain (the one of the callee) that primarily (if not exclusively) > > benefits the other (the one of the caller). This provides > an unfortunate > > negative incentive for implementing this in a callee > domain. I think we > > need to take care to minimize the amount of work required on the > > callee's side to support this. So for example, I am reluctant to > > introduce requirements to do things like suspend/resume, as these > > introduce additional work and state on behalf of the callee's agent. > > While I agree that it is important to consider what the > incentives are > for implementing this stuff, I don't think this should be viewed as > primarily benefiting the caller at the expense of the callee. > At least > not if this is considered an *optional* feature at the callee. > > A callee will offer this capability if he considers it advantageous - > namely if he considers the incoming calls to his benefit, and missing > them to be a loss. If the callee doesn't view incoming calls > that way he > is free to omit the feature. [JRE] Or he may use some other capability, like call queueing / call waiting. > > To the caller who wants to get through, there are various potential > solutions. One is to simply repeatedly reattempt the call. > This can be > done manually by the UAC user, but it can also be implemented > in the UAC > itself. > > Why isn't *that* a sufficient solution? Several reasons: > 1) it increases signaling load on the network > 2) there is no "fairness" - no guarantee that the first to try > is the first to get through, or even that the caller the > callee would most like to talk to gets through first. > 3) the calling user has to stand by awaiting success. There > is no opportunity to do other things and be alerted when > the call can get through. > > Who benefits varies for these: > 1) the network carriers benefit. This may be incentive enough. > 2) using repeat dialing, the UA that can generate call attempts most > frequently has the highest probablility of getting > through quickly. > The has the potential to aggrevate (1). (A clever UAC may even > initiate additional INVITEs before the earlier one has had a > response.) Its not clear if fairness for callers is of benefit > to the callee or not - perhaps if some callers get mad for having > had to wait a long time. To be effective, the queuing must > prevent people from getting through without entering the queue > so long as a queue exists. But that locks out callers that don't > have support for the queuing mechanism. That makes rollout of > the queuing support problematic. > Additional benefit can accrue to the callee if he is given the > opportunity to cherry pick the queue - choosing how it is > ordered. > 3) this clearly is benefit to the caller. The benefit to the callee > is a happier caller. For instance, this might make long call wait > times be more palatable. > > IMO the "caller" call completion service and the "callee" call > completion service are logically separate. As a caller, I just want a > way to complete my calls, and I want it regardless of what > capabilities > the callee does or doesn't have. So the UAC should have a bag > of tricks > that it can use to accomplish call completion. Repeat dialing and the > ccna/ccbs queing are two tricks in the bag. There may be other > possibilities. (E.g. probing with OPTIONS checking for busy.) > > The "callee" call completion service is probably the callee > half of the > queuing mechanism, offered as an option to callers that > support it. It > is there if and when it suits the callee to offer it. [JRE] For PSTN use, the callee generally does not have a say - it is up to the network provider whether it operates CCBS on callee's behalf or not. John _______________________________________________ BLISS mailing list BLISS@ietf.org https://www.ietf.org/mailman/listinfo/bliss From bliss-bounces@ietf.org Tue Jul 29 05:38:45 2008 Return-Path: X-Original-To: bliss-archive@optimus.ietf.org Delivered-To: ietfarch-bliss-archive@core3.amsl.com Received: from [127.0.0.1] (localhost [127.0.0.1]) by core3.amsl.com (Postfix) with ESMTP id 18B2828C231; Tue, 29 Jul 2008 05:38:45 -0700 (PDT) X-Original-To: bliss@core3.amsl.com Delivered-To: bliss@core3.amsl.com Received: from localhost (localhost [127.0.0.1]) by core3.amsl.com (Postfix) with ESMTP id F2FA028C231 for ; Tue, 29 Jul 2008 05:38:43 -0700 (PDT) X-Virus-Scanned: amavisd-new at amsl.com X-Spam-Flag: NO X-Spam-Score: -3.249 X-Spam-Level: X-Spam-Status: No, score=-3.249 tagged_above=-999 required=5 tests=[BAYES_00=-2.599, HELO_EQ_DE=0.35, RCVD_IN_DNSWL_LOW=-1] Received: from mail.ietf.org ([64.170.98.32]) by localhost (core3.amsl.com [127.0.0.1]) (amavisd-new, port 10024) with ESMTP id 5bsldL3YqBbj for ; Tue, 29 Jul 2008 05:38:40 -0700 (PDT) Received: from tcmail12.telekom.de (tcmail12.telekom.de [217.5.214.82]) by core3.amsl.com (Postfix) with ESMTP id 867DA28C1F7 for ; Tue, 29 Jul 2008 05:38:39 -0700 (PDT) Received: from S4DE9JSAANO.ost.t-com.de (S4DE9JSAANO.ost.t-com.de [10.125.177.105]) by tcmail11.telekom.de with ESMTP; Tue, 29 Jul 2008 14:38:45 +0200 Received: from S4DE9JSAAMW.ost.t-com.de ([10.125.177.114]) by S4DE9JSAANO.ost.t-com.de with Microsoft SMTPSVC(6.0.3790.3959); Tue, 29 Jul 2008 14:38:44 +0200 Content-class: urn:content-classes:message MIME-Version: 1.0 X-MimeOLE: Produced By Microsoft Exchange V6.5 Date: Tue, 29 Jul 2008 14:38:37 +0200 Message-Id: In-Reply-To: <488DCD8A.2040202@cisco.com> X-MS-Has-Attach: X-MS-TNEF-Correlator: Thread-Topic: [BLISS] thoughts on draft-ietf-bliss-call-completion Thread-Index: AcjwuE0gGnQ16UlpSYaUTCvLEBmC5AApR+Rg References: <488DCD8A.2040202@cisco.com> From: "Alexeitsev, D" To: X-OriginalArrivalTime: 29 Jul 2008 12:38:44.0281 (UTC) FILETIME=[09CC4290:01C8F178] Cc: bliss@ietf.org Subject: Re: [BLISS] thoughts on draft-ietf-bliss-call-completion X-BeenThere: bliss@ietf.org X-Mailman-Version: 2.1.9 Precedence: list List-Id: "Basic Level of Interoperability for SIP Services \(BLISS\) BoF" List-Unsubscribe: , List-Archive: List-Post: List-Help: List-Subscribe: , Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: 7bit Sender: bliss-bounces@ietf.org Errors-To: bliss-bounces@ietf.org Comments on some of the points inline: Also related, I think we need to consider the security implications of this. So, if a malicious caller calls a busy user many times, they can cause state to build up in the callee's agent. Indeed I think an implementation would almost be required to construct the call-info URI in a way that allowed it to contain all of the needed state, moving the burden to the caller. This isn't discussed at all, afaict, and its a critical design issue. I seem to recall that this was discussed previously. [DA] It was disussed previously and one option was to create the queue state not at the time of busy recognition for the initial INVITE but at the suscription time. In this case the caller would be interested to SUBSCRIBE as fast as possible that there will be as less possible delta between busy and queue creation time. Subscription also consumes resources on the caller side and the resource consumption is not that unballanced for caller and callee. On the HERFP problem, I think the issue is really deeper than that. The issue is really targeting/re-forwarding. The question is, if A calls B and this is retargeted to C, does it make sense to call complete on B or on C? C kind of makes sense; but what if C is busy for a while and during that time, the forwarding rules change and calls to B no longer forward to C. Now, if A called B back they would reach B (who was the person they REALLY wanted to call anyway), but a call completion request would in fact go to C. This seems wrong to me. One might even argue that call completion and forwarding are incompatible. Frankly, if our initial solution basically didn't support it (no Call-Info passed back at all when a call is forwarded), I think that is arguably a feature. I suspect experts on feature interaction have looked at this one; would be curious on the best practices around this in the PSTN. [DA] This is the question, that has no ultimative unswer. One view is that it is easier to control the correlation if the call completion wit call forwarding is done at the B side. And as you said B is the one A really wanted. Things get even more complicated bacause of the different types of call forwarding. Here is one of the PSTN BCPs: For CFU (B): already established bevor the CCxx request: If present it means that B is not available anyway, then independent from the C CCxx indication, no CCxx is presented from B to A established after the CCxx request: CCxx monitoring continues, but change of the state of the B-party is not signalled to A. If CFU is taken off while the CCxx contol timer did not run out, then recall possible is indicated to A, otherwise termination of the queue entry. For CFNR (B): already established bevor the CCNR request: If C provides any of the CCxx indications, than the CCNR is indicated to A from B If C provides no CCxx indications, no CCxx is indicated to A from B established after the CCNR request: CCNR monitoring continues, but change of the state of the B-party is not signalled to A. If CFNR is taken off while the CCNR contol timer did not run out, then recall possible is indicated to A, otherwise termination of the queue entry. For CFB (B): already established bevor the CCBS request: If C provides any of the CCxx indications, than the CCBS is indicated to A from B If C provides no CCxx indications, no CCxx is indicated to A from B established after the CCBS request: CCBS monitoring continues, but change of the state of the B-party is not signalled to A. If CFB is taken off while the CCBS contol timer did not run out, then recall possible is indicated to A, otherwise termination of the queue entry. No CCxx (B): In there is no CCxx service for the B available, then independent from the B's CFx and C's CCxx indication, no CCxx is presented from B to A _______________________________________________ BLISS mailing list BLISS@ietf.org https://www.ietf.org/mailman/listinfo/bliss From bliss-bounces@ietf.org Tue Jul 29 06:26:13 2008 Return-Path: X-Original-To: bliss-archive@optimus.ietf.org Delivered-To: ietfarch-bliss-archive@core3.amsl.com Received: from [127.0.0.1] (localhost [127.0.0.1]) by core3.amsl.com (Postfix) with ESMTP id 6A39E3A6B6B; Tue, 29 Jul 2008 06:26:13 -0700 (PDT) X-Original-To: bliss@core3.amsl.com Delivered-To: bliss@core3.amsl.com Received: from localhost (localhost [127.0.0.1]) by core3.amsl.com (Postfix) with ESMTP id 163823A6B61 for ; Tue, 29 Jul 2008 06:26:12 -0700 (PDT) X-Virus-Scanned: amavisd-new at amsl.com X-Spam-Flag: NO X-Spam-Score: -6.274 X-Spam-Level: X-Spam-Status: No, score=-6.274 tagged_above=-999 required=5 tests=[AWL=0.325, BAYES_00=-2.599, RCVD_IN_DNSWL_MED=-4] Received: from mail.ietf.org ([64.170.98.32]) by localhost (core3.amsl.com [127.0.0.1]) (amavisd-new, port 10024) with ESMTP id dex-7VliAtLX for ; Tue, 29 Jul 2008 06:26:06 -0700 (PDT) Received: from rtp-iport-2.cisco.com (rtp-iport-2.cisco.com [64.102.122.149]) by core3.amsl.com (Postfix) with ESMTP id 219D63A69F9 for ; Tue, 29 Jul 2008 06:26:06 -0700 (PDT) X-IronPort-AV: E=Sophos;i="4.31,272,1215388800"; d="scan'208";a="15852528" Received: from rtp-dkim-2.cisco.com ([64.102.121.159]) by rtp-iport-2.cisco.com with ESMTP; 29 Jul 2008 13:26:17 +0000 Received: from rtp-core-1.cisco.com (rtp-core-1.cisco.com [64.102.124.12]) by rtp-dkim-2.cisco.com (8.12.11/8.12.11) with ESMTP id m6TDQH86002283; Tue, 29 Jul 2008 09:26:17 -0400 Received: from xbh-rtp-211.amer.cisco.com (xbh-rtp-211.cisco.com [64.102.31.102]) by rtp-core-1.cisco.com (8.13.8/8.13.8) with ESMTP id m6TDQHGO023810; Tue, 29 Jul 2008 13:26:17 GMT 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, 29 Jul 2008 09:26:17 -0400 Received: from [161.44.174.168] ([161.44.174.168]) by xfe-rtp-201.amer.cisco.com with Microsoft SMTPSVC(6.0.3790.1830); Tue, 29 Jul 2008 09:26:17 -0400 Message-ID: <488F1AAF.3010307@cisco.com> Date: Tue, 29 Jul 2008 09:27:11 -0400 From: Paul Kyzivat User-Agent: Thunderbird 2.0.0.16 (Windows/20080708) MIME-Version: 1.0 To: "Elwell, John" References: <488DCD8A.2040202@cisco.com> <488E073E.9040904@cisco.com> <0D5F89FAC29E2C41B98A6A762007F5D0EF2FAD@GBNTHT12009MSX.gb002.siemens.net> In-Reply-To: <0D5F89FAC29E2C41B98A6A762007F5D0EF2FAD@GBNTHT12009MSX.gb002.siemens.net> X-OriginalArrivalTime: 29 Jul 2008 13:26:17.0354 (UTC) FILETIME=[AE5CA6A0:01C8F17E] DKIM-Signature: v=1; a=rsa-sha256; q=dns/txt; l=5523; t=1217337977; x=1218201977; c=relaxed/simple; s=rtpdkim2001; h=Content-Type:From:Subject:Content-Transfer-Encoding:MIME-Version; d=cisco.com; i=pkyzivat@cisco.com; z=From:=20Paul=20Kyzivat=20 |Subject:=20Re=3A=20[BLISS]=20thoughts=20on=20draft-ietf-bl iss-call-completion |Sender:=20 |To:=20=22Elwell,=20John=22=20; bh=QqnDklrmwgMrZmLc3vyzHT2gMgAD02zwn9bEgS2QHFQ=; b=o5gW0/rkuYBMO0j2dEi9TEanHtACxm+8U5TromIWDURxBc6wu40O8Gu6Uh gxzB8Tk1pHyocpPJVjh3RZdyDBhW5FliOobiLsxVBUtSittHfgRUM2dj1JbU aN5KbQds7c; Authentication-Results: rtp-dkim-2; header.From=pkyzivat@cisco.com; dkim=pass ( sig from cisco.com/rtpdkim2001 verified; ); Cc: bliss@ietf.org Subject: Re: [BLISS] thoughts on draft-ietf-bliss-call-completion X-BeenThere: bliss@ietf.org X-Mailman-Version: 2.1.9 Precedence: list List-Id: "Basic Level of Interoperability for SIP Services \(BLISS\) BoF" List-Unsubscribe: , List-Archive: List-Post: List-Help: List-Subscribe: , Content-Transfer-Encoding: 7bit Content-Type: text/plain; charset="us-ascii"; Format="flowed" Sender: bliss-bounces@ietf.org Errors-To: bliss-bounces@ietf.org Elwell, John wrote: > > >> -----Original Message----- >> From: bliss-bounces@ietf.org [mailto:bliss-bounces@ietf.org] >> On Behalf Of Paul Kyzivat >> Sent: 28 July 2008 18:52 >> To: Jonathan Rosenberg >> Cc: bliss@ietf.org >> Subject: Re: [BLISS] thoughts on draft-ietf-bliss-call-completion >> >> Lots of comments below. >> >> Jonathan Rosenberg wrote: >>> A key part of the mechanism in here is that it focuses on >> the 'server to >>> server' aspect of the problem. It defines only what would >> need to flow >>> between domains or between organizations to accomplish this feature >>> between organizations. I think that aspect of it needs to >> be emphasized. >>> For example, some diagrams which show this, including >> different models >>> for how the agents can be co-located. >>> >>> Very much related to that, ccbs has an effect of >> introducing work in one >>> domain (the one of the callee) that primarily (if not exclusively) >>> benefits the other (the one of the caller). This provides >> an unfortunate >>> negative incentive for implementing this in a callee >> domain. I think we >>> need to take care to minimize the amount of work required on the >>> callee's side to support this. So for example, I am reluctant to >>> introduce requirements to do things like suspend/resume, as these >>> introduce additional work and state on behalf of the callee's agent. >> While I agree that it is important to consider what the >> incentives are >> for implementing this stuff, I don't think this should be viewed as >> primarily benefiting the caller at the expense of the callee. >> At least >> not if this is considered an *optional* feature at the callee. >> >> A callee will offer this capability if he considers it advantageous - >> namely if he considers the incoming calls to his benefit, and missing >> them to be a loss. If the callee doesn't view incoming calls >> that way he >> is free to omit the feature. > [JRE] Or he may use some other capability, like call queueing / call > waiting. Sure. And these things aren't necessarily mutually exclusive, though they may be in the pstn. (Why can't I have call waiting and cc too? This would however require more explicit control at the receiving end - perhaps visibility to the queue and the option to take another call from the queue even though I am on a call already.) >> To the caller who wants to get through, there are various potential >> solutions. One is to simply repeatedly reattempt the call. >> This can be >> done manually by the UAC user, but it can also be implemented >> in the UAC >> itself. >> >> Why isn't *that* a sufficient solution? Several reasons: >> 1) it increases signaling load on the network >> 2) there is no "fairness" - no guarantee that the first to try >> is the first to get through, or even that the caller the >> callee would most like to talk to gets through first. >> 3) the calling user has to stand by awaiting success. There >> is no opportunity to do other things and be alerted when >> the call can get through. >> >> Who benefits varies for these: >> 1) the network carriers benefit. This may be incentive enough. >> 2) using repeat dialing, the UA that can generate call attempts most >> frequently has the highest probablility of getting >> through quickly. >> The has the potential to aggrevate (1). (A clever UAC may even >> initiate additional INVITEs before the earlier one has had a >> response.) Its not clear if fairness for callers is of benefit >> to the callee or not - perhaps if some callers get mad for having >> had to wait a long time. To be effective, the queuing must >> prevent people from getting through without entering the queue >> so long as a queue exists. But that locks out callers that don't >> have support for the queuing mechanism. That makes rollout of >> the queuing support problematic. >> Additional benefit can accrue to the callee if he is given the >> opportunity to cherry pick the queue - choosing how it is >> ordered. >> 3) this clearly is benefit to the caller. The benefit to the callee >> is a happier caller. For instance, this might make long call wait >> times be more palatable. >> >> IMO the "caller" call completion service and the "callee" call >> completion service are logically separate. As a caller, I just want a >> way to complete my calls, and I want it regardless of what >> capabilities >> the callee does or doesn't have. So the UAC should have a bag >> of tricks >> that it can use to accomplish call completion. Repeat dialing and the >> ccna/ccbs queing are two tricks in the bag. There may be other >> possibilities. (E.g. probing with OPTIONS checking for busy.) >> >> The "callee" call completion service is probably the callee >> half of the >> queuing mechanism, offered as an option to callers that >> support it. It >> is there if and when it suits the callee to offer it. > [JRE] For PSTN use, the callee generally does not have a say - it is up > to the network provider whether it operates CCBS on callee's behalf or > not. In this case we at least pretend that the SP is carrying out the callee's policy. That policy is decided when the callee enrolls for service. Any perhaps there isn't a lot of choice. Yet it was the callee's policy decision to enroll. :-) Paul _______________________________________________ BLISS mailing list BLISS@ietf.org https://www.ietf.org/mailman/listinfo/bliss From bliss-bounces@ietf.org Thu Jul 31 14:06:02 2008 Return-Path: X-Original-To: bliss-archive@optimus.ietf.org Delivered-To: ietfarch-bliss-archive@core3.amsl.com Received: from [127.0.0.1] (localhost [127.0.0.1]) by core3.amsl.com (Postfix) with ESMTP id 325CC3A6CD9; Thu, 31 Jul 2008 14:06:02 -0700 (PDT) X-Original-To: bliss@core3.amsl.com Delivered-To: bliss@core3.amsl.com Received: from localhost (localhost [127.0.0.1]) by core3.amsl.com (Postfix) with ESMTP id 9F02B3A6CD9 for ; Thu, 31 Jul 2008 14:06:01 -0700 (PDT) X-Virus-Scanned: amavisd-new at amsl.com X-Spam-Flag: NO X-Spam-Score: -100.495 X-Spam-Level: X-Spam-Status: No, score=-100.495 tagged_above=-999 required=5 tests=[BAYES_00=-2.599, FH_RELAY_NODNS=1.451, HELO_MISMATCH_COM=0.553, RDNS_NONE=0.1, USER_IN_WHITELIST=-100] Received: from mail.ietf.org ([64.170.98.32]) by localhost (core3.amsl.com [127.0.0.1]) (amavisd-new, port 10024) with ESMTP id v2KqYOWsdpZv for ; Thu, 31 Jul 2008 14:06:00 -0700 (PDT) Received: from mail.amsl.com (mail.amsl.com [IPv6:2001:1890:1112:1::14]) by core3.amsl.com (Postfix) with ESMTP id D05D63A6A05 for ; Thu, 31 Jul 2008 14:06:00 -0700 (PDT) Received: from localhost (localhost [127.0.0.1]) by thunder2.amsl.com (Postfix) with ESMTP id 96D947FC6 for ; Thu, 31 Jul 2008 14:06:16 -0700 (PDT) X-Virus-Scanned: amavisd-new at amsl.com Received: from mail.amsl.com ([64.170.98.20]) by localhost (thunder2.amsl.com [127.0.0.1]) (amavisd-new, port 10024) with ESMTP id raGllLOpB9WI for ; Thu, 31 Jul 2008 14:06:16 -0700 (PDT) Received: from [64.170.98.103] (dv2500r1.c5i.net [64.170.98.103]) by thunder2.amsl.com (Postfix) with ESMTP id 5D99E7FC5 for ; Thu, 31 Jul 2008 14:06:16 -0700 (PDT) Message-ID: <4892294C.5040009@amsl.com> Date: Thu, 31 Jul 2008 14:06:20 -0700 From: Glen User-Agent: Thunderbird 2.0.0.16 (Windows/20080708) MIME-Version: 1.0 To: bliss@ietf.org X-Enigmail-Version: 0.95.6 Subject: [BLISS] Test, please ignore X-BeenThere: bliss@ietf.org X-Mailman-Version: 2.1.9 Precedence: list List-Id: "Basic Level of Interoperability for SIP Services \(BLISS\) BoF" List-Unsubscribe: , List-Archive: List-Post: List-Help: List-Subscribe: , Content-Transfer-Encoding: 7bit Content-Type: text/plain; charset="us-ascii"; Format="flowed" Sender: bliss-bounces@ietf.org Errors-To: bliss-bounces@ietf.org Please excuse this test. We have been asked by Chairperson Shida Schubert to diagnose a problem with this list. Glen (AMS IT Director) _______________________________________________ BLISS mailing list BLISS@ietf.org https://www.ietf.org/mailman/listinfo/bliss From bliss-bounces@ietf.org Thu Jul 31 14:24:57 2008 Return-Path: X-Original-To: bliss-archive@optimus.ietf.org Delivered-To: ietfarch-bliss-archive@core3.amsl.com Received: from [127.0.0.1] (localhost [127.0.0.1]) by core3.amsl.com (Postfix) with ESMTP id DF0213A6C61; Thu, 31 Jul 2008 14:24:57 -0700 (PDT) X-Original-To: bliss@core3.amsl.com Delivered-To: bliss@core3.amsl.com Received: from localhost (localhost [127.0.0.1]) by core3.amsl.com (Postfix) with ESMTP id ECFD13A68BD for ; Thu, 31 Jul 2008 14:24:55 -0700 (PDT) X-Virus-Scanned: amavisd-new at amsl.com X-Spam-Flag: NO X-Spam-Score: -2.265 X-Spam-Level: X-Spam-Status: No, score=-2.265 tagged_above=-999 required=5 tests=[BAYES_00=-2.599, IP_NOT_FRIENDLY=0.334] Received: from mail.ietf.org ([64.170.98.32]) by localhost (core3.amsl.com [127.0.0.1]) (amavisd-new, port 10024) with ESMTP id lXxKzXtRxynF for ; Thu, 31 Jul 2008 14:24:55 -0700 (PDT) Received: from gateway07.websitewelcome.com (gateway07.websitewelcome.com [69.56.142.25]) by core3.amsl.com (Postfix) with SMTP id C78CA3A6D4A for ; Thu, 31 Jul 2008 14:24:43 -0700 (PDT) Received: (qmail 2758 invoked from network); 31 Jul 2008 21:34:18 -0000 Received: from gator465.hostgator.com (69.56.174.130) by gateway07.websitewelcome.com with SMTP; 31 Jul 2008 21:34:18 -0000 Received: from [130.129.65.54] (port=55532 helo=[172.17.5.21]) by gator465.hostgator.com with esmtpa (Exim 4.69) (envelope-from ) id 1KOfdg-00071x-8z for bliss@ietf.org; Thu, 31 Jul 2008 16:24:48 -0500 Message-Id: From: Shida Schubert To: bliss@ietf.org Mime-Version: 1.0 (Apple Message framework v926) Date: Thu, 31 Jul 2008 22:24:51 +0100 X-Mailer: Apple Mail (2.926) X-AntiAbuse: This header was added to track abuse, please include it with any abuse report X-AntiAbuse: Primary Hostname - gator465.hostgator.com X-AntiAbuse: Original Domain - ietf.org X-AntiAbuse: Originator/Caller UID/GID - [47 12] / [47 12] X-AntiAbuse: Sender Address Domain - ntt-at.com Subject: [BLISS] Testing again X-BeenThere: bliss@ietf.org X-Mailman-Version: 2.1.9 Precedence: list List-Id: "Basic Level of Interoperability for SIP Services \(BLISS\) BoF" List-Unsubscribe: , List-Archive: List-Post: List-Help: List-Subscribe: , Content-Transfer-Encoding: 7bit Content-Type: text/plain; charset="us-ascii"; Format="flowed" Sender: bliss-bounces@ietf.org Errors-To: bliss-bounces@ietf.org Dale, came up to me and told me his e-mail hasn't been going through to the mailing list. This is a 2nd test message to see if this is the case for everybody or just for selective members. Please do send me an e-mail if there is anybody else beside Dale that's experiencing the trouble. Many Thanks Shida _______________________________________________ BLISS mailing list BLISS@ietf.org https://www.ietf.org/mailman/listinfo/bliss From bliss-bounces@ietf.org Thu Jul 31 14:24:58 2008 Return-Path: X-Original-To: bliss-archive@optimus.ietf.org Delivered-To: ietfarch-bliss-archive@core3.amsl.com Received: from [127.0.0.1] (localhost [127.0.0.1]) by core3.amsl.com (Postfix) with ESMTP id 418A63A6D4A; Thu, 31 Jul 2008 14:24:58 -0700 (PDT) X-Original-To: bliss@core3.amsl.com Delivered-To: bliss@core3.amsl.com Received: from localhost (localhost [127.0.0.1]) by core3.amsl.com (Postfix) with ESMTP id 8A0DE3A6961 for ; Thu, 31 Jul 2008 14:24:56 -0700 (PDT) X-Virus-Scanned: amavisd-new at amsl.com X-Spam-Flag: NO X-Spam-Score: -2.243 X-Spam-Level: X-Spam-Status: No, score=-2.243 tagged_above=-999 required=5 tests=[AWL=-0.022, BAYES_00=-2.599, DATE_IN_PAST_03_06=0.044, IP_NOT_FRIENDLY=0.334] Received: from mail.ietf.org ([64.170.98.32]) by localhost (core3.amsl.com [127.0.0.1]) (amavisd-new, port 10024) with ESMTP id krFmADazmBBn for ; Thu, 31 Jul 2008 14:24:55 -0700 (PDT) Received: from gateway05.websitewelcome.com (gateway05.websitewelcome.com [69.93.179.12]) by core3.amsl.com (Postfix) with SMTP id EEEC93A6D4B for ; Thu, 31 Jul 2008 14:24:43 -0700 (PDT) Received: (qmail 16836 invoked from network); 31 Jul 2008 21:35:23 -0000 Received: from gator465.hostgator.com (69.56.174.130) by gateway05.websitewelcome.com with SMTP; 31 Jul 2008 21:35:23 -0000 Received: from [130.129.65.54] (port=55532 helo=[172.17.5.21]) by gator465.hostgator.com with esmtpa (Exim 4.69) (envelope-from ) id 1KOfdf-00071x-K6 for bliss@ietf.org; Thu, 31 Jul 2008 16:24:47 -0500 Message-Id: <54B19339-9489-4723-9619-74547DA5E64F@ntt-at.com> From: Shida Schubert To: bliss@ietf.org Mime-Version: 1.0 (Apple Message framework v926) Date: Thu, 31 Jul 2008 19:00:12 +0100 X-Mailer: Apple Mail (2.926) X-AntiAbuse: This header was added to track abuse, please include it with any abuse report X-AntiAbuse: Primary Hostname - gator465.hostgator.com X-AntiAbuse: Original Domain - ietf.org X-AntiAbuse: Originator/Caller UID/GID - [47 12] / [47 12] X-AntiAbuse: Sender Address Domain - ntt-at.com Subject: [BLISS] Test X-BeenThere: bliss@ietf.org X-Mailman-Version: 2.1.9 Precedence: list List-Id: "Basic Level of Interoperability for SIP Services \(BLISS\) BoF" List-Unsubscribe: , List-Archive: List-Post: List-Help: List-Subscribe: , Content-Transfer-Encoding: 7bit Content-Type: text/plain; charset="us-ascii"; Format="flowed" Sender: bliss-bounces@ietf.org Errors-To: bliss-bounces@ietf.org Dale, came up to me and told me his e-mail hasn't been going through to the mailing list. This is a test message to see if this is the case for everybody or if this only applies to Dale. Please do send me an e-mail if there is anybody else beside Dale that hasn't seen their posts going through. Many Thanks Shida _______________________________________________ BLISS mailing list BLISS@ietf.org https://www.ietf.org/mailman/listinfo/bliss