From daemon@optimus.ietf.org Sat Feb 2 14:22:57 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id OAA06476 for ; Sat, 2 Feb 2002 14:22:57 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id OAA09499 for sipping-archive@odin.ietf.org; Sat, 2 Feb 2002 14:22:59 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id NAA08482; Sat, 2 Feb 2002 13:58:52 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id NAA08453 for ; Sat, 2 Feb 2002 13:58:50 -0500 (EST) Received: from bdsl.66.12.12.130.gte.net (bdsl.66.12.12.130.gte.net [66.12.12.130]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id NAA06283 for ; Sat, 2 Feb 2002 13:58:47 -0500 (EST) Received: from txdwillis (bdsl.greycouncil.com [127.0.0.1]) by bdsl.66.12.12.130.gte.net (8.11.6/8.11.6) with SMTP id g12IwJH21151 for ; Sat, 2 Feb 2002 12:58:19 -0600 Message-ID: <007b01c1ac1b$b2595d20$55fa403f@dfw.dynamicsoft.com> From: "Dean Willis" To: Date: Sat, 2 Feb 2002 12:59:09 -0600 MIME-Version: 1.0 Content-Type: multipart/alternative; boundary="----=_NextPart_000_0078_01C1ABE9.67740170" X-Priority: 3 X-MSMail-Priority: Normal X-Mailer: Microsoft Outlook Express 6.00.2600.0000 X-MimeOLE: Produced By Microsoft MimeOLE V6.00.2600.0000 Subject: [Sipping] Test, please ignore Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org This is a multi-part message in MIME format. ------=_NextPart_000_0078_01C1ABE9.67740170 Content-Type: text/plain; charset="iso-8859-1" Content-Transfer-Encoding: quoted-printable Hey, I periodically test these list-things to see if to they're broken = yet. -- Dean ------=_NextPart_000_0078_01C1ABE9.67740170 Content-Type: text/html; charset="iso-8859-1" Content-Transfer-Encoding: quoted-printable
Hey, I periodically test these = list-things to see=20 if to they're broken yet.
 
--
Dean
 
------=_NextPart_000_0078_01C1ABE9.67740170-- _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Mon Feb 4 04:09:00 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id EAA16963 for ; Mon, 4 Feb 2002 04:09:00 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id EAA18357 for sipping-archive@odin.ietf.org; Mon, 4 Feb 2002 04:09:02 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id DAA16675; Mon, 4 Feb 2002 03:45:34 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id DAA16642 for ; Mon, 4 Feb 2002 03:45:32 -0500 (EST) Received: from x263.net ([218.6.2.12]) by ietf.org (8.9.1a/8.9.1a) with SMTP id DAA16587 for ; Mon, 4 Feb 2002 03:45:25 -0500 (EST) Message-Id: <200202040845.DAA16587@ietf.org> From: "L Mi" To: Mime-Version: 1.0 Content-Type: text/html; charset="ISO-8859-1" Date: Mon, 4 Feb 2002 16:41:11 +0800 Reply-To: "L Mi" Content-Transfer-Encoding: 8bit Content-Transfer-Encoding: 8bit Subject: [Sipping] * To help your business succeed Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 8bit Dear friend

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_______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Tue Feb 5 01:58:04 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id BAA15957 for ; Tue, 5 Feb 2002 01:58:03 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id BAA07824 for sipping-archive@odin.ietf.org; Tue, 5 Feb 2002 01:58:05 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id BAA06244; Tue, 5 Feb 2002 01:31:52 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id BAA06210 for ; Tue, 5 Feb 2002 01:31:41 -0500 (EST) Received: from wiprom2mx1.wipro.com (wiprom2mx1.wipro.com [203.197.164.41]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id BAA15536 for ; Tue, 5 Feb 2002 01:31:34 -0500 (EST) Received: from m2vwall2.wipro.com (m2vwall2.wipro.com [164.164.29.236]) by wiprom2mx1.wipro.com (8.11.3/8.11.3) with SMTP id g156V5A27374 for ; Tue, 5 Feb 2002 12:01:05 +0530 (IST) Received: from m1coet3567 ([10.115.6.222]) by ace.mail.wipro.com (Netscape Messaging Server 4.15) with ESMTP id GR1RFN00.W54 for ; Tue, 5 Feb 2002 12:00:59 +0530 Reply-To: From: "VIDHI RASTOGI" To: Date: Tue, 5 Feb 2002 12:04:09 +0530 Message-ID: <001a01c1ae0f$1ec5b360$de06730a@m1coet3567> MIME-Version: 1.0 Content-Type: multipart/mixed; boundary="----=_NextPartTM-000-d05597d1-19e9-11d6-af7e-0080c8048dde" X-Priority: 3 (Normal) X-MSMail-Priority: Normal X-Mailer: Microsoft Outlook CWS, Build 9.0.2416 (9.0.2910.0) Importance: Normal X-MimeOLE: Produced By Microsoft MimeOLE V5.50.4807.1700 Subject: [Sipping] Question regarding CANCEL Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org This is a multi-part message in MIME format. ------=_NextPartTM-000-d05597d1-19e9-11d6-af7e-0080c8048dde Content-Type: text/plain; charset="iso-8859-1" Content-Transfer-Encoding: 7bit Hi, Should the UAC send a CANCEL after INVITE request times out. (ie, the UAC has retransmitted INVITE 7 times and never received any response). Do we need to send CANCEL for any other SIP requests (say REGISTER, INFO) after the REQUEST TIMEOUT? regards Vidhi ------=_NextPartTM-000-d05597d1-19e9-11d6-af7e-0080c8048dde Content-Type: text/plain; name="Wipro_Disclaimer.txt" Content-Disposition: attachment; filename="Wipro_Disclaimer.txt" Content-Transfer-Encoding: 7bit **************************Disclaimer************************************ Information contained in this E-MAIL being proprietary to Wipro Limited is 'privileged' and 'confidential' and intended for use only by the individual or entity to which it is addressed. You are notified that any use, copying or dissemination of the information contained in the E-MAIL in any manner whatsoever is strictly prohibited. ******************************************************************** ------=_NextPartTM-000-d05597d1-19e9-11d6-af7e-0080c8048dde-- _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Tue Feb 5 05:17:47 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id FAA26417 for ; Tue, 5 Feb 2002 05:17:47 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id FAA17600 for sipping-archive@odin.ietf.org; Tue, 5 Feb 2002 05:17:50 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id FAA17204; Tue, 5 Feb 2002 05:06:09 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id FAA17174 for ; Tue, 5 Feb 2002 05:06:07 -0500 (EST) Received: from hindon.hss.co.in ([202.54.26.202]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id FAA26265 for ; Tue, 5 Feb 2002 05:06:01 -0500 (EST) From: mchhabra@hss.hns.com Received: from sandesh.hss.hns.com (sandesh [139.85.242.35]) by hindon.hss.co.in (8.10.0/8.10.0) with SMTP id g15A5hA12574; Tue, 5 Feb 2002 15:35:43 +0530 (IST) Received: by sandesh.hss.hns.com(Lotus SMTP MTA v4.6.3 (733.2 10-16-1998)) id 65256B57.00374870 ; Tue, 5 Feb 2002 15:33:50 +0530 X-Lotus-FromDomain: HSS To: vidhi.rastogi@wipro.com cc: sipping@ietf.org Message-ID: <65256B57.00374858.00@sandesh.hss.hns.com> Date: Tue, 5 Feb 2002 15:35:54 +0530 Subject: Re: [Sipping] Question regarding CANCEL Mime-Version: 1.0 Content-type: multipart/mixed; Boundary="0__=Q34IO2rZi7JqLxkwBoNszBSeEd9vgAzmyo3dm7Nf4ftndExBYQba57hf" Content-Disposition: inline Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org --0__=Q34IO2rZi7JqLxkwBoNszBSeEd9vgAzmyo3dm7Nf4ftndExBYQba57hf Content-type: text/plain; charset=us-ascii Content-Disposition: inline Hi Vidhi, please see my comments below : Rgds, Munish Hughes Software Systems Limited www.hssworld.com "VIDHI RASTOGI" on 02/05/2002 12:04:09 PM Please respond to vidhi.rastogi@wipro.com To: sipping@ietf.org cc: (bcc: Munish Chhabra/HSS) Subject: [Sipping] Question regarding CANCEL Hi, Should the UAC send a CANCEL after INVITE request times out. (ie, the UAC has retransmitted INVITE 7 times and never received any response). Munish : Refer to 14.4.1 Unreliable Transport Protocols in bis04 - A SIP client using an unreliable transport protocol SHOULD retransmit a SIP INVITE request with an interval that starts at T1 seconds, and doubles after each packet transmission. The client ceases retransmissions if it receives a provisional or definitive response, or once it has sent a total of seven request packets. If no response (final or provisional) is received after sending seven request packets, processing continues as if a 481 response was received for that request (no ACK is generated, however), and a CANCEL SHOULD NOT be sent. Do we need to send CANCEL for any other SIP requests (say REGISTER, INFO) after the REQUEST TIMEOUT? Munish : It doesn't really makes sense to send CANCEL for any of these request timeouts either. regards Vidhi --0__=Q34IO2rZi7JqLxkwBoNszBSeEd9vgAzmyo3dm7Nf4ftndExBYQba57hf Content-type: application/octet-stream; name="Wipro_Disclaimer.txt" Content-Disposition: attachment; filename="Wipro_Disclaimer.txt" Content-Description: Text - character set unknown Content-Transfer-Encoding: base64 KioqKioqKioqKioqKioqKioqKioqKioqKipEaXNjbGFpbWVyKioqKioqKioqKioqKioqKioqKioq KioqKioqKioqKioqKioqDQoNCg0KSW5mb3JtYXRpb24gY29udGFpbmVkIGluIHRoaXMgRS1NQUlM IGJlaW5nIHByb3ByaWV0YXJ5IHRvIFdpcHJvIExpbWl0ZWQNCmlzICdwcml2aWxlZ2VkJyBhbmQg J2NvbmZpZGVudGlhbCcgYW5kIGludGVuZGVkIGZvciB1c2Ugb25seSBieSB0aGUNCmluZGl2aWR1 YWwgb3IgZW50aXR5IHRvIHdoaWNoIGl0IGlzIGFkZHJlc3NlZC4gWW91IGFyZSBub3RpZmllZCB0 aGF0IGFueQ0KdXNlLCBjb3B5aW5nIG9yIGRpc3NlbWluYXRpb24gb2YgdGhlIGluZm9ybWF0aW9u IGNvbnRhaW5lZCBpbiB0aGUgRS1NQUlMDQppbiBhbnkgbWFubmVyIHdoYXRzb2V2ZXIgaXMgc3Ry aWN0bHkgcHJvaGliaXRlZC4NCg0KDQoqKioqKioqKioqKioqKioqKioqKioqKioqKioqKioqKioq KioqKioqKioqKioqKioqKioqKioqKioqKioqKioqKioqKg0KDQo= --0__=Q34IO2rZi7JqLxkwBoNszBSeEd9vgAzmyo3dm7Nf4ftndExBYQba57hf-- _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Tue Feb 5 09:46:14 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id JAA01399 for ; Tue, 5 Feb 2002 09:46:14 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id JAA03004 for sipping-archive@odin.ietf.org; Tue, 5 Feb 2002 09:46:16 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id JAA02065; Tue, 5 Feb 2002 09:33:52 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id JAA02033 for ; Tue, 5 Feb 2002 09:33:49 -0500 (EST) Received: from zcars0m9.ca.nortel.com (zcars0m9.nortelnetworks.com [47.129.242.157]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id JAA00971 for ; Tue, 5 Feb 2002 09:33:45 -0500 (EST) Received: from zcars04f.ca.nortel.com (zcars04f.ca.nortel.com [47.129.242.57]) by zcars0m9.ca.nortel.com (Switch-2.2.0/Switch-2.2.0) with ESMTP id g15EXGH22080 for ; Tue, 5 Feb 2002 09:33:16 -0500 (EST) Received: from zcard015.ca.nortel.com (zcard015.ca.nortel.com [47.129.30.7]) by zcars04f.ca.nortel.com (Switch-2.2.0/Switch-2.2.0) with ESMTP id g15EXE308534 for ; Tue, 5 Feb 2002 09:33:14 -0500 (EST) Received: by zcard015.ca.nortel.com with Internet Mail Service (5.5.2653.19) id <1J3391H3>; Tue, 5 Feb 2002 09:33:14 -0500 Message-ID: <4D79C746863DD51197690002A52CDA000102AF9C@zcard0kc.ca.nortel.com> From: "Tom-PT Taylor" To: sipping@ietf.org Date: Tue, 5 Feb 2002 09:33:14 -0500 MIME-Version: 1.0 X-Mailer: Internet Mail Service (5.5.2653.19) Content-Type: multipart/alternative; boundary="----_=_NextPart_001_01C1AE52.0BAD42A0" Subject: [Sipping] Status Check: Subscriber Identification Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org This message is in MIME format. Since your mail reader does not understand this format, some or all of this message may not be legible. ------_=_NextPart_001_01C1AE52.0BAD42A0 Content-Type: text/plain; charset="iso-8859-1" At one point in the thread "calling number blocking at sip/isup gateway" a few of us seemed to agree that it must be possible for the gateway to identify the subscriber in whose name service is being requested. What if any is the intended SIP mechanism to meet this requirement? Tom Taylor taylor@nortelnetworks.com Ph. +1 613 736 0961 (ESN 396 1490) ------_=_NextPart_001_01C1AE52.0BAD42A0 Content-Type: text/html; charset="iso-8859-1" Content-Transfer-Encoding: quoted-printable Status Check: Subscriber Identification

At one point in the thread "calling number = blocking at sip/isup gateway" a few of us seemed to agree that it = must be possible for the gateway to identify the subscriber in whose = name service is being requested.  What if any is the intended SIP = mechanism to meet this requirement?

Tom Taylor
taylor@nortelnetworks.com
Ph. +1 613 736 0961 (ESN 396 1490)
 

------_=_NextPart_001_01C1AE52.0BAD42A0-- _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Tue Feb 5 10:30:38 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id KAA03883 for ; Tue, 5 Feb 2002 10:30:37 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id KAA06678 for sipping-archive@odin.ietf.org; Tue, 5 Feb 2002 10:30:39 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id KAA04925; Tue, 5 Feb 2002 10:04:00 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id KAA04896 for ; Tue, 5 Feb 2002 10:03:58 -0500 (EST) Received: from rtp-msg-core-1.cisco.com (rtp-msg-core-1.cisco.com [161.44.11.97]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id KAA02530 for ; Tue, 5 Feb 2002 10:03:56 -0500 (EST) Received: from cia.cisco.com (localhost [127.0.0.1]) by rtp-msg-core-1.cisco.com (8.11.3/8.9.1) with ESMTP id g15F3he27433; Tue, 5 Feb 2002 10:03:43 -0500 (EST) Received: from MHAMMER-W2K.cisco.com (hrn2-dhcp-161-44-87-145.cisco.com [161.44.87.145]) by cia.cisco.com (Mirapoint) with ESMTP id AAP20486; Tue, 5 Feb 2002 09:58:13 -0500 (EST) Message-Id: <4.3.2.7.2.20020205095425.00b15b48@cia.cisco.com> X-Sender: mhammer@cia.cisco.com X-Mailer: QUALCOMM Windows Eudora Version 4.3.2 Date: Tue, 05 Feb 2002 10:03:23 -0500 To: "Tom-PT Taylor" From: Michael Hammer Subject: Re: [Sipping] Status Check: Subscriber Identification Cc: sipping@ietf.org In-Reply-To: <4D79C746863DD51197690002A52CDA000102AF9C@zcard0kc.ca.norte l.com> Mime-Version: 1.0 Content-Type: text/plain; charset="us-ascii"; format=flowed Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Tom, This problem of having to identify the customer before proceeding to provide service is a characteristic of mobile networks. I suspect that a similar authentication and authorization process will be required. But, that pre-supposes an infrastructure to handle that. I would assume that it would also support wired and wireless mobility. Yet, some of the drafts I have seen suggest that such infrastructure is undesirable. I think the group should decide whether they are serious about this, then bite the bullet and get on with it. Mike At 09:33 AM 2/5/2002 -0500, Tom-PT Taylor wrote: >At one point in the thread "calling number blocking at sip/isup gateway" a >few of us seemed to agree that it must be possible for the gateway to >identify the subscriber in whose name service is being requested. What if >any is the intended SIP mechanism to meet this requirement? > >Tom Taylor >taylor@nortelnetworks.com >Ph. +1 613 736 0961 (ESN 396 1490) > _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@ns.ietf.org Tue Feb 5 14:07:32 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id OAA14158 for ; Tue, 5 Feb 2002 14:07:32 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id OAA26328 for sipping-archive@odin.ietf.org; Tue, 5 Feb 2002 14:07:30 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id NAA24485; Tue, 5 Feb 2002 13:41:54 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id NAA24453 for ; Tue, 5 Feb 2002 13:41:52 -0500 (EST) Received: from nisser.cisco.com (nisser.cisco.com [171.71.176.85]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id NAA13160 for ; Tue, 5 Feb 2002 13:41:49 -0500 (EST) Received: from cisco.com (ssh-sj1.cisco.com [171.68.225.134]) by nisser.cisco.com (8.8.6 (PHNE_14041)/CISCO.SERVER.1.2) with ESMTP id KAA10339; Tue, 5 Feb 2002 10:36:08 -0800 (PST) Message-ID: <3C60261F.394C997@cisco.com> Date: Tue, 05 Feb 2002 13:36:15 -0500 From: Flemming Andreasen X-Mailer: Mozilla 4.79 [en]C-CCK-MCD (Windows NT 5.0; U) X-Accept-Language: en MIME-Version: 1.0 To: Tom-PT Taylor CC: sipping@ietf.org Subject: Re: [Sipping] Status Check: Subscriber Identification References: <4D79C746863DD51197690002A52CDA000102AF9C@zcard0kc.ca.nortel.com> Content-Type: text/plain; charset=us-ascii Content-Transfer-Encoding: 7bit Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit Do you mean authenticate the subscriber or simply provide an identification of the subscriber that subsequent hops can use ? -- Flemming Tom-PT Taylor wrote: > > > At one point in the thread "calling number blocking at sip/isup > gateway" a few of us seemed to agree that it must be possible for the > gateway to identify the subscriber in whose name service is being > requested. What if any is the intended SIP mechanism to meet this > requirement? > > Tom Taylor > taylor@nortelnetworks.com > Ph. +1 613 736 0961 (ESN 396 1490) > -- Flemming Andreasen Cisco Systems _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@ns.ietf.org Tue Feb 5 14:12:14 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id OAA14373 for ; Tue, 5 Feb 2002 14:12:14 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id OAA26483 for sipping-archive@odin.ietf.org; Tue, 5 Feb 2002 14:12:16 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id NAA25102; Tue, 5 Feb 2002 13:56:01 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id NAA25071 for ; Tue, 5 Feb 2002 13:55:59 -0500 (EST) Received: from web11604.mail.yahoo.com (web11604.mail.yahoo.com [216.136.172.56]) by ietf.org (8.9.1a/8.9.1a) with SMTP id NAA13708 for ; Tue, 5 Feb 2002 13:55:56 -0500 (EST) Message-ID: <20020205185558.40616.qmail@web11604.mail.yahoo.com> Received: from [216.51.104.84] by web11604.mail.yahoo.com via HTTP; Tue, 05 Feb 2002 10:55:58 PST Date: Tue, 5 Feb 2002 10:55:58 -0800 (PST) From: Sean Olson Subject: Re: [Sipping] Status Check: Subscriber Identification To: Flemming Andreasen , Tom-PT Taylor Cc: sipping@ietf.org In-Reply-To: <3C60261F.394C997@cisco.com> MIME-Version: 1.0 Content-Type: text/plain; charset=us-ascii Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Given the intended usage of this identifier, a non-authenticated subscriber ID would not be very useful. /sean --- Flemming Andreasen wrote: > Do you mean authenticate the subscriber or simply > provide an > identification of the subscriber that subsequent > hops can use ? > > -- Flemming > > Tom-PT Taylor wrote: > > > > > > > At one point in the thread "calling number > blocking at sip/isup > > gateway" a few of us seemed to agree that it must > be possible for the > > gateway to identify the subscriber in whose name > service is being > > requested. What if any is the intended SIP > mechanism to meet this > > requirement? > > > > Tom Taylor > > taylor@nortelnetworks.com > > Ph. +1 613 736 0961 (ESN 396 1490) > > > > -- > Flemming Andreasen > Cisco Systems > > > > _______________________________________________ > Sipping mailing list > https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application > of SIP > Use sip-implementors@cs.columbia.edu for questions > on current sip > Use sip@ietf.org for new developments of core SIP ===== Sean Olson __________________________________________________ Do You Yahoo!? Send FREE Valentine eCards with Yahoo! Greetings! http://greetings.yahoo.com _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@ns.ietf.org Tue Feb 5 15:42:00 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id PAA18546 for ; Tue, 5 Feb 2002 15:42:00 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id PAA00721 for sipping-archive@odin.ietf.org; Tue, 5 Feb 2002 15:42:03 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id PAA29596; Tue, 5 Feb 2002 15:22:38 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id PAA29565 for ; Tue, 5 Feb 2002 15:22:36 -0500 (EST) Received: from nisser.cisco.com (nisser.cisco.com [171.71.176.85]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id PAA17793 for ; Tue, 5 Feb 2002 15:22:31 -0500 (EST) Received: from cisco.com (ssh-sj1.cisco.com [171.68.225.134]) by nisser.cisco.com (8.8.6 (PHNE_14041)/CISCO.SERVER.1.2) with ESMTP id MAA25315; Tue, 5 Feb 2002 12:08:07 -0800 (PST) Message-ID: <3C603BB3.32178126@cisco.com> Date: Tue, 05 Feb 2002 15:08:19 -0500 From: Flemming Andreasen X-Mailer: Mozilla 4.79 [en]C-CCK-MCD (Windows NT 5.0; U) X-Accept-Language: en MIME-Version: 1.0 To: Sean Olson CC: Tom-PT Taylor , sipping@ietf.org Subject: Re: [Sipping] Status Check: Subscriber Identification References: <20020205185558.40616.qmail@web11604.mail.yahoo.com> Content-Type: text/plain; charset=us-ascii Content-Transfer-Encoding: 7bit Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit Right, however that wasn't my question. The privacy draft, and in particular the Remote-Party-Id is intended to provide the identity information, however it does not specify the authentication mechanism by which the identity is determined, which is why I asked Tom to clarify. -- Flemming Sean Olson wrote: > Given the intended usage of this identifier, a > non-authenticated subscriber ID would not be > very useful. > > /sean > > --- Flemming Andreasen wrote: > > Do you mean authenticate the subscriber or simply > > provide an > > identification of the subscriber that subsequent > > hops can use ? > > > > -- Flemming > > > > Tom-PT Taylor wrote: > > > > > > > > > > > At one point in the thread "calling number > > blocking at sip/isup > > > gateway" a few of us seemed to agree that it must > > be possible for the > > > gateway to identify the subscriber in whose name > > service is being > > > requested. What if any is the intended SIP > > mechanism to meet this > > > requirement? > > > > > > Tom Taylor > > > taylor@nortelnetworks.com > > > Ph. +1 613 736 0961 (ESN 396 1490) > > > > > > > -- > > Flemming Andreasen > > Cisco Systems > > > > > > > > _______________________________________________ > > Sipping mailing list > > https://www1.ietf.org/mailman/listinfo/sipping > > This list is for NEW development of the application > > of SIP > > Use sip-implementors@cs.columbia.edu for questions > > on current sip > > Use sip@ietf.org for new developments of core SIP > > ===== > Sean Olson > > __________________________________________________ > Do You Yahoo!? > Send FREE Valentine eCards with Yahoo! Greetings! > http://greetings.yahoo.com -- Flemming Andreasen Cisco Systems _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@ns.ietf.org Tue Feb 5 16:54:33 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id QAA20827 for ; Tue, 5 Feb 2002 16:54:33 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id QAA04146 for sipping-archive@odin.ietf.org; Tue, 5 Feb 2002 16:54:37 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id QAA03560; Tue, 5 Feb 2002 16:40:37 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id QAA03535 for ; Tue, 5 Feb 2002 16:40:35 -0500 (EST) Received: from mailcity.com (fes-qout.whowhere.com [209.185.123.96]) by ietf.org (8.9.1a/8.9.1a) with SMTP id QAA20487 for ; Tue, 5 Feb 2002 16:40:30 -0500 (EST) Received: from Unknown/Local ([?.?.?.?]) by mailcity.com; Tue Feb 5 13:39:50 2002 To: sipping@ietf.org, vidhi.rastogi@wipro.com Date: Tue, 05 Feb 2002 16:39:50 -0500 From: "Info Van" Message-ID: Mime-Version: 1.0 X-Sent-Mail: on Reply-To: infovaan@lycos.com X-Mailer: MailCity Service X-Priority: 3 Subject: Re: [Sipping] Question regarding CANCEL X-Sender-Ip: 64.66.32.130 Organization: Lycos Mail (http://mail.lycos.com:80) Content-Type: text/plain; charset=us-ascii Content-Language: en Content-Transfer-Encoding: 7bit Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit Yes. Since it is for a "pending" session establishment. Or am I missing something here !! Anyway you are also less than the 10 retries. -iv. --- ---- Info Vaan is the way to go ------- On Tue, 5 Feb 2002 12:04:09 VIDHI RASTOGI wrote: >Hi, > >Should the UAC send a CANCEL after INVITE request times out. (ie, the UAC >has retransmitted INVITE 7 times and never received any response). > >Do we need to send CANCEL for any other SIP requests (say REGISTER, INFO) >after the REQUEST TIMEOUT? > >regards >Vidhi > > _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@ns.ietf.org Tue Feb 5 21:56:02 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id VAA29826 for ; Tue, 5 Feb 2002 21:56:01 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id VAA19924 for sipping-archive@odin.ietf.org; Tue, 5 Feb 2002 21:56:04 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id VAA19244; Tue, 5 Feb 2002 21:45:35 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id VAA19184 for ; Tue, 5 Feb 2002 21:45:30 -0500 (EST) Received: from dan ([211.100.89.77]) by ietf.org (8.9.1a/8.9.1a) with SMTP id VAA29581 for ; Tue, 5 Feb 2002 21:45:23 -0500 (EST) Message-Id: <200202060245.VAA29581@ietf.org> From: "T&F Information Exchange" To: Mime-Version: 1.0 Content-Type: text/plain; charset="gb2312" Date: Wed, 6 Feb 2002 10:45:00 Subject: [Sipping] A professional company providing credit & status report of Chinese company. 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Thank you. 使用极星邮件群发,无须通过邮件服务器,直达对方邮箱,速度绝对一流! 下载网址:http://love2net.51.net/,更多免费的超酷软件等你来下…… ---------------------------------------------------- INFORMATION This message has been sent using a trial-run version of the TSmtpRelayServer Delphi Component. ---------------------------------------------------- _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@ns.ietf.org Tue Feb 5 22:32:49 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id WAA01423 for ; Tue, 5 Feb 2002 22:32:46 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id WAA22538 for sipping-archive@odin.ietf.org; Tue, 5 Feb 2002 22:32:49 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id WAA21677; Tue, 5 Feb 2002 22:18:05 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id WAA21646 for ; Tue, 5 Feb 2002 22:18:02 -0500 (EST) Received: from oak.neustar.com (oak.neustar.com [209.173.53.70]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id WAA00260 for ; Tue, 5 Feb 2002 22:17:59 -0500 (EST) Received: from chiimc01.il.neustar.com (dmz1.va.neustar.com [209.173.53.65]) by oak.neustar.com (8.11.0/8.11.0) with ESMTP id g163Acb21017; Tue, 5 Feb 2002 22:10:38 -0500 Received: by chiimc01.il.neustar.com with Internet Mail Service (5.5.2653.19) id ; Tue, 5 Feb 2002 21:10:38 -0600 Message-ID: <70565611B164D511957A001083FCDD56CAAEAD@VA02> From: "Peterson, Jon" To: "'Flemming Andreasen'" , Sean Olson Cc: Tom-PT Taylor , sipping@ietf.org Subject: RE: [Sipping] Status Check: Subscriber Identification Date: Tue, 5 Feb 2002 21:10:29 -0600 MIME-Version: 1.0 X-Mailer: Internet Mail Service (5.5.2653.19) Content-Type: text/plain; charset="iso-8859-1" Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org There are SIP mechanisms that allow servers to ascertain the identities of remote users - just to pick one out of a hat, let's look at the Digest authentication scheme. When the gateway receives an INVITE, it sends a challenge (401) in response. The user retries the request with a Digest username and password (corresponding to its account at this gateway operator's network). This username and password could be used by the gateway to retrieve a particular phone number and/or caller name corresponding to 'the subscriber in whose name service is being requested', returning to Tom's original question. The gateway could then use this information to populate a CgPN field and whatever other ISUP fields it liked. It might even overwrite any telephone number that happened to appear in the From field of the request with its own data. Or, the gateway could merely retrieve an account number for the user, and record this number in some sort of event block/CDR that it cuts for this call - in this instance it could send a common 'gateway number' rather than an individual subscriber's number in the CgPN, if it were so inclined. If a malicious trace were subsequently required, the gateway operator could find the event block in question and provide the user's account information to the authorities. The point is, none of this requires any changes to SIP (or the SIP-ISUP mappings) that I'm aware of. Jon Peterson NeuStar, Inc. > -----Original Message----- > From: Flemming Andreasen [mailto:fandreas@cisco.com] > Sent: Tuesday, February 05, 2002 12:08 PM > To: Sean Olson > Cc: Tom-PT Taylor; sipping@ietf.org > Subject: Re: [Sipping] Status Check: Subscriber Identification > > > Right, however that wasn't my question. The privacy draft, and in > particular the Remote-Party-Id is intended to provide the identity > information, however it does not specify the authentication > mechanism by > which the identity is determined, which is why I asked Tom to clarify. > > -- Flemming > > Sean Olson wrote: > > > Given the intended usage of this identifier, a > > non-authenticated subscriber ID would not be > > very useful. > > > > /sean > > > > --- Flemming Andreasen wrote: > > > Do you mean authenticate the subscriber or simply > > > provide an > > > identification of the subscriber that subsequent > > > hops can use ? > > > > > > -- Flemming > > > > > > Tom-PT Taylor wrote: > > > > > > > > > > > > > > > At one point in the thread "calling number > > > blocking at sip/isup > > > > gateway" a few of us seemed to agree that it must > > > be possible for the > > > > gateway to identify the subscriber in whose name > > > service is being > > > > requested. What if any is the intended SIP > > > mechanism to meet this > > > > requirement? > > > > > > > > Tom Taylor > > > > taylor@nortelnetworks.com > > > > Ph. +1 613 736 0961 (ESN 396 1490) > > > > > > > > > > -- > > > Flemming Andreasen > > > Cisco Systems > > > > > > > > > > > > _______________________________________________ > > > Sipping mailing list > > > https://www1.ietf.org/mailman/listinfo/sipping > > > This list is for NEW development of the application > > > of SIP > > > Use sip-implementors@cs.columbia.edu for questions > > > on current sip > > > Use sip@ietf.org for new developments of core SIP > > > > ===== > > Sean Olson > > > > __________________________________________________ > > Do You Yahoo!? > > Send FREE Valentine eCards with Yahoo! Greetings! > > http://greetings.yahoo.com > > -- > Flemming Andreasen > Cisco Systems > > > > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP > Use sip-implementors@cs.columbia.edu for questions on current sip > Use sip@ietf.org for new developments of core SIP > _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Wed Feb 6 01:45:18 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id BAA04714 for ; Wed, 6 Feb 2002 01:45:18 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id BAA10195 for sipping-archive@odin.ietf.org; Wed, 6 Feb 2002 01:45:18 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id BAA09510; Wed, 6 Feb 2002 01:32:14 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id BAA09478 for ; Wed, 6 Feb 2002 01:32:10 -0500 (EST) Received: from oak.neustar.com (oak.neustar.com [209.173.53.70]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id BAA04515 for ; Wed, 6 Feb 2002 01:32:09 -0500 (EST) Received: from chiimc01.il.neustar.com (dmz1.va.neustar.com [209.173.53.65]) by oak.neustar.com (8.11.0/8.11.0) with ESMTP id g166L5b24423; Wed, 6 Feb 2002 01:21:05 -0500 Received: by chiimc01.il.neustar.com with Internet Mail Service (5.5.2653.19) id ; Wed, 6 Feb 2002 00:21:05 -0600 Message-ID: <70565611B164D511957A001083FCDD56CAAEAF@VA02> From: "Peterson, Jon" To: "'Juha Heinanen'" , Jonathan Rosenberg Cc: Tom-PT Taylor , "'Mpierce1@aol.com'" , sipping@ietf.org Subject: RE: [Sipping] RE: calling number blocking at sip/isup gateway Date: Wed, 6 Feb 2002 00:20:56 -0600 MIME-Version: 1.0 X-Mailer: Internet Mail Service (5.5.2653.19) Content-Type: text/plain; charset="iso-8859-1" Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org There has been a lot of misleading talk in this thread that half-suggests that support for presentation restriction in SIP is necessary if you want to interwork with the PSTN. I find it hard to believe that there is any existing national network which requires that every communications protocol interworking with the PSTN has a feature by which callers can express that their preference not to reveal their name and telephone number to those they call. I know of some legacy protocols that interwork with the PSTN worldwide today that lack this feature. I also think this would be a very difficult rule to enforce - if a gateway creates properly formed presentation indicators, I doubt these would be subjected to any scrutiny. Because I don't think this is a requirement for PSTN interworking, I do not think it is an absolute dependency in the SIP-ISUP mapping work. We therefore don't include any mapping in the SIP-ISUP draft, although we do say that SIP-ISUP implementations can make use of any general SIP privacy mechanism. All that said, I feel strongly that we should have a way to implement a presentation restriction feature in SIP. However, if there isn't a general presentation restriction feature in SIP, that doesn't in any way prevent us from interworking with the PSTN. If such a general mechanism were defined, then by all means we can determine how to to map it to the SS7 presentation restriction mechanism (easily, no doubt). Note that at least one draft (the early media mechanism) describes the SIP-ISUP mapping relevant to its own mechanism - at least partially because we didn't want the SIP-ISUP draft to have a dependency on SIP extensions defined in other drafts. I guess that although some people may be happy with the DCS sip-privacy mechanism, I'm not sure that it really has general applicability. I currently don't think we need to describe a mapping to this mechanism in the SIP-ISUP draft. Jon Peterson NeuStar, Inc. > -----Original Message----- > From: Juha Heinanen [mailto:jh@lohi.eng.song.fi] > Sent: Wednesday, January 30, 2002 12:28 AM > To: Jonathan Rosenberg > Cc: Tom-PT Taylor; 'Mpierce1@aol.com'; sipping@ietf.org > Subject: Re: [Sipping] RE: calling number blocking at sip/isup gateway > > > Jonathan Rosenberg writes: > > > Well, we have a documented way for that in the privacy > spec right now, > > which is for the proxy to insert the RPID-Privacy header indicating > > calling party, with any kind of privacy, and type of subscriber. > > yes, and i'm prefectly happy with it. the problem is that > the sip-isup > i-d doesn't say a word about it and it is that document that > the ss7/sip > gw vendors will read. > > -- juha > > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP > Use sip-implementors@cs.columbia.edu for questions on current sip > Use sip@ietf.org for new developments of core SIP > _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Wed Feb 6 03:18:44 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id DAA14140 for ; Wed, 6 Feb 2002 03:18:44 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id DAA14839 for sipping-archive@odin.ietf.org; Wed, 6 Feb 2002 03:18:45 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id DAA13372; Wed, 6 Feb 2002 03:00:03 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id DAA13319 for ; Wed, 6 Feb 2002 03:00:01 -0500 (EST) Received: from imo-r10.mx.aol.com (imo-r10.mx.aol.com [152.163.225.106]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id CAA13896 for ; Wed, 6 Feb 2002 02:59:59 -0500 (EST) From: Mpierce1@aol.com Received: from Mpierce1@aol.com by imo-r10.mx.aol.com (mail_out_v31_r1.26.) id c.14a.86a3762 (15888); Wed, 6 Feb 2002 02:54:03 -0500 (EST) Received: from web42.aolmail.aol.com (web42.aolmail.aol.com [205.188.161.3]) by air-id08.mx.aol.com (v83.35) with ESMTP id MAILINID83-0206025402; Wed, 06 Feb 2002 02:54:02 -0500 Date: Wed, 06 Feb 2002 02:54:02 EST Subject: Re: [Sipping] Status Check: Subscriber Identification To: , Cc: Mime-Version: 1.0 Content-Type: text/plain; charset=ISO-8859-1 Content-Transfer-Encoding: 7bit X-Mailer: Unknown (No Version) Message-ID: <14a.86a3762.29923b1a@aol.com> Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit On 5 Feb, Mike Hammer wrote: > Tom, > This problem of having to identify the customer before proceeding to > provide service is a characteristic of mobile networks. I suspect that a > similar authentication and authorization process will be required. But, > that pre-supposes an infrastructure to handle that. I would assume that it > would also support wired and wireless mobility. > Yet, some of the drafts I have seen suggest that such infrastructure is > undesirable. I think the group should decide whether they are serious > about this, then bite the bullet and get on with it. > Mike > At 09:33 AM 2/5/2002 -0500, Tom-PT Taylor wrote: > >At one point in the thread "calling number blocking at sip/isup gateway" a > >few of us seemed to agree that it must be possible for the gateway to > >identify the subscriber in whose name service is being requested. What if > >any is the intended SIP mechanism to meet this requirement? > > > >Tom Taylor > >taylor@nortelnetworks.com It seems to me that the existance of an infrastucture must be assumed in order to handle authentication. It can't be done on a worldwide network supporting everyone without one. Yes, I know there are those who believe any assumed infrastructure is undesirable when one is talking about the Internet, but then, sometimes undesirable things are still necessary. Mike _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Wed Feb 6 04:30:35 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id EAA15106 for ; Wed, 6 Feb 2002 04:30:35 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id EAA18557 for sipping-archive@odin.ietf.org; Wed, 6 Feb 2002 04:30:38 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id EAA17113; Wed, 6 Feb 2002 04:06:39 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id EAA17087 for ; Wed, 6 Feb 2002 04:06:37 -0500 (EST) Received: from mail.szutstar.com ([202.105.137.238]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id EAA14784 for ; Wed, 6 Feb 2002 04:06:32 -0500 (EST) Received: from sznssalbert ([172.16.162.59]) by mail.szutstar.com (8.11.1/8.11.1) with SMTP id g168vrJ13110 for ; Wed, 6 Feb 2002 16:57:53 +0800 Message-ID: <04ce01c1aeed$63fbe7e0$3ba210ac@sznssalbert> From: "wlw" To: Date: Wed, 6 Feb 2002 17:05:14 +0800 MIME-Version: 1.0 Content-Type: multipart/alternative; boundary="----=_NextPart_000_04CB_01C1AF30.71FD9620" X-Priority: 3 X-MSMail-Priority: Normal X-Mailer: Microsoft Outlook Express 5.50.4133.2400 X-MimeOLE: Produced By Microsoft MimeOLE V5.50.4133.2400 Subject: [Sipping] Where can I find draft-ietf-sip-isup-mime-10.txt? Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org This is a multi-part message in MIME format. ------=_NextPart_000_04CB_01C1AF30.71FD9620 Content-Type: text/plain; charset="gb2312" Content-Transfer-Encoding: base64 DQo= ------=_NextPart_000_04CB_01C1AF30.71FD9620 Content-Type: text/html; charset="gb2312" Content-Transfer-Encoding: base64 PCFET0NUWVBFIEhUTUwgUFVCTElDICItLy9XM0MvL0RURCBIVE1MIDQuMCBUcmFuc2l0aW9uYWwv L0VOIj4NCjxIVE1MPjxIRUFEPg0KPE1FVEEgY29udGVudD0idGV4dC9odG1sOyBjaGFyc2V0PWdi MjMxMiIgaHR0cC1lcXVpdj1Db250ZW50LVR5cGU+DQo8TUVUQSBjb250ZW50PSJNU0hUTUwgNS4w MC4zMTAzLjEwMDAiIG5hbWU9R0VORVJBVE9SPg0KPFNUWUxFPjwvU1RZTEU+DQo8L0hFQUQ+DQo8 Qk9EWSBiZ0NvbG9yPSNmZmZmZmY+DQo8RElWPiZuYnNwOzwvRElWPjwvQk9EWT48L0hUTUw+DQo= ------=_NextPart_000_04CB_01C1AF30.71FD9620-- _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Wed Feb 6 04:36:02 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id EAA15168 for ; Wed, 6 Feb 2002 04:36:02 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id EAA19030 for sipping-archive@odin.ietf.org; Wed, 6 Feb 2002 04:36:04 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id EAA17747; Wed, 6 Feb 2002 04:22:43 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id EAA17714 for ; Wed, 6 Feb 2002 04:22:41 -0500 (EST) Received: from lohi.eng.song.fi (lohi.eng.song.fi [195.10.149.18]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id EAA14953 for ; Wed, 6 Feb 2002 04:22:37 -0500 (EST) Received: from lohi.eng.song.fi (localhost [127.0.0.1]) by lohi.eng.song.fi (8.12.1/8.12.1/Debian -5) with ESMTP id g169MZio013258; Wed, 6 Feb 2002 11:22:35 +0200 Received: (from jh@localhost) by lohi.eng.song.fi (8.12.1/8.12.1/Debian -5) id g169MX8F013254; Wed, 6 Feb 2002 11:22:33 +0200 From: Juha Heinanen MIME-Version: 1.0 Content-Type: text/plain; charset=us-ascii Content-Transfer-Encoding: 7bit Message-ID: <15456.62937.161292.304267@lohi.eng.song.fi> Date: Wed, 6 Feb 2002 11:22:33 +0200 To: "Peterson, Jon" Cc: Jonathan Rosenberg , Tom-PT Taylor , "'Mpierce1@aol.com'" , sipping@ietf.org Subject: RE: [Sipping] RE: calling number blocking at sip/isup gateway In-Reply-To: <70565611B164D511957A001083FCDD56CAAEAF@VA02> References: <70565611B164D511957A001083FCDD56CAAEAF@VA02> X-Mailer: VM 6.97 under Emacs 20.7.2 Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit Peterson, Jon writes: > I guess that although some people may be happy with the DCS sip-privacy > mechanism, I'm not sure that it really has general applicability. I > currently don't think we need to describe a mapping to this mechanism in the > SIP-ISUP draft. if presentation restricted mapping is not defined in the sip-isup i-d, then there is no hope for interoperability between uas and sip/isup gws or between proxies and sip/isup gws. it would mean that i as the customer of all this equipment would need to negotiate with each vendor and try to make them implement the same mapping. this would be in practise impossible. therefore, i would not vote for advancing the sip/isup i-d before it specifies how restricted number presentation is handled. what comes to the solution, i'm happy with the one proposed by someone from cisco on this mailing list. -- juha _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Wed Feb 6 04:36:45 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id EAA15190 for ; Wed, 6 Feb 2002 04:36:45 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id EAA19059 for sipping-archive@odin.ietf.org; Wed, 6 Feb 2002 04:36:48 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id EAA17814; Wed, 6 Feb 2002 04:23:45 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id EAA17781 for ; Wed, 6 Feb 2002 04:23:42 -0500 (EST) Received: from pine.neustar.com (pine.neustar.com [209.173.57.70]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id EAA14974 for ; Wed, 6 Feb 2002 04:23:39 -0500 (EST) Received: from chiimc01.il.neustar.com (dmz1.il.neustar.com [209.173.57.65]) by pine.neustar.com (8.11.0/8.11.0) with ESMTP id g169Neb30970; Wed, 6 Feb 2002 03:23:40 -0600 Received: by chiimc01.il.neustar.com with Internet Mail Service (5.5.2653.19) id ; Wed, 6 Feb 2002 03:23:40 -0600 Message-ID: <70565611B164D511957A001083FCDD56CAAEB5@VA02> From: "Peterson, Jon" To: "'wlw'" , sipping@ietf.org Subject: RE: [Sipping] Where can I find draft-ietf-sip-isup-mime-10.txt? Date: Wed, 6 Feb 2002 03:23:31 -0600 MIME-Version: 1.0 X-Mailer: Internet Mail Service (5.5.2653.19) Content-Type: multipart/alternative; boundary="----_=_NextPart_001_01C1AEEF.F143BE50" Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org This message is in MIME format. Since your mail reader does not understand this format, some or all of this message may not be legible. ------_=_NextPart_001_01C1AEEF.F143BE50 Content-Type: text/plain; charset="gb2312" In RFC 3204, thankfully. Jon Peterson NeuStar, Inc. -----Original Message----- From: wlw [mailto:albert.wang@utstar.com] Sent: Wednesday, February 06, 2002 1:05 AM To: sipping@ietf.org Subject: [Sipping] Where can I find draft-ietf-sip-isup-mime-10.txt? ------_=_NextPart_001_01C1AEEF.F143BE50 Content-Type: text/html; charset="gb2312"
In RFC 3204, thankfully.
 
Jon Peterson
NeuStar, Inc.
-----Original Message-----
From: wlw [mailto:albert.wang@utstar.com]
Sent: Wednesday, February 06, 2002 1:05 AM
To: sipping@ietf.org
Subject: [Sipping] Where can I find draft-ietf-sip-isup-mime-10.txt?

 
------_=_NextPart_001_01C1AEEF.F143BE50-- _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Wed Feb 6 05:52:13 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id FAA16354 for ; Wed, 6 Feb 2002 05:52:13 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id FAA23119 for sipping-archive@odin.ietf.org; Wed, 6 Feb 2002 05:52:16 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id FAA21883; Wed, 6 Feb 2002 05:37:09 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id FAA21855 for ; Wed, 6 Feb 2002 05:37:07 -0500 (EST) Received: from albatross.wise.edt.ericsson.se (albatross-ext.wise.edt.ericsson.se [194.237.142.116]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id FAA16033 for ; Wed, 6 Feb 2002 05:37:02 -0500 (EST) Received: from fogerty.lmf.ericsson.se (fogerty.lmf.ericsson.se [131.160.11.6]) by albatross.wise.edt.ericsson.se (8.12.1/8.12.1/WIREfire-1.4) with ESMTP id g16AaEX4014631; Wed, 6 Feb 2002 11:36:14 +0100 (MET) Received: from lmf.ericsson.se (E005004B52C74.lmf.ericsson.se [131.160.30.135]) by fogerty.lmf.ericsson.se (8.12.1/8.12.1/lmf.8.12.1.jcs) with ESMTP id g16AaDr6004454; Wed, 6 Feb 2002 12:36:13 +0200 (EET) Message-ID: <3C610701.D66640A2@lmf.ericsson.se> Date: Wed, 06 Feb 2002 12:35:45 +0200 From: Christer Holmberg X-Mailer: Mozilla 4.61 [en] (WinNT; I) X-Accept-Language: en MIME-Version: 1.0 To: Juha Heinanen CC: "Peterson, Jon" , Jonathan Rosenberg , Tom-PT Taylor , "'Mpierce1@aol.com'" , sipping@ietf.org Subject: Re: [Sipping] RE: calling number blocking at sip/isup gateway References: <70565611B164D511957A001083FCDD56CAAEAF@VA02> <15456.62937.161292.304267@lohi.eng.song.fi> Content-Type: text/plain; charset=us-ascii Content-Transfer-Encoding: 7bit Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit Hi, I think the difficult part is to decide which mechanism to use - then we can argue in where to describe it... Regards, Christer Holmberg Ericsson Finland Juha Heinanen wrote: > > Peterson, Jon writes: > > > I guess that although some people may be happy with the DCS sip-privacy > > mechanism, I'm not sure that it really has general applicability. I > > currently don't think we need to describe a mapping to this mechanism in the > > SIP-ISUP draft. > > if presentation restricted mapping is not defined in the sip-isup i-d, > then there is no hope for interoperability between uas and sip/isup gws > or between proxies and sip/isup gws. it would mean that i as the > customer of all this equipment would need to negotiate with each vendor > and try to make them implement the same mapping. this would be in > practise impossible. therefore, i would not vote for advancing the > sip/isup i-d before it specifies how restricted number presentation is > handled. > > what comes to the solution, i'm happy with the one proposed by someone > from cisco on this mailing list. > > -- juha > > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP > Use sip-implementors@cs.columbia.edu for questions on current sip > Use sip@ietf.org for new developments of core SIP _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Wed Feb 6 06:02:59 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id GAA16627 for ; Wed, 6 Feb 2002 06:02:59 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id GAA24058 for sipping-archive@odin.ietf.org; Wed, 6 Feb 2002 06:03:03 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id FAA22387; Wed, 6 Feb 2002 05:46:12 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id FAA22260 for ; Wed, 6 Feb 2002 05:46:06 -0500 (EST) Received: from oak.neustar.com (oak.neustar.com [209.173.53.70]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id FAA16203 for ; Wed, 6 Feb 2002 05:46:01 -0500 (EST) Received: from chiimc01.il.neustar.com (dmz1.va.neustar.com [209.173.53.65]) by oak.neustar.com (8.11.0/8.11.0) with ESMTP id g16Ae6b29256; Wed, 6 Feb 2002 05:40:06 -0500 Received: by chiimc01.il.neustar.com with Internet Mail Service (5.5.2653.19) id ; Wed, 6 Feb 2002 04:40:06 -0600 Message-ID: <70565611B164D511957A001083FCDD56CAAEB6@VA02> From: "Peterson, Jon" To: "'Juha Heinanen'" , "Peterson, Jon" Cc: Jonathan Rosenberg , Tom-PT Taylor , "'Mpierce1@aol.com'" , sipping@ietf.org Subject: RE: [Sipping] RE: calling number blocking at sip/isup gateway Date: Wed, 6 Feb 2002 04:39:55 -0600 MIME-Version: 1.0 X-Mailer: Internet Mail Service (5.5.2653.19) Content-Type: text/plain; charset="iso-8859-1" Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Some notes below. Jon Peterson NeuStar, Inc. > -----Original Message----- > From: Juha Heinanen [mailto:jh@lohi.eng.song.fi] > Sent: Wednesday, February 06, 2002 1:23 AM > To: Peterson, Jon > Cc: Jonathan Rosenberg; Tom-PT Taylor; 'Mpierce1@aol.com'; > sipping@ietf.org > Subject: RE: [Sipping] RE: calling number blocking at sip/isup gateway > [snip] > > if presentation restricted mapping is not defined in the sip-isup i-d, > then there is no hope for interoperability between uas and > sip/isup gws > or between proxies and sip/isup gws. In fact, there are already numerous gateways that operate and interoperate without the benefit of presentation restriction. > it would mean that i as the > customer of all this equipment would need to negotiate with > each vendor > and try to make them implement the same mapping. this would be in > practise impossible. therefore, i would not vote for advancing the > sip/isup i-d before it specifies how restricted number presentation is > handled. I submit I have illustrated amply that you can define how SIP would interwork with the PSTN without inventing a presentation restriction mechanism for SIP - please don't just assert the contrary, but let me know in what respects your analysis differs from mine. I'll reiterate - the purpose of the SIP-ISUP mapping project (known collectively as SIP-T) is not to add new features to SIP - it's to describe how SIP and SS7, without any modifications on either side, can be interworked. There are numerous bits of data that appear in SS7 that have no SIP corrollary today, over and above presentation restriction indicators. For example, there's an SS7 bit that allows you to specify that a one-way echo canceller is available on a particular port. You might as well argue that there can be no SIP-ISUP interoperability until we define an indicator for the presence of a half-echo canceller somehow in SIP as well. The fact remains that you don't need this information to interwork. Anyway, the real problem is the want of a general privacy mechanism in SIP - this really isn't draft-ietf-sipping-isup-00's problem. If there weren't even an existing proposal for a presentation restriction mechanism, would you still insist that the SIP-ISUP mapping draft should block until someone proposes this feature for SIP? Because FYI, there's nothing comparable to a continuity check feature proposed for SIP yet either. Nor any standard for test tones in user agents to measure slope. The fact remains that SIP and ISUP can usefully be interworked without these things. Finally, when a general presentation restriction mechanism for SIP is standardized, a new component of the SIP-ISUP mapping work (which is now already distributed across four or five documents) could be authored to describe the standard way of mapping presentation restriction. There's nothing magical about being a part of draft-ietf-sipping-isup-00. I don't think this will lead to any interoperability nightmares or undue negotiation with vendors. > > what comes to the solution, i'm happy with the one proposed by someone > from cisco on this mailing list. While I don't have the time to go into a detailed analysis of the existing privacy proposal, suffice it to say that I understand it to be primarily applicable to certain architectures in which user agents are untrusted and a federation of intermediaries under a common policy arbitrarily modifies the signaling provided by user agents, including asserting identity on the user's behalf. I don't think that the SIP-ISUP mapping has any need to acquire a dependency on this network design philosophy. I believe the sip-privacy draft is admirable, and it no doubt serves the purpose for which it was created, but I don't believe it has widespread general applicability for SIP. > > -- juha > _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Wed Feb 6 06:04:07 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id GAA16655 for ; Wed, 6 Feb 2002 06:04:06 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id GAA24120 for sipping-archive@odin.ietf.org; Wed, 6 Feb 2002 06:04:10 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id FAA22763; Wed, 6 Feb 2002 05:50:01 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id FAA22732 for ; Wed, 6 Feb 2002 05:49:58 -0500 (EST) Received: from smtp2.libero.it (smtp2.libero.it [193.70.192.52]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id FAA16307 for ; Wed, 6 Feb 2002 05:49:54 -0500 (EST) Received: from libero.it (193.70.192.63) by smtp2.libero.it (6.0.040) id 3C5840440043CD5A for sipping@ietf.org; Wed, 6 Feb 2002 11:49:20 +0100 Date: Wed, 6 Feb 2002 11:49:20 +0100 Message-Id: MIME-Version: 1.0 Content-Type: text/plain; charset=iso-8859-1 From: "=?utf-8?Q?vitiellopas@libero.it?=" To: sipping@ietf.org X-XaM3-API-Version: 2.5 X-type: 0 X-SenderIP: 193.205.164.90 Content-Transfer-Encoding: 8bit X-MIME-Autoconverted: from base64 to 8bit by optimus.ietf.org id FAA22733 Subject: [Sipping] =?iso-8859-1?Q?dialog_in_conference_server?= Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 8bit hi, I'm implementing a conference server and I have the following question: in a presence server, refresh and NOTIFY requests have the same dialog of the SUBSCRIBE requests, particularly the same Call-ID... is it the same in a conference server, too? Is it mandatory or not? Regards, Pasquale _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Wed Feb 6 09:23:16 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id JAA20724 for ; Wed, 6 Feb 2002 09:23:16 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id JAA03948 for sipping-archive@odin.ietf.org; Wed, 6 Feb 2002 09:23:20 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id JAA03474; Wed, 6 Feb 2002 09:10:01 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id JAA03449 for ; Wed, 6 Feb 2002 09:10:00 -0500 (EST) Received: from cs.columbia.edu (cs.columbia.edu [128.59.16.20]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id JAA20322 for ; Wed, 6 Feb 2002 09:09:55 -0500 (EST) Received: from opus.cs.columbia.edu (opus.cs.columbia.edu [128.59.20.100]) by cs.columbia.edu (8.9.3/8.9.3) with ESMTP id JAA11415; Wed, 6 Feb 2002 09:09:55 -0500 (EST) Received: from cs.columbia.edu (pool-138-89-38-37.mad.east.verizon.net [138.89.38.37]) (authenticated bits=0) by opus.cs.columbia.edu (8.12.1/8.12.1) with ESMTP id g16E9rJ2025443 (version=TLSv1/SSLv3 cipher=RC4-MD5 bits=128 verify=NOT); Wed, 6 Feb 2002 09:09:54 -0500 (EST) Message-ID: <3C613AA6.EB6022FF@cs.columbia.edu> Date: Wed, 06 Feb 2002 09:16:06 -0500 From: Henning Schulzrinne X-Mailer: Mozilla 4.78 [en] (Windows NT 5.0; U) X-Accept-Language: en MIME-Version: 1.0 To: Mpierce1@aol.com CC: mhammer@cisco.com, taylor@nortelnetworks.com, sipping@ietf.org Subject: Re: [Sipping] Status Check: Subscriber Identification References: <14a.86a3762.29923b1a@aol.com> Content-Type: text/plain; charset=us-ascii Content-Transfer-Encoding: 7bit Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit However, this infrastructure may have nothing whatsoever to do with SIP or Internet telephony. For example, there is no reason you couldn't reuse the personal certs that companies, universities, Internet2 and various "public" CAs (Thawte, AT&T, Verisign, etc.) hand out, primarily for email use. All of these infrastructures have limitations, as has been amply discussed here and on the sip list, but the important point is that "infrastructure" may (and probably should not be) service-specific. > > It seems to me that the existance of an infrastucture must be assumed in order to handle authentication. It can't be done on a worldwide network supporting everyone without one. Yes, I know there are those who believe any assumed infrastructure is undesirable when one is talking about the Internet, but then, sometimes undesirable things are still necessary. > _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Wed Feb 6 10:34:50 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id KAA23444 for ; Wed, 6 Feb 2002 10:34:50 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id KAA08403 for sipping-archive@odin.ietf.org; Wed, 6 Feb 2002 10:34:53 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id KAA07050; Wed, 6 Feb 2002 10:19:42 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id KAA07013 for ; Wed, 6 Feb 2002 10:19:40 -0500 (EST) Received: from rtp-msg-core-1.cisco.com (rtp-msg-core-1.cisco.com [161.44.11.97]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id KAA22758 for ; Wed, 6 Feb 2002 10:19:37 -0500 (EST) Received: from cia.cisco.com (localhost [127.0.0.1]) by rtp-msg-core-1.cisco.com (8.11.3/8.9.1) with ESMTP id g16FJOF18825; Wed, 6 Feb 2002 10:19:24 -0500 (EST) Received: from MHAMMER-W2K.cisco.com (hrn2-dhcp-161-44-87-145.cisco.com [161.44.87.145]) by cia.cisco.com (Mirapoint) with ESMTP id AAP30404; Wed, 6 Feb 2002 10:13:54 -0500 (EST) Message-Id: <4.3.2.7.2.20020206101836.00b21250@cia.cisco.com> X-Sender: mhammer@cia.cisco.com X-Mailer: QUALCOMM Windows Eudora Version 4.3.2 Date: Wed, 06 Feb 2002 10:19:05 -0500 To: "wlw" From: Michael Hammer Subject: Re: [Sipping] Where can I find draft-ietf-sip-isup-mime-10.txt? Cc: In-Reply-To: <04ce01c1aeed$63fbe7e0$3ba210ac@sznssalbert> Mime-Version: 1.0 Content-Type: text/plain; charset="us-ascii"; format=flowed Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org If my tracking servers me, isn't that now RFC-3204? Mike At 05:05 PM 2/6/2002 +0800, wlw wrote: > _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Wed Feb 6 10:37:10 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id KAA23570 for ; Wed, 6 Feb 2002 10:37:10 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id KAA08645 for sipping-archive@odin.ietf.org; Wed, 6 Feb 2002 10:37:13 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id KAA07421; Wed, 6 Feb 2002 10:25:19 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id KAA07390 for ; Wed, 6 Feb 2002 10:25:17 -0500 (EST) Received: from rtp-msg-core-1.cisco.com (rtp-msg-core-1.cisco.com [161.44.11.97]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id KAA22998 for ; Wed, 6 Feb 2002 10:25:14 -0500 (EST) Received: from cia.cisco.com (localhost [127.0.0.1]) by rtp-msg-core-1.cisco.com (8.11.3/8.9.1) with ESMTP id g16FP1519219; Wed, 6 Feb 2002 10:25:01 -0500 (EST) Received: from MHAMMER-W2K.cisco.com (hrn2-dhcp-161-44-87-145.cisco.com [161.44.87.145]) by cia.cisco.com (Mirapoint) with ESMTP id AAP30465; Wed, 6 Feb 2002 10:19:31 -0500 (EST) Message-Id: <4.3.2.7.2.20020206102026.00b86148@cia.cisco.com> X-Sender: mhammer@cia.cisco.com X-Mailer: QUALCOMM Windows Eudora Version 4.3.2 Date: Wed, 06 Feb 2002 10:24:43 -0500 To: Henning Schulzrinne From: Michael Hammer Subject: Re: [Sipping] Status Check: Subscriber Identification Cc: Mpierce1@aol.com, taylor@nortelnetworks.com, sipping@ietf.org In-Reply-To: <3C613AA6.EB6022FF@cs.columbia.edu> References: <14a.86a3762.29923b1a@aol.com> Mime-Version: 1.0 Content-Type: text/plain; charset="us-ascii"; format=flowed Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Granted. But, then there is no guarantee that a user can be authenticated by anyone but the local service provider originally subscribed to. At that point forget mobility. The degree of universality of the supporting mechanism will restrict the universality of the service. Think about what roaming looked like in the first version of mobile networks. Mike At 09:16 AM 2/6/2002 -0500, Henning Schulzrinne wrote: >However, this infrastructure may have nothing whatsoever to do with SIP >or Internet telephony. For example, there is no reason you couldn't >reuse the personal certs that companies, universities, Internet2 and >various "public" CAs (Thawte, AT&T, Verisign, etc.) hand out, primarily >for email use. All of these infrastructures have limitations, as has >been amply discussed here and on the sip list, but the important point >is that "infrastructure" may (and probably should not be) >service-specific. > > > > > It seems to me that the existance of an infrastucture must be assumed > in order to handle authentication. It can't be done on a worldwide > network supporting everyone without one. Yes, I know there are those who > believe any assumed infrastructure is undesirable when one is talking > about the Internet, but then, sometimes undesirable things are still necessary. > > _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Wed Feb 6 11:01:57 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id LAA24460 for ; Wed, 6 Feb 2002 11:01:57 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id LAA10031 for sipping-archive@odin.ietf.org; Wed, 6 Feb 2002 11:02:00 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id KAA09046; Wed, 6 Feb 2002 10:46:18 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id KAA09015 for ; Wed, 6 Feb 2002 10:46:15 -0500 (EST) Received: from cs.columbia.edu (cs.columbia.edu [128.59.16.20]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id KAA23842 for ; Wed, 6 Feb 2002 10:46:11 -0500 (EST) Received: from opus.cs.columbia.edu (opus.cs.columbia.edu [128.59.20.100]) by cs.columbia.edu (8.9.3/8.9.3) with ESMTP id KAA22122; Wed, 6 Feb 2002 10:46:13 -0500 (EST) Received: from cs.columbia.edu (cta.cs.columbia.edu [128.59.19.46]) (authenticated bits=0) by opus.cs.columbia.edu (8.12.1/8.12.1) with ESMTP id g16FkAJ2029045 (version=TLSv1/SSLv3 cipher=RC4-MD5 bits=128 verify=NOT); Wed, 6 Feb 2002 10:46:11 -0500 (EST) Message-ID: <3C614FC1.6334FFD7@cs.columbia.edu> Date: Wed, 06 Feb 2002 10:46:09 -0500 From: Henning Schulzrinne Organization: Columbia University X-Mailer: Mozilla 4.76 [en] (Windows NT 5.0; U) X-Accept-Language: en MIME-Version: 1.0 To: Michael Hammer CC: Mpierce1@aol.com, taylor@nortelnetworks.com, sipping@ietf.org Subject: Re: [Sipping] Status Check: Subscriber Identification References: <14a.86a3762.29923b1a@aol.com> <4.3.2.7.2.20020206102026.00b86148@cia.cisco.com> Content-Type: text/plain; charset=us-ascii Content-Transfer-Encoding: 7bit Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit In many cases, what I care about is to identify a small subset of people. Except for malicious call tracing, I don't get much benefit of knowing that Alice Smith (as opposed to somebody impersonating Alice Smith), whom I have never met and don't know, calls. As long as identities are cheap, negative filtering doesn't work. For positive filtering, semi-closed user communities are not ideal, but workable. Michael Hammer wrote: > > Granted. But, then there is no guarantee that a user can be authenticated > by anyone but the local service provider originally subscribed to. At that > point forget mobility. The degree of universality of the supporting > mechanism will restrict the universality of the service. Think about what > roaming looked like in the first version of mobile networks. > _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Wed Feb 6 11:22:46 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id LAA25002 for ; Wed, 6 Feb 2002 11:22:46 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id LAA10948 for sipping-archive@odin.ietf.org; Wed, 6 Feb 2002 11:22:49 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id LAA10198; Wed, 6 Feb 2002 11:05:25 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id LAA10167 for ; Wed, 6 Feb 2002 11:05:22 -0500 (EST) Received: from rtp-msg-core-1.cisco.com (rtp-msg-core-1.cisco.com [161.44.11.97]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id LAA24520 for ; Wed, 6 Feb 2002 11:05:19 -0500 (EST) Received: from cia.cisco.com (localhost [127.0.0.1]) by rtp-msg-core-1.cisco.com (8.11.3/8.9.1) with ESMTP id g16G56b22963; Wed, 6 Feb 2002 11:05:07 -0500 (EST) Received: from MHAMMER-W2K.cisco.com (hrn2-dhcp-161-44-87-145.cisco.com [161.44.87.145]) by cia.cisco.com (Mirapoint) with ESMTP id AAP31068; Wed, 6 Feb 2002 10:59:36 -0500 (EST) Message-Id: <4.3.2.7.2.20020206110421.00bacf00@cia.cisco.com> X-Sender: mhammer@cia.cisco.com X-Mailer: QUALCOMM Windows Eudora Version 4.3.2 Date: Wed, 06 Feb 2002 11:04:48 -0500 To: Henning Schulzrinne From: Michael Hammer Subject: Re: [Sipping] Status Check: Subscriber Identification Cc: Mpierce1@aol.com, taylor@nortelnetworks.com, sipping@ietf.org In-Reply-To: <3C614FC1.6334FFD7@cs.columbia.edu> References: <14a.86a3762.29923b1a@aol.com> <4.3.2.7.2.20020206102026.00b86148@cia.cisco.com> Mime-Version: 1.0 Content-Type: text/plain; charset="us-ascii"; format=flowed Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org OK, but what about billing and regulatory requirements? At 10:46 AM 2/6/2002 -0500, Henning Schulzrinne wrote: >In many cases, what I care about is to identify a small subset of >people. Except for malicious call tracing, I don't get much benefit of >knowing that Alice Smith (as opposed to somebody impersonating Alice >Smith), whom I have never met and don't know, calls. As long as >identities are cheap, negative filtering doesn't work. For positive >filtering, semi-closed user communities are not ideal, but workable. > >Michael Hammer wrote: > > > > Granted. But, then there is no guarantee that a user can be authenticated > > by anyone but the local service provider originally subscribed to. At that > > point forget mobility. The degree of universality of the supporting > > mechanism will restrict the universality of the service. Think about what > > roaming looked like in the first version of mobile networks. > > _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Wed Feb 6 12:21:02 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id MAA26209 for ; Wed, 6 Feb 2002 12:21:01 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id MAA15535 for sipping-archive@odin.ietf.org; Wed, 6 Feb 2002 12:21:05 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id MAA14649; Wed, 6 Feb 2002 12:06:53 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id MAA14619 for ; Wed, 6 Feb 2002 12:06:47 -0500 (EST) Received: from dnsmx1pya.telcordia.com (dnsmx1pya.telcordia.com [128.96.20.31]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id MAA25814 for ; Wed, 6 Feb 2002 12:06:38 -0500 (EST) From: ysong@telcordia.com Received: from notes900.cc.telcordia.com (notes900.cc.telcordia.com [128.96.79.7]) by dnsmx1pya.telcordia.com (8.9.3/8.9.3) with ESMTP id LAA04793; Wed, 6 Feb 2002 11:58:24 -0500 (EST) To: sipping@ietf.org, "Peterson, Jon" X-Mailer: Lotus Notes Release 5.0.5 September 22, 2000 Message-ID: Date: Wed, 6 Feb 2002 11:59:17 -0500 X-MIMETrack: Serialize by Router on notes900/Telcordia(Release 5.0.6a |January 17, 2001) at 02/06/2002 11:59:18 AM MIME-Version: 1.0 Content-type: text/plain; charset=us-ascii Subject: [Sipping] forking for ISUP to SIP call Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Hi, I have a question on forking for an ISUP originated call that is terminating to a SIP device. Suppose the following scenario. PSTN ---- IAM----> SIP-GW ----- INVITE -----> SIP network PSTN <----ACM---- SIP-GW <----- 180----- SIP-device-1 PSTN <----???---- SIP-GW <----- 183----- SIP-device-2 PSTN <----???---- SIP-GW <----- 180----- SIP-device-3 How are the multiple 18xs for different call legs mapped to ISUP? Thanks, YoungSun _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Wed Feb 6 12:26:23 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id MAA26346 for ; Wed, 6 Feb 2002 12:26:23 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id MAA15683 for sipping-archive@odin.ietf.org; Wed, 6 Feb 2002 12:26:26 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id MAA15125; Wed, 6 Feb 2002 12:14:34 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id MAA15094 for ; Wed, 6 Feb 2002 12:14:32 -0500 (EST) Received: from cs.columbia.edu (cs.columbia.edu [128.59.16.20]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id MAA25987 for ; Wed, 6 Feb 2002 12:14:27 -0500 (EST) Received: from opus.cs.columbia.edu (opus.cs.columbia.edu [128.59.20.100]) by cs.columbia.edu (8.9.3/8.9.3) with ESMTP id MAA02169; Wed, 6 Feb 2002 12:14:22 -0500 (EST) Received: from cs.columbia.edu (cta.cs.columbia.edu [128.59.19.46]) (authenticated bits=0) by opus.cs.columbia.edu (8.12.1/8.12.1) with ESMTP id g16HELJ2002210 (version=TLSv1/SSLv3 cipher=RC4-MD5 bits=128 verify=NOT); Wed, 6 Feb 2002 12:14:22 -0500 (EST) Message-ID: <3C61646C.92D21A94@cs.columbia.edu> Date: Wed, 06 Feb 2002 12:14:20 -0500 From: Henning Schulzrinne Organization: Columbia University X-Mailer: Mozilla 4.76 [en] (Windows NT 5.0; U) X-Accept-Language: en MIME-Version: 1.0 To: Michael Hammer CC: Mpierce1@aol.com, taylor@nortelnetworks.com, sipping@ietf.org Subject: Re: [Sipping] Status Check: Subscriber Identification References: <14a.86a3762.29923b1a@aol.com> <4.3.2.7.2.20020206102026.00b86148@cia.cisco.com> <4.3.2.7.2.20020206110421.00bacf00@cia.cisco.com> Content-Type: text/plain; charset=us-ascii Content-Transfer-Encoding: 7bit Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit Regulatory issues are constantly changing (and it's not clear exactly which apply to different services), so let me address the roaming aspect of billing. (Local billing is obviously easier - your SP hands you a suitably signed credential.) I would imagine a model where if Alice, as customer of SP A, move into B's territory, B sees that I'm really A's customer. Assuming A has a relationship with B (otherwise, you don't get roaming, regardless of authentication), B sends a request to A (with its signature) saying "Your customer Alice wants to roam in my network. Here's her signed admission request. Is this legit, has she paid her bills and do you trust her enough to allow her to roam?". A then either agrees or not. If A does, A will pay B, and collect money from Alice. Thus, no need for B to know Alice as an individual. None of this requires more technology than what's available off-the-shelf today. Substitute various clearing houses to avoid N^2 scaling issues. Michael Hammer wrote: > > OK, but what about billing and regulatory requirements? > _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Wed Feb 6 13:38:55 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id NAA27990 for ; Wed, 6 Feb 2002 13:38:54 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id NAA19539 for sipping-archive@odin.ietf.org; Wed, 6 Feb 2002 13:38:57 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id NAA18661; Wed, 6 Feb 2002 13:25:27 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id NAA18630 for ; Wed, 6 Feb 2002 13:25:24 -0500 (EST) Received: from rtp-msg-core-1.cisco.com (rtp-msg-core-1.cisco.com [161.44.11.97]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id NAA27632 for ; Wed, 6 Feb 2002 13:25:21 -0500 (EST) Received: from cia.cisco.com (localhost [127.0.0.1]) by rtp-msg-core-1.cisco.com (8.11.3/8.9.1) with ESMTP id g16IP7x03991; Wed, 6 Feb 2002 13:25:07 -0500 (EST) Received: from MHAMMER-W2K.cisco.com (hrn2-dhcp-161-44-87-145.cisco.com [161.44.87.145]) by cia.cisco.com (Mirapoint) with ESMTP id AAP32841; Wed, 6 Feb 2002 13:19:37 -0500 (EST) Message-Id: <4.3.2.7.2.20020206132158.03a2a038@cia.cisco.com> X-Sender: mhammer@cia.cisco.com X-Mailer: QUALCOMM Windows Eudora Version 4.3.2 Date: Wed, 06 Feb 2002 13:24:47 -0500 To: ysong@telcordia.com From: Michael Hammer Subject: Re: [Sipping] forking for ISUP to SIP call Cc: sipping@ietf.org, "Peterson, Jon" In-Reply-To: Mime-Version: 1.0 Content-Type: text/plain; charset="us-ascii"; format=flowed Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org YoungSun, As the SIP network introduced the forking, shouldn't it resolve the forking on behalf of the PSTN caller? The SIP side of the SIP GW is the client that initiated the SIP call. That agent should resolve the issue and not pass it on to the PSTN. This is not any different than if a switch in the PSTN had invoked a service such as sequential or parallel hunt groups. Mike At 11:59 AM 2/6/2002 -0500, ysong@telcordia.com wrote: >Hi, > >I have a question on forking for an ISUP originated call that is >terminating to a SIP device. >Suppose the following scenario. > >PSTN ---- IAM----> SIP-GW ----- INVITE -----> SIP network >PSTN <----ACM---- SIP-GW <----- 180----- SIP-device-1 >PSTN <----???---- SIP-GW <----- 183----- SIP-device-2 >PSTN <----???---- SIP-GW <----- 180----- SIP-device-3 > >How are the multiple 18xs for different call legs mapped to ISUP? > >Thanks, >YoungSun > > >_______________________________________________ >Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping >This list is for NEW development of the application of SIP >Use sip-implementors@cs.columbia.edu for questions on current sip >Use sip@ietf.org for new developments of core SIP _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Wed Feb 6 16:39:32 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id QAA03181 for ; Wed, 6 Feb 2002 16:39:31 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id QAA28448 for sipping-archive@odin.ietf.org; Wed, 6 Feb 2002 16:39:36 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id QAA27194; Wed, 6 Feb 2002 16:23:16 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id QAA27162 for ; Wed, 6 Feb 2002 16:23:14 -0500 (EST) Received: from mail2.microsoft.com (mail2.microsoft.com [131.107.3.124]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id QAA02603 for ; Wed, 6 Feb 2002 16:23:09 -0500 (EST) Received: from INET-VRS-02.redmond.corp.microsoft.com ([157.54.8.110]) by mail2.microsoft.com with Microsoft SMTPSVC(5.0.2195.4617); Wed, 6 Feb 2002 13:22:27 -0800 Received: from 157.54.6.197 by INET-VRS-02.redmond.corp.microsoft.com (InterScan E-Mail VirusWall NT); Wed, 06 Feb 2002 13:22:41 -0800 Received: from red-imc-04.redmond.corp.microsoft.com ([157.54.2.168]) by inet-hub-06.redmond.corp.microsoft.com with Microsoft SMTPSVC(5.0.2195.2966); Wed, 6 Feb 2002 13:22:26 -0800 Received: from win-imc-01.wingroup.windeploy.ntdev.microsoft.com ([157.54.0.39]) by red-imc-04.redmond.corp.microsoft.com with Microsoft SMTPSVC(5.0.2195.2966); Wed, 6 Feb 2002 13:22:26 -0800 Received: from win-msg-02.wingroup.windeploy.ntdev.microsoft.com ([157.54.0.134]) by win-imc-01.wingroup.windeploy.ntdev.microsoft.com with Microsoft SMTPSVC(6.0.3588.0); Wed, 6 Feb 2002 13:20:06 -0800 X-MimeOLE: Produced By Microsoft Exchange V6.0.6132.0 content-class: urn:content-classes:message MIME-Version: 1.0 Content-Type: text/plain; charset="us-ascii" Subject: RE: [Sipping] Status Check: Subscriber Identification Date: Wed, 6 Feb 2002 13:20:05 -0800 Message-ID: Thread-Topic: [Sipping] Status Check: Subscriber Identification Thread-Index: AcGvMW+v1ON1q3F0QIWDT45N+P2reQAIl1IA From: "Christian Huitema" To: "Henning Schulzrinne" , "Michael Hammer" Cc: , , X-OriginalArrivalTime: 06 Feb 2002 21:20:06.0425 (UTC) FILETIME=[0C75E090:01C1AF54] Content-Transfer-Encoding: 8bit X-MIME-Autoconverted: from quoted-printable to 8bit by optimus.ietf.org id QAA27163 Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 8bit > I would imagine a model where if Alice, as customer of SP A, move into > B's territory, B sees that I'm really A's customer. Assuming A has a > relationship with B (otherwise, you don't get roaming, regardless of > authentication), B sends a request to A (with its signature) saying > "Your customer Alice wants to roam in my network. Here's her signed > admission request. Is this legit, has she paid her bills and do you > trust her enough to allow her to roam?". A then either agrees or not. If > A does, A will pay B, and collect money from Alice. Thus, no need for B > to know Alice as an individual. None of this requires more technology > than what's available off-the-shelf today. Why should this be SIP's problem? There are already solutions for roaming at the network level, e.g. combinations of media access control (PPP, 802.1x) and AAA protocols (Radius). Money is actually being collected that way today. When Alice roams into B, she only gets local IP service because she can perform this authentication. Once she has IP access, she can just as well REGISTER her new location with A's SIP proxy -- using IPSEC or TLS if needed. -- Christian Huitema _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Wed Feb 6 16:42:26 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id QAA03282 for ; Wed, 6 Feb 2002 16:42:26 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id QAA28627 for sipping-archive@odin.ietf.org; Wed, 6 Feb 2002 16:42:30 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id QAA28249; Wed, 6 Feb 2002 16:33:42 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id QAA28219 for ; Wed, 6 Feb 2002 16:33:40 -0500 (EST) Received: from cs.columbia.edu (cs.columbia.edu [128.59.16.20]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id QAA02990 for ; Wed, 6 Feb 2002 16:33:35 -0500 (EST) Received: from opus.cs.columbia.edu (opus.cs.columbia.edu [128.59.20.100]) by cs.columbia.edu (8.9.3/8.9.3) with ESMTP id QAA01337; Wed, 6 Feb 2002 16:33:26 -0500 (EST) Received: from cs.columbia.edu (cta.cs.columbia.edu [128.59.19.46]) (authenticated bits=0) by opus.cs.columbia.edu (8.12.1/8.12.1) with ESMTP id g16LXPJ2011729 (version=TLSv1/SSLv3 cipher=RC4-MD5 bits=128 verify=NOT); Wed, 6 Feb 2002 16:33:25 -0500 (EST) Message-ID: <3C61A124.901F4DF6@cs.columbia.edu> Date: Wed, 06 Feb 2002 16:33:24 -0500 From: Henning Schulzrinne Organization: Columbia University X-Mailer: Mozilla 4.76 [en] (Windows NT 5.0; U) X-Accept-Language: en MIME-Version: 1.0 To: Christian Huitema CC: Michael Hammer , Mpierce1@aol.com, taylor@nortelnetworks.com, sipping@ietf.org Subject: Re: [Sipping] Status Check: Subscriber Identification References: Content-Type: text/plain; charset=us-ascii Content-Transfer-Encoding: 7bit Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit I should have made this clear, but as I said earlier, this is not a SIP-specific problem and thus the solution should not be SIP-specific. I don't think RADIUS does cross-domain authentication yet, but I haven't followed the AAA stuff closely. Christian Huitema wrote: > > > I would imagine a model where if Alice, as customer of SP A, move into > > B's territory, B sees that I'm really A's customer. Assuming A has a > > relationship with B (otherwise, you don't get roaming, regardless of > > authentication), B sends a request to A (with its signature) saying > > "Your customer Alice wants to roam in my network. Here's her signed > > admission request. Is this legit, has she paid her bills and do you > > trust her enough to allow her to roam?". A then either agrees or not. > If > > A does, A will pay B, and collect money from Alice. Thus, no need for > B > > to know Alice as an individual. None of this requires more technology > > than what's available off-the-shelf today. > > Why should this be SIP's problem? There are already solutions for > roaming at the network level, e.g. combinations of media access control > (PPP, 802.1x) and AAA protocols (Radius). Money is actually being > collected that way today. When Alice roams into B, she only gets local > IP service because she can perform this authentication. Once she has IP > access, she can just as well REGISTER her new location with A's SIP > proxy -- using IPSEC or TLS if needed. > > -- Christian Huitema _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Wed Feb 6 20:14:59 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id UAA06055 for ; Wed, 6 Feb 2002 20:14:59 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id UAA07778 for sipping-archive@odin.ietf.org; Wed, 6 Feb 2002 20:15:01 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id TAA06888; Wed, 6 Feb 2002 19:58:01 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id TAA06858 for ; Wed, 6 Feb 2002 19:57:59 -0500 (EST) Received: from bdsl.66.12.12.130.gte.net (bdsl.66.12.12.130.gte.net [66.12.12.130]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id TAA05890 for ; Wed, 6 Feb 2002 19:57:56 -0500 (EST) Received: from txdwillis (bdsl.greycouncil.com [127.0.0.1]) by bdsl.66.12.12.130.gte.net (8.11.6/8.11.6) with SMTP id g170qkH01394; Wed, 6 Feb 2002 18:52:47 -0600 Message-ID: <000f01c1af71$e46f4ff0$9219a8c0@dfw.dynamicsoft.com> From: "Dean Willis" To: "Tom-PT Taylor" Cc: References: <4D79C746863DD51197690002A52CDA000102AF3D@zcard0kc.ca.nortel.com> Subject: Re: [Sipping] RE: calling number blocking at sip/isup gateway Date: Wed, 6 Feb 2002 12:11:59 -0600 MIME-Version: 1.0 Content-Type: text/plain; charset="iso-8859-1" Content-Transfer-Encoding: 7bit X-Priority: 3 X-MSMail-Priority: Normal X-Mailer: Microsoft Outlook Express 6.00.2600.0000 X-MimeOLE: Produced By Microsoft MimeOLE V6.00.2600.0000 Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit I think we call that "authentication", and any UA or proxy MAY issue an appropriate 401/407 to request authentication. Bear in mind that there are at least four identies in every dialog: 1) The "From:' party 2) The "To:" party 3) The requesting party 4) The fiscally responsible party Role 2 can be further subdivied into: 2a: the targeted party 2b: the responding party in traditional telephone calls, #1, #3, and #4 are the same. For collect calls, the caller has roles 1 and 3, and the called has roles 2 and 4. In third-party control, you can get an even wider range of scenarios. SIP, as we currently know it, allows declaration of roles 1 (From:), 2 (To:), and 3 (authenticated identity). We don't have anything for role #4, but let that be handled by applications. There are proposed extensions to role #1 and controlling of disclosure of the identity associated with that role (Privacy draft, which defines Remote party ID). This draft also defines a partial solution for 2, essentially what I called 2b. There is some discussion in 3GPP about the need to address 2a. We may need to find a way to express transitive relationships between roles 3 and 1 -- "by Joe on behalf of Sally", but I don't believe there's been any expression of that yet in IETF. Some of the responses to your question are basically asking what the mechanism for transitively expressing #3 is. Is that what you're asking? My flip response to the question as originally worded would have been "I don't know about the other guys, but I authenticate the requester." But I'm not sure that's actually asnwering the question . . . -- Dean ----- Original Message ----- From: "Tom-PT Taylor" To: "'Michael Hammer'" Cc: "'Juha Heinanen'" ; ; Sent: Tuesday, January 29, 2002 2:43 PM Subject: RE: [Sipping] RE: calling number blocking at sip/isup gateway > "Subscriber" implies that the SIP user has subscribed to some service > somewhere. I suppose this is a reasonable assumption for SIP-ISUP > interworking, though I could imagine scenarios where it might not apply. > Very well, the first SIP requirement becomes: > > (1) SIP must provide a way for the gateway to determine the authenticated > identity of the SIP subscriber in whose name the call is being made. > > > > -----Original Message----- > From: Michael Hammer [mailto:mhammer@cisco.com] > Sent: Tuesday, January 29, 2002 2:09 PM > To: Taylor, Tom-PT [CAR:B800:EXCH] > Cc: 'Juha Heinanen'; 'Mpierce1@aol.com'; sipping@ietf.org > Subject: RE: [Sipping] RE: calling number blocking at sip/isup gateway > > > Tom, > > Given the wireless world and the possibility that there is a distinction > between the host platform and the subscriber/user, wouldn't it be better to > identify the subscriber? There is also technology that allows a user to > log into and personalize terminals for their use. Or in some cases, the > user, the subscriber, and the terminal could all be different. > > Presumably, the SIP UA that originates the call has an identity and a > subscription with a service provider that authorizes the call. Could we > focus terminology on the subscriber as that may be all that is known to the > network? > > Mike > > > At 01:42 PM 1/29/2002 -0500, Tom-PT Taylor wrote: > >Well, the overall requirement is clear, but I think it breaks down into two > >different sets of requirements, on SIP and on the SIP-PSTN gateway. > > > >For SIP: > > > >(1) A requirement to provide some means whereby a PSTN gateway can obtain > >the authenticated identity of the SIP host. I say host rather than user > >because I can't see regulators requiring SIP to provide more than the PSTN > >does now. > > > >(2) Another requirement to provide the means whereby a PSTN gateway can > >determine if the SIP user wishes to restrict the presentation of his/her > SIP > >identity. > > > > > >At the gateway the requirements will be the same whether the signalling > >protocol is SIP, H.323, or something else. > > > >(1) A requirement on the gateway to provide the information within the PSTN > >signalling which satisfies regulatory requirements. > > > >(2) A requirement to report the authenticated identity of the IP host in > >specific cases e.g. for malicious call trace. > > > >(3) A requirement to support billing arrangements. > > > >(4) A requirement to extract appropriate display information and pass it in > >ISUP signalling if the user has not requested privacy. > > > > > > > >-----Original Message----- > >From: Juha Heinanen [mailto:jh@lohi.eng.song.fi] > >Sent: Tuesday, January 29, 2002 11:17 AM > >To: Taylor, Tom-PT [CAR:B800:EXCH] > >Cc: 'Mpierce1@aol.com'; sipping@ietf.org > >Subject: [Sipping] RE: calling number blocking at sip/isup gateway > > > > > >Tom-PT Taylor writes: > > > > > What about UACs > > > which do not have telephone numbers assigned? Will they be barred from > >PSTN > > > access, or should we mandate that the gateway assign some sort of > number > > > which it alone can cross-correlate to the original user@domain if need > be > > > for malicious call trace, etc.? > > > >yes, either way. in finland we are not allowed to gateway any call to > >pstn unless it has a valid from number (either of the user or the > >gateway) and if we place the gw's number there, we must be able to > >answer the question, which of the gw's subscribers actually made the > >call. in addition, the law requires the gw to mark the presentation of > >the from number of all calls from the subscriber as "restricted" if > >asked so by the subscriber. > > > >if sip doesn't have means to implement the above, then there is no > >gatewaying of sip calls to pstn in finland and there is no business case > >for us until sip starts to take over the world, which will take a few > >years at least. all i want is for a proxy a way to mark the from number > >of the invite as "restricted" without touching the from header AND i > >want the chosen way to be clearly documented in the sip-isup i-d so that > >i don't have to tell each sip/ss7/proxy vendor separately, how it is > >done. > > > >what method you choose to do it, is not my business, since i'm just > >a stupid user of the sip protocol. > > > >-- juha > > > > > >_______________________________________________ > >Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > >This list is for NEW development of the application of SIP > >Use sip-implementors@cs.columbia.edu for questions on current sip > >Use sip@ietf.org for new developments of core SIP > > > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP > Use sip-implementors@cs.columbia.edu for questions on current sip > Use sip@ietf.org for new developments of core SIP > > _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Thu Feb 7 00:48:37 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id AAA12410 for ; Thu, 7 Feb 2002 00:48:37 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id AAA20591 for sipping-archive@odin.ietf.org; Thu, 7 Feb 2002 00:48:38 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id AAA20281; Thu, 7 Feb 2002 00:35:01 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id AAA20249 for ; Thu, 7 Feb 2002 00:34:50 -0500 (EST) Received: from penguin-ext.wise.edt.ericsson.se (penguin-ext.wise.edt.ericsson.se [194.237.142.110]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id AAA12214 for ; Thu, 7 Feb 2002 00:34:47 -0500 (EST) Received: from fogerty.lmf.ericsson.se (fogerty.lmf.ericsson.se [131.160.11.6]) by penguin.wise.edt.ericsson.se (8.11.0/8.11.0/WIREfire-1.3) with ESMTP id g175YfC06088; Thu, 7 Feb 2002 06:34:41 +0100 (MET) Received: from lmf.ericsson.se (E005004B52C74.lmf.ericsson.se [131.160.30.135]) by fogerty.lmf.ericsson.se (8.12.1/8.12.1/lmf.8.12.1.jcs) with ESMTP id g175Yer6029942; Thu, 7 Feb 2002 07:34:40 +0200 (EET) Message-ID: <3C6211E3.85CB63C8@lmf.ericsson.se> Date: Thu, 07 Feb 2002 07:34:27 +0200 From: Christer Holmberg X-Mailer: Mozilla 4.61 [en] (WinNT; I) X-Accept-Language: en MIME-Version: 1.0 To: Michael Hammer CC: ysong@telcordia.com, sipping@ietf.org, "Peterson, Jon" Subject: Re: [Sipping] forking for ISUP to SIP call References: <4.3.2.7.2.20020206132158.03a2a038@cia.cisco.com> Content-Type: text/plain; charset=us-ascii Content-Transfer-Encoding: 7bit Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit Hi Michael, The proxy which "introduced" the forking WILL forward ALL PROVISIONAL responses to the SIP GW, but it should only forward ONE final response. Regards, Christer Holmberg Ericsson Finland Michael Hammer wrote: > > YoungSun, > > As the SIP network introduced the forking, shouldn't it resolve the forking > on behalf of the PSTN caller? The SIP side of the SIP GW is the client > that initiated the SIP call. That agent should resolve the issue and not > pass it on to the PSTN. This is not any different than if a switch in the > PSTN had invoked a service such as sequential or parallel hunt groups. > > Mike > > At 11:59 AM 2/6/2002 -0500, ysong@telcordia.com wrote: > > >Hi, > > > >I have a question on forking for an ISUP originated call that is > >terminating to a SIP device. > >Suppose the following scenario. > > > >PSTN ---- IAM----> SIP-GW ----- INVITE -----> SIP network > >PSTN <----ACM---- SIP-GW <----- 180----- SIP-device-1 > >PSTN <----???---- SIP-GW <----- 183----- SIP-device-2 > >PSTN <----???---- SIP-GW <----- 180----- SIP-device-3 > > > >How are the multiple 18xs for different call legs mapped to ISUP? > > > >Thanks, > >YoungSun > > > > > >_______________________________________________ > >Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > >This list is for NEW development of the application of SIP > >Use sip-implementors@cs.columbia.edu for questions on current sip > >Use sip@ietf.org for new developments of core SIP > > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP > Use sip-implementors@cs.columbia.edu for questions on current sip > Use sip@ietf.org for new developments of core SIP _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Thu Feb 7 01:13:22 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id BAA12736 for ; Thu, 7 Feb 2002 01:13:22 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id BAA29478 for sipping-archive@odin.ietf.org; Thu, 7 Feb 2002 01:13:21 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id AAA20871; Thu, 7 Feb 2002 00:58:17 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id AAA20843 for ; Thu, 7 Feb 2002 00:58:15 -0500 (EST) Received: from lohi.eng.song.fi (lohi.eng.song.fi [195.10.149.18]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id AAA12587 for ; Thu, 7 Feb 2002 00:58:13 -0500 (EST) Received: from lohi.eng.song.fi (localhost [127.0.0.1]) by lohi.eng.song.fi (8.12.1/8.12.1/Debian -5) with ESMTP id g175vpio014270; Thu, 7 Feb 2002 07:57:51 +0200 Received: (from jh@localhost) by lohi.eng.song.fi (8.12.1/8.12.1/Debian -5) id g175vorS014266; Thu, 7 Feb 2002 07:57:50 +0200 From: Juha Heinanen MIME-Version: 1.0 Content-Type: text/plain; charset=us-ascii Content-Transfer-Encoding: 7bit Message-ID: <15458.5982.309753.795324@lohi.eng.song.fi> Date: Thu, 7 Feb 2002 07:57:50 +0200 To: Henning Schulzrinne Cc: Christian Huitema , Michael Hammer , Mpierce1@aol.com, taylor@nortelnetworks.com, sipping@ietf.org Subject: Re: [Sipping] Status Check: Subscriber Identification In-Reply-To: <3C61A124.901F4DF6@cs.columbia.edu> References: <3C61A124.901F4DF6@cs.columbia.edu> X-Mailer: VM 6.97 under Emacs 20.7.2 Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit Henning Schulzrinne writes: > I should have made this clear, but as I said earlier, this is not a > SIP-specific problem and thus the solution should not be SIP-specific. I > don't think RADIUS does cross-domain authentication yet, but I haven't > followed the AAA stuff closely. there are several roaming service providers, such as gric and ipass, that proxy radius requests between domains. -- juha _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Thu Feb 7 03:22:52 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id DAA22395 for ; Thu, 7 Feb 2002 03:22:52 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id DAA06439 for sipping-archive@odin.ietf.org; Thu, 7 Feb 2002 03:22:54 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id DAA05479; Thu, 7 Feb 2002 03:03:25 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id DAA05407 for ; Thu, 7 Feb 2002 03:03:20 -0500 (EST) Received: from zcars0m9.ca.nortel.com (zcars0m9.nortelnetworks.com [47.129.242.157]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id DAA22160 for ; Thu, 7 Feb 2002 03:03:17 -0500 (EST) Received: from zcars04f.ca.nortel.com (zcars04f.ca.nortel.com [47.129.242.57]) by zcars0m9.ca.nortel.com (Switch-2.2.0/Switch-2.2.0) with ESMTP id g1782mH11587 for ; Thu, 7 Feb 2002 03:02:48 -0500 (EST) Received: from zcard015.ca.nortel.com (zcard015.ca.nortel.com [47.129.30.7]) by zcars04f.ca.nortel.com (Switch-2.2.0/Switch-2.2.0) with ESMTP id g1782k323363 for ; Thu, 7 Feb 2002 03:02:47 -0500 (EST) Received: by zcard015.ca.nortel.com with Internet Mail Service (5.5.2653.19) id <1J3303VR>; Thu, 7 Feb 2002 03:02:48 -0500 Message-ID: <4D79C746863DD51197690002A52CDA00016A88C0@zcard0kc.ca.nortel.com> From: "Tom-PT Taylor" To: Flemming Andreasen Cc: sipping@ietf.org Subject: RE: [Sipping] Status Check: Subscriber Identification Date: Thu, 7 Feb 2002 03:02:49 -0500 X-Mailer: Internet Mail Service (5.5.2653.19) Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org I mean authenticate. -----Original Message----- From: Flemming Andreasen [mailto:fandreas@cisco.com] Sent: Tuesday, February 05, 2002 7:36 PM To: Taylor, Tom-PT [CAR:B800:EXCH] Cc: sipping@ietf.org Subject: Re: [Sipping] Status Check: Subscriber Identification Do you mean authenticate the subscriber or simply provide an identification of the subscriber that subsequent hops can use ? -- Flemming Tom-PT Taylor wrote: > > > At one point in the thread "calling number blocking at sip/isup > gateway" a few of us seemed to agree that it must be possible for the > gateway to identify the subscriber in whose name service is being > requested. What if any is the intended SIP mechanism to meet this > requirement? > > Tom Taylor > taylor@nortelnetworks.com > Ph. +1 613 736 0961 (ESN 396 1490) > -- Flemming Andreasen Cisco Systems _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Thu Feb 7 03:33:34 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id DAA22570 for ; Thu, 7 Feb 2002 03:33:33 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id DAA07118 for sipping-archive@odin.ietf.org; Thu, 7 Feb 2002 03:33:36 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id DAA05470; Thu, 7 Feb 2002 03:03:25 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id DAA05414 for ; Thu, 7 Feb 2002 03:03:21 -0500 (EST) Received: from zcars0m9.ca.nortel.com (zcars0m9.nortelnetworks.com [47.129.242.157]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id DAA22161 for ; Thu, 7 Feb 2002 03:03:18 -0500 (EST) Received: from zcars04e.ca.nortel.com (zcars04e.ca.nortel.com [47.129.242.56]) by zcars0m9.ca.nortel.com (Switch-2.2.0/Switch-2.2.0) with ESMTP id g1782mH11589; Thu, 7 Feb 2002 03:02:48 -0500 (EST) Received: from zcard015.ca.nortel.com (zcard015.ca.nortel.com [47.129.30.7]) by zcars04e.ca.nortel.com (Switch-2.2.0/Switch-2.2.0) with ESMTP id g1782kg26554; Thu, 7 Feb 2002 03:02:46 -0500 (EST) Received: by zcard015.ca.nortel.com with Internet Mail Service (5.5.2653.19) id <1J3303VQ>; Thu, 7 Feb 2002 03:02:48 -0500 Message-ID: <4D79C746863DD51197690002A52CDA00016A88C1@zcard0kc.ca.nortel.com> From: "Tom-PT Taylor" To: "Peterson, Jon" , "'Flemming Andreasen'" , Sean Olson Cc: sipping@ietf.org Subject: RE: [Sipping] Status Check: Subscriber Identification Date: Thu, 7 Feb 2002 03:02:50 -0500 X-Mailer: Internet Mail Service (5.5.2653.19) Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org I had this in mind, but was confused by the availability of the Remote-Party-Id. What we write into the interworking specification at SG 11 may be of the order of: "an authenticated subscriber ID is obtained by means beyond the scope of this specification", or I may find that people insist on more details. I'll try to get away with the minimum. -----Original Message----- From: Peterson, Jon [mailto:jon.peterson@neustar.biz] Sent: Wednesday, February 06, 2002 4:10 AM To: 'Flemming Andreasen'; Sean Olson Cc: Taylor, Tom-PT [CAR:B800:EXCH]; sipping@ietf.org Subject: RE: [Sipping] Status Check: Subscriber Identification There are SIP mechanisms that allow servers to ascertain the identities of remote users - just to pick one out of a hat, let's look at the Digest authentication scheme. When the gateway receives an INVITE, it sends a challenge (401) in response. The user retries the request with a Digest username and password (corresponding to its account at this gateway operator's network). This username and password could be used by the gateway to retrieve a particular phone number and/or caller name corresponding to 'the subscriber in whose name service is being requested', returning to Tom's original question. The gateway could then use this information to populate a CgPN field and whatever other ISUP fields it liked. It might even overwrite any telephone number that happened to appear in the From field of the request with its own data. Or, the gateway could merely retrieve an account number for the user, and record this number in some sort of event block/CDR that it cuts for this call - in this instance it could send a common 'gateway number' rather than an individual subscriber's number in the CgPN, if it were so inclined. If a malicious trace were subsequently required, the gateway operator could find the event block in question and provide the user's account information to the authorities. The point is, none of this requires any changes to SIP (or the SIP-ISUP mappings) that I'm aware of. Jon Peterson NeuStar, Inc. > -----Original Message----- > From: Flemming Andreasen [mailto:fandreas@cisco.com] > Sent: Tuesday, February 05, 2002 12:08 PM > To: Sean Olson > Cc: Tom-PT Taylor; sipping@ietf.org > Subject: Re: [Sipping] Status Check: Subscriber Identification > > > Right, however that wasn't my question. The privacy draft, and in > particular the Remote-Party-Id is intended to provide the identity > information, however it does not specify the authentication > mechanism by > which the identity is determined, which is why I asked Tom to clarify. > > -- Flemming > > Sean Olson wrote: > > > Given the intended usage of this identifier, a > > non-authenticated subscriber ID would not be > > very useful. > > > > /sean > > > > --- Flemming Andreasen wrote: > > > Do you mean authenticate the subscriber or simply > > > provide an > > > identification of the subscriber that subsequent > > > hops can use ? > > > > > > -- Flemming > > > > > > Tom-PT Taylor wrote: > > > > > > > > > > > > > > > At one point in the thread "calling number > > > blocking at sip/isup > > > > gateway" a few of us seemed to agree that it must > > > be possible for the > > > > gateway to identify the subscriber in whose name > > > service is being > > > > requested. What if any is the intended SIP > > > mechanism to meet this > > > > requirement? > > > > > > > > Tom Taylor > > > > taylor@nortelnetworks.com > > > > Ph. +1 613 736 0961 (ESN 396 1490) > > > > > > > > > > -- > > > Flemming Andreasen > > > Cisco Systems > > > > > > > > > > > > _______________________________________________ > > > Sipping mailing list > > > https://www1.ietf.org/mailman/listinfo/sipping > > > This list is for NEW development of the application > > > of SIP > > > Use sip-implementors@cs.columbia.edu for questions > > > on current sip > > > Use sip@ietf.org for new developments of core SIP > > > > ===== > > Sean Olson > > > > __________________________________________________ > > Do You Yahoo!? > > Send FREE Valentine eCards with Yahoo! Greetings! > > http://greetings.yahoo.com > > -- > Flemming Andreasen > Cisco Systems > > > > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP > Use sip-implementors@cs.columbia.edu for questions on current sip > Use sip@ietf.org for new developments of core SIP > _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Thu Feb 7 09:14:13 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id JAA27339 for ; Thu, 7 Feb 2002 09:14:13 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id JAA21705 for sipping-archive@odin.ietf.org; Thu, 7 Feb 2002 09:14:13 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id JAA21513; Thu, 7 Feb 2002 09:07:33 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id JAA21481 for ; Thu, 7 Feb 2002 09:07:25 -0500 (EST) Received: from cs.columbia.edu (cs.columbia.edu [128.59.16.20]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id JAA26997 for ; Thu, 7 Feb 2002 09:07:23 -0500 (EST) Received: from opus.cs.columbia.edu (opus.cs.columbia.edu [128.59.20.100]) by cs.columbia.edu (8.9.3/8.9.3) with ESMTP id JAA03027; Thu, 7 Feb 2002 09:07:20 -0500 (EST) Received: from cs.columbia.edu (pool-138-89-38-37.mad.east.verizon.net [138.89.38.37]) (authenticated bits=0) by opus.cs.columbia.edu (8.12.1/8.12.1) with ESMTP id g17E7IJ2006304 (version=TLSv1/SSLv3 cipher=RC4-MD5 bits=128 verify=NOT); Thu, 7 Feb 2002 09:07:19 -0500 (EST) Message-ID: <3C628B82.D912E045@cs.columbia.edu> Date: Thu, 07 Feb 2002 09:13:22 -0500 From: Henning Schulzrinne X-Mailer: Mozilla 4.78 [en] (Windows NT 5.0; U) X-Accept-Language: en MIME-Version: 1.0 To: Juha Heinanen CC: Christian Huitema , Michael Hammer , Mpierce1@aol.com, taylor@nortelnetworks.com, sipping@ietf.org Subject: Re: [Sipping] Status Check: Subscriber Identification References: <3C61A124.901F4DF6@cs.columbia.edu> <15458.5982.309753.795324@lohi.eng.song.fi> Content-Type: text/plain; charset=us-ascii Content-Transfer-Encoding: 7bit Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit > there are several roaming service providers, such as gric and ipass, > that proxy radius requests between domains. From what you know, do they worry about carrier authentication? Presumably, they could just set up IPsec tunnels between each other, as the number of associations is finite. _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Thu Feb 7 09:31:17 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id JAA27935 for ; Thu, 7 Feb 2002 09:31:17 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id JAA22836 for sipping-archive@odin.ietf.org; Thu, 7 Feb 2002 09:31:17 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id JAA22139; Thu, 7 Feb 2002 09:23:59 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id JAA22036 for ; Thu, 7 Feb 2002 09:23:54 -0500 (EST) Received: from rtp-msg-core-1.cisco.com (rtp-msg-core-1.cisco.com [161.44.11.97]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id JAA27690 for ; Thu, 7 Feb 2002 09:23:52 -0500 (EST) Received: from cannon.cisco.com (localhost [127.0.0.1]) by rtp-msg-core-1.cisco.com (8.11.3/8.9.1) with ESMTP id g17ENdX01134; Thu, 7 Feb 2002 09:23:39 -0500 (EST) Received: from cisco.com (dhcp-161-44-241-140.cisco.com [161.44.241.140]) by cannon.cisco.com (Mirapoint) with ESMTP id AAB81063 (AUTH pkyzivat); Thu, 7 Feb 2002 09:26:05 -0500 (EST) Message-ID: <3C628C80.33E28B98@cisco.com> Date: Thu, 07 Feb 2002 09:17:36 -0500 From: Paul Kyzivat X-Mailer: Mozilla 4.76 [en]C-CCK-MCD (Windows NT 5.0; U) X-Accept-Language: en MIME-Version: 1.0 To: Dean Willis CC: Tom-PT Taylor , sipping@ietf.org Subject: Re: [Sipping] RE: calling number blocking at sip/isup gateway References: <4D79C746863DD51197690002A52CDA000102AF3D@zcard0kc.ca.nortel.com> <000f01c1af71$e46f4ff0$9219a8c0@dfw.dynamicsoft.com> Content-Type: text/plain; charset=us-ascii Content-Transfer-Encoding: 7bit Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit Dean - I find your dissection of the issues interesting and useful. You point out that sip doesn't deal explicitly with #4, and seem to find that ok. I feel this is probably something that ought to be addressed directly, because otherwise we will have a hodgepodge as bad or worse than the old phone network. - As you point out, a traditional call associates #4 with #1. - Achieving anything else is a hack. The "collect" call, associating #4 with #2, can really only be set up via 3pcc for arbitrary values of #2. The other way it is done is via the "800 number" hack - certain forms of #2 result in #4 always associating with #2. I think that ideally the fiscal responsibility for a call should be negotiated. The caller may indicate either desire or reluctant willingness to pay. Then the callee could either permit the caller to pay, or take on the responsibility itself. In the end, the call should complete if and only if somebody has accepted the responsibility. It is also possible that the responsibility could be shared, but perhaps that is too much to deal with. Such a model would eliminate the ridiculous kludge of 800 numbers. Instead, the caller would simply select the "collect" option when making a call. But a negotiation of this sort requires something in the protocol to handle the negotiation. Paul Dean Willis wrote: > > I think we call that "authentication", and any UA or proxy MAY issue an > appropriate 401/407 to request authentication. > > Bear in mind that there are at least four identies in every dialog: > > 1) The "From:' party > 2) The "To:" party > 3) The requesting party > 4) The fiscally responsible party > > Role 2 can be further subdivied into: > 2a: the targeted party > 2b: the responding party > > in traditional telephone calls, #1, #3, and #4 are the same. > > For collect calls, the caller has roles 1 and 3, and the called has roles 2 > and 4. > > In third-party control, you can get an even wider range of scenarios. > > SIP, as we currently know it, allows declaration of roles 1 (From:), 2 > (To:), and 3 (authenticated identity). We don't have anything for role #4, > but let that be handled by applications. > > There are proposed extensions to role #1 and controlling of disclosure of > the identity associated with that role (Privacy draft, which defines Remote > party ID). This draft also defines a partial solution for 2, essentially > what I called 2b. There is some discussion in 3GPP about the need to address > 2a. > > We may need to find a way to express transitive relationships between roles > 3 and 1 -- "by Joe on behalf of Sally", but I don't believe there's been any > expression of that yet in IETF. > > Some of the responses to your question are basically asking what the > mechanism for transitively expressing #3 is. Is that what you're asking? My > flip response to the question as originally worded would have been "I don't > know about the other guys, but I authenticate the requester." But I'm not > sure that's actually asnwering the question . . . > > -- > Dean > > ----- Original Message ----- > From: "Tom-PT Taylor" > To: "'Michael Hammer'" > Cc: "'Juha Heinanen'" ; ; > > Sent: Tuesday, January 29, 2002 2:43 PM > Subject: RE: [Sipping] RE: calling number blocking at sip/isup gateway > > > "Subscriber" implies that the SIP user has subscribed to some service > > somewhere. I suppose this is a reasonable assumption for SIP-ISUP > > interworking, though I could imagine scenarios where it might not apply. > > Very well, the first SIP requirement becomes: > > > > (1) SIP must provide a way for the gateway to determine the authenticated > > identity of the SIP subscriber in whose name the call is being made. > > > > > > > > -----Original Message----- > > From: Michael Hammer [mailto:mhammer@cisco.com] > > Sent: Tuesday, January 29, 2002 2:09 PM > > To: Taylor, Tom-PT [CAR:B800:EXCH] > > Cc: 'Juha Heinanen'; 'Mpierce1@aol.com'; sipping@ietf.org > > Subject: RE: [Sipping] RE: calling number blocking at sip/isup gateway > > > > > > Tom, > > > > Given the wireless world and the possibility that there is a distinction > > between the host platform and the subscriber/user, wouldn't it be better > to > > identify the subscriber? There is also technology that allows a user to > > log into and personalize terminals for their use. Or in some cases, the > > user, the subscriber, and the terminal could all be different. > > > > Presumably, the SIP UA that originates the call has an identity and a > > subscription with a service provider that authorizes the call. Could we > > focus terminology on the subscriber as that may be all that is known to > the > > network? > > > > Mike > > > > > > At 01:42 PM 1/29/2002 -0500, Tom-PT Taylor wrote: > > >Well, the overall requirement is clear, but I think it breaks down into > two > > >different sets of requirements, on SIP and on the SIP-PSTN gateway. > > > > > >For SIP: > > > > > >(1) A requirement to provide some means whereby a PSTN gateway can obtain > > >the authenticated identity of the SIP host. I say host rather than user > > >because I can't see regulators requiring SIP to provide more than the > PSTN > > >does now. > > > > > >(2) Another requirement to provide the means whereby a PSTN gateway can > > >determine if the SIP user wishes to restrict the presentation of his/her > > SIP > > >identity. > > > > > > > > >At the gateway the requirements will be the same whether the signalling > > >protocol is SIP, H.323, or something else. > > > > > >(1) A requirement on the gateway to provide the information within the > PSTN > > >signalling which satisfies regulatory requirements. > > > > > >(2) A requirement to report the authenticated identity of the IP host in > > >specific cases e.g. for malicious call trace. > > > > > >(3) A requirement to support billing arrangements. > > > > > >(4) A requirement to extract appropriate display information and pass it > in > > >ISUP signalling if the user has not requested privacy. > > > > > > > > > > > >-----Original Message----- > > >From: Juha Heinanen [mailto:jh@lohi.eng.song.fi] > > >Sent: Tuesday, January 29, 2002 11:17 AM > > >To: Taylor, Tom-PT [CAR:B800:EXCH] > > >Cc: 'Mpierce1@aol.com'; sipping@ietf.org > > >Subject: [Sipping] RE: calling number blocking at sip/isup gateway > > > > > > > > >Tom-PT Taylor writes: > > > > > > > What about UACs > > > > which do not have telephone numbers assigned? Will they be barred > from > > >PSTN > > > > access, or should we mandate that the gateway assign some sort of > > number > > > > which it alone can cross-correlate to the original user@domain if > need > > be > > > > for malicious call trace, etc.? > > > > > >yes, either way. in finland we are not allowed to gateway any call to > > >pstn unless it has a valid from number (either of the user or the > > >gateway) and if we place the gw's number there, we must be able to > > >answer the question, which of the gw's subscribers actually made the > > >call. in addition, the law requires the gw to mark the presentation of > > >the from number of all calls from the subscriber as "restricted" if > > >asked so by the subscriber. > > > > > >if sip doesn't have means to implement the above, then there is no > > >gatewaying of sip calls to pstn in finland and there is no business case > > >for us until sip starts to take over the world, which will take a few > > >years at least. all i want is for a proxy a way to mark the from number > > >of the invite as "restricted" without touching the from header AND i > > >want the chosen way to be clearly documented in the sip-isup i-d so that > > >i don't have to tell each sip/ss7/proxy vendor separately, how it is > > >done. > > > > > >what method you choose to do it, is not my business, since i'm just > > >a stupid user of the sip protocol. > > > > > >-- juha > > > > > > > > >_______________________________________________ > > >Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > > >This list is for NEW development of the application of SIP > > >Use sip-implementors@cs.columbia.edu for questions on current sip > > >Use sip@ietf.org for new developments of core SIP > > > > > > _______________________________________________ > > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > > This list is for NEW development of the application of SIP > > Use sip-implementors@cs.columbia.edu for questions on current sip > > Use sip@ietf.org for new developments of core SIP > > > > > > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP > Use sip-implementors@cs.columbia.edu for questions on current sip > Use sip@ietf.org for new developments of core SIP _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Thu Feb 7 10:33:27 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id KAA29888 for ; Thu, 7 Feb 2002 10:33:27 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id KAA26456 for sipping-archive@odin.ietf.org; Thu, 7 Feb 2002 10:33:29 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id KAA25365; Thu, 7 Feb 2002 10:17:59 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id KAA25333 for ; Thu, 7 Feb 2002 10:17:57 -0500 (EST) Received: from rtp-msg-core-1.cisco.com (rtp-msg-core-1.cisco.com [161.44.11.97]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id KAA29573 for ; Thu, 7 Feb 2002 10:17:55 -0500 (EST) Received: from cia.cisco.com (localhost [127.0.0.1]) by rtp-msg-core-1.cisco.com (8.11.3/8.9.1) with ESMTP id g17FHbl05880; Thu, 7 Feb 2002 10:17:37 -0500 (EST) Received: from MHAMMER-W2K.cisco.com (hrn2-dhcp-161-44-87-145.cisco.com [161.44.87.145]) by cia.cisco.com (Mirapoint) with ESMTP id AAP40034; Thu, 7 Feb 2002 10:12:07 -0500 (EST) Message-Id: <4.3.2.7.2.20020207101620.00b1c150@cia.cisco.com> X-Sender: mhammer@cia.cisco.com X-Mailer: QUALCOMM Windows Eudora Version 4.3.2 Date: Thu, 07 Feb 2002 10:17:17 -0500 To: Christer Holmberg From: Michael Hammer Subject: Re: [Sipping] forking for ISUP to SIP call Cc: ysong@telcordia.com, sipping@ietf.org, "Peterson, Jon" In-Reply-To: <3C6211E3.85CB63C8@lmf.ericsson.se> References: <4.3.2.7.2.20020206132158.03a2a038@cia.cisco.com> Mime-Version: 1.0 Content-Type: text/plain; charset="us-ascii"; format=flowed Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Yes, of course. It was not clear where the forking was occurring in the given scenario, and I did not clarify that point. Thanks. Mike At 07:34 AM 2/7/2002 +0200, Christer Holmberg wrote: >Hi Michael, > >The proxy which "introduced" the forking WILL forward ALL PROVISIONAL >responses to the SIP GW, but it should only forward ONE final response. > >Regards, > >Christer Holmberg >Ericsson Finland > > >Michael Hammer wrote: > > > > YoungSun, > > > > As the SIP network introduced the forking, shouldn't it resolve the forking > > on behalf of the PSTN caller? The SIP side of the SIP GW is the client > > that initiated the SIP call. That agent should resolve the issue and not > > pass it on to the PSTN. This is not any different than if a switch in the > > PSTN had invoked a service such as sequential or parallel hunt groups. > > > > Mike > > > > At 11:59 AM 2/6/2002 -0500, ysong@telcordia.com wrote: > > > > >Hi, > > > > > >I have a question on forking for an ISUP originated call that is > > >terminating to a SIP device. > > >Suppose the following scenario. > > > > > >PSTN ---- IAM----> SIP-GW ----- INVITE -----> SIP network > > >PSTN <----ACM---- SIP-GW <----- 180----- SIP-device-1 > > >PSTN <----???---- SIP-GW <----- 183----- SIP-device-2 > > >PSTN <----???---- SIP-GW <----- 180----- SIP-device-3 > > > > > >How are the multiple 18xs for different call legs mapped to ISUP? > > > > > >Thanks, > > >YoungSun > > > > > > > > >_______________________________________________ > > >Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > > >This list is for NEW development of the application of SIP > > >Use sip-implementors@cs.columbia.edu for questions on current sip > > >Use sip@ietf.org for new developments of core SIP > > > > _______________________________________________ > > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > > This list is for NEW development of the application of SIP > > Use sip-implementors@cs.columbia.edu for questions on current sip > > Use sip@ietf.org for new developments of core SIP _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Thu Feb 7 11:12:39 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id LAA01096 for ; Thu, 7 Feb 2002 11:12:39 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id LAA28626 for sipping-archive@odin.ietf.org; Thu, 7 Feb 2002 11:12:41 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id KAA27333; Thu, 7 Feb 2002 10:56:14 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id KAA27303 for ; Thu, 7 Feb 2002 10:56:08 -0500 (EST) Received: from mail3.dynamicsoft.com ([63.113.44.69]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id KAA00475 for ; Thu, 7 Feb 2002 10:56:06 -0500 (EST) Received: from dynamicsoft.com ([63.113.46.76]) by mail3.dynamicsoft.com (8.12.1/8.12.1) with ESMTP id g17Fu96Y016526; Thu, 7 Feb 2002 10:56:09 -0500 (EST) Message-ID: <3C62A36D.794B8E49@dynamicsoft.com> Date: Thu, 07 Feb 2002 10:55:25 -0500 From: Jonathan Rosenberg Organization: dynamicsoft X-Mailer: Mozilla 4.75 [en] (Windows NT 5.0; U) X-Accept-Language: en MIME-Version: 1.0 To: Michael Hammer CC: Christer Holmberg , ysong@telcordia.com, sipping@ietf.org, "Peterson, Jon" Subject: Re: [Sipping] forking for ISUP to SIP call References: <4.3.2.7.2.20020207101620.00b1c150@cia.cisco.com> Content-Type: text/plain; charset=us-ascii Content-Transfer-Encoding: 7bit Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit Michael Hammer wrote: > > Yes, of course. It was not clear where the forking was occurring in the > > given scenario, and I did not clarify that point. Thanks. > > Mike > > At 07:34 AM 2/7/2002 +0200, Christer Holmberg wrote: > > >Hi Michael, > > > >The proxy which "introduced" the forking WILL forward ALL PROVISIONAL > >responses to the SIP GW, but it should only forward ONE final response. Thats not true. It may end up forwarding multiple 2xx responses if the CANCEL and 2xx pass on the wire. -Jonathan R. -- Jonathan D. Rosenberg, Ph.D. 72 Eagle Rock Avenue Chief Scientist First Floor dynamicsoft East Hanover, NJ 07936 jdrosen@dynamicsoft.com FAX: (973) 952-5050 http://www.jdrosen.net PH: (973) 952-5000 http://www.dynamicsoft.com _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Thu Feb 7 11:16:05 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id LAA01178 for ; Thu, 7 Feb 2002 11:16:05 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id LAA28745 for sipping-archive@odin.ietf.org; Thu, 7 Feb 2002 11:16:07 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id KAA27484; Thu, 7 Feb 2002 10:58:15 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id KAA27451 for ; Thu, 7 Feb 2002 10:58:12 -0500 (EST) Received: from ihemail1.firewall.lucent.com (ihemail1.lucent.com [192.11.222.161]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id KAA00552 for ; Thu, 7 Feb 2002 10:58:10 -0500 (EST) Received: from uk0006exch001h.wins.lucent.com (h135-86-160-150.lucent.com [135.86.160.150]) by ihemail1.firewall.lucent.com (Switch-2.1.3/Switch-2.1.0) with ESMTP id g17FwBi07652 for ; Thu, 7 Feb 2002 10:58:11 -0500 (EST) Received: by uk0006exch001h.uk.lucent.com with Internet Mail Service (5.5.2650.21) id <1K49XTP5>; Thu, 7 Feb 2002 15:58:10 -0000 Message-ID: <475FF955A05DD411980D00508B6D5FB00439E8D0@en0033exch001u.uk.lucent.com> From: "Drage, Keith (Keith)" To: "'Juha Heinanen'" , Jonathan Rosenberg , "'Peterson, Jon'" Cc: Tom-PT Taylor , "'Mpierce1@aol.com'" , sipping@ietf.org Subject: RE: [Sipping] RE: calling number blocking at sip/isup gateway Date: Thu, 7 Feb 2002 15:58:09 -0000 MIME-Version: 1.0 X-Mailer: Internet Mail Service (5.5.2650.21) Content-Type: text/plain; charset="iso-8859-1" Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org If mapping of the restriction is not supported, then the owner of the i/w unit has to take a decision on whether to map the identity, and ignore the restriction indicator, to to refuse to map the identity if the restriction indicator indicates restricted. In general in European public networks, the decision would be to restrict the identity at the interworking point. In private networks, the decision may well be different. In my view, these principles are well established. Note that one significant private network signalling protocol (DPNSS) did not support a restriction indicator in its earlier versions. It only invented one when it needed to interwork with DSS1. Keith Keith Drage Lucent Technologies Tel: +44 1793 776249 Email: drage@lucent.com > ---------- > From: Peterson, Jon[SMTP:jon.peterson@neustar.biz] > Sent: 06 February 2002 6:20 > To: 'Juha Heinanen'; Jonathan Rosenberg > Cc: Tom-PT Taylor; 'Mpierce1@aol.com'; sipping@ietf.org > Subject: RE: [Sipping] RE: calling number blocking at sip/isup > gateway > > > There has been a lot of misleading talk in this thread that half-suggests > that support for presentation restriction in SIP is necessary if you want > to > interwork with the PSTN. I find it hard to believe that there is any > existing national network which requires that every communications > protocol > interworking with the PSTN has a feature by which callers can express that > their preference not to reveal their name and telephone number to those > they > call. I know of some legacy protocols that interwork with the PSTN > worldwide > today that lack this feature. I also think this would be a very difficult > rule to enforce - if a gateway creates properly formed presentation > indicators, I doubt these would be subjected to any scrutiny. Because I > don't think this is a requirement for PSTN interworking, I do not think it > is an absolute dependency in the SIP-ISUP mapping work. We therefore don't > include any mapping in the SIP-ISUP draft, although we do say that > SIP-ISUP > implementations can make use of any general SIP privacy mechanism. > > All that said, I feel strongly that we should have a way to implement a > presentation restriction feature in SIP. However, if there isn't a general > presentation restriction feature in SIP, that doesn't in any way prevent > us > from interworking with the PSTN. If such a general mechanism were defined, > then by all means we can determine how to to map it to the SS7 > presentation > restriction mechanism (easily, no doubt). Note that at least one draft > (the > early media mechanism) describes the SIP-ISUP mapping relevant to its own > mechanism - at least partially because we didn't want the SIP-ISUP draft > to > have a dependency on SIP extensions defined in other drafts. > > I guess that although some people may be happy with the DCS sip-privacy > mechanism, I'm not sure that it really has general applicability. I > currently don't think we need to describe a mapping to this mechanism in > the > SIP-ISUP draft. > > Jon Peterson > NeuStar, Inc. > > > -----Original Message----- > > From: Juha Heinanen [mailto:jh@lohi.eng.song.fi] > > Sent: Wednesday, January 30, 2002 12:28 AM > > To: Jonathan Rosenberg > > Cc: Tom-PT Taylor; 'Mpierce1@aol.com'; sipping@ietf.org > > Subject: Re: [Sipping] RE: calling number blocking at sip/isup gateway > > > > > > Jonathan Rosenberg writes: > > > > > Well, we have a documented way for that in the privacy > > spec right now, > > > which is for the proxy to insert the RPID-Privacy header indicating > > > calling party, with any kind of privacy, and type of subscriber. > > > > yes, and i'm prefectly happy with it. the problem is that > > the sip-isup > > i-d doesn't say a word about it and it is that document that > > the ss7/sip > > gw vendors will read. > > > > -- juha > > > > _______________________________________________ > > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > > This list is for NEW development of the application of SIP > > Use sip-implementors@cs.columbia.edu for questions on current sip > > Use sip@ietf.org for new developments of core SIP > > > > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP > Use sip-implementors@cs.columbia.edu for questions on current sip > Use sip@ietf.org for new developments of core SIP > _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@ns.ietf.org Thu Feb 7 14:23:14 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id OAA06594 for ; Thu, 7 Feb 2002 14:23:14 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id OAA11739 for sipping-archive@odin.ietf.org; Thu, 7 Feb 2002 14:23:15 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id OAA10803; Thu, 7 Feb 2002 14:08:51 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id OAA10778 for ; Thu, 7 Feb 2002 14:08:49 -0500 (EST) Received: from lohi.eng.song.fi (lohi.eng.song.fi [195.10.149.18]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id OAA06061 for ; Thu, 7 Feb 2002 14:08:47 -0500 (EST) Received: from harjus.eng.song.fi (harjus.eng.song.fi [195.10.149.20]) by lohi.eng.song.fi (8.12.1/8.12.1/Debian -5) with ESMTP id g17J8Xio014870; Thu, 7 Feb 2002 21:08:33 +0200 From: Juha Heinanen MIME-Version: 1.0 Content-Type: text/plain; charset=us-ascii Content-Transfer-Encoding: 7bit Message-ID: <15458.53425.530724.168725@harjus.eng.song.fi> Date: Thu, 7 Feb 2002 21:08:33 +0200 To: Henning Schulzrinne Cc: Christian Huitema , Michael Hammer , Mpierce1@aol.com, taylor@nortelnetworks.com, sipping@ietf.org Subject: Re: [Sipping] Status Check: Subscriber Identification In-Reply-To: <3C628B82.D912E045@cs.columbia.edu> References: <3C61A124.901F4DF6@cs.columbia.edu> <15458.5982.309753.795324@lohi.eng.song.fi> <3C628B82.D912E045@cs.columbia.edu> X-Mailer: VM 7.00 under Emacs 21.1.1 Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit Henning Schulzrinne writes: > >From what you know, do they worry about carrier authentication? > Presumably, they could just set up IPsec tunnels between each other, as > the number of associations is finite. they could use ipsec tunnels between the member radius servers and the radius server of the clearing house, but i think that currently they simply rely on the server/client password based authentication that is part of the radius protocol. -- juha _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@ns.ietf.org Thu Feb 7 14:32:38 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id OAA06982 for ; Thu, 7 Feb 2002 14:32:38 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id OAA12531 for sipping-archive@odin.ietf.org; Thu, 7 Feb 2002 14:32:40 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id OAA11855; Thu, 7 Feb 2002 14:26:42 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id OAA11824 for ; Thu, 7 Feb 2002 14:26:39 -0500 (EST) Received: from lohi.eng.song.fi (lohi.eng.song.fi [195.10.149.18]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id OAA06667 for ; Thu, 7 Feb 2002 14:26:36 -0500 (EST) Received: from harjus.eng.song.fi (harjus.eng.song.fi [195.10.149.20]) by lohi.eng.song.fi (8.12.1/8.12.1/Debian -5) with ESMTP id g17JQTio014898; Thu, 7 Feb 2002 21:26:29 +0200 From: Juha Heinanen MIME-Version: 1.0 Content-Type: text/plain; charset=us-ascii Content-Transfer-Encoding: 7bit Message-ID: <15458.54501.614455.337088@harjus.eng.song.fi> Date: Thu, 7 Feb 2002 21:26:29 +0200 To: "Peterson, Jon" Cc: Jonathan Rosenberg , Tom-PT Taylor , "'Mpierce1@aol.com'" , sipping@ietf.org Subject: RE: [Sipping] RE: calling number blocking at sip/isup gateway In-Reply-To: <70565611B164D511957A001083FCDD56CAAEB6@VA02> References: <70565611B164D511957A001083FCDD56CAAEB6@VA02> X-Mailer: VM 7.00 under Emacs 21.1.1 Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit Peterson, Jon writes: > I'll reiterate - the purpose of the SIP-ISUP mapping project (known > collectively as SIP-T) is not to add new features to SIP - it's to describe > how SIP and SS7, without any modifications on either side, can be > interworked. in that case, it might be fair to list in the i-d its applicability, i.e., which ss7 capabilities are not supported by this i-d. > Anyway, the real problem is the want of a general privacy mechanism in SIP - > this really isn't draft-ietf-sipping-isup-00's problem. If there weren't > even an existing proposal for a presentation restriction mechanism, would > you still insist that the SIP-ISUP mapping draft should block until someone > proposes this feature for SIP? no, it would be ok to go forward as long as the things that can't be mapped are listed. -- juha _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@ns.ietf.org Thu Feb 7 15:12:12 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id PAA08234 for ; Thu, 7 Feb 2002 15:12:12 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id PAA14416 for sipping-archive@odin.ietf.org; Thu, 7 Feb 2002 15:12:15 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id OAA13465; Thu, 7 Feb 2002 14:55:31 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id OAA13434 for ; Thu, 7 Feb 2002 14:55:29 -0500 (EST) Received: from rtp-msg-core-1.cisco.com (rtp-msg-core-1.cisco.com [161.44.11.97]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id OAA07749 for ; Thu, 7 Feb 2002 14:55:25 -0500 (EST) Received: from cia.cisco.com (localhost [127.0.0.1]) by rtp-msg-core-1.cisco.com (8.11.3/8.9.1) with ESMTP id g17Jt8A29700; Thu, 7 Feb 2002 14:55:09 -0500 (EST) Received: from MHAMMER-W2K.cisco.com (hrn2-dhcp-161-44-87-145.cisco.com [161.44.87.145]) by cia.cisco.com (Mirapoint) with ESMTP id AAP43511; Thu, 7 Feb 2002 14:49:38 -0500 (EST) Message-Id: <4.3.2.7.2.20020207132938.00b57db8@cia.cisco.com> X-Sender: mhammer@cia.cisco.com X-Mailer: QUALCOMM Windows Eudora Version 4.3.2 Date: Thu, 07 Feb 2002 14:54:49 -0500 To: Henning Schulzrinne From: Michael Hammer Subject: Re: [Sipping] Status Check: Subscriber Identification Cc: Juha Heinanen , Christian Huitema , Mpierce1@aol.com, taylor@nortelnetworks.com, sipping@ietf.org In-Reply-To: <3C628B82.D912E045@cs.columbia.edu> References: <3C61A124.901F4DF6@cs.columbia.edu> <15458.5982.309753.795324@lohi.eng.song.fi> Mime-Version: 1.0 Content-Type: text/plain; charset="us-ascii"; format=flowed Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org What I gathered from this thread (using Michael Thomas model) is that: O: originator T: terminator OP: originating proxy/registrar TP: terminating proxy/registrar IP: intermediate proxy cr: credentials au: authorization indicated by sender In a bilateral scenario: Forward path: O--Ocr-->OP--Ocr,OPcr-->TP (HLR-like function) Reverse path: TP--TPcr,TPauOP,TPauO-->OP--TPcr,TPauOP-->0 meaning: OP trust TP from authenticating TP (from TPcr) and embedded transitively trust O from receipt of authorization (TPauO) TP trust OP from authenticating OP (from OPcr) and embedded transitively trust O (from Ocr) O trust TP from authenticating TP (from TPcr) and embedded transitively trust OP (from TPauOP) So the relationships are: O to OP, O to TP, and OP to TP which uses potentially three different security mechanisms. The thread only addresses the OP->TP->O transitive trust case. Other trust requirements are omitted. This can be expanded to the clearinghouse case (Juha comment) by inserting the intermediate proxy into the forward and backward chain: O-->OP-->IP-->TP-->IP-->OP-->O and including the credentials for each along the path. Now the trust chain (for trusting the subscriber only) is: OP->IP->TP->O Now add the actual call: (forward and reverse path) O-->OP-->IP-->TP'-->T'-->TP'-->IP-->OP-->0 where ' indicates domain of callee versus home domain of caller. So now there needs to be trust between O, OP, IP, TP, TP', T'. Repeat the above for every combination of roaming caller and callee. We seem to be relying on three assumptions: 1) The UA and a home proxy/registrar use the same mechanism. 2) The proxies are able to find a common mechanism between any pair. 3) A transitive trust chain is required with proxies vouching for UAs. I need to read more on the current meaning of protocol elements and whether this provides the 'vouching' semantics that would provide this transitivity. Otherwise, a visited proxy cannot trust the UA. Nor can the callee or any other proxy on the path unless they share the same system as the requesting UA. Mike At 09:13 AM 2/7/2002 -0500, Henning Schulzrinne wrote: > > there are several roaming service providers, such as gric and ipass, > > that proxy radius requests between domains. > > From what you know, do they worry about carrier authentication? >Presumably, they could just set up IPsec tunnels between each other, as >the number of associations is finite. _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@ns.ietf.org Thu Feb 7 15:33:38 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id PAA08968 for ; Thu, 7 Feb 2002 15:33:38 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id PAA15951 for sipping-archive@odin.ietf.org; Thu, 7 Feb 2002 15:33:38 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id PAA14595; Thu, 7 Feb 2002 15:17:43 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id PAA14568 for ; Thu, 7 Feb 2002 15:17:41 -0500 (EST) Received: from bdsl.66.12.12.130.gte.net (bdsl.66.12.12.130.gte.net [66.12.12.130]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id PAA08407 for ; Thu, 7 Feb 2002 15:17:38 -0500 (EST) Received: from txdwillis (bdsl.greycouncil.com [127.0.0.1]) by bdsl.66.12.12.130.gte.net (8.11.6/8.11.6) with SMTP id g17KGmH19562; Thu, 7 Feb 2002 14:16:48 -0600 Message-ID: <02f301c1b014$80eb18d0$1df30a0a@dfw.dynamicsoft.com> From: "Dean Willis" To: "Flemming Andreasen" , References: <3C60460A.1C8CCEA5@cisco.com> <000d01c1af71$e3658110$9219a8c0@dfw.dynamicsoft.com> <3C61EA80.2F209E5D@cisco.com> Subject: Re: Signing RPIDS (Was: Re: [Sipping] calling number blocking at sip/isup gateway)] Date: Thu, 7 Feb 2002 14:03:50 -0600 MIME-Version: 1.0 Content-Type: text/plain; charset="iso-8859-1" Content-Transfer-Encoding: 7bit X-Priority: 3 X-MSMail-Priority: Normal X-Mailer: Microsoft Outlook Express 6.00.2600.0000 X-MimeOLE: Produced By Microsoft MimeOLE V6.00.2600.0000 Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit Warning, indentations may be bogus, message forwarded too many times. > > On Monday 28 January 2002 03:21 pm, Flemming Andreasen wrote: > > > In an almost two year old version of the draft, we had the ability to > > > sign the Remote-Party-ID (rather than just having the "screen" > > > parameter), however, it was subsequently pointed out that it was > > > subject to cut-and-past attacks as you mention, and that it really > > > was part of a much larger SIP security problem that should be solved > > > instead. > Dean Wrote: > > The problem is that the ability to use the RPID as an "introduced by" > > token is contingent on being able to sign it, > > That is your contention. While I agree it is a way of solving the > problem > in general, it is also a complicated solution that may be overkill for > several commercially viable architectures. I assume you are talking > public-key cryptography here which means that we not only incur a > computational overhead, but we also have to solve the key > distribution problem and decide upon a proper signature scheme. It's an either-or question of "trust the network" or "do the crypto". One simply can't trust the network all the time, which leaves the option of "do the crypto". The cert management modem in browser TLS is directly applicable. Every browser is shipped with a bunch of root certs. Every operator has a cert signed by a root cert. So your "network authenticated" ID can simply be signed with the operator cert, which answers the questions of "who vouches for this RPID". > > as well as context > > associate it with the session being introduced. Hashing in To, From, > > and Call-ID headers provides context (which I believe was covered in > > the two-year-old draft). > > I don't believe we ever settled on which headers should be covered by > the > signature. The important observation we did make however was, that the > issue of authenticating parts of a SIP message was a general problem and > should be solved as such. We should not have individual headers come up > with their own authentication schemes, etc, but rather take the usual > route > of solving the more general problem. I'm still of that opinion. I believe there's another draft in circulation that talks about which headers need to be protected by digest authentication. That should suffice for RPID vouching as well. > > The traditional mechanism for bounding replays > > is to apply temporal information. > > I believe Michael objected to temmporizing the signing in that earlier > > draft because of the deifficulty of clock synchronization and of > > describing durations. > > > > > > I think it was more than that, but I'll let Mike speak to that. > > > > However, I think that it might be worth considering what COULD be > > gained from a time-stamped RPID signature. While one might not wish to > > use the resultant programatically as a final authenticator, the > > information might still be useful in making a HUMAN-scale > > interpretation of the RPID. > > > > > You are planning on defining a full-fledged signature scheme in > order to make the authentication information trustworthy, yet you > recognize > the practical problems and thus don't really want to rely on it anyway ? > I recognize the practical difficulties of implementing a signature mechanism. I believe it's solvable. However, I don't believe it's an adequate replacement for all subscriber authentication questions, primarily for reasons of backward compatibility and subscriber certificate management. Same reason that HTTP has challenge-response as well as TLS. You CAN use your browser with a user cert, but you are not required to do so. -- > Dean > > -- > Flemming Andreasen > Cisco Systems > > > _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@ns.ietf.org Thu Feb 7 15:35:07 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id PAA09043 for ; Thu, 7 Feb 2002 15:35:07 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id PAA16026 for sipping-archive@odin.ietf.org; Thu, 7 Feb 2002 15:35:08 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id PAA14641; Thu, 7 Feb 2002 15:18:09 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id PAA14614 for ; Thu, 7 Feb 2002 15:18:07 -0500 (EST) Received: from bdsl.66.12.12.130.gte.net (bdsl.66.12.12.130.gte.net [66.12.12.130]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id PAA08413 for ; Thu, 7 Feb 2002 15:18:03 -0500 (EST) Received: from txdwillis (bdsl.greycouncil.com [127.0.0.1]) by bdsl.66.12.12.130.gte.net (8.11.6/8.11.6) with SMTP id g17KGkH19559; Thu, 7 Feb 2002 14:16:46 -0600 Message-ID: <02f201c1b014$808949c0$1df30a0a@dfw.dynamicsoft.com> From: "Dean Willis" To: "Henning Schulzrinne" , "Michael Hammer" Cc: , , References: <14a.86a3762.29923b1a@aol.com> <4.3.2.7.2.20020206102026.00b86148@cia.cisco.com> <4.3.2.7.2.20020206110421.00bacf00@cia.cisco.com> Subject: Re: [Sipping] Status Check: Subscriber Identification Date: Thu, 7 Feb 2002 13:55:27 -0600 MIME-Version: 1.0 Content-Type: text/plain; charset="iso-8859-1" Content-Transfer-Encoding: 7bit X-Priority: 3 X-MSMail-Priority: Normal X-Mailer: Microsoft Outlook Express 6.00.2600.0000 X-MimeOLE: Produced By Microsoft MimeOLE V6.00.2600.0000 Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit You mean "the stuff we do authentication for"? -- Dean ----- Original Message ----- From: "Michael Hammer" To: "Henning Schulzrinne" Cc: ; ; Sent: Wednesday, February 06, 2002 10:04 AM Subject: Re: [Sipping] Status Check: Subscriber Identification > OK, but what about billing and regulatory requirements? > > At 10:46 AM 2/6/2002 -0500, Henning Schulzrinne wrote: > >In many cases, what I care about is to identify a small subset of > >people. Except for malicious call tracing, I don't get much benefit of > >knowing that Alice Smith (as opposed to somebody impersonating Alice > >Smith), whom I have never met and don't know, calls. As long as > >identities are cheap, negative filtering doesn't work. For positive > >filtering, semi-closed user communities are not ideal, but workable. > > > >Michael Hammer wrote: > > > > > > Granted. But, then there is no guarantee that a user can be authenticated > > > by anyone but the local service provider originally subscribed to. At that > > > point forget mobility. The degree of universality of the supporting > > > mechanism will restrict the universality of the service. Think about what > > > roaming looked like in the first version of mobile networks. > > > > > > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP > Use sip-implementors@cs.columbia.edu for questions on current sip > Use sip@ietf.org for new developments of core SIP > > _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@ns.ietf.org Thu Feb 7 15:45:49 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id PAA09335 for ; Thu, 7 Feb 2002 15:45:49 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id PAA16368 for sipping-archive@odin.ietf.org; Thu, 7 Feb 2002 15:45:50 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id PAA16107; Thu, 7 Feb 2002 15:37:14 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id PAA16083 for ; Thu, 7 Feb 2002 15:37:12 -0500 (EST) Received: from rtp-msg-core-1.cisco.com (rtp-msg-core-1.cisco.com [161.44.11.97]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id PAA09114 for ; Thu, 7 Feb 2002 15:37:11 -0500 (EST) Received: from cia.cisco.com (localhost [127.0.0.1]) by rtp-msg-core-1.cisco.com (8.11.3/8.9.1) with ESMTP id g17KavB03523; Thu, 7 Feb 2002 15:36:58 -0500 (EST) Received: from MHAMMER-W2K.cisco.com (hrn2-dhcp-161-44-87-145.cisco.com [161.44.87.145]) by cia.cisco.com (Mirapoint) with ESMTP id AAP44029; Thu, 7 Feb 2002 15:31:28 -0500 (EST) Message-Id: <4.3.2.7.2.20020207153413.00b18bb8@cia.cisco.com> X-Sender: mhammer@cia.cisco.com X-Mailer: QUALCOMM Windows Eudora Version 4.3.2 Date: Thu, 07 Feb 2002 15:36:39 -0500 To: "Dean Willis" From: Michael Hammer Subject: Re: [Sipping] Status Check: Subscriber Identification Cc: "Henning Schulzrinne" , , , In-Reply-To: <02f201c1b014$808949c0$1df30a0a@dfw.dynamicsoft.com> References: <14a.86a3762.29923b1a@aol.com> <4.3.2.7.2.20020206102026.00b86148@cia.cisco.com> <4.3.2.7.2.20020206110421.00bacf00@cia.cisco.com> Mime-Version: 1.0 Content-Type: text/plain; charset="us-ascii"; format=flowed Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Yes. I am assuming that a service provider needs positive filtering (if I understand that) to determine when to permit signaling and bearer connections to proceed to use resources of their network. Mike At 01:55 PM 2/7/2002 -0600, Dean Willis wrote: >You mean "the stuff we do authentication for"? > >-- >Dean > >----- Original Message ----- >From: "Michael Hammer" >To: "Henning Schulzrinne" >Cc: ; ; >Sent: Wednesday, February 06, 2002 10:04 AM >Subject: Re: [Sipping] Status Check: Subscriber Identification > > > > OK, but what about billing and regulatory requirements? > > > > At 10:46 AM 2/6/2002 -0500, Henning Schulzrinne wrote: > > >In many cases, what I care about is to identify a small subset of > > >people. Except for malicious call tracing, I don't get much benefit of > > >knowing that Alice Smith (as opposed to somebody impersonating Alice > > >Smith), whom I have never met and don't know, calls. As long as > > >identities are cheap, negative filtering doesn't work. For positive > > >filtering, semi-closed user communities are not ideal, but workable. > > > > > >Michael Hammer wrote: > > > > > > > > Granted. But, then there is no guarantee that a user can be >authenticated > > > > by anyone but the local service provider originally subscribed to. At >that > > > > point forget mobility. The degree of universality of the supporting > > > > mechanism will restrict the universality of the service. Think about >what > > > > roaming looked like in the first version of mobile networks. > > > > > > > > > > _______________________________________________ > > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > > This list is for NEW development of the application of SIP > > Use sip-implementors@cs.columbia.edu for questions on current sip > > Use sip@ietf.org for new developments of core SIP > > > > _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@ns.ietf.org Fri Feb 8 00:36:46 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id AAA20798 for ; Fri, 8 Feb 2002 00:36:46 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id AAA07593 for sipping-archive@odin.ietf.org; Fri, 8 Feb 2002 00:36:45 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id AAA06668; Fri, 8 Feb 2002 00:17:50 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id AAA06638 for ; Fri, 8 Feb 2002 00:17:48 -0500 (EST) Received: from mail3.dynamicsoft.com ([63.113.44.69]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id AAA20451 for ; Fri, 8 Feb 2002 00:17:47 -0500 (EST) Received: from dynamicsoft.com ([63.113.46.67]) by mail3.dynamicsoft.com (8.12.1/8.12.1) with ESMTP id g185Hx6Y017365; Fri, 8 Feb 2002 00:18:02 -0500 (EST) Message-ID: <3C635F5A.4281D075@dynamicsoft.com> Date: Fri, 08 Feb 2002 00:17:14 -0500 From: Jonathan Rosenberg Organization: dynamicsoft X-Mailer: Mozilla 4.75 [en] (Windows NT 5.0; U) X-Accept-Language: en MIME-Version: 1.0 To: vitiellopas@libero.it CC: sipping@ietf.org Subject: Re: [Sipping] dialog in conference server References: Content-Type: text/plain; charset=us-ascii Content-Transfer-Encoding: 7bit Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit The question cannot be answered without more information on what conference model you are talking about, and which message exchanges. -Jonathan R. vitiellopas@libero.it wrote: > > hi, > > I'm implementing a conference server and I have the following question: > > in a presence server, refresh and NOTIFY requests have the same dialog > of the SUBSCRIBE requests, particularly the same Call-ID... > is it the same in a conference server, too? > Is it mandatory or not? > > Regards, Pasquale > > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP > Use sip-implementors@cs.columbia.edu for questions on current sip > Use sip@ietf.org for new developments of core SIP -- Jonathan D. Rosenberg, Ph.D. 72 Eagle Rock Avenue Chief Scientist First Floor dynamicsoft East Hanover, NJ 07936 jdrosen@dynamicsoft.com FAX: (973) 952-5050 http://www.jdrosen.net PH: (973) 952-5000 http://www.dynamicsoft.com _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Fri Feb 8 01:56:07 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id BAA21678 for ; Fri, 8 Feb 2002 01:56:06 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id BAA19376 for sipping-archive@odin.ietf.org; Fri, 8 Feb 2002 01:56:07 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id BAA19033; Fri, 8 Feb 2002 01:39:38 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id BAA19004 for ; Fri, 8 Feb 2002 01:39:36 -0500 (EST) Received: from bdsl.66.12.12.130.gte.net (bdsl.66.12.12.130.gte.net [66.12.12.130]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id BAA21482 for ; Fri, 8 Feb 2002 01:39:34 -0500 (EST) Received: from txdwillis (bdsl.greycouncil.com [127.0.0.1]) by bdsl.66.12.12.130.gte.net (8.11.6/8.11.6) with SMTP id g186cwH25706; Fri, 8 Feb 2002 00:38:58 -0600 Message-ID: <06d601c1b06b$6bca64e0$1df30a0a@dfw.dynamicsoft.com> From: "Dean Willis" To: "Michael Hammer" Cc: References: <14a.86a3762.29923b1a@aol.com> <4.3.2.7.2.20020206102026.00b86148@cia.cisco.com> <4.3.2.7.2.20020206110421.00bacf00@cia.cisco.com> <4.3.2.7.2.20020207153413.00b18bb8@cia.cisco.com> Subject: Re: [Sipping] Status Check: Subscriber Identification Date: Fri, 8 Feb 2002 00:39:55 -0600 MIME-Version: 1.0 Content-Type: text/plain; charset="iso-8859-1" Content-Transfer-Encoding: 7bit X-Priority: 3 X-MSMail-Priority: Normal X-Mailer: Microsoft Outlook Express 6.00.2600.0000 X-MimeOLE: Produced By Microsoft MimeOLE V6.00.2600.0000 Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit Well, I don't know about the other guys, but I tend to request authentication headers (if not present, use 401/407 to get) rather than trust any kind of calling-party-ID header to provide positive identification of the user. That sort of implies that the authentication framework needs to be either global or provide an attributable and verifiable transitive model (Cisco's proxy said this guy was Mike Hammer, and I can prove Cisco's proxy said it). -- Dean ----- Original Message ----- From: "Michael Hammer" To: "Dean Willis" Cc: "Henning Schulzrinne" ; ; ; Sent: Thursday, February 07, 2002 2:36 PM Subject: Re: [Sipping] Status Check: Subscriber Identification > Yes. I am assuming that a service provider needs positive filtering (if I > understand that) to determine when to permit signaling and bearer > connections to proceed to use resources of their network. > > Mike > > > At 01:55 PM 2/7/2002 -0600, Dean Willis wrote: > >You mean "the stuff we do authentication for"? > > > >-- > >Dean > > > >----- Original Message ----- > >From: "Michael Hammer" > >To: "Henning Schulzrinne" > >Cc: ; ; > >Sent: Wednesday, February 06, 2002 10:04 AM > >Subject: Re: [Sipping] Status Check: Subscriber Identification > > > > > > > OK, but what about billing and regulatory requirements? > > > > > > At 10:46 AM 2/6/2002 -0500, Henning Schulzrinne wrote: > > > >In many cases, what I care about is to identify a small subset of > > > >people. Except for malicious call tracing, I don't get much benefit of > > > >knowing that Alice Smith (as opposed to somebody impersonating Alice > > > >Smith), whom I have never met and don't know, calls. As long as > > > >identities are cheap, negative filtering doesn't work. For positive > > > >filtering, semi-closed user communities are not ideal, but workable. > > > > > > > >Michael Hammer wrote: > > > > > > > > > > Granted. But, then there is no guarantee that a user can be > >authenticated > > > > > by anyone but the local service provider originally subscribed to. At > >that > > > > > point forget mobility. The degree of universality of the supporting > > > > > mechanism will restrict the universality of the service. Think about > >what > > > > > roaming looked like in the first version of mobile networks. > > > > > > > > > > > > > > _______________________________________________ > > > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > > > This list is for NEW development of the application of SIP > > > Use sip-implementors@cs.columbia.edu for questions on current sip > > > Use sip@ietf.org for new developments of core SIP > > > > > > > > > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP > Use sip-implementors@cs.columbia.edu for questions on current sip > Use sip@ietf.org for new developments of core SIP > > _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Fri Feb 8 03:38:27 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id DAA01190 for ; Fri, 8 Feb 2002 03:38:26 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id DAA24106 for sipping-archive@odin.ietf.org; Fri, 8 Feb 2002 03:38:28 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id DAA23309; Fri, 8 Feb 2002 03:25:05 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id DAA23279 for ; Fri, 8 Feb 2002 03:25:03 -0500 (EST) Received: from mail.szutstar.com ([202.105.137.238]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id DAA01051 for ; Fri, 8 Feb 2002 03:24:58 -0500 (EST) Received: from sznssalbert ([172.16.162.59]) by mail.szutstar.com (8.11.1/8.11.1) with SMTP id g188Gwu08184 for ; Fri, 8 Feb 2002 16:16:58 +0800 Message-ID: <02c001c1b07a$06c05050$3ba210ac@sznssalbert> From: "wlw" To: Date: Fri, 8 Feb 2002 16:24:28 +0800 MIME-Version: 1.0 Content-Type: multipart/alternative; boundary="----=_NextPart_000_02BD_01C1B0BD.14C81910" X-Priority: 3 X-MSMail-Priority: Normal X-Mailer: Microsoft Outlook Express 5.50.4133.2400 X-MimeOLE: Produced By Microsoft MimeOLE V5.50.4133.2400 Subject: [Sipping] Can all SIP response such as 180 RING with SDP? Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org This is a multi-part message in MIME format. ------=_NextPart_000_02BD_01C1B0BD.14C81910 Content-Type: text/plain; charset="gb2312" Content-Transfer-Encoding: base64 VGhhbmtzDQo= ------=_NextPart_000_02BD_01C1B0BD.14C81910 Content-Type: text/html; charset="gb2312" Content-Transfer-Encoding: base64 PCFET0NUWVBFIEhUTUwgUFVCTElDICItLy9XM0MvL0RURCBIVE1MIDQuMCBUcmFuc2l0aW9uYWwv L0VOIj4NCjxIVE1MPjxIRUFEPg0KPE1FVEEgY29udGVudD0idGV4dC9odG1sOyBjaGFyc2V0PWdi MjMxMiIgaHR0cC1lcXVpdj1Db250ZW50LVR5cGU+DQo8TUVUQSBjb250ZW50PSJNU0hUTUwgNS4w MC4zMTAzLjEwMDAiIG5hbWU9R0VORVJBVE9SPg0KPFNUWUxFPjwvU1RZTEU+DQo8L0hFQUQ+DQo8 Qk9EWSBiZ0NvbG9yPSNmZmZmZmY+DQo8RElWPjxGT05UIHNpemU9Mj5UaGFua3M8L0ZPTlQ+PC9E SVY+PC9CT0RZPjwvSFRNTD4NCg== ------=_NextPart_000_02BD_01C1B0BD.14C81910-- _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Fri Feb 8 03:51:33 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id DAA01331 for ; Fri, 8 Feb 2002 03:51:33 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id DAA24572 for sipping-archive@odin.ietf.org; Fri, 8 Feb 2002 03:51:35 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id DAA24375; Fri, 8 Feb 2002 03:44:50 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id DAA24348 for ; Fri, 8 Feb 2002 03:44:48 -0500 (EST) Received: from smtp2.libero.it (smtp2.libero.it [193.70.192.52]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id DAA01264 for ; Fri, 8 Feb 2002 03:44:45 -0500 (EST) Received: from libero.it (193.70.192.41) by smtp2.libero.it (6.0.040) id 3C584044005B8245; Fri, 8 Feb 2002 09:43:56 +0100 Date: Fri, 8 Feb 2002 09:43:56 +0100 Message-Id: MIME-Version: 1.0 Content-Type: text/plain; charset=iso-8859-1 From: "=?utf-8?Q?vitiellopas@libero.it?=" To: jdrosen@dynamicsoft.com Cc: sipping@ietf.org X-XaM3-API-Version: 2.5 X-type: 0 X-SenderIP: 193.205.164.90 Content-Transfer-Encoding: 8bit X-MIME-Autoconverted: from base64 to 8bit by optimus.ietf.org id DAA24349 Subject: [Sipping] =?iso-8859-1?Q?Re:_[Sipping]_dialog_in_conference_server?= Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 8bit >I'm implementing a Dial-in conference server. this server, like explained in the draft-khartabil-sip-conferencing- 00.txt (conferencing using SIP), "handles REGISTER, SUBSCRIBE for partecipants event and sends NOTIFY". Well, the issue is the following one: Must these requests, exchanged between the server and the same conference partecipant, have the same Call-ID or not? In other words, during the following scenario: CP CPS |--REGISTER---->| |<---200 OK-----| |--SUBSCRIBE--->| |<---200 OK-----| |<---NOTIFY-----| |-----200 OK--->| must the REGISTER, SUBSCRIBE and NOTIFY requests have the same Call-ID? Is it mandatory or not? Thanks, Pasquale Jonathan Rosenberg wrote: The question cannot be answered without more information on what > conference model you are talking about, and which message exchanges. > > -Jonathan R. > > vitiellopas@libero.it wrote: > > > > hi, > > > > I'm implementing a conference server and I have the following questi on: > > > > in a presence server, refresh and NOTIFY requests have the same dial og > > of the SUBSCRIBE requests, particularly the same Call-ID... > > is it the same in a conference server, too? > > Is it mandatory or not? > > > > Regards, Pasquale > > > > _______________________________________________ > > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > > This list is for NEW development of the application of SIP > > Use sip-implementors@cs.columbia.edu for questions on current sip > > Use sip@ietf.org for new developments of core SIP > > -- > Jonathan D. Rosenberg, Ph.D. 72 Eagle Rock Avenue > Chief Scientist First Floor > dynamicsoft East Hanover, NJ 07936 > jdrosen@dynamicsoft.com FAX: (973) 952-5050 > http://www.jdrosen.net PH: (973) 952-5000 > http://www.dynamicsoft.com > > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP > Use sip-implementors@cs.columbia.edu for questions on current sip > Use sip@ietf.org for new developments of core SIP > _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Fri Feb 8 08:46:18 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id IAA06288 for ; Fri, 8 Feb 2002 08:46:18 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id IAA07797 for sipping-archive@odin.ietf.org; Fri, 8 Feb 2002 08:46:21 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id IAA07428; Fri, 8 Feb 2002 08:35:00 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id IAA07401 for ; Fri, 8 Feb 2002 08:34:58 -0500 (EST) Received: from gorilla.mchh.siemens.de (gorilla.mchh.siemens.de [194.138.158.18]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id IAA06007 for ; Fri, 8 Feb 2002 08:34:54 -0500 (EST) Received: from moody.mchh.siemens.de (mail2.mchh.siemens.de [194.138.158.226]) by gorilla.mchh.siemens.de (8.9.3/8.9.3) with ESMTP id OAA01586 for ; Fri, 8 Feb 2002 14:34:49 +0100 (MET) Received: from icn.siemens.de ([139.21.148.41]) by moody.mchh.siemens.de (8.9.1/8.9.1) with ESMTP id OAA25671 for ; Fri, 8 Feb 2002 14:34:47 +0100 (MET) Message-ID: <3C63D3F8.8F4AC103@icn.siemens.de> Date: Fri, 08 Feb 2002 14:34:48 +0100 From: Peter =?iso-8859-1?Q?P=E4ppinghaus?= Organization: Siemens AG X-Mailer: Mozilla 4.73 [en] (Windows NT 5.0; U) X-Accept-Language: en,de MIME-Version: 1.0 To: Sipping@ietf.org X-Priority: 2 (High) Content-Type: text/plain; charset=iso-8859-1 Content-Transfer-Encoding: 8bit Content-Transfer-Encoding: 8bit Subject: [Sipping] Flaw in loose routing? Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 8bit Hi, I have a few questions and comments on the bis-07 draft of SIP, mainly concerning the new loose routing mechanism. 1) Unless I misunderstand something, I believe there is a flaw in the loose routing mechanism, leading to the consequence that a loose routing proxy may have to handle a request differently depending on whether the previous hop proxy is a strict or a loose router. Presumably this is an undesired effect. I'll try to explain the problem using a modification of the Basic SIP Trapezoid example of section 16.11.1.1 in bis-07. Assume the initial INVITE sent by U1 (cf. line 2983f.) has a preloaded Route header, and so the following request is forwarded from U1 to proxy P1. INVITE sip:callee@domain.com SIP/2.0 Route: Route: Contact: sip:caller@u1.example.com P1 then forwards the following request to proxy P2. INVITE sip:callee@domain.com SIP/2.0 Route: Contact: sip:caller@u1.example.com Record-Route: P2 processes this request according to sections 16.3, 16.4 and 16.5 of bis-07. Let us assume the request was successfully validated (section 16.3) and processing of section 16.4 follows. The processing of lines 2456-2484 effectively is a no-op in this case (P2 being responsible for domain.com). Now P2 runs a location service yielding several possible destinations to route the request to. But since there is a Route header present in the request, P2 is not allowed to fork, P2 MUST choose a single destination, place it into the Request-URI and continue according to section 16.5. In a second scenario assume that proxy P1 is a strict router, U1 knows this, and hence the initial INVITE sent from U1 to P1 looks as follows. INVITE sip:p1.example.com SIP/2.0 Route: Route: sip:callee@domain.com Contact: sip:caller@u1.example.com P1 then forwards the following request to proxy P2. INVITE sip:p2.domain.com SIP/2.0 Route: sip:callee@domain.com Contact: sip:caller@u1.example.com Record-Route: P2 after successfully validating the request follows the procedures of section 16.4. By analysing the Request-URI, P2 discovers itself, concludes that the previous hop was a strict router, and according to lines 2456-2460 of bis-07, P2 modifies the request to the following form. INVITE sip:callee@domain.com SIP/2.0 Contact: sip:caller@u1.example.com Record-Route: Note that this request differs from the request received by P2 in the first scenario by the missing Route header. Consequently P2 can now use all the results of its location service and fork the request to several destinations. In my opinion it seems that in the procedures of sections 16.4 and 16.5 a loose routing proxy removes a topmost Route header addressing itself much too late. One can consider the processing of lines 2452-2474 as preprocessing. If checking and possibly removing the topmost Route header would be done immediately after this preprocessing stage, and before determining the destination set, then the problem outlined above would disappear. Maybe I overlook some other problem being created by such an early removal of an own Route header. 2) Semantics and manipulation of Request-URI and Route headers are now clearly separated in SIP. In view of this, I find the title of section 16.4 ("Making a Routing Decision") very confusing. Determining the destination set (and accordingly updating the Request-URI) is what section 16.4 really is about. This confusion is enhanced by wording like "At this point, the proxy must decide where to forward the request." (line 2452), "next-hop location" (line 2455), "the element MAY use whatever mechanism it desires to determine where to send the request." (line 2486f.). Only after reading section 16.5, it became clear to me that the next-hop routing decision (namely possibly updating the Route header and forwarding the request according to the topmost Route header, if present) is the subject of section 16.5 and not of section 16.4. Regards, Peter Paeppinghaus -- Dr. Peter P鋚pinghaus Siemens AG | Phone: +49 89 - 722 40065 ICM N PG U ID A1 | Fax: +49 89 - 722 58726 Hofmannstr. 51 | Visitors: Building 1702 / Room 530 D - 81359 M黱chen | Email: peter.paeppinghaus@icn.siemens.de _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Fri Feb 8 09:10:11 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id JAA07198 for ; Fri, 8 Feb 2002 09:10:11 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id JAA09240 for sipping-archive@odin.ietf.org; Fri, 8 Feb 2002 09:10:14 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id IAA08544; Fri, 8 Feb 2002 08:58:40 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id IAA08520 for ; Fri, 8 Feb 2002 08:58:38 -0500 (EST) Received: from commserver.iperia.com ([63.84.167.3]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id IAA06708 for ; Fri, 8 Feb 2002 08:58:35 -0500 (EST) Received: by commserver.erictest.com with Internet Mail Service (5.5.2653.19) id ; Fri, 8 Feb 2002 08:58:14 -0500 Message-ID: <1A69639B9B6AD511812B00B0D0DE19F60A33CE@commserver.erictest.com> From: Gordon Ledgard To: Sipping@ietf.org Date: Fri, 8 Feb 2002 08:58:13 -0500 MIME-Version: 1.0 X-Mailer: Internet Mail Service (5.5.2653.19) Content-Type: text/plain; charset="iso-8859-1" Subject: [Sipping] Registered sip extensions. Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Hello sipping. Does anyone know where I can find a concise, consolidated, complete catalog of current [aliteration stops here] sip extension names... the text tags that can appear in the "Supports" header? Or does everybody just go mucking through all the drafts and proposals and make up their own list? Where are they registered? Thanks, Gordon R. Ledgard IPeria, Inc. _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Fri Feb 8 10:51:42 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id KAA12256 for ; Fri, 8 Feb 2002 10:51:42 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id KAA14798 for sipping-archive@odin.ietf.org; Fri, 8 Feb 2002 10:51:44 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id KAA13983; Fri, 8 Feb 2002 10:35:34 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id KAA13877 for ; Fri, 8 Feb 2002 10:35:13 -0500 (EST) Received: from rtp-msg-core-1.cisco.com (rtp-msg-core-1.cisco.com [161.44.11.97]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id KAA11350 for ; Fri, 8 Feb 2002 10:35:08 -0500 (EST) Received: from cia.cisco.com (localhost [127.0.0.1]) by rtp-msg-core-1.cisco.com (8.11.3/8.9.1) with ESMTP id g18FYtq28405; Fri, 8 Feb 2002 10:34:55 -0500 (EST) Received: from MHAMMER-W2K.cisco.com (hrn2-dhcp-161-44-87-145.cisco.com [161.44.87.145]) by cia.cisco.com (Mirapoint) with ESMTP id AAP50063; Fri, 8 Feb 2002 10:29:20 -0500 (EST) Message-Id: <4.3.2.7.2.20020208103100.00b20150@cia.cisco.com> X-Sender: mhammer@cia.cisco.com X-Mailer: QUALCOMM Windows Eudora Version 4.3.2 Date: Fri, 08 Feb 2002 10:34:02 -0500 To: Jonathan Rosenberg From: Michael Hammer Subject: Re: [Sipping] forking for ISUP to SIP call Cc: Christer Holmberg , ysong@telcordia.com, sipping@ietf.org, "Peterson, Jon" In-Reply-To: <3C62A36D.794B8E49@dynamicsoft.com> References: <4.3.2.7.2.20020207101620.00b1c150@cia.cisco.com> Mime-Version: 1.0 Content-Type: text/plain; charset="us-ascii"; format=flowed Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Jonathan, Let's not get off on a tangent. Let's just say that all normal SIP processing occurs as defined by your draft. Christer was noting where the forking occurs might be at the PSTN GW as well as in the SIP network. My point was that the nuances of SIP operations do not need to be propagated into the PSTN. The PSTN GW is the endpoint that originates or terminates SIP traffic. Mike At 10:55 AM 2/7/2002 -0500, Jonathan Rosenberg wrote: >Michael Hammer wrote: > > > > Yes, of course. It was not clear where the forking was occurring in the > > > > given scenario, and I did not clarify that point. Thanks. > > > > Mike > > > > At 07:34 AM 2/7/2002 +0200, Christer Holmberg wrote: > > > > >Hi Michael, > > > > > >The proxy which "introduced" the forking WILL forward ALL PROVISIONAL > > >responses to the SIP GW, but it should only forward ONE final response. > >Thats not true. It may end up forwarding multiple 2xx responses if the >CANCEL and 2xx pass on the wire. > >-Jonathan R. > > >-- >Jonathan D. Rosenberg, Ph.D. 72 Eagle Rock Avenue >Chief Scientist First Floor >dynamicsoft East Hanover, NJ 07936 >jdrosen@dynamicsoft.com FAX: (973) 952-5050 >http://www.jdrosen.net PH: (973) 952-5000 >http://www.dynamicsoft.com > >_______________________________________________ >Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping >This list is for NEW development of the application of SIP >Use sip-implementors@cs.columbia.edu for questions on current sip >Use sip@ietf.org for new developments of core SIP _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Fri Feb 8 10:54:03 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id KAA12310 for ; Fri, 8 Feb 2002 10:54:03 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id KAA15060 for sipping-archive@odin.ietf.org; Fri, 8 Feb 2002 10:54:05 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id KAA14130; Fri, 8 Feb 2002 10:37:41 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id KAA14084 for ; Fri, 8 Feb 2002 10:37:38 -0500 (EST) Received: from bdsl.66.12.12.130.gte.net (bdsl.66.12.12.130.gte.net [66.12.12.130]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id KAA11506 for ; Fri, 8 Feb 2002 10:37:33 -0500 (EST) Received: from txdwillis (bdsl.greycouncil.com [127.0.0.1]) by bdsl.66.12.12.130.gte.net (8.11.6/8.11.6) with SMTP id g18Fb5H29065; Fri, 8 Feb 2002 09:37:05 -0600 Message-ID: <07df01c1b0b6$98fbcda0$1df30a0a@dfw.dynamicsoft.com> From: "Dean Willis" To: "Gordon Ledgard" , References: <1A69639B9B6AD511812B00B0D0DE19F60A33CE@commserver.erictest.com> Subject: Re: [Sipping] Registered sip extensions. Date: Fri, 8 Feb 2002 09:38:03 -0600 MIME-Version: 1.0 Content-Type: text/plain; charset="iso-8859-1" Content-Transfer-Encoding: 7bit X-Priority: 3 X-MSMail-Priority: Normal X-Mailer: Microsoft Outlook Express 6.00.2600.0000 X-MimeOLE: Produced By Microsoft MimeOLE V6.00.2600.0000 Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit I'm in the midst of proposing an IANA registry process to the ADs . . . -- Dean ----- Original Message ----- From: "Gordon Ledgard" To: Sent: Friday, February 08, 2002 7:58 AM Subject: [Sipping] Registered sip extensions. > > Hello sipping. > > Does anyone know where I can find a concise, consolidated, > complete catalog of current [aliteration stops here] sip > extension names... the text tags that can appear in the > "Supports" header? Or does everybody just go mucking through > all the drafts and proposals and make up their own list? > > Where are they registered? > > Thanks, > Gordon R. Ledgard > IPeria, Inc. > > > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP > Use sip-implementors@cs.columbia.edu for questions on current sip > Use sip@ietf.org for new developments of core SIP > > _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Fri Feb 8 11:03:46 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id LAA12752 for ; Fri, 8 Feb 2002 11:03:46 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id LAA16180 for sipping-archive@odin.ietf.org; Fri, 8 Feb 2002 11:03:48 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id KAA14740; Fri, 8 Feb 2002 10:50:35 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id KAA14711 for ; Fri, 8 Feb 2002 10:50:33 -0500 (EST) Received: from commserver.iperia.com ([63.84.167.3]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id KAA12229 for ; Fri, 8 Feb 2002 10:50:31 -0500 (EST) Received: by commserver.erictest.com with Internet Mail Service (5.5.2653.19) id ; Fri, 8 Feb 2002 10:50:11 -0500 Message-ID: <1A69639B9B6AD511812B00B0D0DE19F60A33D1@commserver.erictest.com> From: Gordon Ledgard To: "'Sipping@ietf.org'" Subject: FW: [Sipping] Registered sip extensions. Date: Fri, 8 Feb 2002 10:50:11 -0500 MIME-Version: 1.0 X-Mailer: Internet Mail Service (5.5.2653.19) Content-Type: text/plain; charset="iso-8859-1" Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org -----Original Message----- From: Gordon Ledgard Sent: Friday, February 08, 2002 10:49 AM To: 'Dean Willis' Subject: RE: [Sipping] Registered sip extensions. Interesting, thanks. On that same note, we were trying to find a list of the languages for the "Accepts Language" header. In the IANA pages, after some considerable nosing around, you finally find a link called: http://www.iana.org/assignments/language-tags which points you to a woefully short and incomplete list of languages. Does anyone know where I can find the complete list? P.S. I'm just dying to use i-klingon here. SIP/2.0 200 GOHCKH , anyone? Thanks, Gordon -----Original Message----- From: Dean Willis [mailto:dean.willis@softarmor.com] Sent: Friday, February 08, 2002 10:38 AM To: Gordon Ledgard; Sipping@ietf.org Subject: Re: [Sipping] Registered sip extensions. I'm in the midst of proposing an IANA registry process to the ADs . . . -- Dean ----- Original Message ----- From: "Gordon Ledgard" To: Sent: Friday, February 08, 2002 7:58 AM Subject: [Sipping] Registered sip extensions. > > Hello sipping. > > Does anyone know where I can find a concise, consolidated, > complete catalog of current [aliteration stops here] sip > extension names... the text tags that can appear in the > "Supports" header? Or does everybody just go mucking through > all the drafts and proposals and make up their own list? > > Where are they registered? > > Thanks, > Gordon R. Ledgard > IPeria, Inc. > > > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP > Use sip-implementors@cs.columbia.edu for questions on current sip > Use sip@ietf.org for new developments of core SIP > > _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Fri Feb 8 11:19:04 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id LAA13291 for ; Fri, 8 Feb 2002 11:19:02 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id LAA17301 for sipping-archive@odin.ietf.org; Fri, 8 Feb 2002 11:19:04 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id LAA16681; Fri, 8 Feb 2002 11:09:33 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id LAA16651 for ; Fri, 8 Feb 2002 11:09:31 -0500 (EST) Received: from acmepacket.com (mail1.acmepacket.com [63.67.143.10]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id LAA12931 for ; Fri, 8 Feb 2002 11:09:28 -0500 (EST) Received: from BobP [63.67.143.2] by acmepacket.com (SMTPD32-7.05) id A83A40350370; Fri, 08 Feb 2002 11:09:30 -0500 Message-ID: <006a01c1b0b9$c0733aa0$2300000a@acmepacket.com> From: "Bob Penfield" To: =?iso-8859-1?Q?Peter_P=E4ppinghaus?= , References: <3C63D3F8.8F4AC103@icn.siemens.de> Subject: Re: [Sipping] Flaw in loose routing? Date: Fri, 8 Feb 2002 11:00:38 -0500 MIME-Version: 1.0 Content-Type: text/plain; charset="iso-8859-1" Content-Transfer-Encoding: 8bit X-Priority: 3 X-MSMail-Priority: Normal X-Mailer: Microsoft Outlook Express 5.50.4133.2400 X-MimeOLE: Produced By Microsoft MimeOLE V5.50.4133.2400 Content-Transfer-Encoding: 8bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 8bit The rules in 12.2.1.1 for populating the Route header indicate that the next hop (first URI in the route set) is included in the Route header. However, the procedures in 16.4 do not indicate that the receive proxy should check to the top-most Route for its own address. If the top-most route is the receiver's address, and it is the only Route, the receiver should be able to put multiple entries in the destination set. Is there a specific reason that the next hop (first URI in the route set) is included in the Route header? If it must be there, then the receiver should check the top-most route for it own address. cheers, (-:bob Robert F. Penfield Chief Software Architect Acme Packet, Inc. 130 New Boston Street Woburn, MA 01801 bpenfield@acmepacket.com ----- Original Message ----- From: "Peter P鋚pinghaus" To: Sent: Friday, February 08, 2002 8:34 AM Subject: [Sipping] Flaw in loose routing? > Hi, > > I have a few questions and comments on the bis-07 draft of SIP, mainly > concerning the new loose routing mechanism. > > 1) Unless I misunderstand something, I believe there is a flaw in the > loose routing mechanism, leading to the consequence that a loose routing > proxy may have to handle a request differently depending on whether the > previous hop proxy is a strict or a loose router. Presumably this is an > undesired effect. > > I'll try to explain the problem using a modification of the Basic SIP > Trapezoid example of section 16.11.1.1 in bis-07. > > Assume the initial INVITE sent by U1 (cf. line 2983f.) has a preloaded > Route header, and so the following request is forwarded from U1 to proxy > P1. > > INVITE sip:callee@domain.com SIP/2.0 > Route: > Route: > Contact: sip:caller@u1.example.com > > P1 then forwards the following request to proxy P2. > > INVITE sip:callee@domain.com SIP/2.0 > Route: > Contact: sip:caller@u1.example.com > Record-Route: > > P2 processes this request according to sections 16.3, 16.4 and 16.5 of > bis-07. Let us assume the request was successfully validated (section > 16.3) and processing of section 16.4 follows. The processing of lines > 2456-2484 effectively is a no-op in this case (P2 being responsible for > domain.com). Now P2 runs a location service yielding several possible > destinations to route the request to. But since there is a Route header > present in the request, P2 is not allowed to fork, P2 MUST choose a > single destination, place it into the Request-URI and continue according > to section 16.5. > > In a second scenario assume that proxy P1 is a strict router, U1 knows > this, and hence the initial INVITE sent from U1 to P1 looks as follows. > > INVITE sip:p1.example.com SIP/2.0 > Route: > Route: sip:callee@domain.com > Contact: sip:caller@u1.example.com > > P1 then forwards the following request to proxy P2. > > INVITE sip:p2.domain.com SIP/2.0 > Route: sip:callee@domain.com > Contact: sip:caller@u1.example.com > Record-Route: > > P2 after successfully validating the request follows the procedures of > section 16.4. By analysing the Request-URI, P2 discovers itself, > concludes that the previous hop was a strict router, and according to > lines 2456-2460 of bis-07, P2 modifies the request to the following > form. > > INVITE sip:callee@domain.com SIP/2.0 > Contact: sip:caller@u1.example.com > Record-Route: > > Note that this request differs from the request received by P2 in the > first scenario by the missing Route header. Consequently P2 can now use > all the results of its location service and fork the request to several > destinations. > > In my opinion it seems that in the procedures of sections 16.4 and 16.5 > a loose routing proxy removes a topmost Route header addressing itself > much too late. One can consider the processing of lines 2452-2474 as > preprocessing. If checking and possibly removing the topmost Route > header would be done immediately after this preprocessing stage, and > before determining the destination set, then the problem outlined above > would disappear. > > Maybe I overlook some other problem being created by such an early > removal of an own Route header. > > 2) Semantics and manipulation of Request-URI and Route headers are now > clearly separated in SIP. In view of this, I find the title of section > 16.4 ("Making a Routing Decision") very confusing. Determining the > destination set (and accordingly updating the Request-URI) is what > section 16.4 really is about. This confusion is enhanced by wording like > "At this point, the proxy must decide where to forward the request." > (line 2452), "next-hop location" (line 2455), "the element MAY use > whatever mechanism it desires to determine where to send the request." > (line 2486f.). Only after reading section 16.5, it became clear to me > that the next-hop routing decision (namely possibly updating the Route > header and forwarding the request according to the topmost Route header, > if present) is the subject of section 16.5 and not of section 16.4. > > Regards, Peter Paeppinghaus > > -- > Dr. Peter P鋚pinghaus > Siemens AG | Phone: +49 89 - 722 40065 > ICM N PG U ID A1 | Fax: +49 89 - 722 58726 > Hofmannstr. 51 | Visitors: Building 1702 / Room 530 > D - 81359 M黱chen | Email: peter.paeppinghaus@icn.siemens.de > > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP > Use sip-implementors@cs.columbia.edu for questions on current sip > Use sip@ietf.org for new developments of core SIP > _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Fri Feb 8 11:20:09 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id LAA13352 for ; Fri, 8 Feb 2002 11:20:09 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id LAA17348 for sipping-archive@odin.ietf.org; Fri, 8 Feb 2002 11:20:11 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id LAA16880; Fri, 8 Feb 2002 11:12:11 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id LAA16842 for ; Fri, 8 Feb 2002 11:12:09 -0500 (EST) Received: from rtp-msg-core-1.cisco.com (rtp-msg-core-1.cisco.com [161.44.11.97]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id LAA13081 for ; Fri, 8 Feb 2002 11:12:06 -0500 (EST) Received: from cia.cisco.com (localhost [127.0.0.1]) by rtp-msg-core-1.cisco.com (8.11.3/8.9.1) with ESMTP id g18GBsD02854; Fri, 8 Feb 2002 11:11:54 -0500 (EST) Received: from MHAMMER-W2K.cisco.com (hrn2-dhcp-161-44-87-145.cisco.com [161.44.87.145]) by cia.cisco.com (Mirapoint) with ESMTP id AAP50542; Fri, 8 Feb 2002 11:06:25 -0500 (EST) Message-Id: <4.3.2.7.2.20020208103755.00b27aa8@cia.cisco.com> X-Sender: mhammer@cia.cisco.com X-Mailer: QUALCOMM Windows Eudora Version 4.3.2 Date: Fri, 08 Feb 2002 11:11:07 -0500 To: "Dean Willis" From: Michael Hammer Subject: Re: [Sipping] Status Check: Subscriber Identification Cc: In-Reply-To: <06d601c1b06b$6bca64e0$1df30a0a@dfw.dynamicsoft.com> References: <14a.86a3762.29923b1a@aol.com> <4.3.2.7.2.20020206102026.00b86148@cia.cisco.com> <4.3.2.7.2.20020206110421.00bacf00@cia.cisco.com> <4.3.2.7.2.20020207153413.00b18bb8@cia.cisco.com> Mime-Version: 1.0 Content-Type: text/plain; charset="us-ascii"; format=flowed Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Dean, Agree with you fully there. The privacy draft unfortunately punts when it comes to how the authentication is done. In looking at the authentication headers, is it possible to specify what header types are covered by the algorithm, or is that assumed to be every header except the authentication header and certain headers that get modified or new ones inserted along the signaling path? I was looking in the bis draft to see which headers those were, but was having trouble finding it. Mike p.s. In the privacy draft, I was amused by the sections titled Untrusted UAC Behavior and Untrusted UAS Behavior. Seemed like those could be one sentence long: "Same as Trusted UAx Behavior, but don't expect that any of the rules will be followed." :-) At 12:39 AM 2/8/2002 -0600, Dean Willis wrote: >Well, I don't know about the other guys, but I tend to request >authentication headers (if not present, use 401/407 to get) rather than >trust any kind of calling-party-ID header to provide positive identification >of the user. That sort of implies that the authentication framework needs to >be either global or provide an attributable and verifiable transitive model >(Cisco's proxy said this guy was Mike Hammer, and I can prove Cisco's proxy >said it). > >-- >Dean > > >----- Original Message ----- >From: "Michael Hammer" >To: "Dean Willis" >Cc: "Henning Schulzrinne" ; ; >; >Sent: Thursday, February 07, 2002 2:36 PM >Subject: Re: [Sipping] Status Check: Subscriber Identification > > > > Yes. I am assuming that a service provider needs positive filtering (if I > > understand that) to determine when to permit signaling and bearer > > connections to proceed to use resources of their network. > > > > Mike > > > > > > At 01:55 PM 2/7/2002 -0600, Dean Willis wrote: > > >You mean "the stuff we do authentication for"? > > > > > >-- > > >Dean > > > > > >----- Original Message ----- > > >From: "Michael Hammer" > > >To: "Henning Schulzrinne" > > >Cc: ; ; > > >Sent: Wednesday, February 06, 2002 10:04 AM > > >Subject: Re: [Sipping] Status Check: Subscriber Identification > > > > > > > > > > OK, but what about billing and regulatory requirements? > > > > > > > > At 10:46 AM 2/6/2002 -0500, Henning Schulzrinne wrote: > > > > >In many cases, what I care about is to identify a small subset of > > > > >people. Except for malicious call tracing, I don't get much benefit >of > > > > >knowing that Alice Smith (as opposed to somebody impersonating Alice > > > > >Smith), whom I have never met and don't know, calls. As long as > > > > >identities are cheap, negative filtering doesn't work. For positive > > > > >filtering, semi-closed user communities are not ideal, but workable. > > > > > > > > > >Michael Hammer wrote: > > > > > > > > > > > > Granted. But, then there is no guarantee that a user can be > > >authenticated > > > > > > by anyone but the local service provider originally subscribed to. >At > > >that > > > > > > point forget mobility. The degree of universality of the >supporting > > > > > > mechanism will restrict the universality of the service. Think >about > > >what > > > > > > roaming looked like in the first version of mobile networks. > > > > > > > > > > > > > > > > > > _______________________________________________ > > > > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > > > > This list is for NEW development of the application of SIP > > > > Use sip-implementors@cs.columbia.edu for questions on current sip > > > > Use sip@ietf.org for new developments of core SIP > > > > > > > > > > > > > > _______________________________________________ > > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > > This list is for NEW development of the application of SIP > > Use sip-implementors@cs.columbia.edu for questions on current sip > > Use sip@ietf.org for new developments of core SIP > > > > _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Fri Feb 8 12:08:19 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id MAA14979 for ; Fri, 8 Feb 2002 12:08:18 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id MAA22661 for sipping-archive@odin.ietf.org; Fri, 8 Feb 2002 12:08:21 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id MAA22102; Fri, 8 Feb 2002 12:02:02 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id MAA22071 for ; Fri, 8 Feb 2002 12:01:59 -0500 (EST) Received: from nisser.cisco.com (nisser.cisco.com [171.71.176.85]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id MAA14766 for ; Fri, 8 Feb 2002 12:01:54 -0500 (EST) Received: from cisco.com (ssh-sj1.cisco.com [171.68.225.134]) by nisser.cisco.com (8.8.6 (PHNE_14041)/CISCO.SERVER.1.2) with ESMTP id JAA11665; Fri, 8 Feb 2002 09:01:18 -0800 (PST) Message-ID: <3C640460.E245AC27@cisco.com> Date: Fri, 08 Feb 2002 12:01:21 -0500 From: Flemming Andreasen X-Mailer: Mozilla 4.79 [en]C-CCK-MCD (Windows NT 5.0; U) X-Accept-Language: en MIME-Version: 1.0 To: Michael Hammer CC: Dean Willis , sipping@ietf.org Subject: Re: [Sipping] Status Check: Subscriber Identification References: <14a.86a3762.29923b1a@aol.com> <4.3.2.7.2.20020206102026.00b86148@cia.cisco.com> <4.3.2.7.2.20020206110421.00bacf00@cia.cisco.com> <4.3.2.7.2.20020207153413.00b18bb8@cia.cisco.com> <4.3.2.7.2.20020208103755.00b27aa8@cia.cisco.com> Content-Type: text/plain; charset=us-ascii Content-Transfer-Encoding: 7bit Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit Michael Hammer wrote: > Dean, > > Agree with you fully there. The privacy draft unfortunately punts when it > comes to how the authentication is done. That was a conscious decision motivated by the fact that authentication and identity representation are two different things and there can be many different ways of performing the authentication. The original apprach of defining just one authentication mechanism was clearly not general enough here, so you either start defining all possible mechanisms, or recognize that this is really a separate problem that should be dealt with separately. We opted for the latter. > In looking at the authentication > headers, is it possible to specify what header types are covered by the > algorithm, or is that assumed to be every header except the authentication > header and certain headers that get modified or new ones inserted along the > signaling path? I was looking in the bis draft to see which headers those > were, but was having trouble finding it. > > Mike > > p.s. In the privacy draft, I was amused by the sections titled Untrusted > UAC Behavior and Untrusted UAS Behavior. Seemed like those could be one > sentence long: "Same as Trusted UAx Behavior, but don't expect that any of > the rules will be followed." :-) You probably didn't read them very carefully then since: 1. A trusted UA SHOULD include a calling party Remote-Party-Id whereas no such requirement exists for an untrusted UA 2. A trusted UA has to apply the requested privacy for any Remote-Party-Id prior to crossing a trust boundary, whereas an untrusted UA does not. While you could cut down on the text here, experience shows that it doesn't further correct implementation. -- Flemming -- Flemming Andreasen Cisco Systems _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Fri Feb 8 12:34:48 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id MAA16571 for ; Fri, 8 Feb 2002 12:34:47 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id MAA24647 for sipping-archive@odin.ietf.org; Fri, 8 Feb 2002 12:34:50 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id MAA23970; Fri, 8 Feb 2002 12:28:12 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id MAA23938 for ; Fri, 8 Feb 2002 12:28:10 -0500 (EST) Received: from mail3.dynamicsoft.com ([63.113.44.69]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id MAA16125 for ; Fri, 8 Feb 2002 12:28:06 -0500 (EST) Received: from dynamicsoft.com ([63.113.46.67]) by mail3.dynamicsoft.com (8.12.1/8.12.1) with ESMTP id g18HSF6Y017825; Fri, 8 Feb 2002 12:28:16 -0500 (EST) Message-ID: <3C640A82.DF033EA6@dynamicsoft.com> Date: Fri, 08 Feb 2002 12:27:30 -0500 From: Jonathan Rosenberg Organization: dynamicsoft X-Mailer: Mozilla 4.75 [en] (Windows NT 5.0; U) X-Accept-Language: en MIME-Version: 1.0 To: Bob Penfield CC: Peter =?iso-8859-1?Q?P=E4ppinghaus?= , Sipping@ietf.org Subject: Re: [Sipping] Flaw in loose routing? References: <006a01c1b0b9$c0733aa0$2300000a@acmepacket.com> Content-Type: text/plain; charset=us-ascii Content-Transfer-Encoding: 7bit Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit Bob Penfield wrote: > > The rules in 12.2.1.1 for populating the Route header indicate that the > next > hop (first URI in the route set) is included in the Route header. > However, > the procedures in 16.4 do not indicate that the receive proxy should > check > to the top-most Route for its own address. If the top-most route is the > receiver's address, and it is the only Route, the receiver should be > able to > put multiple entries in the destination set. They do say it, in bullet 5: In absence of a policy for forwarding a request through specific next hops, the proxy MUST inspect the topmost Route header field value. If that value indicates this proxy, the proxy MUST remove the value from the copy (removing the Route header field if that was the only value). > > Is there a specific reason that the next hop (first URI in the route > set) is > included in the Route header? If it must be there, then the receiver > should > check the top-most route for it own address. The reason it stays is that the URI might contain state or other useful information for the proxy. When it record-routes, it needs to get this information back. Furthermore, only the element that created the route header is ultimately qualified to know that its been reached. Thus, remove on receive is much better. -Jonathan R. -- Jonathan D. Rosenberg, Ph.D. 72 Eagle Rock Avenue Chief Scientist First Floor dynamicsoft East Hanover, NJ 07936 jdrosen@dynamicsoft.com FAX: (973) 952-5050 http://www.jdrosen.net PH: (973) 952-5000 http://www.dynamicsoft.com _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Fri Feb 8 12:42:16 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id MAA16915 for ; Fri, 8 Feb 2002 12:42:16 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id MAA25180 for sipping-archive@odin.ietf.org; Fri, 8 Feb 2002 12:42:17 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id MAA24029; Fri, 8 Feb 2002 12:29:14 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id MAA23998 for ; Fri, 8 Feb 2002 12:29:12 -0500 (EST) Received: from nisser.cisco.com (nisser.cisco.com [171.71.176.85]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id MAA16181 for ; Fri, 8 Feb 2002 12:29:08 -0500 (EST) Received: from cisco.com (ssh-sj1.cisco.com [171.68.225.134]) by nisser.cisco.com (8.8.6 (PHNE_14041)/CISCO.SERVER.1.2) with ESMTP id JAA02936; Fri, 8 Feb 2002 09:28:34 -0800 (PST) Message-ID: <3C640AC6.96C6292A@cisco.com> Date: Fri, 08 Feb 2002 12:28:38 -0500 From: Flemming Andreasen X-Mailer: Mozilla 4.79 [en]C-CCK-MCD (Windows NT 5.0; U) X-Accept-Language: en MIME-Version: 1.0 To: Dean Willis CC: sipping@ietf.org Subject: Re: Signing RPIDS (Was: Re: [Sipping] calling number blocking at sip/isup gateway)] References: <3C60460A.1C8CCEA5@cisco.com> <000d01c1af71$e3658110$9219a8c0@dfw.dynamicsoft.com> <3C61EA80.2F209E5D@cisco.com> <02f301c1b014$80eb18d0$1df30a0a@dfw.dynamicsoft.com> Content-Type: text/plain; charset=us-ascii Content-Transfer-Encoding: 7bit Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit Dean Willis wrote: > Warning, indentations may be bogus, message forwarded too many times. > > > > On Monday 28 January 2002 03:21 pm, Flemming Andreasen wrote: > > > > In an almost two year old version of the draft, we had the ability > to > > > > sign the Remote-Party-ID (rather than just having the "screen" > > > > parameter), however, it was subsequently pointed out that it was > > > > subject to cut-and-past attacks as you mention, and that it really > > > > was part of a much larger SIP security problem that should be solved > > > > instead. > > Dean Wrote: > > > The problem is that the ability to use the RPID as an "introduced by" > > > token is contingent on being able to sign it, > > > > That is your contention. While I agree it is a way of solving the > > problem > > in general, it is also a complicated solution that may be overkill for > > several commercially viable architectures. I assume you are talking > > public-key cryptography here which means that we not only incur a > > computational overhead, but we also have to solve the key > > distribution problem and decide upon a proper signature scheme. > > It's an either-or question of "trust the network" or "do the crypto". One > simply can't trust the network all the time, which leaves the option of "do > the crypto". If we want to have a completely general solution, then I agree. However do note, that this is not the architecture currently assumed and described in the privacy draft (Section 3). > The cert management modem in browser TLS is directly > applicable. Every browser is shipped with a bunch of root certs. Every > operator has a cert signed by a root cert. So your "network authenticated" > ID can simply be signed with the operator cert, which answers the questions > of "who vouches for this RPID". It's not enough to have the root certificates. You still need to obtain the operator certificate in order to verify such a signature, not to mention making sure the certificate has not been revoked. Also, as has been noted before, public-key based signatures carry a non-negligible computational overhead which you need to take into consideration here (consider the effect of having all messages going through the proxy) > > I believe there's another draft in circulation that talks about which > headers need to be protected by digest authentication. That should suffice > for RPID vouching as well. > Which draft are you referring to here ? > > > However, I think that it might be worth considering what COULD be > > > gained from a time-stamped RPID signature. While one might not wish to > > > use the resultant programatically as a final authenticator, the > > > information might still be useful in making a HUMAN-scale > > > interpretation of the RPID. > > > > > > > > You are planning on defining a full-fledged signature scheme in > > order to make the authentication information trustworthy, yet you > > recognize > > the practical problems and thus don't really want to rely on it anyway ? > > > > I recognize the practical difficulties of implementing a signature > mechanism. I believe it's solvable. OK - I'd like to hear more about the replay protection solution at least. > However, I don't believe it's an > adequate replacement for all subscriber authentication questions, primarily > for reasons of backward compatibility and subscriber certificate management. > Same reason that HTTP has challenge-response as well as TLS. You CAN use > your browser with a user cert, but you are not required to do so. > Fair enough. Which methods do you propose we should have ? -- Flemming -- Flemming Andreasen Cisco Systems _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Fri Feb 8 12:46:27 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id MAA17052 for ; Fri, 8 Feb 2002 12:46:27 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id MAA25397 for sipping-archive@odin.ietf.org; Fri, 8 Feb 2002 12:46:28 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id MAA25026; Fri, 8 Feb 2002 12:39:53 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id MAA24997 for ; Fri, 8 Feb 2002 12:39:51 -0500 (EST) Received: from acmepacket.com (mail1.acmepacket.com [63.67.143.10]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id MAA16783 for ; Fri, 8 Feb 2002 12:39:49 -0500 (EST) Received: from BobP [63.67.143.2] by acmepacket.com (SMTPD32-7.05) id AD661F300240; Fri, 08 Feb 2002 12:39:50 -0500 Message-ID: <000d01c1b0c6$51b1bb20$2300000a@acmepacket.com> From: "Bob Penfield" To: "Jonathan Rosenberg" Cc: =?iso-8859-1?Q?Peter_P=E4ppinghaus?= , References: <006a01c1b0b9$c0733aa0$2300000a@acmepacket.com> <3C640A82.DF033EA6@dynamicsoft.com> Subject: Re: [Sipping] Flaw in loose routing? Date: Fri, 8 Feb 2002 12:30:36 -0500 MIME-Version: 1.0 Content-Type: text/plain; charset="iso-8859-1" Content-Transfer-Encoding: 7bit X-Priority: 3 X-MSMail-Priority: Normal X-Mailer: Microsoft Outlook Express 5.50.4133.2400 X-MimeOLE: Produced By Microsoft MimeOLE V5.50.4133.2400 Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit ----- Original Message ----- From: "Jonathan Rosenberg" > > Bob Penfield wrote: > > > > The rules in 12.2.1.1 for populating the Route header indicate that the > > next > > hop (first URI in the route set) is included in the Route header. > > However, > > the procedures in 16.4 do not indicate that the receive proxy should > > check > > to the top-most Route for its own address. If the top-most route is the > > receiver's address, and it is the only Route, the receiver should be > > able to > > put multiple entries in the destination set. > > They do say it, in bullet 5: > > In absence of a policy for forwarding a request through specific next > hops, the proxy MUST inspect the topmost Route header field value. If > that value indicates this proxy, the proxy MUST remove the > value from the copy (removing the Route header field if that was the > only value). That's in 16.5, I mean the procedures in 16.4. It says that the proxy computes a set of destinations. But it also says if there is a route header, then is should only put one destination in that set (line 2487). Shouldn't it check the route header for its own address BEFORE it consults the location service to construct a set of destinations? > > > > > Is there a specific reason that the next hop (first URI in the route > > set) is > > included in the Route header? If it must be there, then the receiver > > should > > check the top-most route for it own address. > > The reason it stays is that the URI might contain state or other useful > information for the proxy. When it record-routes, it needs to get this > information back. Furthermore, only the element that created the route > header is ultimately qualified to know that its been reached. Thus, > remove on receive is much better. > OK. thanks. _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Fri Feb 8 12:52:51 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id MAA17332 for ; Fri, 8 Feb 2002 12:52:51 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id MAA25789 for sipping-archive@odin.ietf.org; Fri, 8 Feb 2002 12:52:52 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id MAA25076; Fri, 8 Feb 2002 12:40:16 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id MAA25046 for ; Fri, 8 Feb 2002 12:40:13 -0500 (EST) Received: from mail3.dynamicsoft.com ([63.113.44.69]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id MAA16806 for ; Fri, 8 Feb 2002 12:40:12 -0500 (EST) Received: from dynamicsoft.com ([63.113.46.67]) by mail3.dynamicsoft.com (8.12.1/8.12.1) with ESMTP id g18HeP6Y017841; Fri, 8 Feb 2002 12:40:26 -0500 (EST) Message-ID: <3C640D5C.2A7DDE93@dynamicsoft.com> Date: Fri, 08 Feb 2002 12:39:40 -0500 From: Jonathan Rosenberg Organization: dynamicsoft X-Mailer: Mozilla 4.75 [en] (Windows NT 5.0; U) X-Accept-Language: en MIME-Version: 1.0 To: Dean Willis CC: Gordon Ledgard , Sipping@ietf.org Subject: Re: [Sipping] Registered sip extensions. References: <07df01c1b0b6$98fbcda0$1df30a0a@dfw.dynamicsoft.com> Content-Type: text/plain; charset=us-ascii Content-Transfer-Encoding: 7bit Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit bis already specifies an iana registry for option tags. The problem is that this registry is itself created only when bis goes to rfc, and similarly extensions which define option tags would have an iana registration in them, which would also get added when those go to rfc. -Jonathan R. Dean Willis wrote: > > I'm in the midst of proposing an IANA registry process to the ADs . . . > > -- > Dean > > ----- Original Message ----- > From: "Gordon Ledgard" > To: > Sent: Friday, February 08, 2002 7:58 AM > Subject: [Sipping] Registered sip extensions. > > > > > Hello sipping. > > > > Does anyone know where I can find a concise, consolidated, > > complete catalog of current [aliteration stops here] sip > > extension names... the text tags that can appear in the > > "Supports" header? Or does everybody just go mucking through > > all the drafts and proposals and make up their own list? > > > > Where are they registered? > > > > Thanks, > > Gordon R. Ledgard > > IPeria, Inc. > > > > > > _______________________________________________ > > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > > This list is for NEW development of the application of SIP > > Use sip-implementors@cs.columbia.edu for questions on current sip > > Use sip@ietf.org for new developments of core SIP > > > > > > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP > Use sip-implementors@cs.columbia.edu for questions on current sip > Use sip@ietf.org for new developments of core SIP -- Jonathan D. Rosenberg, Ph.D. 72 Eagle Rock Avenue Chief Scientist First Floor dynamicsoft East Hanover, NJ 07936 jdrosen@dynamicsoft.com FAX: (973) 952-5050 http://www.jdrosen.net PH: (973) 952-5000 http://www.dynamicsoft.com _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Fri Feb 8 14:39:30 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id OAA21748 for ; Fri, 8 Feb 2002 14:39:30 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id OAA01467 for sipping-archive@odin.ietf.org; Fri, 8 Feb 2002 14:39:32 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id OAA00628; Fri, 8 Feb 2002 14:28:33 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id OAA00599 for ; Fri, 8 Feb 2002 14:28:31 -0500 (EST) Received: from mail3.dynamicsoft.com ([63.113.44.69]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id OAA21228 for ; Fri, 8 Feb 2002 14:28:28 -0500 (EST) Received: from dynamicsoft.com ([63.113.46.30]) by mail3.dynamicsoft.com (8.12.1/8.12.1) with ESMTP id g18JSf6Y017977; Fri, 8 Feb 2002 14:28:41 -0500 (EST) Message-ID: <3C6426BA.94257AB0@dynamicsoft.com> Date: Fri, 08 Feb 2002 14:27:54 -0500 From: Jonathan Rosenberg Organization: dynamicsoft X-Mailer: Mozilla 4.75 [en] (Windows NT 5.0; U) X-Accept-Language: en MIME-Version: 1.0 To: Dean Willis CC: Michael Hammer , sipping@ietf.org Subject: Re: [Sipping] Status Check: Subscriber Identification References: <06d601c1b06b$6bca64e0$1df30a0a@dfw.dynamicsoft.com> Content-Type: text/plain; charset=us-ascii Content-Transfer-Encoding: 7bit Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit I think people get bent out of shape on the privacy drafts because they think it is trying to solve problems it doesn't. Practically speaking, IMHO, the privacy R-PID provides a way for a domain with a trust relationship with a user, to authenticate that user (using digest or whatever), and then pass on that identity to other elements within the domain for use. Similarly, if a request comes in and is not authenticated, it provides a way to convey that as well. It can work between domains only if a trust relationship is there to make sure that the proper interpretation and treatment are given to this information. Generally, that won't be the case. -Jonathan R. Dean Willis wrote: > > Well, I don't know about the other guys, but I tend to request > authentication headers (if not present, use 401/407 to get) rather than > trust any kind of calling-party-ID header to provide positive > identification > of the user. That sort of implies that the authentication framework > needs to > be either global or provide an attributable and verifiable transitive > model > (Cisco's proxy said this guy was Mike Hammer, and I can prove Cisco's > proxy > said it). > > -- > Dean > > ----- Original Message ----- > From: "Michael Hammer" > To: "Dean Willis" > Cc: "Henning Schulzrinne" ; ; > ; > Sent: Thursday, February 07, 2002 2:36 PM > Subject: Re: [Sipping] Status Check: Subscriber Identification > > > Yes. I am assuming that a service provider needs positive filtering > (if I > > understand that) to determine when to permit signaling and bearer > > connections to proceed to use resources of their network. > > > > Mike > > > > > > At 01:55 PM 2/7/2002 -0600, Dean Willis wrote: > > >You mean "the stuff we do authentication for"? > > > > > >-- > > >Dean > > > > > >----- Original Message ----- > > >From: "Michael Hammer" > > >To: "Henning Schulzrinne" > > >Cc: ; ; > > > >Sent: Wednesday, February 06, 2002 10:04 AM > > >Subject: Re: [Sipping] Status Check: Subscriber Identification > > > > > > > > > > OK, but what about billing and regulatory requirements? > > > > > > > > At 10:46 AM 2/6/2002 -0500, Henning Schulzrinne wrote: > > > > >In many cases, what I care about is to identify a small subset of > > > > >people. Except for malicious call tracing, I don't get much > benefit > of > > > > >knowing that Alice Smith (as opposed to somebody impersonating > Alice > > > > >Smith), whom I have never met and don't know, calls. As long as > > > > >identities are cheap, negative filtering doesn't work. For > positive > > > > >filtering, semi-closed user communities are not ideal, but > workable. > > > > > > > > > >Michael Hammer wrote: > > > > > > > > > > > > Granted. But, then there is no guarantee that a user can be > > >authenticated > > > > > > by anyone but the local service provider originally subscribed > to. > At > > >that > > > > > > point forget mobility. The degree of universality of the > supporting > > > > > > mechanism will restrict the universality of the service. > Think > about > > >what > > > > > > roaming looked like in the first version of mobile networks. > > > > > > > > > > > > > > > > > > _______________________________________________ > > > > Sipping mailing list > https://www1.ietf.org/mailman/listinfo/sipping > > > > This list is for NEW development of the application of SIP > > > > Use sip-implementors@cs.columbia.edu for questions on current sip > > > > Use sip@ietf.org for new developments of core SIP > > > > > > > > > > > > > > _______________________________________________ > > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > > This list is for NEW development of the application of SIP > > Use sip-implementors@cs.columbia.edu for questions on current sip > > Use sip@ietf.org for new developments of core SIP > > > > > > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP > Use sip-implementors@cs.columbia.edu for questions on current sip > Use sip@ietf.org for new developments of core SIP -- Jonathan D. Rosenberg, Ph.D. 72 Eagle Rock Avenue Chief Scientist First Floor dynamicsoft East Hanover, NJ 07936 jdrosen@dynamicsoft.com FAX: (973) 952-5050 http://www.jdrosen.net PH: (973) 952-5000 http://www.dynamicsoft.com _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Fri Feb 8 16:20:34 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id QAA25299 for ; Fri, 8 Feb 2002 16:20:33 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id QAA06092 for sipping-archive@odin.ietf.org; Fri, 8 Feb 2002 16:20:37 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id QAA05688; Fri, 8 Feb 2002 16:08:35 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id QAA05659 for ; Fri, 8 Feb 2002 16:08:33 -0500 (EST) Received: from mail3.dynamicsoft.com ([63.113.44.69]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id QAA25018 for ; Fri, 8 Feb 2002 16:08:29 -0500 (EST) Received: from dynamicsoft.com ([63.113.46.30]) by mail3.dynamicsoft.com (8.12.1/8.12.1) with ESMTP id g18L8i6Y018088; Fri, 8 Feb 2002 16:08:45 -0500 (EST) Message-ID: <3C643E2D.7FFADE2B@dynamicsoft.com> Date: Fri, 08 Feb 2002 16:07:57 -0500 From: Jonathan Rosenberg Organization: dynamicsoft X-Mailer: Mozilla 4.75 [en] (Windows NT 5.0; U) X-Accept-Language: en MIME-Version: 1.0 To: infovaan@lycos.com CC: sipping@ietf.org, vidhi.rastogi@wipro.com Subject: Re: [Sipping] Question regarding CANCEL References: Content-Type: text/plain; charset=us-ascii Content-Transfer-Encoding: 7bit Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit Info Van wrote: > > Yes. Since it is for a "pending" session establishment. Or am I missing > something here !! Anyway you are also less than the 10 retries. Sorry, no. The spec is quite explicit. CANCEL only gets sent if a provisional is received, period. If there is a timeout, its like you got a 481, but you send no ACK, no CANCEL. -Jonathan R. > On Tue, 5 Feb 2002 12:04:09 > VIDHI RASTOGI wrote: > >Hi, > > > >Should the UAC send a CANCEL after INVITE request times out. (ie, the > UAC > >has retransmitted INVITE 7 times and never received any response). > > > >Do we need to send CANCEL for any other SIP requests (say REGISTER, > INFO) > >after the REQUEST TIMEOUT? > > > >regards > >Vidhi > > > > > > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP > Use sip-implementors@cs.columbia.edu for questions on current sip > Use sip@ietf.org for new developments of core SIP -- Jonathan D. Rosenberg, Ph.D. 72 Eagle Rock Avenue Chief Scientist First Floor dynamicsoft East Hanover, NJ 07936 jdrosen@dynamicsoft.com FAX: (973) 952-5050 http://www.jdrosen.net PH: (973) 952-5000 http://www.dynamicsoft.com _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Fri Feb 8 16:44:38 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id QAA25839 for ; Fri, 8 Feb 2002 16:44:38 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id QAA07175 for sipping-archive@odin.ietf.org; Fri, 8 Feb 2002 16:44:41 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id QAA06905; Fri, 8 Feb 2002 16:32:42 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id QAA06869 for ; Fri, 8 Feb 2002 16:32:40 -0500 (EST) Received: from rtp-msg-core-1.cisco.com (rtp-msg-core-1.cisco.com [161.44.11.97]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id QAA25608 for ; Fri, 8 Feb 2002 16:32:36 -0500 (EST) Received: from cia.cisco.com (localhost [127.0.0.1]) by rtp-msg-core-1.cisco.com (8.11.3/8.9.1) with ESMTP id g18LWNp07429; Fri, 8 Feb 2002 16:32:23 -0500 (EST) Received: from MHAMMER-W2K.cisco.com (hrn2-dhcp-161-44-87-145.cisco.com [161.44.87.145]) by cia.cisco.com (Mirapoint) with ESMTP id AAP54138; Fri, 8 Feb 2002 16:26:53 -0500 (EST) Message-Id: <4.3.2.7.2.20020208155909.00b84d78@cia.cisco.com> X-Sender: mhammer@cia.cisco.com X-Mailer: QUALCOMM Windows Eudora Version 4.3.2 Date: Fri, 08 Feb 2002 16:31:36 -0500 To: Flemming Andreasen From: Michael Hammer Subject: Re: [Sipping] Status Check: Subscriber Identification Cc: Dean Willis , sipping@ietf.org In-Reply-To: <3C640460.E245AC27@cisco.com> References: <14a.86a3762.29923b1a@aol.com> <4.3.2.7.2.20020206102026.00b86148@cia.cisco.com> <4.3.2.7.2.20020206110421.00bacf00@cia.cisco.com> <4.3.2.7.2.20020207153413.00b18bb8@cia.cisco.com> <4.3.2.7.2.20020208103755.00b27aa8@cia.cisco.com> Mime-Version: 1.0 Content-Type: text/plain; charset="us-ascii"; format=flowed Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org At 12:01 PM 2/8/2002 -0500, Flemming Andreasen wrote: >Michael Hammer wrote: > > > Dean, > > > > Agree with you fully there. The privacy draft unfortunately punts when it > > comes to how the authentication is done. > >That was a conscious decision motivated by the fact that authentication and >identity representation are two different things and there can be many >different >ways of performing the authentication. The original apprach of defining >just one >authentication mechanism was clearly not general enough here, so you >either start >defining all possible mechanisms, or recognize that this is really a separate >problem that should be dealt with separately. We opted for the latter. I was able to infer that and have no problem with that. I shouldn't have said "unfortunately". I was simply noting that more pieces of the puzzle are needed to complete the picture. > > In looking at the authentication > > headers, is it possible to specify what header types are covered by the > > algorithm, or is that assumed to be every header except the authentication > > header and certain headers that get modified or new ones inserted along the > > signaling path? I was looking in the bis draft to see which headers those > > were, but was having trouble finding it. > > > > Mike > > > > p.s. In the privacy draft, I was amused by the sections titled Untrusted > > UAC Behavior and Untrusted UAS Behavior. Seemed like those could be one > > sentence long: "Same as Trusted UAx Behavior, but don't expect that any of > > the rules will be followed." :-) > >You probably didn't read them very carefully then since: >1. A trusted UA SHOULD include a calling party Remote-Party-Id whereas no such >requirement exists for an untrusted UA >2. A trusted UA has to apply the requested privacy for any Remote-Party-Id >prior >to crossing a trust boundary, whereas an untrusted UA does not. > >While you could cut down on the text here, experience shows that it doesn't >further correct implementation. > >-- Flemming That got buried. The problem I had in reading this draft related to the fact that trust involves two parties, where one is "trusting" and the other is "trusted." If you look at Figure 3, it indicates that both UAs are trusted, but in describing the trust relationships below the diagram B) it indicates that there really is not any trust there. The UA can add or not add protocol elements and perform actions on existing protocol elements based on whether it trusts the next hop. (Perhaps it should be called "Trusting UA Behavior"?) The proxy can trust the data it receives based on the ability to authenticate the information, either in-band or out-of-band, or based on a dependency like lower layer transport encryption used. The actions you define above seem to be self-defining. In other words, if the UA properly takes protocol actions X, Y, and Z, then it is acting in a trust-worthy manner. If it doesn't, then it is not. The proxy upon examining the message can see whether the syntax was properly formed. But, additional verification is needed to determine whether the information is bogus or not. The fact that early in the document it mentions that one proxy may not trust another proxy, yet there is no equivalent section present for "untrusted proxy behavior", means that it is possible to describe that in one section. >-- >Flemming Andreasen >Cisco Systems _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Fri Feb 8 17:40:51 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id RAA26874 for ; Fri, 8 Feb 2002 17:40:51 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id RAA10081 for sipping-archive@odin.ietf.org; Fri, 8 Feb 2002 17:40:55 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id RAA09038; Fri, 8 Feb 2002 17:18:42 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id RAA08931 for ; Fri, 8 Feb 2002 17:18:36 -0500 (EST) Received: from mail-blue.research.att.com (mail-blue.research.att.com [135.207.30.102]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id RAA26431 for ; Fri, 8 Feb 2002 17:18:32 -0500 (EST) Received: from alliance.research.att.com (alliance.research.att.com [135.207.26.26]) by mail-blue.research.att.com (Postfix) with ESMTP id 406F54CE14; Fri, 8 Feb 2002 17:18:35 -0500 (EST) Received: from fish.research.att.com (fish.research.att.com [135.207.27.137]) by alliance.research.att.com (8.8.7/8.8.7) with ESMTP id RAA17713; Fri, 8 Feb 2002 17:18:32 -0500 (EST) From: William Marshall Received: (from wtm@localhost) by fish.research.att.com (SGI-8.9.3/8.8.5) id RAA77227; Fri, 8 Feb 2002 17:19:59 -0500 (EST) Date: Fri, 8 Feb 2002 17:19:59 -0500 (EST) Message-Id: <200202082219.RAA77227@fish.research.att.com> To: jdrosen@dynamicsoft.com Cc: dean.willis@softarmor.com, mhammer@cisco.com, sipping@ietf.org Subject: [Sipping] (no subject) Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Jonathan wrote: > It can work between domains only if a trust relationship is there to > make sure that the proper interpretation and treatment are given to this > information. Generally, that won't be the case. A few years ago we ran out of carrier codes (CIC-codes) for long distance carriers. It seems the numbering plan only allowed for 900 of them. So you now have to dial 10-1-xxxx instead of 10-xxx (usually advertised as 10-10-xxx, for the early carriers). So its really hard to say anything about "generally" when there are so many pairwise combinations. It is likely that in a few deployments, like within 3GPP and within PacketCable to mention just two, this trust relationship will be there. I think those few deployments are enough to make the solution interesting and useful. But I think I'd have to agree that "generally, [the trust relationship] won't be the case." Bill Marshall wtm@research.att.com -----original message----- Date: Fri, 08 Feb 2002 14:27:54 -0500 From: Jonathan Rosenberg To: Dean Willis Cc: Michael Hammer , sipping@ietf.org Subject: Re: [Sipping] Status Check: Subscriber Identification I think people get bent out of shape on the privacy drafts because they think it is trying to solve problems it doesn't. Practically speaking, IMHO, the privacy R-PID provides a way for a domain with a trust relationship with a user, to authenticate that user (using digest or whatever), and then pass on that identity to other elements within the domain for use. Similarly, if a request comes in and is not authenticated, it provides a way to convey that as well. It can work between domains only if a trust relationship is there to make sure that the proper interpretation and treatment are given to this information. Generally, that won't be the case. -Jonathan R. Dean Willis wrote: > > Well, I don't know about the other guys, but I tend to request > authentication headers (if not present, use 401/407 to get) rather than > trust any kind of calling-party-ID header to provide positive > identification > of the user. That sort of implies that the authentication framework > needs to > be either global or provide an attributable and verifiable transitive > model > (Cisco's proxy said this guy was Mike Hammer, and I can prove Cisco's > proxy > said it). > > -- > Dean > > ----- Original Message ----- > From: "Michael Hammer" > To: "Dean Willis" > Cc: "Henning Schulzrinne" ; ; > ; > Sent: Thursday, February 07, 2002 2:36 PM > Subject: Re: [Sipping] Status Check: Subscriber Identification > > > Yes. I am assuming that a service provider needs positive filtering > (if I > > understand that) to determine when to permit signaling and bearer > > connections to proceed to use resources of their network. > > > > Mike > > > > > > At 01:55 PM 2/7/2002 -0600, Dean Willis wrote: > > >You mean "the stuff we do authentication for"? > > > > > >-- > > >Dean > > > > > >----- Original Message ----- > > >From: "Michael Hammer" > > >To: "Henning Schulzrinne" > > >Cc: ; ; > > > >Sent: Wednesday, February 06, 2002 10:04 AM > > >Subject: Re: [Sipping] Status Check: Subscriber Identification > > > > > > > > > > OK, but what about billing and regulatory requirements? > > > > > > > > At 10:46 AM 2/6/2002 -0500, Henning Schulzrinne wrote: > > > > >In many cases, what I care about is to identify a small subset of > > > > >people. Except for malicious call tracing, I don't get much > benefit > of > > > > >knowing that Alice Smith (as opposed to somebody impersonating > Alice > > > > >Smith), whom I have never met and don't know, calls. As long as > > > > >identities are cheap, negative filtering doesn't work. For > positive > > > > >filtering, semi-closed user communities are not ideal, but > workable. > > > > > > > > > >Michael Hammer wrote: > > > > > > > > > > > > Granted. But, then there is no guarantee that a user can be > > >authenticated > > > > > > by anyone but the local service provider originally subscribed > to. > At > > >that > > > > > > point forget mobility. The degree of universality of the > supporting > > > > > > mechanism will restrict the universality of the service. > Think > about > > >what > > > > > > roaming looked like in the first version of mobile networks. > > > > > > > > > > > > > > > > > > _______________________________________________ > > > > Sipping mailing list > https://www1.ietf.org/mailman/listinfo/sipping > > > > This list is for NEW development of the application of SIP > > > > Use sip-implementors@cs.columbia.edu for questions on current sip > > > > Use sip@ietf.org for new developments of core SIP > > > > > > > > > > > > > > _______________________________________________ > > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > > This list is for NEW development of the application of SIP > > Use sip-implementors@cs.columbia.edu for questions on current sip > > Use sip@ietf.org for new developments of core SIP > > > > > > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP > Use sip-implementors@cs.columbia.edu for questions on current sip > Use sip@ietf.org for new developments of core SIP -- Jonathan D. Rosenberg, Ph.D. 72 Eagle Rock Avenue Chief Scientist First Floor dynamicsoft East Hanover, NJ 07936 jdrosen@dynamicsoft.com FAX: (973) 952-5050 http://www.jdrosen.net PH: (973) 952-5000 http://www.dynamicsoft.com _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Fri Feb 8 18:37:40 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id SAA27468 for ; Fri, 8 Feb 2002 18:37:40 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id SAA12312 for sipping-archive@odin.ietf.org; Fri, 8 Feb 2002 18:36:58 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id SAA11194; Fri, 8 Feb 2002 18:10:01 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id SAA11163 for ; Fri, 8 Feb 2002 18:09:58 -0500 (EST) Received: from imr2.ericy.com (imr2.ericy.com [198.24.6.3]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id SAA27196 for ; Fri, 8 Feb 2002 18:09:55 -0500 (EST) Received: from mr6.exu.ericsson.se (mr6att.ericy.com [138.85.224.157]) by imr2.ericy.com (8.11.3/8.11.3) with ESMTP id g18N9IS06794; Fri, 8 Feb 2002 17:09:18 -0600 (CST) Received: from newman.exu.ericsson.se (newman.exu.ericsson.se [138.85.75.179]) by mr6.exu.ericsson.se (8.11.3/8.11.3) with ESMTP id g18N9HZ18284; Fri, 8 Feb 2002 17:09:17 -0600 (CST) Received: from qpop.exu.ericsson.se (qpop [138.85.75.72]) by newman.exu.ericsson.se (8.7.5/8.7.3) with ESMTP id RAA06575; Fri, 8 Feb 2002 17:09:16 -0600 (CST) Received: from ericsson.com ([138.85.159.104]) by qpop.exu.ericsson.se (8.9.1/8.9.1) with ESMTP id RAA28351; Fri, 8 Feb 2002 17:09:14 -0600 (CST) Message-ID: <3C645A19.B736CE6A@ericsson.com> Date: Fri, 08 Feb 2002 15:07:06 -0800 From: Tony Johansson X-Mailer: Mozilla 4.79 [en] (Windows NT 5.0; U) X-Accept-Language: en MIME-Version: 1.0 To: "Harri Hakala (LMF)" CC: Sipping@ietf.org, pcalhoun@bstormnetworks.com, kempf@docomolabs-usa.com, matt@ipverse.com, henrik.basilier@ericsson.com References: <0154633DAF7BD4119C360002A513AA0301F946EE@efijont102> Content-Type: multipart/alternative; boundary="------------9A549142C77EB00AD19D5E0C" Subject: [Sipping] Re: Comments on draft--calhoun-sip-aaa-reqs-03.txt Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org --------------9A549142C77EB00AD19D5E0C Content-Type: text/plain; charset=us-ascii Content-Transfer-Encoding: 7bit Hello Harri, Thanks a lot for your comments. I found the comments very useful, so based on your text below I have the following text proposals, which I would like to update the draft with: Add a new accounting section: "2.3.3 Session accounting During the SIP session establishment, the SIP server MAY initiate an accounting session towards the Accounting server in the home or visited network. SIP servers MAY generate several accounting records during the SIP session. At the end of the SIP session, a final accounting record should be issued that includes a summary of the SIP session. The accounting records should include accounting information that is necessary for billing and inter-network settlement purpose. Accounting information such as user identities (called and calling party Ids), SIP call Id, used media types, request-URI, QoS parameters (requested/negotiated quality of service), accounting correlation Id, and SIP session duration should be recorded. Any change during the SIP session that MAY affect the charge of SIP session (e.g. change of media type, additional service requested, call redirection) should be reflected in an accounting record. For example during session forwarding, the initial calling party (A) MAY incur the charges from A to B while the forwarding party MAY incur charges due to the 'forwarded' session. Also, if the called party requests additional media components with regard to the initial request from calling party then the called party MAY be charged for these additional components. Parameters for accounting generation SHOULD be configurable in the SIP servers, or alternatively, the AAA server MAY indicate interim accounting information to the SIP server, which advises the SIP server when to generate the interim accounting records and if a cumulative or non-cumulative accounting model MUST be used. It should be noted that the traditional telco world currently makes use of, and prefers, non-cumulative accounting information. If SIP extensions (e.g. instant messaging, presence service) are used, some of the request or response messages in these extensions MAY trigger generation of accounting information. For example, a SIP SUBSCRIBE MAY trigger generation of a one-time event accounting record, session based instant messaging MAY trigger generation of accounting session and SIP INFO methods MAY trigger generation of an interim accounting message. Accounting data related to bearer payload (e.g. number of transmitted packets, used bandwidth) is assumed to be handled by other mechanisms than SIP and is not included in these requirements. However a unique accounting correlation Id MAY exist, which correlates the SIP session with specific bearer(s)in the access network. If the unique correlation Id exist, it MUST be included in all accounting records to be able to correlate the accounting information from SIP session to used bearer in the access network." Update of the requirements in section 3.0: "3.0 Requirements From the previous section, we can extract the following requirements for SIP-AAA interworking: 1. A basic requirement for Service Providers is to be able to integrate different networks for more efficient operations. IETF AAA efforts support this idea by striving for a single AAA architecture and protocol set. Whether access is supported by PPP, Mobile-IP, MEGACO or SIP, a common architecture and protocol set SHOULD be used. 2. The AAAH MAY authenticate a user based on the corresponding SIP REGISTER method. 3. The AAAH MAY authenticate a user session (the end result of a successful SIP INVITE method), implementing whatever policy it wishes, such as taking into account time of day, distance, etc. 4. The AAA infrastructure MUST be able to distribute (push or pull) subscriber/service/security profiles to the relevant SIP servers, based on policies 5. The AAAH MUST be able to update the entry for the assigned SIP server for the user performing SIP registration. 6. The AAAH MUST be able to provide the assigned server of the user to the SIP entities requesting it. 7. The AAAH MUST be able to initiate de-registration of the user. 8. The AAA infrastructure or the Home entry SIP Proxy MUST be able to allocate a SIP server for a subscriber at registration time, based on policies, load distribution etc. 9. The scheme supported for the authentication between the SIP servers and the AAA infrastructure MUST be flexible enough to accommodate current authentication mechanisms, e.g. using Subscriber Identity Module (SIM) card, and possible future authentication mechanisms. 10. Ability for SIP Servers to provide the duration of a session, the parties involved, and other relevant information to the visited and home AAA servers as accounting information. 11. The AAA protocol MUST provide support for both cumulative, and non-cumulative, accounting models. 12. The AAA accounting messages MUST separate the "session duration time" information from those generated via additional services (e.g. 3-way calling, etc.) 13. There MUST be support for accounting transfers where the information contained in the accounting data has a direct bearing on the establishment, progression and termination of a session. 14. There MUST be support for accounting transfers where the information contained in the accounting data does NOT have a direct bearing on the establishment, progression and termination of a session. 15. SIP servers MUST be able to provide relevant accounting information for billing and inter-network settlement purpose to home and visited AAA servers. Both one time event accounting records and session based (START, INTERIM, STOP records) accounting MUST be supported. 16. Any change during the SIP session that affects the charge of SIP session MUST be reflected in accounting messages. 17. The AAA accounting protocol MUST support accounting per media component (e.g. voice, video). The AAA accounting Protocol MUST enable different parties to be charged per media component. 18. Both cumulative and non-cumulative accounting model MUST be supported. 19. Parameters for accounting generation must either be configurable in the SIP servers or communicated by the AAA servers. 20. If possible the accounting for SIP session MUST be correlated to the bearer in the access network." Comments are very much encouraged. /Tony "Harri Hakala (LMF)" wrote: > Hi, > > I have a few comments on the draft-calhoun-sip-aaa-reqs-03. > http://search.ietf.org/internet-drafts/draft-calhoun-sip-aaa-reqs-03.txt > > Since this draft describes the AAA requirements for IP > telephony/Multimedia, I would like to add a few accounting requirements > to the draft. > I think also that it would be good state what kind of events may > trigger generation of accounting records and what kind of information > needs to be reported. > > I made a proposal for introductory text for body of the draft describing > the accounting mechanism and listed some accounting requirements > in the requirement section. > > I have not been following the Sipping list discussions so far, so please > excuse me if some of my points have been discussed already or if > some of my points are out of scope of this list. > > Section 2.3 Invitation > ---------------------- > Sections 2.3.1 Invitation terminating in the mobile node and > 2.3.2 Invitation originating from the mobile node > ------------------------------------------------------------ > I would like to change the description of accounting in these > chapters in the following way: > > During the SIP session establishment, the SIP serves MAY initiate > accounting session towards Accounting server in the home or visited > network. SIP servers MAY generate several accounting records during > the SIP session. At the end of the SIP session, a final accounting > record should be issued that includes a summary of the SIP session. > > The accounting records should include accounting information that is > necessary for billing and inter-network settlement purpose. Accounting > information such as user identities (called and calling party Ids), > SIP call Id, used media types, request-URI, QoS parameters (requested/ > negotiated quality of service), accounting correlation Id, SIP session > duration should be recorded. > > Any change during the SIP session that MAY affect the charge of SIP > session (e.g. change of media type, additional service requested, > call redirection) should be recorded as an accounting > record. For example during session forwarding, the initial calling party > (A) MAY incur the charges from A to B while the forwarding party MAY incur > charges due to the 'forwarded' session. Also, if the called party requests > additional media components with regard to the initial request from calling > party then called party MAY be charged for these additional components. > > AAA server MAY also return at the SIP session establishment the > accounting-interim-information to the SIP server, which advices the SIP > server when to generate the interim accounting records and if cumulative or > non-cumulative accounting model MUST be used. The traditional telco world > currently makes use of, and prefers, non-cumulative accounting information, > therefore any interim accounting information MAY be non-cumulative. > > If the SIP extensions (for example Instant messaging, presence service) are > used some of the request or response messages in these extension MAY trigger > generation of accounting information. For example SUBSCRIBE MAY trigger > generation of one time event accounting record, session based instant messaging > MAY trigger generation of accounting session and INFO message MAY trigger > generation of interim message. It MUST be configurable in the SIP servers or > AAA server MAY inform the SIP servers when to generate accounting records or > which SIP messages triggers generation of accounting information. > > Accounting data related to bearer payload (e.g. number of transmitted > packets, used bandwidth) is assumed to be handled by other mechanisms > than SIP and is not included in these requirements. However a unique > accounting correlation Id MAY exist, which correlates the SIP session with > specific bearer(s)in the access network. If the unique correlation Id exist, > it MUST be included in all accounting records to be able to correlate the > accounting information from SIP session to used bearer in the access network. > > 3.0 Requirements > ---------------- > Here is my proposal for requirements. > > XX. SIP servers MUST be able to provide relevant accounting information > for billing and inter-network settlement purpose to home and visited > AAA servers. Both one time event accounting records and session base > (START, INTERIM, STOP records) accounting MUST be supported. > > XX. Any change during the SIP session that affects the charge of SIP session > MUST be recorded over AAA accounting protocol. > > XX. It MUST be possible to divide the accounting of the SIP session into > different parts, so that different parties can incur charges for different > parts of the SIP session. > > XX. The AAA accounting protocol MUST support accounting per media component > (e.g. voice, video). AAA accounting Protocol MUST enable different parties > to be charged per media component. > > XX. Both cumulative and non-cumulative accounting model MUST be supported. > > XX. It MUST be option to configure the SIP servers when to generate accounting > records. Alternatively the AAA server MAY inform SIP servers when to > generate accounting records. > > XX. If possible the accounting for SIP session MUST be correlated to the > bearer in the access network. > > XX. There MUST be support for real-time and non-real time accounting. > > Regards............Harri --------------9A549142C77EB00AD19D5E0C Content-Type: text/html; charset=us-ascii Content-Transfer-Encoding: 7bit Hello Harri,

Thanks a lot for your comments. I found the comments very useful, so based on your text below I have the following text proposals, which I would like to update the draft with:

Add a new accounting section:

"2.3.3  Session accounting

   During the SIP session establishment, the SIP server MAY initiate an
   accounting session towards the Accounting server in the home or
   visited network. SIP servers MAY generate several accounting records
   during the SIP session. At the end of the SIP session, a final
   accounting record should be issued that includes a summary of the SIP
   session.

   The accounting records should include accounting information that is
   necessary for billing and inter-network settlement purpose.
   Accounting information such as user identities (called and calling
   party Ids), SIP call Id, used media types, request-URI, QoS
   parameters (requested/negotiated quality of service), accounting
   correlation Id, and SIP session duration should be recorded.

   Any change during the SIP session that MAY affect the charge of SIP
   session (e.g. change of media type, additional service requested,
   call redirection) should be reflected in an accounting record. For
   example during session forwarding, the initial calling party (A) MAY
   incur the charges from A to B while the forwarding party MAY incur
   charges due to the 'forwarded' session. Also, if the called party
   requests additional media components with regard to the initial
   request from calling party then the called party MAY be charged for
   these additional components.

   Parameters for accounting generation SHOULD be configurable in the
   SIP servers, or alternatively, the AAA server MAY indicate interim
   accounting information to the SIP server, which advises the SIP
   server when to generate the interim accounting records and if a
   cumulative or non-cumulative accounting model MUST be used.  It
   should be noted that the traditional telco world currently makes use
   of, and prefers, non-cumulative accounting information.

   If SIP extensions (e.g. instant messaging, presence service) are
   used, some of the request or response messages in these extensions
   MAY trigger generation of accounting information. For example, a SIP
   SUBSCRIBE MAY trigger generation of a one-time event accounting
   record, session based instant messaging MAY trigger generation of
   accounting session and SIP INFO methods MAY trigger generation of an
   interim accounting message.

   Accounting data related to bearer payload (e.g. number of transmitted
   packets, used bandwidth) is assumed to be handled by other mechanisms
   than SIP and is not included in these requirements. However a unique
   accounting correlation Id MAY exist, which correlates the SIP session
   with specific bearer(s)in the access network. If the unique
   correlation Id exist, it MUST be included in all accounting records
   to be able to correlate the accounting information from SIP session
   to used bearer in the access network."
 

Update of the requirements in section 3.0:

"3.0  Requirements

   From the previous section, we can extract the following requirements
   for SIP-AAA interworking:

       1. A basic requirement for Service Providers is to be able to
         integrate different networks for more efficient operations.
         IETF AAA efforts support this idea by striving for a single AAA
         architecture and protocol set. Whether access is supported by
         PPP, Mobile-IP, MEGACO or SIP, a common architecture and
         protocol set SHOULD be used.

       2. The AAAH MAY authenticate a user based on the corresponding
         SIP REGISTER method.

       3. The AAAH MAY authenticate a user session (the end result of a
         successful SIP INVITE method), implementing whatever policy it
         wishes, such as taking into account time of day, distance, etc.

       4. The AAA infrastructure MUST be able to distribute (push or
         pull) subscriber/service/security profiles to the relevant SIP
         servers, based on policies

       5. The AAAH MUST be able to update the entry for the assigned SIP
         server for the user performing SIP registration.
 
      6. The AAAH MUST be able to provide the assigned server of the
         user to the SIP entities requesting it.

       7. The AAAH MUST be able to initiate de-registration of the user.

       8. The AAA infrastructure or the Home entry SIP Proxy MUST be
         able to allocate a SIP server for a subscriber at registration
         time, based on policies, load distribution etc.

       9. The scheme supported for the authentication between the SIP
         servers and the AAA infrastructure MUST be flexible enough to
         accommodate current authentication mechanisms, e.g. using
         Subscriber Identity Module (SIM) card, and possible future
         authentication mechanisms.

       10. Ability for SIP Servers to provide the duration of a session,
         the parties involved, and other relevant information to the
         visited and home AAA servers as accounting information.

       11. The AAA protocol MUST provide support for both cumulative,
         and non-cumulative, accounting models.

       12. The AAA accounting messages MUST separate the "session
         duration time" information from those generated via additional
         services (e.g. 3-way calling, etc.)

       13. There MUST be support for accounting transfers where the
         information contained in the accounting data has a direct
         bearing on the establishment, progression and termination of a
         session.

       14. There MUST be support for accounting transfers where the
         information contained in the accounting data does NOT have a
         direct bearing on the establishment, progression and
         termination of a session.

       15. SIP servers MUST be able to provide relevant accounting
         information for billing and inter-network settlement purpose to
         home and visited AAA servers. Both one time event accounting
         records and session based (START, INTERIM, STOP records)
         accounting MUST be supported.

       16. Any change during the SIP session that affects the charge of
         SIP session MUST be reflected in accounting messages.

       17. The AAA accounting protocol MUST support accounting per media
         component (e.g. voice, video).  The AAA accounting Protocol
         MUST enable different parties to be charged per media component.

      18. Both cumulative and non-cumulative accounting model MUST be
         supported.

       19. Parameters for accounting generation must either be
         configurable in the SIP servers or communicated by the AAA
         servers.

       20. If possible the accounting for SIP session MUST be correlated
         to the bearer in the access network."
 

Comments are very much encouraged.
 

/Tony
 
 
 
 

"Harri Hakala (LMF)" wrote:

Hi,

I have a few comments on the draft-calhoun-sip-aaa-reqs-03.
http://search.ietf.org/internet-drafts/draft-calhoun-sip-aaa-reqs-03.txt

Since this draft describes the AAA requirements for IP
telephony/Multimedia, I would like to add a few accounting requirements
to the draft.
I think also that it would be good state what kind of events may
trigger generation of accounting records and what kind of information
needs to be reported.

I made a proposal for introductory text for body of the draft describing
the accounting mechanism and listed some accounting requirements
in the requirement section.

I have not been following the Sipping list discussions so far, so please
excuse me if some of my points have been discussed already or if
some of my points are out of scope of this list.

Section 2.3 Invitation
----------------------
Sections 2.3.1 Invitation terminating in the mobile node and
         2.3.2 Invitation originating from the mobile node
------------------------------------------------------------
I would like to change the description of accounting in these
chapters in the following way:

During the SIP session establishment, the SIP serves MAY initiate
accounting session towards Accounting server in the home or visited
network. SIP servers MAY generate several accounting records during
the SIP session. At the end of the SIP session, a final accounting
record should be issued that includes a summary of the SIP session.

The accounting records should include accounting information that is
necessary for billing and inter-network settlement purpose. Accounting
information such as user identities (called and calling party Ids),
SIP call Id, used media types, request-URI, QoS parameters (requested/
negotiated quality of service), accounting correlation Id, SIP session
duration should be recorded.

Any change during the SIP session that MAY affect the charge of SIP
session (e.g. change of media type, additional service requested,
call redirection) should be recorded as an accounting
record. For example during session forwarding, the initial calling party
(A) MAY incur the charges from A to B while the forwarding party MAY incur
charges due to the 'forwarded' session. Also, if the called party requests
additional media components with regard to the initial request from calling
party then called party MAY be charged for these additional components.

AAA server MAY also return at the SIP session establishment the
accounting-interim-information to the SIP server, which advices the SIP
server when to generate the interim accounting records and if cumulative or
non-cumulative accounting model MUST be used. The traditional telco world
currently makes use of, and prefers, non-cumulative accounting information,
therefore any interim accounting information MAY be non-cumulative.

If the SIP extensions (for example Instant messaging, presence service) are
used some of the request or response messages in these extension MAY trigger
generation of accounting information. For example SUBSCRIBE MAY trigger
generation of one time event accounting record, session based instant messaging
MAY trigger generation of accounting session and INFO message MAY trigger
generation of interim message. It MUST be configurable in the SIP servers or
AAA server MAY inform the SIP servers when to generate accounting records or
which SIP messages triggers generation of accounting information.

Accounting data related to bearer payload (e.g. number of transmitted
packets, used bandwidth) is assumed to be handled by other mechanisms
than SIP and is not included in these requirements. However a unique
accounting correlation Id MAY exist, which correlates the SIP session with
specific bearer(s)in the access network. If the unique correlation Id exist,
it MUST be included in all accounting records to be able to correlate the
accounting information from SIP session to used bearer in the access network.

3.0 Requirements
----------------
Here is my proposal for requirements.

XX. SIP servers MUST be able to provide relevant accounting information
    for billing and inter-network settlement purpose to home and visited
    AAA servers. Both one time event accounting records and session base
    (START, INTERIM, STOP records) accounting MUST be supported.

XX. Any change during the SIP session that affects the charge of SIP session
    MUST be recorded over AAA accounting protocol.

XX. It MUST be possible to divide the accounting of the SIP session into
    different parts, so that different parties can incur charges for different
    parts of the SIP session.

XX. The AAA accounting protocol MUST support accounting per media component
    (e.g. voice, video). AAA accounting Protocol MUST enable different parties
    to be charged per media component.

XX. Both cumulative and non-cumulative accounting model MUST be supported.

XX. It MUST be option to configure the SIP servers when to generate accounting
    records. Alternatively the AAA server MAY inform SIP servers when to
    generate accounting records.

XX. If possible the accounting for SIP session MUST be correlated to the
    bearer in the access network.

XX. There MUST be support for real-time and non-real time accounting.

Regards............Harri

--------------9A549142C77EB00AD19D5E0C-- _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Sat Feb 9 03:58:45 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id DAA13213 for ; Sat, 9 Feb 2002 03:58:44 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id DAA10801 for sipping-archive@odin.ietf.org; Sat, 9 Feb 2002 03:58:47 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id DAA10559; Sat, 9 Feb 2002 03:41:48 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id DAA10456 for ; Sat, 9 Feb 2002 03:41:43 -0500 (EST) Received: from lohi.eng.song.fi (lohi.eng.song.fi [195.10.149.18]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id DAA13135 for ; Sat, 9 Feb 2002 03:41:40 -0500 (EST) Received: from harjus.eng.song.fi (harjus.eng.song.fi [195.10.149.20]) by lohi.eng.song.fi (8.12.1/8.12.1/Debian -5) with ESMTP id g198fcio016695; Sat, 9 Feb 2002 10:41:38 +0200 From: Juha Heinanen MIME-Version: 1.0 Content-Type: text/plain; charset=us-ascii Content-Transfer-Encoding: 7bit Message-ID: <15460.57538.457422.986299@harjus.eng.song.fi> Date: Sat, 9 Feb 2002 10:41:38 +0200 To: Jonathan Rosenberg Cc: sipping@ietf.org In-Reply-To: <3C6493F7.3C902424@dynamicsoft.com> References: <3C6493F7.3C902424@dynamicsoft.com> X-Mailer: VM 7.00 under Emacs 21.1.1 Content-Transfer-Encoding: 7bit Subject: [Sipping] [Sip] Anonymization of To & revisiting tags alone for dialog ID Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit Jonathan Rosenberg writes: > The spec has generally assumed that the r-uri equals the To field for > new requests. The exceptions are register, and pre-loaded routes, both > of which are mentioned in bis-07. moved to sipping, this reminds me of one thing regarding register that i have never remembered to ask. bis-07 says that in the register mmessage The Request-URI names the domain of the location service for which the registration is meant (for example, sip:chicago.com ). 1514 and figure 2 in bis-07 shows the register message going *directly* from the UA to the registrar without going through any proxies. now my questions are: 1) how does the UA know which host in the domain, in the example above, within chicago.com, acts as the registrar of that domain? 2) is there an srv record that points the UA to the correct host and if not, why? 3) why can't the UA simply send also the register message to its outbound proxy that would them handle the dns lookup and find the host acting as the registrar? what i'm after here is making the UA as simple as possible. -- juha _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Sat Feb 9 12:39:30 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id MAA18157 for ; Sat, 9 Feb 2002 12:39:30 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id MAA26523 for sipping-archive@odin.ietf.org; Sat, 9 Feb 2002 12:39:34 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id MAA25679; Sat, 9 Feb 2002 12:25:51 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id MAA25573 for ; Sat, 9 Feb 2002 12:25:46 -0500 (EST) Received: from t-0n045voi2os9b ([202.99.47.134]) by ietf.org (8.9.1a/8.9.1a) with SMTP id MAA17639 for ; Sat, 9 Feb 2002 12:25:39 -0500 (EST) Message-Id: <200202091725.MAA17639@ietf.org> From: "T&F" To: Mime-Version: 1.0 Content-Type: text/plain; charset="gb2312" Date: Sun, 10 Feb 2002 01:25:11 Subject: [Sipping] A professional company providing credit&status report of Chinese company Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org A professional company providing credit & status report of Chinese company. 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T&F China 使用极星邮件群发,无须通过邮件服务器,直达对方邮箱,速度绝对一流! 下载网址:http://love2net.51.net/,更多免费的超酷软件等你来下…… ---------------------------------------------------- INFORMATION This message has been sent using a trial-run version of the TSmtpRelayServer Delphi Component. ---------------------------------------------------- _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Sun Feb 10 03:09:17 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id DAA03742 for ; Sun, 10 Feb 2002 03:09:17 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id DAA00383 for sipping-archive@odin.ietf.org; Sun, 10 Feb 2002 03:09:20 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id CAA29527; Sun, 10 Feb 2002 02:51:07 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id CAA29497 for ; Sun, 10 Feb 2002 02:51:05 -0500 (EST) Received: from bdsl.66.12.12.130.gte.net (bdsl.66.12.12.130.gte.net [66.12.12.130]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id CAA03625 for ; Sun, 10 Feb 2002 02:51:02 -0500 (EST) Received: from C1893415A (bdsl.greycouncil.com [127.0.0.1]) by bdsl.66.12.12.130.gte.net (8.11.6/8.11.6) with SMTP id g1A7oQH09933; Sun, 10 Feb 2002 01:50:26 -0600 Message-ID: <006001c1b207$9b4b19b0$133fed0c@C1893415A> From: "Dean Willis" To: "Michael Hammer" Cc: References: <14a.86a3762.29923b1a@aol.com> <4.3.2.7.2.20020206102026.00b86148@cia.cisco.com> <4.3.2.7.2.20020206110421.00bacf00@cia.cisco.com> <4.3.2.7.2.20020207153413.00b18bb8@cia.cisco.com> <4.3.2.7.2.20020208103755.00b27aa8@cia.cisco.com> Subject: Re: [Sipping] Status Check: Subscriber Identification Date: Sun, 10 Feb 2002 01:50:27 -0600 MIME-Version: 1.0 Content-Type: text/plain; charset="iso-8859-1" Content-Transfer-Encoding: 7bit X-Priority: 3 X-MSMail-Priority: Normal X-Mailer: Microsoft Outlook Express 6.00.2600.0000 X-MimeOLE: Produced By Microsoft MimeOLE V6.00.2600.0000 Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit Unfortunately, despite telling you "who" more precisely than the RPID headers do at this time, digest authentication as currently practiced leaves a lot lacking in the area of integrity protecting request headers, leaving it open to a variety of extended replay attacks. There has been a lot of work in this area (several drafts discussed at previoud IETF meeting) and we currently have a design team in the SIP working group worrying away at the problem. -- Dean ----- Original Message ----- From: "Michael Hammer" To: "Dean Willis" Cc: Sent: Friday, February 08, 2002 10:11 AM Subject: Re: [Sipping] Status Check: Subscriber Identification > Dean, > > Agree with you fully there. The privacy draft unfortunately punts when it > comes to how the authentication is done. In looking at the authentication > headers, is it possible to specify what header types are covered by the > algorithm, or is that assumed to be every header except the authentication > header and certain headers that get modified or new ones inserted along the > signaling path? I was looking in the bis draft to see which headers those > were, but was having trouble finding it. > > Mike > > p.s. In the privacy draft, I was amused by the sections titled Untrusted > UAC Behavior and Untrusted UAS Behavior. Seemed like those could be one > sentence long: "Same as Trusted UAx Behavior, but don't expect that any of > the rules will be followed." :-) > > > At 12:39 AM 2/8/2002 -0600, Dean Willis wrote: > > >Well, I don't know about the other guys, but I tend to request > >authentication headers (if not present, use 401/407 to get) rather than > >trust any kind of calling-party-ID header to provide positive identification > >of the user. That sort of implies that the authentication framework needs to > >be either global or provide an attributable and verifiable transitive model > >(Cisco's proxy said this guy was Mike Hammer, and I can prove Cisco's proxy > >said it). > > > >-- > >Dean > > > > > >----- Original Message ----- > >From: "Michael Hammer" > >To: "Dean Willis" > >Cc: "Henning Schulzrinne" ; ; > >; > >Sent: Thursday, February 07, 2002 2:36 PM > >Subject: Re: [Sipping] Status Check: Subscriber Identification > > > > > > > Yes. I am assuming that a service provider needs positive filtering (if I > > > understand that) to determine when to permit signaling and bearer > > > connections to proceed to use resources of their network. > > > > > > Mike > > > > > > > > > At 01:55 PM 2/7/2002 -0600, Dean Willis wrote: > > > >You mean "the stuff we do authentication for"? > > > > > > > >-- > > > >Dean > > > > > > > >----- Original Message ----- > > > >From: "Michael Hammer" > > > >To: "Henning Schulzrinne" > > > >Cc: ; ; > > > >Sent: Wednesday, February 06, 2002 10:04 AM > > > >Subject: Re: [Sipping] Status Check: Subscriber Identification > > > > > > > > > > > > > OK, but what about billing and regulatory requirements? > > > > > > > > > > At 10:46 AM 2/6/2002 -0500, Henning Schulzrinne wrote: > > > > > >In many cases, what I care about is to identify a small subset of > > > > > >people. Except for malicious call tracing, I don't get much benefit > >of > > > > > >knowing that Alice Smith (as opposed to somebody impersonating Alice > > > > > >Smith), whom I have never met and don't know, calls. As long as > > > > > >identities are cheap, negative filtering doesn't work. For positive > > > > > >filtering, semi-closed user communities are not ideal, but workable. > > > > > > > > > > > >Michael Hammer wrote: > > > > > > > > > > > > > > Granted. But, then there is no guarantee that a user can be > > > >authenticated > > > > > > > by anyone but the local service provider originally subscribed to. > >At > > > >that > > > > > > > point forget mobility. The degree of universality of the > >supporting > > > > > > > mechanism will restrict the universality of the service. Think > >about > > > >what > > > > > > > roaming looked like in the first version of mobile networks. > > > > > > > > > > > > > > > > > > > > > > _______________________________________________ > > > > > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > > > > > This list is for NEW development of the application of SIP > > > > > Use sip-implementors@cs.columbia.edu for questions on current sip > > > > > Use sip@ietf.org for new developments of core SIP > > > > > > > > > > > > > > > > > > > _______________________________________________ > > > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > > > This list is for NEW development of the application of SIP > > > Use sip-implementors@cs.columbia.edu for questions on current sip > > > Use sip@ietf.org for new developments of core SIP > > > > > > > > _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Sun Feb 10 03:30:33 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id DAA03913 for ; Sun, 10 Feb 2002 03:30:33 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id DAA00973 for sipping-archive@odin.ietf.org; Sun, 10 Feb 2002 03:30:36 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id DAA00515; Sun, 10 Feb 2002 03:12:36 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id DAA00486 for ; Sun, 10 Feb 2002 03:12:34 -0500 (EST) Received: from bdsl.66.12.12.130.gte.net (bdsl.66.12.12.130.gte.net [66.12.12.130]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id DAA03780 for ; Sun, 10 Feb 2002 03:12:31 -0500 (EST) Received: from C1893415A (bdsl.greycouncil.com [127.0.0.1]) by bdsl.66.12.12.130.gte.net (8.11.6/8.11.6) with SMTP id g1A8BuH10058; Sun, 10 Feb 2002 02:11:56 -0600 Message-ID: <00aa01c1b20a$9c02f0a0$133fed0c@C1893415A> From: "Dean Willis" To: "Flemming Andreasen" Cc: References: <3C60460A.1C8CCEA5@cisco.com> <000d01c1af71$e3658110$9219a8c0@dfw.dynamicsoft.com> <3C61EA80.2F209E5D@cisco.com> <02f301c1b014$80eb18d0$1df30a0a@dfw.dynamicsoft.com> <3C640AC6.96C6292A@cisco.com> Subject: Re: Signing RPIDS (Was: Re: [Sipping] calling number blocking at sip/isup gateway)] Date: Sun, 10 Feb 2002 02:11:57 -0600 MIME-Version: 1.0 Content-Type: text/plain; charset="iso-8859-1" Content-Transfer-Encoding: 7bit X-Priority: 3 X-MSMail-Priority: Normal X-Mailer: Microsoft Outlook Express 6.00.2600.0000 X-MimeOLE: Produced By Microsoft MimeOLE V6.00.2600.0000 Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit > > I believe there's another draft in circulation that talks about which > > headers need to be protected by digest authentication. That should suffice > > for RPID vouching as well. > > > > Which draft are you referring to here ? I think I'm thinking of the the pnonce spin-off work that James Undery is supposed to be editing, but I seem to recall seeing another document recently that was just discussing which headers need to be protected. IETF draft searching is down right now, but if it comes back to me I'll let you know. > > I recognize the practical difficulties of implementing a signature > > mechanism. I believe it's solvable. > > OK - I'd like to hear more about the replay protection solution at least. Well, if one is willing to accept roughly synchronized clocks (look, for example, at the nanosecond-level synchronization required for CDMA), then one can easily reduce the replay window to human-acceptable time scales, like 30 seconds or so. One might consider fast perfect replay to be a security flaw, but I've seen enough networks where it happens spontaneously to realize it isn't much of a problem. > > However, I don't believe it's an > > adequate replacement for all subscriber authentication questions, primarily > > for reasons of backward compatibility and subscriber certificate management. > > Same reason that HTTP has challenge-response as well as TLS. You CAN use > > your browser with a user cert, but you are not required to do so. > > Fair enough. Which methods do you propose we should have ? Well, clearly digest has a lot of installed base. 3GPP is leaning towards an EAP-derived system that can reuse the key management of AKA. The AKA mechanism is interesting, because they use sequences of one-time keys generated by a smart-card, which would appear to reduce the probability of replay attack. Assume that we can cobine the AKA key management with ppnonce approaches, and it could be interesting -- any message that was being replayed could only be replayed with the AKA session-key lifetime. -- Dean _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Sun Feb 10 19:47:43 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id TAA12707 for ; Sun, 10 Feb 2002 19:47:43 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id TAA00774 for sipping-archive@odin.ietf.org; Sun, 10 Feb 2002 19:47:45 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id TAA00218; Sun, 10 Feb 2002 19:31:02 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id TAA00111 for ; Sun, 10 Feb 2002 19:30:57 -0500 (EST) Received: from huamin ([61.241.122.146]) by ietf.org (8.9.1a/8.9.1a) with SMTP id TAA12446 for ; Sun, 10 Feb 2002 19:30:53 -0500 (EST) Message-Id: <200202110030.TAA12446@ietf.org> From: "黃硖碼" To: Mime-Version: 1.0 Content-Type: text/plain; charset="gb2312" Date: Mon, 11 Feb 2002 08:31:14 Subject: [Sipping] 黄页号码库给您拜年,祝您马年马到成功! Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org 尊敬的先生/女士: 您好! 如果這封信影響到您的工作,或占用了您的時間,請您立即刪除。同時深表謙意。 如果您能抽出一點時間來了解,可使您和您的企業達到成功的頂峰, 《黃頁號碼庫》致力於建立全國工商企業及社會機關團體的基本數據資料庫,除了114電話查詢的功能之外,還有其它更多的服務如:單位電話查詢、產品服務關鍵字、行業分類等查詢以及其它更多的增值服務。 《黃頁號碼庫》具有強大的交互性和便利快捷的特性,是一個面向企業並以"智能查詢"為主導的電子信息查詢平臺。 《黃頁號碼庫》是一個由全國各省黃頁號碼庫組成的網站群體,每個省的黃頁號碼庫又是以地區黃頁號碼庫為獨立經營對像的聯合體,全國黃頁號碼庫具有強大的信息搜集與信息綜合的能力。 《黃頁號碼庫》数千家企事业单位信息等你查询,同时提供企业登陆 http://www.china168.com 《 欢迎加盟黄页号码库网络联盟 》 祝新年快樂,萬事如意! 使用极星邮件群发,无须通过邮件服务器,直达对方邮箱,速度绝对一流! 下载网址:http://love2net.51.net/,更多免费的超酷软件等你来下…… ---------------------------------------------------- INFORMATION This message has been sent using a trial-run version of the TSmtpRelayServer Delphi Component. ---------------------------------------------------- _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@ns.ietf.org Mon Feb 11 05:55:14 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id FAA28086 for ; Mon, 11 Feb 2002 05:55:14 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id FAA00567 for sipping-archive@odin.ietf.org; Mon, 11 Feb 2002 05:55:14 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id FAA28712; Mon, 11 Feb 2002 05:29:15 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id FAA28688 for ; Mon, 11 Feb 2002 05:29:14 -0500 (EST) Received: from igate2.vodafone.co.uk (igate2.vodafone.co.uk [194.62.232.66]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id FAA27804 for ; Mon, 11 Feb 2002 05:29:09 -0500 (EST) From: duncan.mills@vf.vodafone.co.uk Received: by igate2.vodafone.co.uk; (8.8.8/1.3/10May95) id KAA03733; Mon, 11 Feb 2002 10:28:39 GMT Received: from mimesweeper1.vfl.vodafone (mimesweeper1 [10.33.32.67]) by mailguard2 (4.6.1.91) with ESMTP id for ; Mon, 11 Feb 2002 10:26:24 GMT Received: from putney.vfl.vodafone (unverified) by mimesweeper1.vfl.vodafone (Content Technologies SMTPRS 2.0.15) with ESMTP id for ; Mon, 11 Feb 2002 10:02:25 +0000 Received: by putney.vfl.vodafone with Internet Mail Service (5.5.2650.21) id <1XGYA8N9>; Mon, 11 Feb 2002 10:02:24 -0000 Message-Id: To: sipping@ietf.org Date: Mon, 11 Feb 2002 09:54:30 -0000 X-MS-TNEF-Correlator: X-Mailer: Internet Mail Service (5.5.2650.21) MIME-Version: 1.0 (Generated by Clearswift ES version 4.6.1.109) Content-Type: multipart/mixed; boundary="----_=_NextPart_000_01C1B2E3.32891710" Subject: [Sipping] SIP INFO method for DTMF Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org This message is in MIME format. Since your mail reader does not understand this format, some or all of this message may not be legible. ------_=_NextPart_000_01C1B2E3.32891710 Content-Type: text/plain; charset="iso-8859-1" IETF SIPPING experts, 3GPP standards people have been looking at various potential solutions for carrying DTMF in the IP Multimedia Subsystem (IMS). One such solution is the use of the SIP INFO method. I have read the expired internet-draft, written by Jiri Kuthan (whom I tried to contact first, but the e-mail address was no longer a valid one) and there is support for such a solution in 3GPP. I was involved in a telephone conference last week with other 3GPP delegates, in which we questioned why this draft expired? Whilst we have highlighted some potential problems with using the draft that Jiri wrote, we did at least see the INFO method as a feasible option (as well as the RTP payload option - I've had a fair amount of e-mail discussion with Henning on RFC 2833). 3GPP may run into problems in the Radio Access Network (RAN) area if we choose the RTP solution for DTMF. As such, we are considering writing an internet-draft on the use of the SIP INFO method for DTMF. However, before doing this, I thought it wise to 'test the water'. The RTP solution is well documented, and we're currently looking into those RAN issues. In the meantime, I'm asking for opinions on the general concept of using the SIP INFO method for DTMF. Best Regards, Duncan Duncan Mills UMTS Standards Technology Team Core Network Development Vodafone Ltd (Networks) Tel: +44 1635 676074 Fax: +44 1635 234445 mailto:duncan.mills@vf.vodafone.co.uk ------_=_NextPart_000_01C1B2E3.32891710 Content-Type: application/ms-tnef Content-Transfer-Encoding: base64 eJ8+IhkKAQaQCAAEAAAAAAABAAEAAQeQBgAIAAAA5AQAAAAAAADoAAEIgAcAGAAAAElQTS5NaWNy b3NvZnQgTWFpbC5Ob3RlADEIAQWAAwAOAAAA0gcCAAsACQA2AB4AAQBEAQEggAMADgAAANIHAgAL AAoAAgAVAAEACAEBCYABACEAAABGQTc4RjFFOENCMUVENjExQUIzNjAwQTBDOUU5NzYzOQBRBwEE gAEAGQAAAFNJUCBJTkZPIG1ldGhvZCBmb3IgRFRNRgCLBwENgAQAAgAAAAIAAgABA5AGADALAAAx AAAAAwAAgAggBgAAAAAAwAAAAAAAAEYAAAAAUoUAAPATAAAeAAGACCAGAAAAAADAAAAAAAAARgAA AABUhQAAAQAAAAQAAAA4LjUACwACgAggBgAAAAAAwAAAAAAAAEYAAAAABoUAAAAAAAADAAOACCAG AAAAAADAAAAAAAAARgAAAAABhQAAAAAAAAsABIAIIAYAAAAAAMAAAAAAAABGAAAAAAOFAAAAAAAA CwAFgAggBgAAAAAAwAAAAAAAAEYAAAAADoUAAAAAAAADAAaACCAGAAAAAADAAAAAAAAARgAAAAAQ hQAAAAAAAAMAB4AIIAYAAAAAAMAAAAAAAABGAAAAABGFAAAAAAAAAwAIgAggBgAAAAAAwAAAAAAA AEYAAAAAGIUAAAAAAAAeAAmACCAGAAAAAADAAAAAAAAARgAAAAA2hQAAAQAAAAEAAAAAAAAAHgAK gAggBgAAAAAAwAAAAAAAAEYAAAAAN4UAAAEAAAABAAAAAAAAAB4AC4AIIAYAAAAAAMAAAAAAAABG AAAAADiFAAABAAAAAQAAAAAAAAALAAyACyAGAAAAAADAAAAAAAAARgAAAAAAiAAAAAAAAAsADYAL IAYAAAAAAMAAAAAAAABGAAAAAAWIAAAAAAAAAgEJEAEAAABSBQAATgUAADoIAABMWkZ1/jPkMgMA CgByY3BnMTI1FjIA+Atgbg4QMDMznQH3IAKkA+MCAGNoCsDgc2V0MCAHEwKDAFD/AvIQ2QdtAoMO UANUEo8TnMYzBNYCAHBycRRxEPe9CFBtDeAGBAXgAoM0FIebF50Vr30KgAjIIDsJYvwxOQ5QCcMd cgoyHXEdI/sJqw4gOB6/CmAdJB/gHaVvIZEfnwn2DjA1AoAKgXUOYwBQCwMLtSBJRVQCRgYASVBQ SU5HECBleHAEkHRzLCcKogqECoAzRyXgIHO5AZBuZAsRBCAmYG8LUFRlIBEAdijAYgnhIIkJAG9r C4BnIGEFQJp2CsBpCGAoUW90CfDudAcxJ8AG8HUrAAIgBCBLAhAFwGMKwHJ5KcJEjFRNJaALgCB0 aCjAsyXQBdB1bCsAB4BkBzAhBgB1YnN5J9BlbUggKEkF4CkuJspP6m4owHMkYGgrRy0ABCDnLUIq gCjAb2YtMyXBJWBwTkZPIAeALUAEcC7OICVgKNQgIGFkLTMmQT5pICEtASrQBKARQC1kxHJhAYAs IHcFEAJAISlRYnkgSjSwaSDOSyuAEQADoCh3MyAu4O0zgHQIgTQhbywgAiEA0N0FQGY0sCfQNeBi K4A0NH4tAMADETQQNZAHkAQgd/phBCBuOGAJAA8gBJAp8JkqIWxpNCACIGUpKfDfKAAtMiAgMTIw YHAqsAAg/yvjMGMuQDC4A6AngjNTOuL9C4B2BvApADTiO7Eq0CiwLnAzIDAxOIFmPPFuY98owAtg J9A18AngazXwNiBvMJAqwDzhP0MgAQAosGf/KgAHkDXgLRE3gA3gMJBCoPwgcQpQJ9AroTTRN4A2 kK8tQDFBNZM0dj8zYFdE4M5sQnMo1ETgZ2g8AEiwvyrQNCArUAeAKqkXkG8CYP8u0DrBQwIqgCnC LUJGpDch/wVANrM2ACrBNeFLoTwRKgH/KLBCYhEwKMAtQzK4KfAEIPcuQEHQOvBpSnEx4AUwMQI2 KDrxQqBsOkExVFJUuyewCrB5CQA0EVBVLSVgfidIYjQRT6ELcDuhBGB1/wIwMeI59S4gBPAqgACQ MRE9QvNICfADACnRMRFSRvxDICKgD2AvTCeDAMA2kL5yVEA08jhgSkctFVI0EDsqYBFwY0IgBBEH wHR3wwWwQtAoUkFOPHEz8f8tADIARTEQ8CmQMcFRhjC39yvyLMIzUUE9QhDwTRNcAf84cgCQBIEp wjYCKcNY4zU5z1aSMX8yi14qSG9CoCkA3nI5UQEQBbBLoW9LRAQA9zXgN9EzIHVI8S0AQoEEAHtO UThgJ0RROZQ64DUhJ/0vW1RdPjFBUQNmEFVQB4DdNRFkNeA8kkKgJ1+yCHD5QfF0bDaQKYZZAzMR YsH/W7ExMTBgB5AzUi0kB4AAcPst4maxJy7gOvApsyvyKJD/C4Aro2JFO4AwMDWgAyA4gf9CIAUx YvFLKGNvXkZXeld0ukJoQlJEISgxJrtEVEDfLDALkHX/CvV5fmMAQQww8RISczMyLLB4owXQAxAP R9AKIAvFJtRiIFVN3lQF8SfmaZUFkGg7IAkA/mc2kH9wVBBXdAhQPQFbJrZEZUEJAHBrwld0VgRw UzWwQWJMdDQgKFslc/4pe3EOUCbUe4AW4A8CAdAsNTd9I4RIMYARbDoBDIIgKzQ0IDE2ADM1IDY3 NjA3DxmhFuAm1BmQIEZheMmGzTIzhzA0NXtxGaB/hGaKUC3AFFFYcQMQOFA6GmR4oy4YoHzxQHZm Fi5AQILELgWgLnVr/3txAUAPBotBAtF9lnooHEECAJCQAAAeAHAAAQAAABkAAABTSVAgSU5GTyBt ZXRob2QgZm9yIERUTUYAAAAAAgFxAAEAAAAWAAAAAcGy4hnxGrdirB7LEdaaXgAQWqhj2gAACwAC AAEAAAADAN4/r28AAEAAOQAQYZAZ4rLBAQMA8T8JBAAAHgAxQAEAAAAHAAAATUlMTFNEAAADABpA AAAAAB4AMEABAAAABwAAAE1JTExTRAAAAwAZQAAAAAADAP0/5AQAAAMAJgAAAAAAAwA2AAAAAAAD AIAQ/////wIBRwABAAAAPAAAAGM9R0I7YT1HT0xEIDQwMDtwPVZPREFGT05FO2w9SEFNUFNURUFE LTAyMDIxMTA5NTQzMFotMjc0OTg0AAIB+T8BAAAAVQAAAAAAAADcp0DIwEIQGrS5CAArL+GCAQAA AAAAAAAvTz1WT0RBRk9ORS9PVT1WTElNSVRFRC9DTj1FWENIQU5HRSBSRUNJUElFTlRTL0NOPU1J TExTRAAAAAAeAPg/AQAAAB8AAABNaWxscywgRHVuY2FuLCBELCAgRW5naW5lZXJpbmcAAB4AOEAB AAAABwAAAE1JTExTRAAAAgH7PwEAAABVAAAAAAAAANynQMjAQhAatLkIACsv4YIBAAAAAAAAAC9P PVZPREFGT05FL09VPVZMSU1JVEVEL0NOPUVYQ0hBTkdFIFJFQ0lQSUVOVFMvQ049TUlMTFNEAAAA AB4A+j8BAAAAHwAAAE1pbGxzLCBEdW5jYW4sIEQsICBFbmdpbmVlcmluZwAAHgA5QAEAAAAHAAAA TUlMTFNEAABAAAcwrJPZceCywQFAAAgwEBeJMuOywQEeAD0AAQAAAAEAAAAAAAAAHgAdDgEAAAAZ AAAAU0lQIElORk8gbWV0aG9kIGZvciBEVE1GAAAAAB4ANRABAAAAQgAAADxEREMzRDkyMUZCNjdE MjExQUFEMTAwQTBDOUU1OEY1MzBCMkM5REIwQEhhbXBzdGVhZC52Zmwudm9kYWZvbmU+AAAACwAp AAAAAAALACMAAAAAAAMABhDOmsdYAwAHEJIEAAADABAQAAAAAAMAERAAAAAAHgAIEAEAAABlAAAA SUVURlNJUFBJTkdFWFBFUlRTLDNHUFBTVEFOREFSRFNQRU9QTEVIQVZFQkVFTkxPT0tJTkdBVFZB UklPVVNQT1RFTlRJQUxTT0xVVElPTlNGT1JDQVJSWUlOR0RUTUZJTlRIRQAAAAACAX8AAQAAAEIA AAA8RERDM0Q5MjFGQjY3RDIxMUFBRDEwMEEwQzlFNThGNTMwQjJDOURCMEBIYW1wc3RlYWQudmZs LnZvZGFmb25lPgAAAN3I ------_=_NextPart_000_01C1B2E3.32891710-- _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@ns.ietf.org Mon Feb 11 06:52:20 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id GAA28585 for ; Mon, 11 Feb 2002 06:52:20 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id GAA02909 for sipping-archive@odin.ietf.org; Mon, 11 Feb 2002 06:52:21 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id GAA02688; Mon, 11 Feb 2002 06:40:06 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id GAA02657 for ; Mon, 11 Feb 2002 06:40:04 -0500 (EST) Received: from mr.tuwien.ac.at (mr.tuwien.ac.at [128.130.2.10]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id GAA28444 for ; Mon, 11 Feb 2002 06:40:02 -0500 (EST) Received: from eidolon (eidolon.ikn.tuwien.ac.at [128.131.88.99]) by mr.tuwien.ac.at (8.12.0/8.12.0) with SMTP id g1BBdLSq008004 for ; Mon, 11 Feb 2002 12:39:27 +0100 (MET) From: "Igor Miladinovic" To: Date: Mon, 11 Feb 2002 12:41:24 +0100 Message-ID: MIME-Version: 1.0 Content-Type: text/plain; charset="iso-8859-1" Content-Transfer-Encoding: 7bit X-Priority: 3 (Normal) X-MSMail-Priority: Normal X-Mailer: Microsoft Outlook IMO, Build 9.0.2416 (9.0.2911.0) X-MIMEOLE: Produced By Microsoft MimeOLE V5.50.4522.1200 Importance: Normal X-Virus-Scanned: by amavisd-milter (http://amavis.org/) Content-Transfer-Encoding: 7bit Subject: [Sipping] New draft about SIP conferencing avaiable Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit Hi, We have submitted a new draft to the IESG. It describes a SIP extension for discovery of participants identities in a multiparty conference. The draft is available at: http://search.ietf.org/internet-drafts/draft-miladinovic-sip-multiparty-ext- 00.txt or http://www.ikn.tuwien.ac.at/ftw-a1/download.html Your comments are welcome. Regards, Igor -- DI Igor Miladinovic Vienna University of Technology Institute of Communication Networks Call Control and Signaling Favoritenstrasse 9/388 A-1040 Vienna, Austria tel. +43-1-58801-38844 fax. +43-1-58801-38898 http://www.ikn.tuwien.ac.at _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@ns.ietf.org Mon Feb 11 07:09:36 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id HAA28914 for ; Mon, 11 Feb 2002 07:09:36 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id HAA04006 for sipping-archive@odin.ietf.org; Mon, 11 Feb 2002 07:09:37 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id GAA03127; Mon, 11 Feb 2002 06:56:50 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id GAA03062 for ; Mon, 11 Feb 2002 06:56:46 -0500 (EST) Received: from CNRI.Reston.VA.US (localhost [127.0.0.1]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id GAA28724; Mon, 11 Feb 2002 06:56:44 -0500 (EST) Message-Id: <200202111156.GAA28724@ietf.org> Mime-Version: 1.0 Content-Type: Multipart/Mixed; Boundary="NextPart" To: IETF-Announce: ; CC: enum@ietf.org Cc: sipping@ietf.org From: Internet-Drafts@ietf.org Reply-to: Internet-Drafts@ietf.org Date: Mon, 11 Feb 2002 06:56:44 -0500 Subject: [Sipping] I-D ACTION:draft-ietf-sipping-e164-01.txt Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org --NextPart A New Internet-Draft is available from the on-line Internet-Drafts directories. This draft is a work item of the Session Initiation Proposal Investigation Working Group of the IETF. Title : Using ENUM for SIP Applications Author(s) : J. Peterson, H. Liu, J. Yu, B. Campbell Filename : draft-ietf-sipping-e164-01.txt Pages : 35 Date : 08-Feb-02 Although SIP was clearly one of the primary applications for which ENUM was created, there is nevertheless widespread uncertainty about the demarcation of SIP location services from ENUM. This document attempts to sharpen the distinction between location services provided by the two protocols, illustrating how the two protocols might work in concert and clarifying the authoring and processing of ENUM records for SIP applications A URL for this Internet-Draft is: http://www.ietf.org/internet-drafts/draft-ietf-sipping-e164-01.txt To remove yourself from the IETF Announcement list, send a message to ietf-announce-request with the word unsubscribe in the body of the message. Internet-Drafts are also available by anonymous FTP. Login with the username "anonymous" and a password of your e-mail address. After logging in, type "cd internet-drafts" and then "get draft-ietf-sipping-e164-01.txt". A list of Internet-Drafts directories can be found in http://www.ietf.org/shadow.html or ftp://ftp.ietf.org/ietf/1shadow-sites.txt Internet-Drafts can also be obtained by e-mail. Send a message to: mailserv@ietf.org. In the body type: "FILE /internet-drafts/draft-ietf-sipping-e164-01.txt". NOTE: The mail server at ietf.org can return the document in MIME-encoded form by using the "mpack" utility. To use this feature, insert the command "ENCODING mime" before the "FILE" command. To decode the response(s), you will need "munpack" or a MIME-compliant mail reader. Different MIME-compliant mail readers exhibit different behavior, especially when dealing with "multipart" MIME messages (i.e. documents which have been split up into multiple messages), so check your local documentation on how to manipulate these messages. Below is the data which will enable a MIME compliant mail reader implementation to automatically retrieve the ASCII version of the Internet-Draft. --NextPart Content-Type: Multipart/Alternative; Boundary="OtherAccess" --OtherAccess Content-Type: Message/External-body; access-type="mail-server"; server="mailserv@ietf.org" Content-Type: text/plain Content-ID: <20020208130646.I-D@ietf.org> ENCODING mime FILE /internet-drafts/draft-ietf-sipping-e164-01.txt --OtherAccess Content-Type: Message/External-body; name="draft-ietf-sipping-e164-01.txt"; site="ftp.ietf.org"; access-type="anon-ftp"; directory="internet-drafts" Content-Type: text/plain Content-ID: <20020208130646.I-D@ietf.org> --OtherAccess-- --NextPart-- _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@ns.ietf.org Mon Feb 11 08:11:26 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id IAA00292 for ; Mon, 11 Feb 2002 08:11:26 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id IAA06698 for sipping-archive@odin.ietf.org; Mon, 11 Feb 2002 08:11:26 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id HAA05641; Mon, 11 Feb 2002 07:52:14 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id HAA05610 for ; Mon, 11 Feb 2002 07:52:12 -0500 (EST) Received: from zcars0m9.ca.nortel.com (zcars0m9.nortelnetworks.com [47.129.242.157]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id HAA29685 for ; Mon, 11 Feb 2002 07:52:11 -0500 (EST) Received: from zcars04f.ca.nortel.com (zcars04f.ca.nortel.com [47.129.242.57]) by zcars0m9.ca.nortel.com (Switch-2.2.0/Switch-2.2.0) with ESMTP id g1BCpZi02208 for ; Mon, 11 Feb 2002 07:51:36 -0500 (EST) Received: from zcard015.ca.nortel.com (zcard015.ca.nortel.com [47.129.30.7]) by zcars04f.ca.nortel.com (Switch-2.2.0/Switch-2.2.0) with ESMTP id g1BCpYV22769 for ; Mon, 11 Feb 2002 07:51:34 -0500 (EST) Received: by zcard015.ca.nortel.com with Internet Mail Service (5.5.2653.19) id <1J3PBN9N>; Mon, 11 Feb 2002 07:51:37 -0500 Message-ID: <4D79C746863DD51197690002A52CDA000102AFCF@zcard0kc.ca.nortel.com> From: "Tom-PT Taylor" To: "'duncan.mills@vf.vodafone.co.uk'" , sipping@ietf.org Date: Mon, 11 Feb 2002 07:51:44 -0500 X-Mailer: Internet Mail Service (5.5.2653.19) Subject: [Sipping] RE: SIP INFO method for DTMF Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org The key point to bear in mind is that your architecture has to make sure someone can't bring down network equipment simply by pressing their keypad repeatedly. The INFO method may produce unacceptably high volumes of messaging on the signalling path unless filtering is applied at the point of ingress. > -----Original Message----- > From: duncan.mills@vf.vodafone.co.uk > [mailto:duncan.mills@vf.vodafone.co.uk] > Sent: Monday, February 11, 2002 10:55 AM > To: sipping@ietf.org > Subject: SIP INFO method for DTMF > > IETF SIPPING experts, > > 3GPP standards people have been looking at various potential solutions for > carrying DTMF in the IP Multimedia Subsystem (IMS). > > One such solution is the use of the SIP INFO method. I have read the > expired internet-draft, written by Jiri Kuthan (whom I tried to contact > first, but the e-mail address was no longer a valid one) and there is > support for such a solution in 3GPP. I was involved in a telephone > conference last week with other 3GPP delegates, in which we questioned why > this draft expired? Whilst we have highlighted some potential problems > with using the draft that Jiri wrote, we did at least see the INFO method > as a feasible option (as well as the RTP payload option - I've had a fair > amount of e-mail discussion with Henning on RFC 2833). > > 3GPP may run into problems in the Radio Access Network (RAN) area if we > choose the RTP solution for DTMF. As such, we are considering writing an > internet-draft on the use of the SIP INFO method for DTMF. However, > before doing this, I thought it wise to 'test the water'. > > The RTP solution is well documented, and we're currently looking into > those RAN issues. In the meantime, I'm asking for opinions on the general > concept of using the SIP INFO method for DTMF. > > > Best Regards, > > Duncan > > > > > Duncan Mills > UMTS Standards > Technology Team > Core Network Development > Vodafone Ltd (Networks) > > Tel: +44 1635 676074 > Fax: +44 1635 234445 > mailto:duncan.mills@vf.vodafone.co.uk > _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@ns.ietf.org Mon Feb 11 09:52:41 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id JAA04270 for ; Mon, 11 Feb 2002 09:52:41 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id JAA11558 for sipping-archive@odin.ietf.org; Mon, 11 Feb 2002 09:52:42 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id JAA11056; Mon, 11 Feb 2002 09:40:15 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id JAA11025 for ; Mon, 11 Feb 2002 09:40:12 -0500 (EST) Received: from pmesmtp02.wcom.com (pmesmtp02.wcom.com [199.249.20.2]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id JAA03843 for ; Mon, 11 Feb 2002 09:40:11 -0500 (EST) Received: from CONVERSION-DAEMON by firewall.wcom.com (PMDF V5.2-32 #42257) id <0GRD00I01I1QOS@firewall.wcom.com> for sipping@ietf.org; Mon, 11 Feb 2002 14:39:26 +0000 (GMT) Received: from dgismtp01.wcomnet.com ([166.38.58.141]) by firewall.wcom.com (PMDF V5.2-32 #42257) with ESMTP id <0GRD00H7BI1PG0@firewall.wcom.com>; Mon, 11 Feb 2002 14:39:26 +0000 (GMT) Received: from dgismtp01.wcomnet.com by dgismtp01.wcomnet.com (PMDF V5.2-33 #42262) with SMTP id <0GRD00B01I1P5K@dgismtp01.wcomnet.com>; Mon, 11 Feb 2002 14:39:25 +0000 (GMT) Received: from ajohnston ([166.42.33.191]) by dgismtp01.wcomnet.com (PMDF V5.2-33 #42262) with ESMTP id <0GRD008EQI19KK@dgismtp01.wcomnet.com>; Mon, 11 Feb 2002 14:39:10 +0000 (GMT) Date: Mon, 11 Feb 2002 08:38:49 -0600 From: Alan Johnston Subject: RE: [Sipping] RE: SIP INFO method for DTMF In-reply-to: <4D79C746863DD51197690002A52CDA000102AFCF@zcard0kc.ca.nortel.com> To: "'Tom-PT Taylor'" , duncan.mills@vf.vodafone.co.uk, sipping@ietf.org Message-id: <000601c1b309$d2437b80$bf212aa6@ajohnston> MIME-version: 1.0 X-MIMEOLE: Produced By Microsoft MimeOLE V6.00.2600.0000 X-Mailer: Microsoft Outlook, Build 10.0.3416 Content-type: text/plain; charset=US-ASCII Content-transfer-encoding: 7bit Importance: Normal X-Priority: 3 (Normal) X-MSMail-priority: Normal Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit If you must send DTMF over SIP, then the NOTIFY method is preferred. See: http://search.ietf.org/internet-drafts/draft-mahy-sipping-signaled-digit s-00.txt Thanks, Alan Johnston WorldCom sip:alan@siptest.wcom.com > -----Original Message----- > From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] > On Behalf Of Tom-PT Taylor > Sent: Monday, February 11, 2002 6:52 AM > To: 'duncan.mills@vf.vodafone.co.uk'; sipping@ietf.org > Subject: [Sipping] RE: SIP INFO method for DTMF > > > The key point to bear in mind is that your architecture has > to make sure someone can't bring down network equipment > simply by pressing their keypad repeatedly. The INFO method > may produce unacceptably high volumes of messaging on the > signalling path unless filtering is applied at the point of ingress. > > > -----Original Message----- > > From: duncan.mills@vf.vodafone.co.uk > > [mailto:duncan.mills@vf.vodafone.co.uk] > > Sent: Monday, February 11, 2002 10:55 AM > > To: sipping@ietf.org > > Subject: SIP INFO method for DTMF > > > > IETF SIPPING experts, > > > > 3GPP standards people have been looking at various > potential solutions > > for carrying DTMF in the IP Multimedia Subsystem (IMS). > > > > One such solution is the use of the SIP INFO method. I > have read the > > expired internet-draft, written by Jiri Kuthan (whom I tried to > > contact first, but the e-mail address was no longer a valid > one) and > > there is support for such a solution in 3GPP. I was involved in a > > telephone conference last week with other 3GPP delegates, > in which we > > questioned why this draft expired? Whilst we have highlighted some > > potential problems with using the draft that Jiri wrote, we did at > > least see the INFO method as a feasible option (as well as the RTP > > payload option - I've had a fair amount of e-mail discussion with > > Henning on RFC 2833). > > > > 3GPP may run into problems in the Radio Access Network > (RAN) area if > > we choose the RTP solution for DTMF. As such, we are considering > > writing an internet-draft on the use of the SIP INFO method > for DTMF. > > However, before doing this, I thought it wise to 'test the water'. > > > > The RTP solution is well documented, and we're currently > looking into > > those RAN issues. In the meantime, I'm asking for opinions on the > > general concept of using the SIP INFO method for DTMF. > > > > > > Best Regards, > > > > Duncan > > > > > > > > > > Duncan Mills > > UMTS Standards > > Technology Team > > Core Network Development > > Vodafone Ltd (Networks) > > > > Tel: +44 1635 676074 > > Fax: +44 1635 234445 > > mailto:duncan.mills@vf.vodafone.co.uk > > > > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP > Use sip-implementors@cs.columbia.edu for questions on current > sip Use sip@ietf.org for new developments of core SIP > _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@ns.ietf.org Mon Feb 11 10:33:23 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id KAA05911 for ; Mon, 11 Feb 2002 10:33:23 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id KAA14387 for sipping-archive@odin.ietf.org; Mon, 11 Feb 2002 10:33:24 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id KAA13149; Mon, 11 Feb 2002 10:14:06 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id KAA13118 for ; Mon, 11 Feb 2002 10:14:04 -0500 (EST) Received: from acmepacket.com (mail1.acmepacket.com [63.67.143.10]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id KAA05020 for ; Mon, 11 Feb 2002 10:14:02 -0500 (EST) Received: from BobP [63.67.143.2] by acmepacket.com (SMTPD32-7.05) id AFBAE2CB02E6; Mon, 11 Feb 2002 10:14:02 -0500 Message-ID: <00f501c1b30e$a5428a40$2300000a@acmepacket.com> From: "Bob Penfield" To: "Juha Heinanen" Cc: References: <3C6493F7.3C902424@dynamicsoft.com> <15460.57538.457422.986299@harjus.eng.song.fi> Subject: Re: [Sipping] Register question (was [Sip] Anonymization of To & revisiting tags alone for dialog ID) Date: Mon, 11 Feb 2002 10:13:22 -0500 MIME-Version: 1.0 Content-Type: text/plain; charset="iso-8859-1" Content-Transfer-Encoding: 7bit X-Priority: 3 X-MSMail-Priority: Normal X-Mailer: Microsoft Outlook Express 5.50.4133.2400 X-MimeOLE: Produced By Microsoft MimeOLE V5.50.4133.2400 Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit ----- Original Message ----- From: "Juha Heinanen" > bis-07 says that in the register mmessage > > The Request-URI names the domain of the location service for which the > registration is meant (for example, sip:chicago.com ). 1514 > > and figure 2 in bis-07 shows the register message going *directly* from > the UA to the registrar without going through any proxies. > > now my questions are: > > 1) how does the UA know which host in the domain, in > the example above, within chicago.com, acts as the registrar of that > domain? If the UA can do DNS, it follows the procedures in: http://bdsl.greycouncil.com/sipwg/drafts/draft-ietf-sip-srv-04.txt > > 2) is there an srv record that points the UA to the correct host and if > not, why? yes. see above draft > > 3) why can't the UA simply send also the register message to its > outbound proxy that would them handle the dns lookup and find the host > acting as the registrar? The UA IS allowed to send the register request to its outbound proxy. The proxy will locate the registrar and send it on when it sees that it is not the target of the Request_URI in accordance with section 16.4 of bis-07. _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@ns.ietf.org Mon Feb 11 10:45:29 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id KAA06426 for ; Mon, 11 Feb 2002 10:45:29 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id KAA15145 for sipping-archive@odin.ietf.org; Mon, 11 Feb 2002 10:45:27 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id KAA13869; Mon, 11 Feb 2002 10:30:05 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id KAA13816 for ; Mon, 11 Feb 2002 10:30:02 -0500 (EST) Received: from lohi.eng.song.fi (lohi.eng.song.fi [195.10.149.18]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id KAA05802 for ; Mon, 11 Feb 2002 10:29:59 -0500 (EST) Received: from harjus.eng.song.fi (harjus.eng.song.fi [195.10.149.20]) by lohi.eng.song.fi (8.12.1/8.12.1/Debian -5) with ESMTP id g1BFTxio019060; Mon, 11 Feb 2002 17:29:59 +0200 From: Juha Heinanen MIME-Version: 1.0 Content-Type: text/plain; charset=us-ascii Content-Transfer-Encoding: 7bit Message-ID: <15463.58231.77286.234405@harjus.eng.song.fi> Date: Mon, 11 Feb 2002 17:29:59 +0200 To: "Bob Penfield" Cc: Subject: Re: [Sipping] Register question (was [Sip] Anonymization of To & revisiting tags alone for dialog ID) In-Reply-To: <00f501c1b30e$a5428a40$2300000a@acmepacket.com> References: <3C6493F7.3C902424@dynamicsoft.com> <15460.57538.457422.986299@harjus.eng.song.fi> <00f501c1b30e$a5428a40$2300000a@acmepacket.com> X-Mailer: VM 7.01 under Emacs 21.1.1 Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit Bob Penfield writes: > > 1) how does the UA know which host in the domain, in > > the example above, within chicago.com, acts as the registrar of that > > domain? > > If the UA can do DNS, it follows the procedures in: > http://bdsl.greycouncil.com/sipwg/drafts/draft-ietf-sip-srv-04.txt from that i-d i have found how a sip ua finds a proxy server of domain chicago.com, not how a sip ua finds a registrar of domain chicago.com. what if the proxy server of a domain is a different host than the registrar? -- juha _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@ns.ietf.org Mon Feb 11 10:52:44 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id KAA06851 for ; Mon, 11 Feb 2002 10:52:43 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id KAA15490 for sipping-archive@odin.ietf.org; Mon, 11 Feb 2002 10:52:45 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id KAA14227; Mon, 11 Feb 2002 10:31:11 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id KAA14165 for ; Mon, 11 Feb 2002 10:31:06 -0500 (EST) Received: from mail2.intervoice-brite.com (mail2.intervoice-brite.com [204.62.8.15]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id KAA05853 for ; Mon, 11 Feb 2002 10:31:02 -0500 (EST) Received: from DALNTMS02.ivbi.com (dalntms02.ivbi.com [151.214.90.44]) by mail2.intervoice-brite.com (Build 101 8.9.3/NT-8.9.3) with ESMTP id JAA01264 for ; Mon, 11 Feb 2002 09:34:06 -0600 Received: from 172.16.16.64 (unverified) by DALNTMS02.ivbi.com (Content Technologies SMTPRS 4.2.5) with SMTP id for ; Mon, 11 Feb 2002 09:14:34 -0600 Received: from bculpepperpc ([151.214.151.115]) by 172.16.16.64; Mon, 11 Feb 2002 09:30:20 -0600 From: "Bert Culpepper" To: "'Alan Johnston'" , "'Tom-PT Taylor'" , , Subject: RE: [Sipping] RE: SIP INFO method for DTMF Date: Mon, 11 Feb 2002 10:30:15 -0500 Message-ID: <003001c1b311$02593060$7397d697@bculpepperpc> MIME-Version: 1.0 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: 7bit X-Priority: 3 (Normal) X-MSMail-Priority: Normal X-Mailer: Microsoft Outlook CWS, Build 9.0.2416 (9.0.2911.0) X-MimeOLE: Produced By Microsoft MimeOLE V5.50.4910.0300 Importance: Normal In-Reply-To: <000601c1b309$d2437b80$bf212aa6@ajohnston> Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit I agree with the comments below when it's desired to transmit actual tone data in an IP network. If on the other hand, it's desired to have "user indications" transmitted to points in the SIP network in addition to the endpoints involved in the multi-media session, I believe a suitable approach is one myself and a couple of others have proposed. See http://search.ietf.org/internet-drafts/draft-culpepper-sip-key-events-00.txt . This is an initial draft and if thought worthwhile by the community, will be revised as necessary. Best regards, Bert > -----Original Message----- > From: sipping-admin@ietf.org > [mailto:sipping-admin@ietf.org]On Behalf Of > Alan Johnston > Sent: Monday, February 11, 2002 9:39 AM > To: 'Tom-PT Taylor'; duncan.mills@vf.vodafone.co.uk; sipping@ietf.org > Subject: RE: [Sipping] RE: SIP INFO method for DTMF > > > If you must send DTMF over SIP, then the NOTIFY method is preferred. > See: > > http://search.ietf.org/internet-drafts/draft-mahy-sipping-sign > aled-digit > s-00.txt > > Thanks, > Alan Johnston > WorldCom > sip:alan@siptest.wcom.com > > > -----Original Message----- > > From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] > > On Behalf Of Tom-PT Taylor > > Sent: Monday, February 11, 2002 6:52 AM > > To: 'duncan.mills@vf.vodafone.co.uk'; sipping@ietf.org > > Subject: [Sipping] RE: SIP INFO method for DTMF > > > > > > The key point to bear in mind is that your architecture has > > to make sure someone can't bring down network equipment > > simply by pressing their keypad repeatedly. The INFO method > > may produce unacceptably high volumes of messaging on the > > signalling path unless filtering is applied at the point of ingress. > > > > > -----Original Message----- > > > From: duncan.mills@vf.vodafone.co.uk > > > [mailto:duncan.mills@vf.vodafone.co.uk] > > > Sent: Monday, February 11, 2002 10:55 AM > > > To: sipping@ietf.org > > > Subject: SIP INFO method for DTMF > > > > > > IETF SIPPING experts, > > > > > > 3GPP standards people have been looking at various > > potential solutions > > > for carrying DTMF in the IP Multimedia Subsystem (IMS). > > > > > > One such solution is the use of the SIP INFO method. I > > have read the > > > expired internet-draft, written by Jiri Kuthan (whom I tried to > > > contact first, but the e-mail address was no longer a valid > > one) and > > > there is support for such a solution in 3GPP. I was > involved in a > > > telephone conference last week with other 3GPP delegates, > > in which we > > > questioned why this draft expired? Whilst we have > highlighted some > > > potential problems with using the draft that Jiri wrote, > we did at > > > least see the INFO method as a feasible option (as well > as the RTP > > > payload option - I've had a fair amount of e-mail discussion with > > > Henning on RFC 2833). > > > > > > 3GPP may run into problems in the Radio Access Network > > (RAN) area if > > > we choose the RTP solution for DTMF. As such, we are considering > > > writing an internet-draft on the use of the SIP INFO method > > for DTMF. > > > However, before doing this, I thought it wise to 'test the water'. > > > > > > The RTP solution is well documented, and we're currently > > looking into > > > those RAN issues. In the meantime, I'm asking for > opinions on the > > > general concept of using the SIP INFO method for DTMF. > > > > > > > > > Best Regards, > > > > > > Duncan > > > > > > > > > > > > > > > Duncan Mills > > > UMTS Standards > > > Technology Team > > > Core Network Development > > > Vodafone Ltd (Networks) > > > > > > Tel: +44 1635 676074 > > > Fax: +44 1635 234445 > > > mailto:duncan.mills@vf.vodafone.co.uk > > > > > > > _______________________________________________ > > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > > This list is for NEW development of the application of SIP > > Use sip-implementors@cs.columbia.edu for questions on current > > sip Use sip@ietf.org for new developments of core SIP > > > > > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP > Use sip-implementors@cs.columbia.edu for questions on current sip > Use sip@ietf.org for new developments of core SIP _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@ns.ietf.org Mon Feb 11 11:02:58 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id LAA07246 for ; Mon, 11 Feb 2002 11:02:58 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id LAA16236 for sipping-archive@odin.ietf.org; Mon, 11 Feb 2002 11:03:00 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id KAA15409; Mon, 11 Feb 2002 10:50:52 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id KAA15367 for ; Mon, 11 Feb 2002 10:50:49 -0500 (EST) Received: from mgw-x1.nokia.com (mgw-x1.nokia.com [131.228.20.21]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id KAA06768 for ; Mon, 11 Feb 2002 10:50:46 -0500 (EST) Received: from esvir05nok.ntc.nokia.com (esvir05nokt.ntc.nokia.com [172.21.143.37]) by mgw-x1.nokia.com (Switch-2.1.0/Switch-2.1.0) with ESMTP id g1BFoM309095 for ; Mon, 11 Feb 2002 17:50:22 +0200 (EET) Received: from esebh002.NOE.Nokia.com (unverified) by esvir05nok.ntc.nokia.com (Content Technologies SMTPRS 4.2.5) with ESMTP id ; Mon, 11 Feb 2002 17:50:47 +0200 Received: from esebe013.NOE.Nokia.com ([172.21.138.52]) by esebh002.NOE.Nokia.com with Microsoft SMTPSVC(5.0.2195.3779); Mon, 11 Feb 2002 17:50:47 +0200 Received: from essrv103nok149183.ntc.nokia.com ([172.21.149.183]) by esebe013.NOE.Nokia.com with Microsoft SMTPSVC(5.0.2195.2966); Mon, 11 Feb 2002 17:50:46 +0200 Subject: Re: [Sipping] Register question (was [Sip] Anonymization of To & revisiting tags alone for dialog ID) From: Aki Niemi To: ext Juha Heinanen Cc: Bob Penfield , sipping@ietf.org In-Reply-To: <15463.58231.77286.234405@harjus.eng.song.fi> References: <15463.58231.77286.234405@harjus.eng.song.fi> Content-Type: text/plain Content-Transfer-Encoding: 7bit X-Mailer: Evolution/1.0.2 Date: 11 Feb 2002 18:50:36 +0200 Message-Id: <1013446236.9427.249.camel@essrv103nok149183.ntc.nokia.com> Mime-Version: 1.0 X-OriginalArrivalTime: 11 Feb 2002 15:50:46.0901 (UTC) FILETIME=[DEEE8A50:01C1B313] Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit I would argue, that this depends on the configuration of chicago.com's network. The server at that domain receiving the request may accept and process it by itself, or may forward it to a server that does. Similarly, I do not have to know the exact Presence Server address at chicago.com, when I SUBSCRIBE to alice@chicago.com's presence. Br, Aki On Mon, 2002-02-11 at 17:29, ext Juha Heinanen wrote: > Bob Penfield writes: > > > > 1) how does the UA know which host in the domain, in > > > the example above, within chicago.com, acts as the registrar of that > > > domain? > > > > If the UA can do DNS, it follows the procedures in: > > http://bdsl.greycouncil.com/sipwg/drafts/draft-ietf-sip-srv-04.txt > > from that i-d i have found how a sip ua finds a proxy server of domain > chicago.com, not how a sip ua finds a registrar of domain chicago.com. > what if the proxy server of a domain is a different host than the > registrar? > > -- juha > > > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP > Use sip-implementors@cs.columbia.edu for questions on current sip > Use sip@ietf.org for new developments of core SIP > _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@ns.ietf.org Mon Feb 11 11:07:26 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id LAA07389 for ; Mon, 11 Feb 2002 11:07:25 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id LAA16528 for sipping-archive@odin.ietf.org; Mon, 11 Feb 2002 11:07:27 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id KAA15597; Mon, 11 Feb 2002 10:56:19 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id KAA15556 for ; Mon, 11 Feb 2002 10:56:16 -0500 (EST) Received: from lohi.eng.song.fi (lohi.eng.song.fi [195.10.149.18]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id KAA06967 for ; Mon, 11 Feb 2002 10:56:13 -0500 (EST) Received: from harjus.eng.song.fi (harjus.eng.song.fi [195.10.149.20]) by lohi.eng.song.fi (8.12.1/8.12.1/Debian -5) with ESMTP id g1BFuDio019085; Mon, 11 Feb 2002 17:56:13 +0200 From: Juha Heinanen MIME-Version: 1.0 Content-Type: text/plain; charset=us-ascii Content-Transfer-Encoding: 7bit Message-ID: <15463.59805.825763.352963@harjus.eng.song.fi> Date: Mon, 11 Feb 2002 17:56:13 +0200 To: Aki Niemi Cc: Bob Penfield , sipping@ietf.org Subject: Re: [Sipping] Register question (was [Sip] Anonymization of To & revisiting tags alone for dialog ID) In-Reply-To: <1013446236.9427.249.camel@essrv103nok149183.ntc.nokia.com> References: <15463.58231.77286.234405@harjus.eng.song.fi> <1013446236.9427.249.camel@essrv103nok149183.ntc.nokia.com> X-Mailer: VM 7.01 under Emacs 21.1.1 Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit Aki Niemi writes: > I would argue, that this depends on the configuration of chicago.com's > network. The server at that domain receiving the request may accept and > process it by itself, or may forward it to a server that does. yes, that is what i thought too, i.e., that the proxy serving chicago.com would know who is the registrar of chicago.com. it would be nice though if i would not need to manually configure in the proxy, who is the registrar of that domain. that would be possible if there would be a separate srv entry for registars. -- juha _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@ns.ietf.org Mon Feb 11 11:21:23 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id LAA07922 for ; Mon, 11 Feb 2002 11:21:22 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id LAA17226 for sipping-archive@odin.ietf.org; Mon, 11 Feb 2002 11:21:24 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id LAA16389; Mon, 11 Feb 2002 11:05:09 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id LAA16281 for ; Mon, 11 Feb 2002 11:05:03 -0500 (EST) Received: from web11607.mail.yahoo.com (web11607.mail.yahoo.com [216.136.172.59]) by ietf.org (8.9.1a/8.9.1a) with SMTP id LAA07304 for ; Mon, 11 Feb 2002 11:05:01 -0500 (EST) Message-ID: <20020211160503.95506.qmail@web11607.mail.yahoo.com> Received: from [12.237.5.37] by web11607.mail.yahoo.com via HTTP; Mon, 11 Feb 2002 08:05:03 PST Date: Mon, 11 Feb 2002 08:05:03 -0800 (PST) From: Sean Olson Subject: Re: [Sipping] New draft about SIP conferencing avaiable To: Igor Miladinovic , sipping@ietf.org In-Reply-To: MIME-Version: 1.0 Content-Type: text/plain; charset=us-ascii Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org How do you deal with the situation where a "conference" is identified by a r-uri and not by a pre-existing dialog? (How do you know the appropriate Call-ID to place in the CONF request?) /sean --- Igor Miladinovic wrote: > Hi, > > We have submitted a new draft to the IESG. It > describes a SIP extension for > discovery of participants identities in a multiparty > conference. The draft > is available at: > > http://search.ietf.org/internet-drafts/draft-miladinovic-sip-multiparty-ext- > 00.txt > or > http://www.ikn.tuwien.ac.at/ftw-a1/download.html > > Your comments are welcome. > > Regards, > > Igor > > -- > DI Igor Miladinovic > Vienna University of Technology > Institute of Communication Networks > Call Control and Signaling > Favoritenstrasse 9/388 > A-1040 Vienna, Austria > > tel. +43-1-58801-38844 > fax. +43-1-58801-38898 > http://www.ikn.tuwien.ac.at > > > _______________________________________________ > Sipping mailing list > https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application > of SIP > Use sip-implementors@cs.columbia.edu for questions > on current sip > Use sip@ietf.org for new developments of core SIP ===== Sean Olson __________________________________________________ Do You Yahoo!? Send FREE Valentine eCards with Yahoo! Greetings! http://greetings.yahoo.com _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@ns.ietf.org Mon Feb 11 11:26:30 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id LAA08167 for ; Mon, 11 Feb 2002 11:26:26 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id LAA17545 for sipping-archive@odin.ietf.org; Mon, 11 Feb 2002 11:26:28 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id LAA16857; Mon, 11 Feb 2002 11:15:01 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id LAA16826 for ; Mon, 11 Feb 2002 11:14:59 -0500 (EST) Received: from mgw-x2.nokia.com (mgw-x2.nokia.com [131.228.20.22]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id LAA07641 for ; Mon, 11 Feb 2002 11:14:56 -0500 (EST) Received: from esvir02nok.ntc.nokia.com (esvir02nokt.ntc.nokia.com [172.21.143.34]) by mgw-x2.nokia.com (Switch-2.1.0/Switch-2.1.0) with ESMTP id g1BGF3Z04814 for ; Mon, 11 Feb 2002 18:15:03 +0200 (EET) Received: from esebh001.NOE.Nokia.com (unverified) by esvir02nok.ntc.nokia.com (Content Technologies SMTPRS 4.2.5) with ESMTP id ; Mon, 11 Feb 2002 18:14:56 +0200 Received: from esebe013.NOE.Nokia.com ([172.21.138.52]) by esebh001.NOE.Nokia.com with Microsoft SMTPSVC(5.0.2195.3779); Mon, 11 Feb 2002 18:14:56 +0200 Received: from essrv103nok149183.ntc.nokia.com ([172.21.149.183]) by esebe013.NOE.Nokia.com with Microsoft SMTPSVC(5.0.2195.2966); Mon, 11 Feb 2002 18:14:56 +0200 Subject: Re: [Sipping] Register question (was [Sip] Anonymization of To &revisiting tags alone for dialog ID) From: Aki Niemi To: ext Juha Heinanen Cc: Bob Penfield , sipping@ietf.org In-Reply-To: <15463.59805.825763.352963@harjus.eng.song.fi> References: <15463.59805.825763.352963@harjus.eng.song.fi> Content-Type: text/plain Content-Transfer-Encoding: 7bit X-Mailer: Evolution/1.0.2 Date: 11 Feb 2002 19:14:46 +0200 Message-Id: <1013447686.8741.266.camel@essrv103nok149183.ntc.nokia.com> Mime-Version: 1.0 X-OriginalArrivalTime: 11 Feb 2002 16:14:56.0288 (UTC) FILETIME=[3ED57200:01C1B317] Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit On Mon, 2002-02-11 at 17:56, ext Juha Heinanen wrote: > Aki Niemi writes: > > > I would argue, that this depends on the configuration of chicago.com's > > network. The server at that domain receiving the request may accept and > > process it by itself, or may forward it to a server that does. > > yes, that is what i thought too, i.e., that the proxy serving > chicago.com would know who is the registrar of chicago.com. > > it would be nice though if i would not need to manually configure in the > proxy, who is the registrar of that domain. that would be possible if > there would be a separate srv entry for registars. However, I would consider announcing the SRV records of all of my network servers to the rest of the Internet more expensive than the configuration of the domain's authority proxy. Br, Aki _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@ns.ietf.org Mon Feb 11 11:55:32 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id LAA09806 for ; Mon, 11 Feb 2002 11:55:32 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id LAA19651 for sipping-archive@odin.ietf.org; Mon, 11 Feb 2002 11:55:34 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id LAA18970; Mon, 11 Feb 2002 11:44:00 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id LAA18926 for ; Mon, 11 Feb 2002 11:43:57 -0500 (EST) Received: from mr.tuwien.ac.at (mr.tuwien.ac.at [128.130.2.10]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id LAA09132 for ; Mon, 11 Feb 2002 11:43:53 -0500 (EST) Received: from eidolon (eidolon.ikn.tuwien.ac.at [128.131.88.99]) by mr.tuwien.ac.at (8.12.0/8.12.0) with SMTP id g1BGhoSq003002; Mon, 11 Feb 2002 17:43:50 +0100 (MET) From: "Igor Miladinovic" To: , "Sean Olson" Cc: "Johannes Stadler" Subject: RE: [Sipping] New draft about SIP conferencing avaiable Date: Mon, 11 Feb 2002 17:45:52 +0100 Message-ID: MIME-Version: 1.0 Content-Type: text/plain; charset="US-ASCII" Content-Transfer-Encoding: 7bit X-Priority: 3 (Normal) X-MSMail-Priority: Normal X-Mailer: Microsoft Outlook IMO, Build 9.0.2416 (9.0.2911.0) In-reply-to: <20020211160503.95506.qmail@web11607.mail.yahoo.com> X-MIMEOLE: Produced By Microsoft MimeOLE V5.50.4522.1200 Importance: Normal X-Virus-Scanned: by amavisd-milter (http://amavis.org/) Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit > How do you deal with the situation where a > "conference" is identified by a r-uri and not by > a pre-existing dialog? (How do you know the > appropriate Call-ID to place in the CONF request?) > > /sean The conference server stores data (at least the sip-address) of each participant in a conference, which is identified by a requestURI. This data contains also the Call-ID of the corresponding dialog (determined by the initial INVITE). -- Igor > > --- Igor Miladinovic > wrote: > > Hi, > > > > We have submitted a new draft to the IESG. It > > describes a SIP extension for > > discovery of participants identities in a multiparty > > conference. The draft > > is available at: > > > > > http://search.ietf.org/internet-drafts/draft-miladinovic-sip-multi > party-ext- > > 00.txt > > or > > http://www.ikn.tuwien.ac.at/ftw-a1/download.html > > > > Your comments are welcome. > > > > Regards, > > > > Igor > > > > -- > > DI Igor Miladinovic > > Vienna University of Technology > > Institute of Communication Networks > > Call Control and Signaling > > Favoritenstrasse 9/388 > > A-1040 Vienna, Austria > > > > tel. +43-1-58801-38844 > > fax. +43-1-58801-38898 > > http://www.ikn.tuwien.ac.at > > > > > > _______________________________________________ > > Sipping mailing list > > https://www1.ietf.org/mailman/listinfo/sipping > > This list is for NEW development of the application > > of SIP > > Use sip-implementors@cs.columbia.edu for questions > > on current sip > > Use sip@ietf.org for new developments of core SIP > > > ===== > Sean Olson > > __________________________________________________ > Do You Yahoo!? > Send FREE Valentine eCards with Yahoo! Greetings! > http://greetings.yahoo.com > > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP > Use sip-implementors@cs.columbia.edu for questions on current sip > Use sip@ietf.org for new developments of core SIP > _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Mon Feb 11 12:15:00 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id MAA11081 for ; Mon, 11 Feb 2002 12:15:00 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id MAA21953 for sipping-archive@odin.ietf.org; Mon, 11 Feb 2002 12:15:02 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id MAA21381; Mon, 11 Feb 2002 12:03:34 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id MAA21347 for ; Mon, 11 Feb 2002 12:03:31 -0500 (EST) Received: from bdsl.66.12.12.130.gte.net (bdsl.66.12.12.130.gte.net [66.12.12.130]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id MAA10352 for ; Mon, 11 Feb 2002 12:03:28 -0500 (EST) Received: from dwillis (bdsl.greycouncil.com [127.0.0.1]) by bdsl.66.12.12.130.gte.net (8.11.6/8.11.6) with SMTP id g1BH2qw07737; Mon, 11 Feb 2002 11:02:52 -0600 Message-ID: <010501c1b31e$19f8a180$d2036e3f@dfw.dynamicsoft.com> From: "Dean Willis" To: "Juha Heinanen" Cc: References: <3C6493F7.3C902424@dynamicsoft.com><15460.57538.457422.986299@harjus.eng.song.fi><00f501c1b30e$a5428a40$2300000a@acmepacket.com> <15463.58231.77286.234405@harjus.eng.song.fi> Subject: Re: [Sipping] Register question (was [Sip] Anonymization of To & revisiting tags alone for dialog ID) Date: Mon, 11 Feb 2002 11:03:57 -0600 MIME-Version: 1.0 Content-Type: text/plain; charset="iso-8859-1" Content-Transfer-Encoding: 7bit X-Priority: 3 X-MSMail-Priority: Normal X-Mailer: Microsoft Outlook Express 6.00.2600.0000 X-MimeOLE: Produced By Microsoft MimeOLE V6.00.2600.0000 Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit ----- Original Message ----- From: "Juha Heinanen" To: "Bob Penfield" Cc: Sent: Monday, February 11, 2002 9:29 AM Subject: Re: [Sipping] Register question (was [Sip] Anonymization of To & revisiting tags alone for dialog ID) > Bob Penfield writes: > > > > 1) how does the UA know which host in the domain, in > > > the example above, within chicago.com, acts as the registrar of that > > > domain? > > > > If the UA can do DNS, it follows the procedures in: > > http://bdsl.greycouncil.com/sipwg/drafts/draft-ietf-sip-srv-04.txt > > from that i-d i have found how a sip ua finds a proxy server of domain > chicago.com, not how a sip ua finds a registrar of domain chicago.com. > what if the proxy server of a domain is a different host than the > registrar? > > -- juha That's chicago.com's problem. The most likely configuration is that their externally visible proxy either IS a registrar, or has message-specific routing to send REGISTER messages on to a registrar. This configuration, however, is not required. Some phones allow their registrar to be configured seperately from their home proxy, so one could build a diffeent configuration. How does you mail client know where your POP server is, or where your SMTP server is? You tell it. Personally, I use nine of each, depending on which of my multiple personalities is in control at the moment, but most people just have one. -- Dean _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Mon Feb 11 12:33:44 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id MAA12039 for ; Mon, 11 Feb 2002 12:33:44 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id MAA23006 for sipping-archive@odin.ietf.org; Mon, 11 Feb 2002 12:33:47 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id MAA22186; Mon, 11 Feb 2002 12:19:54 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id MAA22155 for ; Mon, 11 Feb 2002 12:19:51 -0500 (EST) Received: from igate2.vodafone.co.uk (igate2.vodafone.co.uk [194.62.232.66]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id MAA11397 for ; Mon, 11 Feb 2002 12:19:47 -0500 (EST) From: duncan.mills@vf.vodafone.co.uk Received: by igate2.vodafone.co.uk; (8.8.8/1.3/10May95) id RAA23657; Mon, 11 Feb 2002 17:19:19 GMT Received: from mimesweeper1.vfl.vodafone (mimesweeper1 [10.33.32.67]) by mailguard1 (4.6.1.91) with ESMTP id for ; Mon, 11 Feb 2002 17:19:04 GMT Received: from putney.vfl.vodafone (unverified) by mimesweeper1.vfl.vodafone (Content Technologies SMTPRS 2.0.15) with ESMTP id ; Mon, 11 Feb 2002 17:19:24 +0000 Received: by putney.vfl.vodafone with Internet Mail Service (5.5.2650.21) id <1XSFPBWD>; Mon, 11 Feb 2002 17:19:24 -0000 Message-Id: To: bert.culpepper@intervoice-brite.com, alan.johnston@wcom.com, taylor@nortelnetworks.com, sipping@ietf.org Subject: RE: [Sipping] RE: SIP INFO method for DTMF Date: Mon, 11 Feb 2002 17:15:20 -0000 X-Mailer: Internet Mail Service (5.5.2650.21) MIME-Version: 1.0 (Generated by Clearswift ES version 4.6.1.109) Content-Type: text/plain; charset="iso-8859-1" Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Hi All, Thanks for your rapid responses. The interested parties in this DTMF discussion in 3GPP have already considered the SUBSCRIBE/NOTIFY solution and rejected it as unsuitable. Firstly, it appeared to us that for every mobile originated voice call, the network would have to SUBSCRIBE to the UA making the call, on the off-chance that at some point during the call the user wishes to send DTMF in NOTIFY messages. This was seen as unacceptable. Any views? Regards, Duncan -----Original Message----- From: Bert Culpepper [mailto:bert.culpepper@intervoice-brite.com] Sent: 11 February 2002 15:30 To: 'Alan Johnston'; 'Tom-PT Taylor'; duncan.mills@vf.vodafone.co.uk; sipping@ietf.org Subject: RE: [Sipping] RE: SIP INFO method for DTMF I agree with the comments below when it's desired to transmit actual tone data in an IP network. If on the other hand, it's desired to have "user indications" transmitted to points in the SIP network in addition to the endpoints involved in the multi-media session, I believe a suitable approach is one myself and a couple of others have proposed. See http://search.ietf.org/internet-drafts/draft-culpepper-sip-key-events-00.txt . This is an initial draft and if thought worthwhile by the community, will be revised as necessary. Best regards, Bert > -----Original Message----- > From: sipping-admin@ietf.org > [mailto:sipping-admin@ietf.org]On Behalf Of > Alan Johnston > Sent: Monday, February 11, 2002 9:39 AM > To: 'Tom-PT Taylor'; duncan.mills@vf.vodafone.co.uk; sipping@ietf.org > Subject: RE: [Sipping] RE: SIP INFO method for DTMF > > > If you must send DTMF over SIP, then the NOTIFY method is preferred. > See: > > http://search.ietf.org/internet-drafts/draft-mahy-sipping-sign > aled-digit > s-00.txt > > Thanks, > Alan Johnston > WorldCom > sip:alan@siptest.wcom.com > > > -----Original Message----- > > From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] > > On Behalf Of Tom-PT Taylor > > Sent: Monday, February 11, 2002 6:52 AM > > To: 'duncan.mills@vf.vodafone.co.uk'; sipping@ietf.org > > Subject: [Sipping] RE: SIP INFO method for DTMF > > > > > > The key point to bear in mind is that your architecture has > > to make sure someone can't bring down network equipment > > simply by pressing their keypad repeatedly. The INFO method > > may produce unacceptably high volumes of messaging on the > > signalling path unless filtering is applied at the point of ingress. > > > > > -----Original Message----- > > > From: duncan.mills@vf.vodafone.co.uk > > > [mailto:duncan.mills@vf.vodafone.co.uk] > > > Sent: Monday, February 11, 2002 10:55 AM > > > To: sipping@ietf.org > > > Subject: SIP INFO method for DTMF > > > > > > IETF SIPPING experts, > > > > > > 3GPP standards people have been looking at various > > potential solutions > > > for carrying DTMF in the IP Multimedia Subsystem (IMS). > > > > > > One such solution is the use of the SIP INFO method. I > > have read the > > > expired internet-draft, written by Jiri Kuthan (whom I tried to > > > contact first, but the e-mail address was no longer a valid > > one) and > > > there is support for such a solution in 3GPP. I was > involved in a > > > telephone conference last week with other 3GPP delegates, > > in which we > > > questioned why this draft expired? Whilst we have > highlighted some > > > potential problems with using the draft that Jiri wrote, > we did at > > > least see the INFO method as a feasible option (as well > as the RTP > > > payload option - I've had a fair amount of e-mail discussion with > > > Henning on RFC 2833). > > > > > > 3GPP may run into problems in the Radio Access Network > > (RAN) area if > > > we choose the RTP solution for DTMF. As such, we are considering > > > writing an internet-draft on the use of the SIP INFO method > > for DTMF. > > > However, before doing this, I thought it wise to 'test the water'. > > > > > > The RTP solution is well documented, and we're currently > > looking into > > > those RAN issues. In the meantime, I'm asking for > opinions on the > > > general concept of using the SIP INFO method for DTMF. > > > > > > > > > Best Regards, > > > > > > Duncan > > > > > > > > > > > > > > > Duncan Mills > > > UMTS Standards > > > Technology Team > > > Core Network Development > > > Vodafone Ltd (Networks) > > > > > > Tel: +44 1635 676074 > > > Fax: +44 1635 234445 > > > mailto:duncan.mills@vf.vodafone.co.uk > > > > > > > _______________________________________________ > > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > > This list is for NEW development of the application of SIP > > Use sip-implementors@cs.columbia.edu for questions on current > > sip Use sip@ietf.org for new developments of core SIP > > > > > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP > Use sip-implementors@cs.columbia.edu for questions on current sip > Use sip@ietf.org for new developments of core SIP _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Mon Feb 11 12:41:49 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id MAA12314 for ; Mon, 11 Feb 2002 12:41:44 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id MAA23314 for sipping-archive@odin.ietf.org; Mon, 11 Feb 2002 12:41:45 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id MAA22359; Mon, 11 Feb 2002 12:27:29 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id MAA22328 for ; Mon, 11 Feb 2002 12:27:26 -0500 (EST) Received: from exchange1.nuera.com ([12.105.228.79]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id MAA11739 for ; Mon, 11 Feb 2002 12:27:23 -0500 (EST) Received: by exchange1.nuera.com with Internet Mail Service (5.5.2653.19) id <1SJWQ43K>; Mon, 11 Feb 2002 09:26:57 -0800 Message-ID: From: "Fairlie-Cuninghame, Robert" To: "'Bert Culpepper'" , "'Alan Johnston'" , "'Tom-PT Taylor'" , duncan.mills@vf.vodafone.co.uk, sipping@ietf.org Subject: RE: [Sipping] RE: SIP INFO method for DTMF Date: Mon, 11 Feb 2002 09:26:49 -0800 MIME-Version: 1.0 X-Mailer: Internet Mail Service (5.5.2653.19) Content-Type: text/plain; charset="iso-8859-1" Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Hi all, In response to Tom Taylor's point that a user could flood the control plane by pressing a key too many times: In our key-events draft [draft-culpepper-sip-key-events-00.txt], the event indications can simply be aggregated to maintain a maximum NOTIFY rate. As an aside, it seems strange to me the number of people that are still very interested in pushing the signaled-digits draft [draft-mahy-sipping-signaled-digits-00.txt] - which only supports the transport of RFC2833/DTMF tones - at a time when we are trying to move away from PSTN-centric user interfaces!! The key-events draft is designed to support keyboards, function keys as well as other device specific buttons (such as a "Double Latte" button). It is designed to build a framework that will support the breadth and depth of SIP devices to come (whilst still scaling down to a simple implementation for devices supporting only simple interfaces - e.g., a telephone dialpad). It is not reasonable to develop a separate event framework to receive event notification for every new device that is developed. Signaled-digits does not scale - it only supports DTMF. To Rohan Mahy's credit, I think this draft was a great conceptual leap forward from the previous INFO approach however, I don't think that embedding RFC2833 in NOTIFY's is the correct way forward. The SUBSCRIBE/NOTIFY mechanism is essentially the same between the two drafts except that the key-events drafts tries to address the problem of associating a subscription with a single endpoint rather than all end-points on the device. This is essential for the mechanism to be workable on devices such as SIP PSTN gateways or multi-line SIP phones. Also, please note: Our hope (that is, the authors of the Key-Events draft) is that the SIP-Events draft eventually solves this in a consistent manner. As Bert said, this is only an initial draft and we will update the draft to keep step with the Events draft and the IETF community. We are keen to get feedback on what people think about this draft. Regards, Robert. > -----Original Message----- > From: Bert Culpepper [mailto:bert.culpepper@intervoice-brite.com] > Sent: Monday, February 11, 2002 3:30 PM > To: 'Alan Johnston'; 'Tom-PT Taylor'; duncan.mills@vf.vodafone.co.uk; > sipping@ietf.org > Subject: RE: [Sipping] RE: SIP INFO method for DTMF > > > I agree with the comments below when it's desired to transmit > actual tone > data in an IP network. If on the other hand, it's desired to > have "user > indications" transmitted to points in the SIP network in > addition to the > endpoints involved in the multi-media session, I believe a > suitable approach > is one myself and a couple of others have proposed. See > http://search.ietf.org/internet-drafts/draft-culpepper-sip-key > -events-00.txt > . This is an initial draft and if thought worthwhile by the > community, will > be revised as necessary. > > Best regards, > Bert > > > -----Original Message----- > > From: sipping-admin@ietf.org > > [mailto:sipping-admin@ietf.org]On Behalf Of > > Alan Johnston > > Sent: Monday, February 11, 2002 9:39 AM > > To: 'Tom-PT Taylor'; duncan.mills@vf.vodafone.co.uk; > sipping@ietf.org > > Subject: RE: [Sipping] RE: SIP INFO method for DTMF > > > > > > If you must send DTMF over SIP, then the NOTIFY method is preferred. > > See: > > > > http://search.ietf.org/internet-drafts/draft-mahy-sipping-sign > > aled-digit > > s-00.txt > > > > Thanks, > > Alan Johnston > > WorldCom > > sip:alan@siptest.wcom.com > > > > > -----Original Message----- > > > From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] > > > On Behalf Of Tom-PT Taylor > > > Sent: Monday, February 11, 2002 6:52 AM > > > To: 'duncan.mills@vf.vodafone.co.uk'; sipping@ietf.org > > > Subject: [Sipping] RE: SIP INFO method for DTMF > > > > > > > > > The key point to bear in mind is that your architecture has > > > to make sure someone can't bring down network equipment > > > simply by pressing their keypad repeatedly. The INFO method > > > may produce unacceptably high volumes of messaging on the > > > signalling path unless filtering is applied at the point > of ingress. > > > > > > > -----Original Message----- > > > > From: duncan.mills@vf.vodafone.co.uk > > > > [mailto:duncan.mills@vf.vodafone.co.uk] > > > > Sent: Monday, February 11, 2002 10:55 AM > > > > To: sipping@ietf.org > > > > Subject: SIP INFO method for DTMF > > > > > > > > IETF SIPPING experts, > > > > > > > > 3GPP standards people have been looking at various > > > potential solutions > > > > for carrying DTMF in the IP Multimedia Subsystem (IMS). > > > > > > > > One such solution is the use of the SIP INFO method. I > > > have read the > > > > expired internet-draft, written by Jiri Kuthan (whom I tried to > > > > contact first, but the e-mail address was no longer a valid > > > one) and > > > > there is support for such a solution in 3GPP. I was > > involved in a > > > > telephone conference last week with other 3GPP delegates, > > > in which we > > > > questioned why this draft expired? Whilst we have > > highlighted some > > > > potential problems with using the draft that Jiri wrote, > > we did at > > > > least see the INFO method as a feasible option (as well > > as the RTP > > > > payload option - I've had a fair amount of e-mail > discussion with > > > > Henning on RFC 2833). > > > > > > > > 3GPP may run into problems in the Radio Access Network > > > (RAN) area if > > > > we choose the RTP solution for DTMF. As such, we are > considering > > > > writing an internet-draft on the use of the SIP INFO method > > > for DTMF. > > > > However, before doing this, I thought it wise to 'test > the water'. > > > > > > > > The RTP solution is well documented, and we're currently > > > looking into > > > > those RAN issues. In the meantime, I'm asking for > > opinions on the > > > > general concept of using the SIP INFO method for DTMF. > > > > > > > > > > > > Best Regards, > > > > > > > > Duncan > > > > > > > > > > > > > > > > > > > > Duncan Mills > > > > UMTS Standards > > > > Technology Team > > > > Core Network Development > > > > Vodafone Ltd (Networks) > > > > > > > > Tel: +44 1635 676074 > > > > Fax: +44 1635 234445 > > > > mailto:duncan.mills@vf.vodafone.co.uk > > > > > > > > > > _______________________________________________ > > > Sipping mailing list > https://www1.ietf.org/mailman/listinfo/sipping > > > This list is for NEW development of the application of SIP > > > Use sip-implementors@cs.columbia.edu for questions on current > > > sip Use sip@ietf.org for new developments of core SIP > > > > > > > > > _______________________________________________ > > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > > This list is for NEW development of the application of SIP > > Use sip-implementors@cs.columbia.edu for questions on current sip > > Use sip@ietf.org for new developments of core SIP > > > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP > Use sip-implementors@cs.columbia.edu for questions on current sip > Use sip@ietf.org for new developments of core SIP > _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Mon Feb 11 12:57:34 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id MAA12643 for ; Mon, 11 Feb 2002 12:57:34 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id MAA23830 for sipping-archive@odin.ietf.org; Mon, 11 Feb 2002 12:57:35 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id MAA23291; Mon, 11 Feb 2002 12:41:13 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id MAA23260 for ; Mon, 11 Feb 2002 12:41:09 -0500 (EST) Received: from cerbrus.convedia.com (cerbrus.convedia.com [216.129.93.50]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id MAA12279 for ; Mon, 11 Feb 2002 12:41:07 -0500 (EST) Received: from ladon.convedia.com (ladon [192.168.208.15]) by cerbrus.convedia.com (8.9.3+Sun/8.9.3) with ESMTP id JAA15717; Mon, 11 Feb 2002 09:39:18 -0800 (PST) Received: from laptop52 (laptop52.convedia.com [192.168.208.151]) by ladon.convedia.com (8.9.3+Sun/8.9.3) with SMTP id JAA18533; Mon, 11 Feb 2002 09:40:35 -0800 (PST) From: "Garland Sharratt" To: Cc: Date: Mon, 11 Feb 2002 09:41:14 -0800 Message-ID: MIME-Version: 1.0 Content-Type: multipart/mixed; boundary="----=_NextPart_000_0031_01C1B2E0.3EDBDFA0" X-Priority: 3 (Normal) X-MSMail-Priority: Normal X-Mailer: Microsoft Outlook IMO, Build 9.0.2416 (9.0.2911.0) Importance: Normal X-MimeOLE: Produced By Microsoft MimeOLE V6.00.2600.0000 In-Reply-To: X-MS-TNEF-Correlator: Subject: [Sipping] RE: SIP INFO method for DTMF Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org This is a multi-part message in MIME format. ------=_NextPart_000_0031_01C1B2E0.3EDBDFA0 Content-Type: text/plain; charset="iso-8859-1" Content-Transfer-Encoding: 7bit Duncan: From an architectural point of view, the clean way to carry post-dial DTMF is in RTP using RFC 2833, because this preserves the separation of media (audio) from signaling. Some reasons for keeping the separation include: * the network is architected along this separation, with app servers and softswitches handling signaling and media gateways and media servers (Media Resource Functions) handling media; and * when post-dial DTMF is replaced with speech later, it will no longer fit in the signaling. Of course they may be very valid reasons in particular cases to do otherwise. Garland > -----Original Message----- > From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] On > Behalf Of duncan.mills@vf.vodafone.co.uk > Sent: Monday, February 11, 2002 01:55 > To: sipping@ietf.org > Subject: SIP INFO method for DTMF > > IETF SIPPING experts, > > 3GPP standards people have been looking at various potential solutions for > carrying DTMF in the IP Multimedia Subsystem (IMS). > > One such solution is the use of the SIP INFO method. I have read the > expired internet-draft, written by Jiri Kuthan (whom I tried to contact > first, but the e-mail address was no longer a valid one) and there is > support for such a solution in 3GPP. I was involved in a telephone > conference last week with other 3GPP delegates, in which we questioned why > this draft expired? Whilst we have highlighted some potential problems > with using the draft that Jiri wrote, we did at least see the INFO method > as a feasible option (as well as the RTP payload option - I've had a fair > amount of e-mail discussion with Henning on RFC 2833). > > 3GPP may run into problems in the Radio Access Network (RAN) area if we > choose the RTP solution for DTMF. As such, we are considering writing an > internet-draft on the use of the SIP INFO method for DTMF. However, > before doing this, I thought it wise to 'test the water'. > > The RTP solution is well documented, and we're currently looking into > those RAN issues. In the meantime, I'm asking for opinions on the general > concept of using the SIP INFO method for DTMF. > > > Best Regards, > > Duncan > > > > > Duncan Mills > UMTS Standards > Technology Team > Core Network Development > Vodafone Ltd (Networks) > > Tel: +44 1635 676074 > Fax: +44 1635 234445 > mailto:duncan.mills@vf.vodafone.co.uk > ------=_NextPart_000_0031_01C1B2E0.3EDBDFA0 Content-Type: application/ms-tnef; name="winmail.dat" Content-Disposition: attachment; filename="winmail.dat" Content-Transfer-Encoding: base64 eJ8+Ig4RAQaQCAAEAAAAAAABAAEAAQeQBgAIAAAA5AQAAAAAAADoAAEIgAcAGAAAAElQTS5NaWNy b3NvZnQgTWFpbC5Ob3RlADEIAQ2ABAACAAAAAgACAAEGgAMADgAAANIHAgALAAkAKQAAAAEAGQEB A5AGACQMAAAkAAAACwACAAEAAAALACMAAAAAAAMAJgAAAAAACwApAAAAAAADAC4AAAAAAAMANgAA AAAAHgBwAAEAAAAZAAAAU0lQIElORk8gbWV0aG9kIGZvciBEVE1GAAAAAAIBcQABAAAAGwAAAAHB suIZ8Rq3YqweyxHWml4AEFqoY9oAEAjMIAACAR0MAQAAABwAAABTTVRQOkdTSEFSUkFUVEBDT05W RURJQS5DT00ACwABDgAAAABAAAYOAF6lRCOzwQECAQoOAQAAABgAAAAAAAAAqwKbb9RX1BGOkwBg 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Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Mon Feb 11 13:44:22 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id NAA13734 for ; Mon, 11 Feb 2002 13:44:22 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id NAA25896 for sipping-archive@odin.ietf.org; Mon, 11 Feb 2002 13:44:23 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id NAA25401; Mon, 11 Feb 2002 13:32:28 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id NAA25370 for ; Mon, 11 Feb 2002 13:32:26 -0500 (EST) Received: from mail3.dynamicsoft.com ([63.113.44.69]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id NAA13437 for ; Mon, 11 Feb 2002 13:32:24 -0500 (EST) Received: from dynamicsoft.com ([63.113.46.52]) by mail3.dynamicsoft.com (8.12.1/8.12.1) with ESMTP id g1BIWU6Y024930; Mon, 11 Feb 2002 13:32:31 -0500 (EST) Message-ID: <3C680E0E.6FB20E30@dynamicsoft.com> Date: Mon, 11 Feb 2002 13:31:42 -0500 From: Jonathan Rosenberg Organization: dynamicsoft X-Mailer: Mozilla 4.75 [en] (Windows NT 5.0; U) X-Accept-Language: en MIME-Version: 1.0 To: Garland Sharratt CC: duncan.mills@vf.vodafone.co.uk, sipping@ietf.org Subject: Re: [Sipping] RE: SIP INFO method for DTMF References: Content-Type: text/plain; charset=us-ascii Content-Transfer-Encoding: 7bit Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit Let me put in my two cents here. There has been fierce religion on this issue because DTMF plays so many roles in a network. Its most common use is truly media. Between gateways, for example, it needs to be placed into the media stream at the terminating gateway. This requires synchronization with the media stream. That strongly argues for rfc2833. The use of INFO or SUBSCRIBE for DTMF in this usage is bad; it is a load on the network elements. The other use is to signal IP applications. In this use, SIP mechanisms do have a benefit. We may want applications that are not on the path of the media stream to find out about the events. THere may be no issues with media synchronization (if there are, vxml types of solutions with forked media may be needed, as I have long advocated). But, for simple DTMF-only apps, rfc2833 is expensive and provides little gain. Thus, my proposal is: separate solutions for very different problems. When an originator detects DTMF, it always inserts it as rfc2833 into the media stream. This makes sure the e2e media applications work. However, if there are applications not on the media path that need this information, they can SUBSCRIBE to the UA, to receive input on events to drive the application. I think this is broader than DTMF. For example, my presence server needs an "is typing" indication, to support the common IM feature, where you know when your peer is typing. I would propose that this use the same SUBSCRIBE/NOTIFY mechanism. In the 3gpp case, this would mean rfc2833 as the baseline, with no SUBSCRIBE/NOTIFY on a call unless an app needed it. Using rfc2833 instead of INFO is really, really, really important. The usage of INFO for e2e transport of DTMF suffers many problems: 1. because of the separation of media and signaling paths, AND because of the longer signaling paths, AND because of the use of retransmits on the signaling path, the overall latency for transfer of DTMF through INFO will be huge compared to rfc2833. Thus, by using INFO, the jitter buffers at the receiver will need to be monstrous to properly re-insert DTMF into the output media stream. This affects overall voice latency and perceived user quality. 2. The load on the proxies will be huge. Given the frequency of DTMF usage in a call, we could be talking like 20% increase on overall proxy loading, for NOTHING. The proxies don't need to look at the INFO. 3. Because of latency concerns, the originating UA will need to send an INFO for each detected digit. This will be a very large load on the network. With rfc2833, there are no iimplications on the proxy network. Applications that need DTMF, which use SUBSCRIBE, could even provide digit maps, so they only get NOTIFY when the proper event happens. This reduces traffic load in the application case even. This is a change in position for me, since I have long advocated rfc2833 alone for both usages. Now that, in my mind, I have been able to cleanly separate them, I see little to lose from allowing SUB/NOT for application usage (but not e2e transport), although I still think you will need the full media stream for many apps (any voice reco). Thanks, Jonathan R. Garland Sharratt wrote: > > Duncan: > > >From an architectural point of view, the clean way to carry post-dial > DTMF is in RTP using RFC 2833, because this preserves the separation of > media (audio) from signaling. > > Some reasons for keeping the separation include: > * the network is architected along this separation, with app > servers and softswitches handling signaling and media gateways and media > servers (Media Resource Functions) handling media; and > * when post-dial DTMF is replaced with speech later, it will no > longer fit in the signaling. > > Of course they may be very valid reasons in particular cases to do > otherwise. > > Garland > > > -----Original Message----- > > From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] > > On Behalf Of duncan.mills@vf.vodafone.co.uk > > Sent: Monday, February 11, 2002 01:55 > > To: sipping@ietf.org > > Subject: SIP INFO method for DTMF > > > > IETF SIPPING experts, > > > > 3GPP standards people have been looking at various potential solutions > > for carrying DTMF in the IP Multimedia Subsystem (IMS). > > > > One such solution is the use of the SIP INFO method. I have read the > > expired internet-draft, written by Jiri Kuthan (whom I tried to > > contact first, but the e-mail address was no longer a valid one) and > > there is support for such a solution in 3GPP. I was involved in a > > telephone conference last week with other 3GPP delegates, in which we > > questioned why this draft expired? Whilst we have highlighted some > > potential problems with using the draft that Jiri wrote, we did at > > least see the INFO method as a feasible option (as well as the RTP > > payload option - I've had a fair amount of e-mail discussion with > > Henning on RFC 2833). > > > > 3GPP may run into problems in the Radio Access Network (RAN) area if > > we choose the RTP solution for DTMF. As such, we are considering > > writing an internet-draft on the use of the SIP INFO method for DTMF. > > However, before doing this, I thought it wise to 'test the water'. > > > > The RTP solution is well documented, and we're currently looking into > > those RAN issues. In the meantime, I'm asking for opinions on the > > general concept of using the SIP INFO method for DTMF. > > > > > > Best Regards, > > > > Duncan > > > > > > > > > > Duncan Mills > > UMTS Standards > > Technology Team > > Core Network Development > > Vodafone Ltd (Networks) > > > > Tel: +44 1635 676074 > > Fax: +44 1635 234445 > > mailto:duncan.mills@vf.vodafone.co.uk > > -- Jonathan D. Rosenberg, Ph.D. 72 Eagle Rock Avenue Chief Scientist First Floor dynamicsoft East Hanover, NJ 07936 jdrosen@dynamicsoft.com FAX: (973) 952-5050 http://www.jdrosen.net PH: (973) 952-5000 http://www.dynamicsoft.com _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Mon Feb 11 13:47:55 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id NAA13815 for ; Mon, 11 Feb 2002 13:47:55 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id NAA26037 for sipping-archive@odin.ietf.org; Mon, 11 Feb 2002 13:47:56 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id NAA25623; Mon, 11 Feb 2002 13:35:22 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id NAA25520 for ; Mon, 11 Feb 2002 13:35:17 -0500 (EST) Received: from mail3.dynamicsoft.com ([63.113.44.69]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id NAA13491 for ; Mon, 11 Feb 2002 13:35:15 -0500 (EST) Received: from dynamicsoft.com ([63.113.46.52]) by mail3.dynamicsoft.com (8.12.1/8.12.1) with ESMTP id g1BIZO6Y024933; Mon, 11 Feb 2002 13:35:24 -0500 (EST) Message-ID: <3C680EBD.1A1C6BA8@dynamicsoft.com> Date: Mon, 11 Feb 2002 13:34:37 -0500 From: Jonathan Rosenberg Organization: dynamicsoft X-Mailer: Mozilla 4.75 [en] (Windows NT 5.0; U) X-Accept-Language: en MIME-Version: 1.0 To: duncan.mills@vf.vodafone.co.uk CC: bert.culpepper@intervoice-brite.com, alan.johnston@wcom.com, taylor@nortelnetworks.com, sipping@ietf.org Subject: Re: [Sipping] RE: SIP INFO method for DTMF References: Content-Type: text/plain; charset=us-ascii Content-Transfer-Encoding: 7bit Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit duncan.mills@vf.vodafone.co.uk wrote: > > Hi All, > > Thanks for your rapid responses. The interested parties in this DTMF > discussion in 3GPP have already considered the SUBSCRIBE/NOTIFY solution > and > rejected it as unsuitable. > > Firstly, it appeared to us that for every mobile originated voice call, > the > network would have to SUBSCRIBE to the UA making the call, on the > off-chance > that at some point during the call the user wishes to send DTMF in > NOTIFY > messages. This was seen as unacceptable. Yes, of course. SUB/NOT alone is an awful solution, even worse than INFO alone. The proposal is that its rfc2833 for normal e2e usage, and then allow applications to SUBSCRIBE as needed. I have posted more details on that in a follow up note. THanks, Jonathan R. -- Jonathan D. Rosenberg, Ph.D. 72 Eagle Rock Avenue Chief Scientist First Floor dynamicsoft East Hanover, NJ 07936 jdrosen@dynamicsoft.com FAX: (973) 952-5050 http://www.jdrosen.net PH: (973) 952-5000 http://www.dynamicsoft.com _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Mon Feb 11 14:49:52 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id OAA15268 for ; Mon, 11 Feb 2002 14:49:52 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id OAA29115 for sipping-archive@odin.ietf.org; Mon, 11 Feb 2002 14:49:54 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id OAA28490; Mon, 11 Feb 2002 14:31:58 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id OAA28456 for ; Mon, 11 Feb 2002 14:31:56 -0500 (EST) Received: from mail3.dynamicsoft.com ([63.113.44.69]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id OAA14794 for ; Mon, 11 Feb 2002 14:31:53 -0500 (EST) Received: from dynamicsoft.com ([63.113.46.52]) by mail3.dynamicsoft.com (8.12.1/8.12.1) with ESMTP id g1BJW06Y025339; Mon, 11 Feb 2002 14:32:01 -0500 (EST) Message-ID: <3C681C02.DFB417BE@dynamicsoft.com> Date: Mon, 11 Feb 2002 14:31:14 -0500 From: Jonathan Rosenberg Organization: dynamicsoft X-Mailer: Mozilla 4.75 [en] (Windows NT 5.0; U) X-Accept-Language: en MIME-Version: 1.0 To: Juha Heinanen CC: Aki Niemi , Bob Penfield , sipping@ietf.org Subject: Re: [Sipping] Register question (was [Sip] Anonymization of To & revisiting tags alone for dialog ID) References: <15463.59805.825763.352963@harjus.eng.song.fi> Content-Type: text/plain; charset=us-ascii Content-Transfer-Encoding: 7bit Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit Juha Heinanen wrote: > > Aki Niemi writes: > > > I would argue, that this depends on the configuration of > chicago.com's > > network. The server at that domain receiving the request may accept > and > > process it by itself, or may forward it to a server that does. > > yes, that is what i thought too, i.e., that the proxy serving > chicago.com would know who is the registrar of chicago.com. > > it would be nice though if i would not need to manually configure in the > proxy, who is the registrar of that domain. that would be possible if > there would be a separate srv entry for registars. Where do you draw the line for demux? Should we have separate SRV entries for presence servers? What about specific packages for SUBSCRIBE, so that presence goes to one server, conferencing models to another, and so on? I think all of this is best hidden within the providers network, with a front proxy either processing things locally, or sending them to the server(s) its been configured to talk to. That provides for a much richer set of security. I see little gained from exposing this to clients trying to reach that domain. -Jonathan R. -- Jonathan D. Rosenberg, Ph.D. 72 Eagle Rock Avenue Chief Scientist First Floor dynamicsoft East Hanover, NJ 07936 jdrosen@dynamicsoft.com FAX: (973) 952-5050 http://www.jdrosen.net PH: (973) 952-5000 http://www.dynamicsoft.com _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Mon Feb 11 14:50:07 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id OAA15301 for ; Mon, 11 Feb 2002 14:50:07 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id OAA29170 for sipping-archive@odin.ietf.org; Mon, 11 Feb 2002 14:50:09 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id OAA28181; Mon, 11 Feb 2002 14:30:27 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id OAA28142 for ; Mon, 11 Feb 2002 14:30:25 -0500 (EST) Received: from web11608.mail.yahoo.com (web11608.mail.yahoo.com [216.136.172.60]) by ietf.org (8.9.1a/8.9.1a) with SMTP id OAA14760 for ; Mon, 11 Feb 2002 14:30:22 -0500 (EST) Message-ID: <20020211193023.4907.qmail@web11608.mail.yahoo.com> Received: from [216.51.102.66] by web11608.mail.yahoo.com via HTTP; Mon, 11 Feb 2002 11:30:23 PST Date: Mon, 11 Feb 2002 11:30:23 -0800 (PST) From: Sean Olson Subject: RE: [Sipping] New draft about SIP conferencing avaiable To: Igor Miladinovic , sipping@ietf.org Cc: Johannes Stadler In-Reply-To: MIME-Version: 1.0 Content-Type: text/plain; charset=us-ascii Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org What I was wondering was how someone would construct the CONF request without knowing a priori the correct Call-ID to use? I know how I would build this -- just use the Request-URI and ignore the To/From/Call-ID. But the draft appears to imply something different when speaking of returning a 481 response. Perhaps I misread that section or misunderstood your intentions(?) /sean --- Igor Miladinovic wrote: > > How do you deal with the situation where a > > "conference" is identified by a r-uri and not by > > a pre-existing dialog? (How do you know the > > appropriate Call-ID to place in the CONF request?) > > > > /sean > > The conference server stores data (at least the > sip-address) of each > participant in a conference, which is identified by > a requestURI. This data > contains also the Call-ID of the corresponding > dialog (determined by the > initial INVITE). > > -- > Igor > > > > > > --- Igor Miladinovic > > > wrote: > > > Hi, > > > > > > We have submitted a new draft to the IESG. It > > > describes a SIP extension for > > > discovery of participants identities in a > multiparty > > > conference. The draft > > > is available at: > > > > > > > > > http://search.ietf.org/internet-drafts/draft-miladinovic-sip-multi > > party-ext- > > > 00.txt > > > or > > > http://www.ikn.tuwien.ac.at/ftw-a1/download.html > > > > > > Your comments are welcome. > > > > > > Regards, > > > > > > Igor > > > > > > -- > > > DI Igor Miladinovic > > > Vienna University of Technology > > > Institute of Communication Networks > > > Call Control and Signaling > > > Favoritenstrasse 9/388 > > > A-1040 Vienna, Austria > > > > > > tel. +43-1-58801-38844 > > > fax. +43-1-58801-38898 > > > http://www.ikn.tuwien.ac.at > > > > > > > > > _______________________________________________ > > > Sipping mailing list > > > https://www1.ietf.org/mailman/listinfo/sipping > > > This list is for NEW development of the > application > > > of SIP > > > Use sip-implementors@cs.columbia.edu for > questions > > > on current sip > > > Use sip@ietf.org for new developments of core > SIP > > > > > > ===== > > Sean Olson > > > > __________________________________________________ > > Do You Yahoo!? > > Send FREE Valentine eCards with Yahoo! Greetings! > > http://greetings.yahoo.com > > > > _______________________________________________ > > Sipping mailing list > https://www1.ietf.org/mailman/listinfo/sipping > > This list is for NEW development of the > application of SIP > > Use sip-implementors@cs.columbia.edu for questions > on current sip > > Use sip@ietf.org for new developments of core SIP > > > > > _______________________________________________ > Sipping mailing list > https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application > of SIP > Use sip-implementors@cs.columbia.edu for questions > on current sip > Use sip@ietf.org for new developments of core SIP ===== Sean Olson __________________________________________________ Do You Yahoo!? Send FREE Valentine eCards with Yahoo! Greetings! http://greetings.yahoo.com _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Mon Feb 11 15:07:56 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id PAA15739 for ; Mon, 11 Feb 2002 15:07:56 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id PAA00182 for sipping-archive@odin.ietf.org; Mon, 11 Feb 2002 15:07:56 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id OAA29393; Mon, 11 Feb 2002 14:56:37 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id OAA29364 for ; Mon, 11 Feb 2002 14:56:35 -0500 (EST) Received: from lohi.eng.song.fi (lohi.eng.song.fi [195.10.149.18]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id OAA15512 for ; Mon, 11 Feb 2002 14:56:29 -0500 (EST) Received: from harjus.eng.song.fi (harjus.eng.song.fi [195.10.149.20]) by lohi.eng.song.fi (8.12.1/8.12.1/Debian -5) with ESMTP id g1BJuOio019237; Mon, 11 Feb 2002 21:56:24 +0200 From: Juha Heinanen MIME-Version: 1.0 Content-Type: text/plain; charset=us-ascii Content-Transfer-Encoding: 7bit Message-ID: <15464.8680.349426.641688@harjus.eng.song.fi> Date: Mon, 11 Feb 2002 21:56:24 +0200 To: Jonathan Rosenberg Cc: Aki Niemi , Bob Penfield , sipping@ietf.org Subject: Re: [Sipping] Register question (was [Sip] Anonymization of To & revisiting tags alone for dialog ID) In-Reply-To: <3C681C02.DFB417BE@dynamicsoft.com> References: <15463.59805.825763.352963@harjus.eng.song.fi> <3C681C02.DFB417BE@dynamicsoft.com> X-Mailer: VM 7.01 under Emacs 21.1.1 Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit Jonathan Rosenberg writes: > Where do you draw the line for demux? Should we have separate SRV > entries for presence servers? ok, it is fine with me for the proxy of the domain to know where the different servers of the domain are. my question was in the first place prompted by figure 2 of bis-07 that shows the ua sending the register message directly to the registrar without having no automatic means to find out who the registrar is. -- juha _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Mon Feb 11 15:37:55 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id PAA16547 for ; Mon, 11 Feb 2002 15:37:54 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id PAA01572 for sipping-archive@odin.ietf.org; Mon, 11 Feb 2002 15:37:55 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id PAA00798; Mon, 11 Feb 2002 15:25:38 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id PAA00767 for ; Mon, 11 Feb 2002 15:25:35 -0500 (EST) Received: from mail2.intervoice-brite.com (mail2.intervoice-brite.com [204.62.8.15]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id PAA16237 for ; Mon, 11 Feb 2002 15:25:33 -0500 (EST) Received: from DALNTMS02.ivbi.com (dalntms02.ivbi.com [151.214.90.44]) by mail2.intervoice-brite.com (Build 101 8.9.3/NT-8.9.3) with ESMTP id OAA02812 for ; Mon, 11 Feb 2002 14:28:38 -0600 Received: from 172.16.16.64 (unverified) by DALNTMS02.ivbi.com (Content Technologies SMTPRS 4.2.5) with SMTP id for ; Mon, 11 Feb 2002 14:09:05 -0600 Received: from bculpepperpc ([151.214.151.115]) by 172.16.16.64; Mon, 11 Feb 2002 14:24:53 -0600 From: "Bert Culpepper" To: , , , Subject: RE: [Sipping] RE: SIP INFO method for DTMF Date: Mon, 11 Feb 2002 15:24:47 -0500 Message-ID: <003901c1b33a$27251020$7397d697@bculpepperpc> MIME-Version: 1.0 Content-Type: text/plain; charset="iso-8859-1" Content-Transfer-Encoding: 7bit X-Priority: 3 (Normal) X-MSMail-Priority: Normal X-Mailer: Microsoft Outlook CWS, Build 9.0.2416 (9.0.2911.0) X-MimeOLE: Produced By Microsoft MimeOLE V5.50.4910.0300 Importance: Normal In-Reply-To: Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit duncan.mills@vf.vodafone.co.uk wrote in part: > > Thanks for your rapid responses. The interested parties in this DTMF > discussion in 3GPP have already considered the > SUBSCRIBE/NOTIFY solution and > rejected it as unsuitable. > > Firstly, it appeared to us that for every mobile originated > voice call, the > network would have to SUBSCRIBE to the UA making the call, on > the off-chance > that at some point during the call the user wishes to send > DTMF in NOTIFY > messages. This was seen as unacceptable. > > Any views? > Well, if one was to presume subscription state for DTMF was established in a device using a non-SUBSCRIBE mechanism, sending DTMF in NOTIFYs would be equivalent to using INFO in many respects. However, this suffers from the issues Jonathan pointed out. Also, as Jonathan says actual DTMF needs to be transmitted end-to-end, with latency, etc. issues mitigated. For this there's rfc2833. Anyway...it seems you have something else in mind other than simply transporting DTMF, that you think the DTMF in INFO approach would work for. If so, I'd like to hear what that might be, as I'm interested in a standard solution for my problem set. Now given what's written above concerning 3GPP, is it not possible to create subscriptions (through some means) during the registration process? In that way, SUBSCRIBEs don't have to be sent to each UA at each call origination/termination. Subsequent NOTIFYs could use the Path mechanism to traverse only those proxies that were required. Just some thoughts here... Regards, Bert _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Mon Feb 11 15:40:51 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id PAA16611 for ; Mon, 11 Feb 2002 15:40:50 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id PAA01667 for sipping-archive@odin.ietf.org; Mon, 11 Feb 2002 15:40:51 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id PAA00989; Mon, 11 Feb 2002 15:30:01 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id PAA00948 for ; Mon, 11 Feb 2002 15:29:59 -0500 (EST) Received: from mail3.dynamicsoft.com ([63.113.44.69]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id PAA16370 for ; Mon, 11 Feb 2002 15:29:58 -0500 (EST) Received: from dynamicsoft.com ([63.113.46.52]) by mail3.dynamicsoft.com (8.12.1/8.12.1) with ESMTP id g1BKUB6Y025469; Mon, 11 Feb 2002 15:30:11 -0500 (EST) Message-ID: <3C6829A5.11815E70@dynamicsoft.com> Date: Mon, 11 Feb 2002 15:29:25 -0500 From: Jonathan Rosenberg Organization: dynamicsoft X-Mailer: Mozilla 4.75 [en] (Windows NT 5.0; U) X-Accept-Language: en MIME-Version: 1.0 To: Juha Heinanen CC: Aki Niemi , Bob Penfield , sipping@ietf.org Subject: Re: [Sipping] Register question (was [Sip] Anonymization of To & revisiting tags alone for dialog ID) References: <15464.8680.349426.641688@harjus.eng.song.fi> Content-Type: text/plain; charset=us-ascii Content-Transfer-Encoding: 7bit Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit Juha Heinanen wrote: > > Jonathan Rosenberg writes: > > > Where do you draw the line for demux? Should we have separate SRV > > entries for presence servers? > > ok, it is fine with me for the proxy of the domain to know where the > different servers of the domain are. my question was in the first place > prompted by figure 2 of bis-07 that shows the ua sending the register > message directly to the registrar without having no automatic means to > find out who the registrar is. Ah, now that is a good point. Interestingly, the text never even refers to the figure. The figure is OK if the text would say that this is a logical model. However, that will probably confuse people, so I will enter a bug to make it represent physical reality (REG also goes to proxy first), and also make sure the figure is discussed in the text. THanks, Jonathan R. -- Jonathan D. Rosenberg, Ph.D. 72 Eagle Rock Avenue Chief Scientist First Floor dynamicsoft East Hanover, NJ 07936 jdrosen@dynamicsoft.com FAX: (973) 952-5050 http://www.jdrosen.net PH: (973) 952-5000 http://www.dynamicsoft.com _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Mon Feb 11 21:53:28 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id VAA23304 for ; Mon, 11 Feb 2002 21:53:27 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id VAA16577 for sipping-archive@odin.ietf.org; Mon, 11 Feb 2002 21:53:30 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id VAA16352; Mon, 11 Feb 2002 21:37:27 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id VAA16321 for ; Mon, 11 Feb 2002 21:37:24 -0500 (EST) Received: from nisser.cisco.com (nisser.cisco.com [171.71.176.85]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id VAA23178 for ; Mon, 11 Feb 2002 21:37:21 -0500 (EST) Received: from cisco.com (ssh-sj1.cisco.com [171.68.225.134]) by nisser.cisco.com (8.8.6 (PHNE_14041)/CISCO.SERVER.1.2) with ESMTP id SAA20825; Mon, 11 Feb 2002 18:36:44 -0800 (PST) Message-ID: <3C687FB8.E80F6E6E@cisco.com> Date: Mon, 11 Feb 2002 21:36:40 -0500 From: Flemming Andreasen X-Mailer: Mozilla 4.79 [en]C-CCK-MCD (Windows NT 5.0; U) X-Accept-Language: en MIME-Version: 1.0 To: Dean Willis CC: sipping@ietf.org Subject: Re: Signing RPIDS (Was: Re: [Sipping] calling number blocking at sip/isup gateway)] References: <3C60460A.1C8CCEA5@cisco.com> <000d01c1af71$e3658110$9219a8c0@dfw.dynamicsoft.com> <3C61EA80.2F209E5D@cisco.com> <02f301c1b014$80eb18d0$1df30a0a@dfw.dynamicsoft.com> <3C640AC6.96C6292A@cisco.com> <00aa01c1b20a$9c02f0a0$133fed0c@C1893415A> Content-Type: text/plain; charset=us-ascii Content-Transfer-Encoding: 7bit Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit Dean Willis wrote: > > > > I recognize the practical difficulties of implementing a signature > > > mechanism. I believe it's solvable. > > > > OK - I'd like to hear more about the replay protection solution at least. > > Well, if one is willing to accept roughly synchronized clocks (look, for > example, at the nanosecond-level synchronization required for CDMA), then > one can easily reduce the replay window to human-acceptable time scales, > like 30 seconds or so. One might consider fast perfect replay to be a > security flaw, but I've seen enough networks where it happens spontaneously > to realize it isn't much of a problem. Trouble is, that if you really want to be general, then you cannot assume synchronized clocks (how many PC's do you think it would with). While I agree that controlled operator environments can probably get away with such a requirement, what do you propose for the rest ? > > > > > However, I don't believe it's an > > > adequate replacement for all subscriber authentication questions, > primarily > > > for reasons of backward compatibility and subscriber certificate > management. > > > Same reason that HTTP has challenge-response as well as TLS. You CAN use > > > your browser with a user cert, but you are not required to do so. > > > > Fair enough. Which methods do you propose we should have ? > > Well, clearly digest has a lot of installed base. OK, but how is that going to help you determine the authenticity of a Remote-Party-ID ? It seems more like you would simply use it to determine the identity in the first place. > 3GPP is leaning towards an > EAP-derived system that can reuse the key management of AKA. The AKA > mechanism is interesting, because they use sequences of one-time keys > generated by a smart-card, which would appear to reduce the probability of > replay attack. Assume that we can cobine the AKA key management with ppnonce > approaches, and it could be interesting -- any message that was being > replayed could only be replayed with the AKA session-key lifetime. > I'm not familiar with either of this, but are you proposing that they should be part of the privacy draft ? -- Flemming -- Flemming Andreasen Cisco Systems _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Tue Feb 12 04:52:25 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id EAA06971 for ; Tue, 12 Feb 2002 04:52:25 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id EAA11694 for sipping-archive@odin.ietf.org; Tue, 12 Feb 2002 04:52:27 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id EAA11203; Tue, 12 Feb 2002 04:39:01 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id EAA11176 for ; Tue, 12 Feb 2002 04:38:59 -0500 (EST) Received: from pop.tuwien.ac.at (pop.tuwien.ac.at [128.130.2.60]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id EAA06878 for ; Tue, 12 Feb 2002 04:38:57 -0500 (EST) Received: from pioneer (ftw-nat.ftw.tuwien.ac.at [128.130.90.80]) by pop.tuwien.ac.at (8.12.0/8.12.0) with SMTP id g1C9cqWw024579; Tue, 12 Feb 2002 10:38:53 +0100 (MET) Reply-To: From: "Igor Miladinovic" To: "'Sean Olson'" , Cc: "'Johannes Stadler'" Subject: AW: [Sipping] New draft about SIP conferencing avaiable Date: Tue, 12 Feb 2002 10:38:16 +0100 Message-ID: <003101c1b3a9$000977f0$3b00a8c0@pioneer> MIME-Version: 1.0 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: 7bit X-Priority: 3 (Normal) X-MSMail-Priority: Normal X-Mailer: Microsoft Outlook CWS, Build 9.0.2416 (9.0.2911.0) Importance: Normal X-MimeOLE: Produced By Microsoft MimeOLE V5.50.4522.1200 In-Reply-To: <20020211193023.4907.qmail@web11608.mail.yahoo.com> X-Virus-Scanned: by amavisd-milter (http://amavis.org/) Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit > What I was wondering was how someone would construct > the CONF request without knowing a priori the correct > Call-ID to use? I know how I would build this -- > just use the Request-URI and ignore the > To/From/Call-ID. But the draft appears to imply > something different > when speaking of returning a 481 response. Perhaps > I misread that section or misunderstood your > intentions(?) > > /sean Our intention is that a CONF request is created just like a BYE request. Generally, in a conference each participant can send/receive a BYE request every time. Because the BYE request presumes knowing of the From/To/Call-ID, we suppose that it is also available for a CONF request. Do you agree with this? -- Igor > > --- Igor Miladinovic > wrote: > > > How do you deal with the situation where a > > > "conference" is identified by a r-uri and not by > > > a pre-existing dialog? (How do you know the > > > appropriate Call-ID to place in the CONF request?) > > > > > > /sean > > > > The conference server stores data (at least the > > sip-address) of each > > participant in a conference, which is identified by > > a requestURI. This data > > contains also the Call-ID of the corresponding > > dialog (determined by the > > initial INVITE). > > > > -- > > Igor > > > > > > > > > > --- Igor Miladinovic > > > > > wrote: > > > > Hi, > > > > > > > > We have submitted a new draft to the IESG. It > > > > describes a SIP extension for > > > > discovery of participants identities in a > > multiparty > > > > conference. The draft > > > > is available at: > > > > > > > > > > > > > > http://search.ietf.org/internet-drafts/draft-miladinovic-sip-multi > > > party-ext- > > > > 00.txt > > > > or > > > > http://www.ikn.tuwien.ac.at/ftw-a1/download.html > > > > > > > > Your comments are welcome. > > > > > > > > Regards, > > > > > > > > Igor > > > > > > > > -- > > > > DI Igor Miladinovic > > > > Vienna University of Technology > > > > Institute of Communication Networks > > > > Call Control and Signaling > > > > Favoritenstrasse 9/388 > > > > A-1040 Vienna, Austria > > > > > > > > tel. +43-1-58801-38844 > > > > fax. +43-1-58801-38898 > > > > http://www.ikn.tuwien.ac.at > > > > > > > > > > > > _______________________________________________ > > > > Sipping mailing list > > > > https://www1.ietf.org/mailman/listinfo/sipping > > > > This list is for NEW development of the > > application > > > > of SIP > > > > Use sip-implementors@cs.columbia.edu for > > questions > > > > on current sip > > > > Use sip@ietf.org for new developments of core > > SIP > > > > > > > > > ===== > > > Sean Olson > > > > > > __________________________________________________ > > > Do You Yahoo!? > > > Send FREE Valentine eCards with Yahoo! Greetings! > > > http://greetings.yahoo.com > > > > > > _______________________________________________ > > > Sipping mailing list > > https://www1.ietf.org/mailman/listinfo/sipping > > > This list is for NEW development of the > > application of SIP > > > Use sip-implementors@cs.columbia.edu for questions > > on current sip > > > Use sip@ietf.org for new developments of core SIP > > > > > > > > > _______________________________________________ > > Sipping mailing list > > https://www1.ietf.org/mailman/listinfo/sipping > > This list is for NEW development of the application > > of SIP > > Use sip-implementors@cs.columbia.edu for questions > > on current sip > > Use sip@ietf.org for new developments of core SIP > > > ===== > Sean Olson > > __________________________________________________ > Do You Yahoo!? > Send FREE Valentine eCards with Yahoo! Greetings! > http://greetings.yahoo.com > > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP > Use sip-implementors@cs.columbia.edu for questions on current sip > Use sip@ietf.org for new developments of core SIP > _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Tue Feb 12 09:48:10 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id JAA13625 for ; Tue, 12 Feb 2002 09:48:10 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id JAA24705 for sipping-archive@odin.ietf.org; Tue, 12 Feb 2002 09:48:13 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id JAA24134; Tue, 12 Feb 2002 09:33:41 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id JAA24104 for ; Tue, 12 Feb 2002 09:33:39 -0500 (EST) Received: from mailserver.sylantro.com ([65.200.90.207]) by ietf.org (8.9.1a/8.9.1a) with SMTP id JAA13225 for ; Tue, 12 Feb 2002 09:33:36 -0500 (EST) Received: from 172.16.128.12 by mailserver.sylantro.com with ESMTP ( Tumbleweed MMS SMTP Relay (MMS v4.7)); Tue, 12 Feb 2002 06:28:25 -0800 X-Server-Uuid: 59490da2-986c-11d3-91ca-00104b9c3900 Received: by mailserver.sylantro.com with Internet Mail Service ( 5.5.2653.19) id <1Y4ATXA5>; Tue, 12 Feb 2002 06:28:25 -0800 Message-ID: <79FEAA5FABA7D411BF580001023D1BBD9CFC29@mailserver.sylantro.com> From: "Venkatesh Venkataramanan" To: "'sipping@ietf.org'" Date: Tue, 12 Feb 2002 06:28:23 -0800 MIME-Version: 1.0 X-Mailer: Internet Mail Service (5.5.2653.19) X-WSS-ID: 1077F903188701-01-01 Content-Type: text/plain; charset=us-ascii Content-Transfer-Encoding: 7bit Content-Transfer-Encoding: 7bit Subject: [Sipping] UAS recovery mechanism from timeouts of responses that establish a dialog Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit Hi, I have a question regarding the same... Say, UA1 issues an INVITE to UA2. UA2 responds with a 100 Trying, followed by a 180 Ringing. On answer, it issues a 200 response. Say for some reason (short network outage in the subnet coresponding to UA2), the 200 OK response timesout. Per bis07, UA2 SHOULD issue a BYE to terminate this dialog. There is a possibility that this BYE reaches UA1, which in turn could reject the BYE as UA1 doesn't consider the INVITE transaction to be complete? Hence it returns a 4xx response for the same? How do UA2 and UA1 recover from such a scenario? Regards, Venkatesh _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Tue Feb 12 10:27:54 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id KAA15130 for ; Tue, 12 Feb 2002 10:27:54 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id KAA27196 for sipping-archive@odin.ietf.org; Tue, 12 Feb 2002 10:27:57 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id KAA25960; Tue, 12 Feb 2002 10:06:25 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id KAA25928 for ; Tue, 12 Feb 2002 10:06:20 -0500 (EST) Received: from exchange1.nuera.com ([12.105.228.79]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id KAA14232 for ; Tue, 12 Feb 2002 10:06:17 -0500 (EST) Received: by exchange1.nuera.com with Internet Mail Service (5.5.2653.19) id <1YT0CCBG>; Tue, 12 Feb 2002 07:05:49 -0800 Message-ID: From: "Fairlie-Cuninghame, Robert" To: "'Garland Sharratt'" , duncan.mills@vf.vodafone.co.uk Cc: sipping@ietf.org Subject: RE: [Sipping] RE: SIP INFO method for DTMF Date: Tue, 12 Feb 2002 07:05:48 -0800 MIME-Version: 1.0 X-Mailer: Internet Mail Service (5.5.2653.19) Content-Type: text/plain; charset="iso-8859-1" Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Garland, There are many reasons why RFC2833 is not sufficient or desirable for user application control - which is what the signaled digits and key-events drafts are trying to address. We listed many reasons for this in our key-events draft. Jonathan and Bert have mentioned just a couple recently in other emails. However it should be re-iterated that these SUB/NOT proposals _ARE NOT_ attempting to replace RFC2833 for end-to-end DTMF transport. Indeed there are many scenarios where RFC2833 AND the SUB/NOT methods need to be deployed together. For instance, in the text-book example of the pre-paid phone card, a user must have end-to-end DTMF transport so that they can interact with their destination endpoint (e.g. calling a bank or airline); the pre-paid calling card application is NOT on the media path at the time (perhaps the signaling and/or media path between the caller and caller is using end-to-end encryption) and yet needs to listen for certain trigger events (such as "***"). This is a simple example but shows that it is not always desirable OR EVEN POSSIBLE for an application server to use the media plane via fork DTMF (and that the two schemes are complementary not exclusive). The SUB/NOT mechanism is simply a method whereby an entity (e.g. application server) - not necessarily on the media path - can request notification of user interface activity relating to either a particular SIP dialog or to the device itself. As Jonathan says, the end-to-end DTMF transport and on-demand user appellation control are two entirely different beasts - each with differing requirements for both functionality and performance. It, therefore, does not seem unreasonable to me that these are supplied by different mechanisms. Regards, Robert. > -----Original Message----- > From: Garland Sharratt [mailto:gsharratt@convedia.com] > Sent: Monday, February 11, 2002 5:41 PM > To: duncan.mills@vf.vodafone.co.uk > Cc: sipping@ietf.org > Subject: [Sipping] RE: SIP INFO method for DTMF > > Duncan: > > From an architectural point of view, the clean way to carry > post-dial DTMF is in RTP using RFC 2833, because this > preserves the separation of media (audio) from signaling. > > Some reasons for keeping the separation include: > * the network is architected along this separation, with app > servers and softswitches handling signaling and media > gateways and media servers (Media Resource Functions) > handling media; and > * when post-dial DTMF is replaced with speech later, it will > no longer fit in the signaling. > > Of course they may be very valid reasons in particular cases > to do otherwise. > > Garland > > -----Original Message----- > From: sipping-admin@ietf.org > [mailto:sipping-admin@ietf.org] On Behalf Of > duncan.mills@vf.vodafone.co.uk > Sent: Monday, February 11, 2002 01:55 > To: sipping@ietf.org > Subject: SIP INFO method for DTMF > > IETF SIPPING experts, > > 3GPP standards people have been looking at various potential > solutions for carrying DTMF in the IP Multimedia Subsystem (IMS). > > One such solution is the use of the SIP INFO method. I have > read the expired internet-draft, written by Jiri Kuthan (whom > I tried to contact first, but the e-mail address was no > longer a valid one) and there is support for such a solution > in 3GPP. I was involved in a telephone conference last week > with other 3GPP delegates, in which we questioned why this > draft expired? Whilst we have highlighted some potential > problems with using the draft that Jiri wrote, we did at > least see the INFO method as a feasible option (as well as > the RTP payload option - I've had a fair amount of e-mail > discussion with Henning on RFC 2833). > > 3GPP may run into problems in the Radio Access Network (RAN) > area if we choose the RTP solution for DTMF. As such, we are > considering writing an internet-draft on the use of the SIP > INFO method for DTMF. However, before doing this, I thought > it wise to 'test the water'. > > The RTP solution is well documented, and we're currently > looking into those RAN issues. In the meantime, I'm asking > for opinions on the general concept of using the SIP INFO > method for DTMF. > > > Best Regards, > > Duncan > > > > > Duncan Mills > UMTS Standards > Technology Team > Core Network Development > Vodafone Ltd (Networks) > > Tel: +44 1635 676074 > Fax: +44 1635 234445 > mailto:duncan.mills@vf.vodafone.co.uk > _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Tue Feb 12 11:49:44 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id LAA18065 for ; Tue, 12 Feb 2002 11:49:44 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id LAA01582 for sipping-archive@odin.ietf.org; Tue, 12 Feb 2002 11:49:46 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id LAA00948; Tue, 12 Feb 2002 11:34:21 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id LAA00841 for ; Tue, 12 Feb 2002 11:34:15 -0500 (EST) Received: from web11601.mail.yahoo.com (web11601.mail.yahoo.com [216.136.172.53]) by ietf.org (8.9.1a/8.9.1a) with SMTP id LAA17416 for ; Tue, 12 Feb 2002 11:34:12 -0500 (EST) Message-ID: <20020212163414.15535.qmail@web11601.mail.yahoo.com> Received: from [216.51.103.23] by web11601.mail.yahoo.com via HTTP; Tue, 12 Feb 2002 08:34:14 PST Date: Tue, 12 Feb 2002 08:34:14 -0800 (PST) From: Sean Olson Subject: Re: AW: [Sipping] New draft about SIP conferencing avaiable To: igor.miladinovic@tuwien.ac.at, sipping@ietf.org Cc: "'Johannes Stadler'" In-Reply-To: <003101c1b3a9$000977f0$3b00a8c0@pioneer> MIME-Version: 1.0 Content-Type: text/plain; charset=us-ascii Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org It was my understanding that the CONF request was designed to be used to determine the participants of a conference call before joining that conference call. That is, at the point in time at which you construct the CONF request, you may not know the Call-ID for the conference. But more importantly, there probably will not be a unique Call-ID for the conference as in the case of a conference bridge. It's a small nit. You just need to be careful in how you phrase the section describing when a 481 is sent. On a different topic, it would be nice if there was some generic request that could be sent to obtain information about an ongoing call. The participants of that call is just one such piece of information. This kinda falls in line with the call events package that was proposed to be used with SUBSCRIBE/NOTIFY. By doing an immediate fetch with that event package, you could get the participants without inventing a new method. And of course, the type of information you could retrieve could be expanded over time to include things like who is the moderator of this call, how long has this call been going on, etc. cheers, sean --- Igor Miladinovic wrote: > > > What I was wondering was how someone would > construct > > the CONF request without knowing a priori the > correct > > Call-ID to use? I know how I would build this -- > > just use the Request-URI and ignore the > > To/From/Call-ID. But the draft appears to imply > > something different > > when speaking of returning a 481 response. Perhaps > > I misread that section or misunderstood your > > intentions(?) > > > > /sean > > Our intention is that a CONF request is created just > like a BYE request. > Generally, in a conference each participant can > send/receive a BYE request > every time. Because the BYE request presumes knowing > of the From/To/Call-ID, > we suppose that it is also available for a CONF > request. Do you agree with > this? > > -- > Igor > > > > > --- Igor Miladinovic > > > wrote: > > > > How do you deal with the situation where a > > > > "conference" is identified by a r-uri and not > by > > > > a pre-existing dialog? (How do you know the > > > > appropriate Call-ID to place in the CONF > request?) > > > > > > > > /sean > > > > > > The conference server stores data (at least the > > > sip-address) of each > > > participant in a conference, which is identified > by > > > a requestURI. This data > > > contains also the Call-ID of the corresponding > > > dialog (determined by the > > > initial INVITE). > > > > > > -- > > > Igor > > > > > > > > > > > > > > --- Igor Miladinovic > > > > > > > wrote: > > > > > Hi, > > > > > > > > > > We have submitted a new draft to the IESG. > It > > > > > describes a SIP extension for > > > > > discovery of participants identities in a > > > multiparty > > > > > conference. The draft > > > > > is available at: > > > > > > > > > > > > > > > > > > > > http://search.ietf.org/internet-drafts/draft-miladinovic-sip-multi > > > > party-ext- > > > > > 00.txt > > > > > or > > > > > > http://www.ikn.tuwien.ac.at/ftw-a1/download.html > > > > > > > > > > Your comments are welcome. > > > > > > > > > > Regards, > > > > > > > > > > Igor > > > > > > > > > > -- > > > > > DI Igor Miladinovic > > > > > Vienna University of Technology > > > > > Institute of Communication Networks > > > > > Call Control and Signaling > > > > > Favoritenstrasse 9/388 > > > > > A-1040 Vienna, Austria > > > > > > > > > > tel. +43-1-58801-38844 > > > > > fax. +43-1-58801-38898 > > > > > http://www.ikn.tuwien.ac.at > > > > > > > > > > > > > > > > _______________________________________________ > > > > > Sipping mailing list > > > > > > https://www1.ietf.org/mailman/listinfo/sipping > > > > > This list is for NEW development of the > > > application > > > > > of SIP > > > > > Use sip-implementors@cs.columbia.edu for > > > questions > > > > > on current sip > > > > > Use sip@ietf.org for new developments of > core > > > SIP > > > > > > > > > > > > ===== > > > > Sean Olson > > > > > > > > > __________________________________________________ > > > > Do You Yahoo!? > > > > Send FREE Valentine eCards with Yahoo! > Greetings! > > > > http://greetings.yahoo.com > > > > > > > > > _______________________________________________ > > > > Sipping mailing list > > > https://www1.ietf.org/mailman/listinfo/sipping > > > > This list is for NEW development of the > > > application of SIP > > > > Use sip-implementors@cs.columbia.edu for > questions > > > on current sip > > > > Use sip@ietf.org for new developments of core > SIP > > > > > > > > > > > > > _______________________________________________ > > > Sipping mailing list > > > https://www1.ietf.org/mailman/listinfo/sipping > > > This list is for NEW development of the > application > > > of SIP > > > Use sip-implementors@cs.columbia.edu for > questions > > > on current sip > > > Use sip@ietf.org for new developments of core > SIP > > > > > > ===== > > Sean Olson > > > > __________________________________________________ > > Do You Yahoo!? > > Send FREE Valentine eCards with Yahoo! Greetings! > > http://greetings.yahoo.com > > > > _______________________________________________ > > Sipping mailing list > https://www1.ietf.org/mailman/listinfo/sipping > > This list is for NEW development of the > application of SIP > > Use sip-implementors@cs.columbia.edu for questions > on current sip > > Use sip@ietf.org for new developments of core SIP > > > > > _______________________________________________ > Sipping mailing list > https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application > of SIP > Use sip-implementors@cs.columbia.edu for questions > on current sip > Use sip@ietf.org for new developments of core SIP ===== Sean Olson __________________________________________________ Do You Yahoo!? Send FREE Valentine eCards with Yahoo! Greetings! http://greetings.yahoo.com _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Tue Feb 12 12:02:13 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id MAA18582 for ; Tue, 12 Feb 2002 12:02:13 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id MAA03456 for sipping-archive@odin.ietf.org; Tue, 12 Feb 2002 12:02:15 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id LAA01556; Tue, 12 Feb 2002 11:49:20 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id LAA01525 for ; Tue, 12 Feb 2002 11:49:18 -0500 (EST) Received: from mgw-dax1.ext.nokia.com (mgw-dax1.ext.nokia.com [63.78.179.216]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id LAA18029 for ; Tue, 12 Feb 2002 11:49:03 -0500 (EST) From: Dirk.Trossen@nokia.com Received: from davir01nok.americas.nokia.com (davir01nok.americas.nokia.com [172.18.242.84]) by mgw-dax1.ext.nokia.com (Switch-2.1.0/Switch-2.1.0) with ESMTP id g1CGnIQ19488 for ; Tue, 12 Feb 2002 10:49:18 -0600 (CST) Received: from daebh002.NOE.Nokia.com (unverified) by davir01nok.americas.nokia.com (Content Technologies SMTPRS 4.2.5) with ESMTP id ; Tue, 12 Feb 2002 10:49:04 -0600 Received: from bsebe001.NOE.Nokia.com ([172.19.160.13]) by daebh002.NOE.Nokia.com with Microsoft SMTPSVC(5.0.2195.2966); Tue, 12 Feb 2002 10:49:04 -0600 content-class: urn:content-classes:message MIME-Version: 1.0 Content-Type: text/plain; charset="iso-8859-1" Subject: RE: AW: [Sipping] New draft about SIP conferencing avaiable X-MimeOLE: Produced By Microsoft Exchange V6.0.5762.3 Date: Tue, 12 Feb 2002 11:49:03 -0500 Message-ID: Thread-Topic: AW: [Sipping] New draft about SIP conferencing avaiable Thread-Index: AcGz46iokmPHNPUQSkCJaleWH1zLkAAAJBPA To: , , Cc: X-OriginalArrivalTime: 12 Feb 2002 16:49:04.0723 (UTC) FILETIME=[2E359230:01C1B3E5] Content-Transfer-Encoding: 8bit X-MIME-Autoconverted: from quoted-printable to 8bit by optimus.ietf.org id LAA01526 Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 8bit Hi Sean, > On a different topic, it would be nice if there > was some generic request that could be sent to > obtain information about an ongoing call. The > participants of that call is just one such piece > of information. This kinda falls in line with the > call events package that was proposed to be used > with SUBSCRIBE/NOTIFY. By doing an immediate fetch > with that event package, you could get the > participants without inventing a new method. And > of course, the type of information you could retrieve > could be expanded over time to include things like > who is the moderator of this call, how long has this > call been going on, etc. This goes in the direction of course control of an ongoing call or conference. I do agree that this might be worthwhile to pursue. Looking into SIP events to realize (some of) this functionality is worthwhile to pursue, too. However, as a first step it should be clear what functionality is desired in this context. Obtaining the (public) part of a call or conference information, e.g., containing the current participants and call information, is only one part of the story. Finding out who is the moderator and being notified when he/she changed is another part. It might also be desired to provide generic functionality to realize moderator functionality, aka floor control. Regards, Dirk _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Tue Feb 12 12:27:06 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id MAA19357 for ; Tue, 12 Feb 2002 12:27:06 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id MAA04652 for sipping-archive@odin.ietf.org; Tue, 12 Feb 2002 12:27:08 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id MAA03991; Tue, 12 Feb 2002 12:12:04 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id MAA03959 for ; Tue, 12 Feb 2002 12:12:01 -0500 (EST) Received: from web11605.mail.yahoo.com (web11605.mail.yahoo.com [216.136.172.57]) by ietf.org (8.9.1a/8.9.1a) with SMTP id MAA18892 for ; Tue, 12 Feb 2002 12:11:58 -0500 (EST) Message-ID: <20020212171200.15879.qmail@web11605.mail.yahoo.com> Received: from [216.51.103.156] by web11605.mail.yahoo.com via HTTP; Tue, 12 Feb 2002 09:12:00 PST Date: Tue, 12 Feb 2002 09:12:00 -0800 (PST) From: Sean Olson Subject: RE: AW: [Sipping] New draft about SIP conferencing avaiable To: Dirk.Trossen@nokia.com, igor.miladinovic@tuwien.ac.at, sipping@ietf.org Cc: johannes.stadler@tuwien.ac.at In-Reply-To: MIME-Version: 1.0 Content-Type: text/plain; charset=us-ascii Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Hi Dirk, Sure. I don't think we should tackle all these problems at the start. But perhaps we can come up with a more generic way of approaching the conference participant problem that will lend itself to solving these other problems in the future. /sean --- Dirk.Trossen@nokia.com wrote: > Hi Sean, > > > On a different topic, it would be nice if there > > was some generic request that could be sent to > > obtain information about an ongoing call. The > > participants of that call is just one such piece > > of information. This kinda falls in line with the > > call events package that was proposed to be used > > with SUBSCRIBE/NOTIFY. By doing an immediate fetch > > with that event package, you could get the > > participants without inventing a new method. And > > of course, the type of information you could > retrieve > > could be expanded over time to include things like > > who is the moderator of this call, how long has > this > > call been going on, etc. > > This goes in the direction of course control of an > ongoing > call or conference. I do agree that this might be > worthwhile to pursue. Looking into SIP events to > realize > (some of) this functionality is worthwhile to > pursue, too. > However, as a first step it should be clear what > functionality > is desired in this context. > > Obtaining the (public) part of a call or conference > information, > e.g., containing the current participants and call > information, > is only one part of the story. Finding out who is > the moderator > and being notified when he/she changed is another > part. It might > also be desired to provide generic functionality to > realize moderator > functionality, aka floor control. > > Regards, > > > Dirk ===== Sean Olson __________________________________________________ Do You Yahoo!? Send FREE Valentine eCards with Yahoo! Greetings! http://greetings.yahoo.com _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Tue Feb 12 13:32:25 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id NAA21433 for ; Tue, 12 Feb 2002 13:32:25 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id NAA07679 for sipping-archive@odin.ietf.org; Tue, 12 Feb 2002 13:32:26 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id NAA06827; Tue, 12 Feb 2002 13:19:06 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id NAA06799 for ; Tue, 12 Feb 2002 13:19:05 -0500 (EST) Received: from Mitel.COM ([216.191.234.70]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id NAA20988 for ; Tue, 12 Feb 2002 13:19:03 -0500 (EST) From: Tom_Gray@Mitel.COM Received: from kanmta01.software.mitel.com (kanmta01.kanata.mitel.com [134.199.37.58]) by Mitel.COM (V8/MAIL-RELAY-2.1) with SMTP id NAA04365 for ; Tue, 12 Feb 2002 13:16:48 -0500 (EST) Received: by kanmta01.software.mitel.com(Lotus SMTP MTA v4.6.7 (934.1 12-30-1999)) id 85256B5E.00646642 ; Tue, 12 Feb 2002 13:16:37 -0500 X-Lotus-FromDomain: MITEL To: sipping@ietf.org, Tom_Gray@Mitel.COM Message-ID: <85256B5E.00646562.00@kanmta01.software.mitel.com> Date: Tue, 12 Feb 2002 13:16:40 -0500 Mime-Version: 1.0 Content-type: text/plain; charset=us-ascii Content-Disposition: inline Subject: [Sipping] Invite with Participant Header - Conferece Server Ebhavior Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org From: Tom Gray@MITEL on 02/12/2002 01:16 PM In the SIP Extensions for Conferencing draft (http://search.ietf.org/internet-drafts/draft-miladinovic-sip-multiparty-ext-00.txt), section 4.4 contains the text: 'When a dial-in conference server receives an INVITE request from a user that wants to join in an existing conference and when the conference server allows this user to join, it SHOULD check if there is a participant header field in this request. If yes, the value of this header field is compared with the participant list of the conference server. If there is no difference, the 200 (OK) response is sent without the participant header field. If there is a difference or if the participant header field is missing in the INVITE request, the conference server SHOULD insert the participant header field with an actual list of participants in the 200 (OK) response.' What is the point of the comparison of the participant list contained in the INVITE with the list held at the conference server? This will add to the overhead at the conference server for no reason that is obvious to me. This is especially true given that the specification is listed with strength SHOULD and the UAC cannot rely on this behavior being carried out. It would seem that if the UAC wanted a comparison of this sort for some reason, it should just send the INVITE without a participant header, receive the 200OK with the current list of participants and then do the comparison itself - although I do not see why it would want to do this. Also, is there one too many SHOULDs in the passage, Doing the comparison without also complying with the specification on the provision of participant lists does not seem to make sense. tg _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Tue Feb 12 15:34:40 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id PAA25580 for ; Tue, 12 Feb 2002 15:34:40 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id PAA12806 for sipping-archive@odin.ietf.org; Tue, 12 Feb 2002 15:34:42 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id PAA11898; Tue, 12 Feb 2002 15:17:02 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id PAA11859 for ; Tue, 12 Feb 2002 15:16:59 -0500 (EST) Received: from cerbrus.convedia.com (cerbrus.convedia.com [216.129.93.50]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id PAA24731 for ; Tue, 12 Feb 2002 15:16:56 -0500 (EST) Received: from ladon.convedia.com (ladon [192.168.208.15]) by cerbrus.convedia.com (8.9.3+Sun/8.9.3) with ESMTP id MAA19752; Tue, 12 Feb 2002 12:15:09 -0800 (PST) Received: from laptop52 (laptop52.convedia.com [192.168.208.151]) by ladon.convedia.com (8.9.3+Sun/8.9.3) with SMTP id MAA01706; Tue, 12 Feb 2002 12:16:25 -0800 (PST) From: "Garland Sharratt" To: "Fairlie-Cuninghame, Robert" Cc: Subject: RE: [Sipping] RE: SIP INFO method for DTMF Date: Tue, 12 Feb 2002 12:17:03 -0800 Message-ID: MIME-Version: 1.0 Content-Type: text/plain; charset="iso-8859-1" Content-Transfer-Encoding: 7bit X-Priority: 3 (Normal) X-MSMail-Priority: Normal X-Mailer: Microsoft Outlook IMO, Build 9.0.2416 (9.0.2911.0) Importance: Normal In-Reply-To: X-MimeOLE: Produced By Microsoft MimeOLE V6.00.2600.0000 Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit Good points. Two thoughts: 1. My statement was based on the these assumptions: (a) there is a media resource function (MRF) component in the media path, doing the DTMF detection either from audio or from 2833, and under the control of the application running in a feature server or app server; and (b) events get reported from the MRF to the application using either MGCP events or SIP NOTIFY. (Physically the MRF is either a media server or built into a media gateway. If in a media gateway, digits need to be conveyed somehow, in the signaling path, from the media gateway controller to the application.) (In the calling card example, during the talking state, the MRF is only listening for "***" or a long "#". In the initial, IVR stage the MRF is playing announcements and collecting digits.) 2. I definitely support the use of SIP SUBSCRIBE/NOTIFY for enabling digit detection and reporting digits from an MRF (or app server component) to an application. I have been trying to drive, through the ISC, the use of SIP SUBSCRIBE/NOTIFY as a SIP framework for implementing the functionality of MGCP signals and events, especially for use with MRF to application interaction. Garland -----Original Message----- From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org]On Behalf Of Fairlie-Cuninghame, Robert Sent: Tuesday, February 12, 2002 07:06 To: 'Garland Sharratt'; duncan.mills@vf.vodafone.co.uk Cc: sipping@ietf.org Subject: RE: [Sipping] RE: SIP INFO method for DTMF Garland, There are many reasons why RFC2833 is not sufficient or desirable for user application control - which is what the signaled digits and key-events drafts are trying to address. We listed many reasons for this in our key-events draft. Jonathan and Bert have mentioned just a couple recently in other emails. However it should be re-iterated that these SUB/NOT proposals _ARE NOT_ attempting to replace RFC2833 for end-to-end DTMF transport. Indeed there are many scenarios where RFC2833 AND the SUB/NOT methods need to be deployed together. For instance, in the text-book example of the pre-paid phone card, a user must have end-to-end DTMF transport so that they can interact with their destination endpoint (e.g. calling a bank or airline); the pre-paid calling card application is NOT on the media path at the time (perhaps the signaling and/or media path between the caller and caller is using end-to-end encryption) and yet needs to listen for certain trigger events (such as "***"). This is a simple example but shows that it is not always desirable OR EVEN POSSIBLE for an application server to use the media plane via fork DTMF (and that the two schemes are complementary not exclusive). The SUB/NOT mechanism is simply a method whereby an entity (e.g. application server) - not necessarily on the media path - can request notification of user interface activity relating to either a particular SIP dialog or to the device itself. As Jonathan says, the end-to-end DTMF transport and on-demand user appellation control are two entirely different beasts - each with differing requirements for both functionality and performance. It, therefore, does not seem unreasonable to me that these are supplied by different mechanisms. Regards, Robert. > -----Original Message----- > From: Garland Sharratt [mailto:gsharratt@convedia.com] > Sent: Monday, February 11, 2002 5:41 PM > To: duncan.mills@vf.vodafone.co.uk > Cc: sipping@ietf.org > Subject: [Sipping] RE: SIP INFO method for DTMF > > Duncan: > > From an architectural point of view, the clean way to carry > post-dial DTMF is in RTP using RFC 2833, because this > preserves the separation of media (audio) from signaling. > > Some reasons for keeping the separation include: > * the network is architected along this separation, with app > servers and softswitches handling signaling and media > gateways and media servers (Media Resource Functions) > handling media; and > * when post-dial DTMF is replaced with speech later, it will > no longer fit in the signaling. > > Of course they may be very valid reasons in particular cases > to do otherwise. > > Garland > > -----Original Message----- > From: sipping-admin@ietf.org > [mailto:sipping-admin@ietf.org] On Behalf Of > duncan.mills@vf.vodafone.co.uk > Sent: Monday, February 11, 2002 01:55 > To: sipping@ietf.org > Subject: SIP INFO method for DTMF > > IETF SIPPING experts, > > 3GPP standards people have been looking at various potential > solutions for carrying DTMF in the IP Multimedia Subsystem (IMS). > > One such solution is the use of the SIP INFO method. I have > read the expired internet-draft, written by Jiri Kuthan (whom > I tried to contact first, but the e-mail address was no > longer a valid one) and there is support for such a solution > in 3GPP. I was involved in a telephone conference last week > with other 3GPP delegates, in which we questioned why this > draft expired? Whilst we have highlighted some potential > problems with using the draft that Jiri wrote, we did at > least see the INFO method as a feasible option (as well as > the RTP payload option - I've had a fair amount of e-mail > discussion with Henning on RFC 2833). > > 3GPP may run into problems in the Radio Access Network (RAN) > area if we choose the RTP solution for DTMF. As such, we are > considering writing an internet-draft on the use of the SIP > INFO method for DTMF. However, before doing this, I thought > it wise to 'test the water'. > > The RTP solution is well documented, and we're currently > looking into those RAN issues. In the meantime, I'm asking > for opinions on the general concept of using the SIP INFO > method for DTMF. > > > Best Regards, > > Duncan > > > > > Duncan Mills > UMTS Standards > Technology Team > Core Network Development > Vodafone Ltd (Networks) > > Tel: +44 1635 676074 > Fax: +44 1635 234445 > mailto:duncan.mills@vf.vodafone.co.uk > _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Wed Feb 13 03:37:30 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id DAA23375 for ; Wed, 13 Feb 2002 03:37:30 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id DAA21167 for sipping-archive@odin.ietf.org; Wed, 13 Feb 2002 03:37:32 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id DAA20334; Wed, 13 Feb 2002 03:21:55 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id DAA20304 for ; Wed, 13 Feb 2002 03:21:53 -0500 (EST) Received: from bdsl.66.12.12.130.gte.net (bdsl.66.12.12.130.gte.net [66.12.12.130]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id DAA22971 for ; Wed, 13 Feb 2002 03:21:50 -0500 (EST) Received: from C1893415A (bdsl.greycouncil.com [127.0.0.1]) by bdsl.66.12.12.130.gte.net (8.11.6/8.11.6) with SMTP id g1D8LAw22338; Wed, 13 Feb 2002 02:21:10 -0600 Message-ID: <00a501c1b467$5d40e610$133fed0c@C1893415A> From: "Dean Willis" To: "Flemming Andreasen" Cc: "Sipping \(E-mail\)" References: <3C60460A.1C8CCEA5@cisco.com> <000d01c1af71$e3658110$9219a8c0@dfw.dynamicsoft.com> <3C61EA80.2F209E5D@cisco.com> <02f301c1b014$80eb18d0$1df30a0a@dfw.dynamicsoft.com> <3C640AC6.96C6292A@cisco.com> <00aa01c1b20a$9c02f0a0$133fed0c@C1893415A> <3C687FB8.E80F6E6E@cisco.com> Subject: Re: Signing RPIDS (Was: Re: [Sipping] calling number blocking at sip/isup gateway)] Date: Wed, 13 Feb 2002 02:20:57 -0600 MIME-Version: 1.0 Content-Type: text/plain; charset="iso-8859-1" Content-Transfer-Encoding: 7bit X-Priority: 3 X-MSMail-Priority: Normal X-Mailer: Microsoft Outlook Express 6.00.2600.0000 X-MIMEOLE: Produced By Microsoft MimeOLE V6.00.2600.0000 Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit Well, if 80% of the SIP clients are CDMA or W-CDMA (which by definitition have synched clocks) that solves a large part of the problem. Of the otehr 20%, the 80% that know they have bad clocks could just ignore the signing on RPIDS. Look, we're talking primarily about a PROXY MODERATED signature. If you're willing to presume IPSEC and magic key management and all sorts of other stuff on proxies, chances are the clock is synched up too . . . but at least this approach lets MY operator proxy evaluate the signing by YOUR operator's proxy. I don't understand the emphasis on the crippled RPID mechanism of the privacy draft. Is there some sort of patent somebody is trying to submarine into the works? -- Dean ----- Original Message ----- From: "Flemming Andreasen" To: "Dean Willis" Cc: Sent: Monday, February 11, 2002 8:36 PM Subject: Re: Signing RPIDS (Was: Re: [Sipping] calling number blocking at sip/isup gateway)] > > > Dean Willis wrote: > > > > > > > I recognize the practical difficulties of implementing a signature > > > > mechanism. I believe it's solvable. > > > > > > OK - I'd like to hear more about the replay protection solution at least. > > > > Well, if one is willing to accept roughly synchronized clocks (look, for > > example, at the nanosecond-level synchronization required for CDMA), then > > one can easily reduce the replay window to human-acceptable time scales, > > like 30 seconds or so. One might consider fast perfect replay to be a > > security flaw, but I've seen enough networks where it happens spontaneously > > to realize it isn't much of a problem. > > Trouble is, that if you really want to be general, then you cannot assume > synchronized clocks (how many PC's do you think it would with). While I agree > that controlled operator environments can probably get away with such a > requirement, what do you propose for the rest ? > > > > > > > > > > However, I don't believe it's an > > > > adequate replacement for all subscriber authentication questions, > > primarily > > > > for reasons of backward compatibility and subscriber certificate > > management. > > > > Same reason that HTTP has challenge-response as well as TLS. You CAN use > > > > your browser with a user cert, but you are not required to do so. > > > > > > Fair enough. Which methods do you propose we should have ? > > > > Well, clearly digest has a lot of installed base. > > OK, but how is that going to help you determine the authenticity of a > Remote-Party-ID ? It seems more like you would simply use it to determine the > identity in the first place. > > > > 3GPP is leaning towards an > > EAP-derived system that can reuse the key management of AKA. The AKA > > mechanism is interesting, because they use sequences of one-time keys > > generated by a smart-card, which would appear to reduce the probability of > > replay attack. Assume that we can cobine the AKA key management with ppnonce > > approaches, and it could be interesting -- any message that was being > > replayed could only be replayed with the AKA session-key lifetime. > > > > I'm not familiar with either of this, but are you proposing that they should be > part of the privacy draft ? > > -- Flemming > > -- > Flemming Andreasen > Cisco Systems > > > _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Wed Feb 13 05:50:54 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id FAA26538 for ; Wed, 13 Feb 2002 05:50:54 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id FAA26724 for sipping-archive@odin.ietf.org; Wed, 13 Feb 2002 05:50:57 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id FAA26305; Wed, 13 Feb 2002 05:36:02 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id FAA26261 for ; Wed, 13 Feb 2002 05:35:59 -0500 (EST) Received: from pop.tuwien.ac.at (pop.tuwien.ac.at [128.130.2.60]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id FAA26105 for ; Wed, 13 Feb 2002 05:35:55 -0500 (EST) Received: from pioneer (ftw-nat.ftw.tuwien.ac.at [128.130.90.80]) by pop.tuwien.ac.at (8.12.0/8.12.0) with SMTP id g1DAZqWw004371; Wed, 13 Feb 2002 11:35:52 +0100 (MET) Reply-To: From: "Igor Miladinovic" To: "'Sean Olson'" , Cc: "'Johannes Stadler'" Subject: AW: AW: [Sipping] New draft about SIP conferencing avaiable Date: Wed, 13 Feb 2002 11:35:16 +0100 Message-ID: <000601c1b47a$20c5d390$3b00a8c0@pioneer> MIME-Version: 1.0 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: 7bit X-Priority: 3 (Normal) X-MSMail-Priority: Normal X-Mailer: Microsoft Outlook CWS, Build 9.0.2416 (9.0.2911.0) In-Reply-To: <20020212163414.15535.qmail@web11601.mail.yahoo.com> X-MimeOLE: Produced By Microsoft MimeOLE V5.50.4522.1200 Importance: Normal X-Virus-Scanned: by amavisd-milter (http://amavis.org/) Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit Hi, > It was my understanding that the CONF request > was designed to be used to determine the participants > of a conference call before joining that conference > call. That is, at the point in time at which you > construct the CONF request, you may not know the > Call-ID for the conference. But more importantly, > there probably will not be a unique Call-ID for > the conference as in the case of a conference > bridge. Actually, the purpose of the CONF request is to update participants lists in a established conference. To determine the conference participants before joining, we use the "participant" header in the INVITE request. Regards, Igor _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Wed Feb 13 10:28:43 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id KAA07517 for ; Wed, 13 Feb 2002 10:28:43 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id KAA09706 for sipping-archive@odin.ietf.org; Wed, 13 Feb 2002 10:28:45 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id JAA08242; Wed, 13 Feb 2002 09:59:22 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id JAA08211 for ; Wed, 13 Feb 2002 09:59:20 -0500 (EST) Received: from nisser.cisco.com (nisser.cisco.com [171.71.176.85]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id JAA05858 for ; Wed, 13 Feb 2002 09:59:17 -0500 (EST) Received: from cisco.com (ssh-sj1.cisco.com [171.68.225.134]) by nisser.cisco.com (8.8.6 (PHNE_14041)/CISCO.SERVER.1.2) with ESMTP id GAA17241; Wed, 13 Feb 2002 06:58:42 -0800 (PST) Message-ID: <3C6A7F1A.84202868@cisco.com> Date: Wed, 13 Feb 2002 09:58:34 -0500 From: Flemming Andreasen X-Mailer: Mozilla 4.79 [en]C-CCK-MCD (Windows NT 5.0; U) X-Accept-Language: en MIME-Version: 1.0 To: Dean Willis CC: "Sipping (E-mail)" Subject: Re: Signing RPIDS (Was: Re: [Sipping] calling number blocking at sip/isup gateway)] References: <3C60460A.1C8CCEA5@cisco.com> <000d01c1af71$e3658110$9219a8c0@dfw.dynamicsoft.com> <3C61EA80.2F209E5D@cisco.com> <02f301c1b014$80eb18d0$1df30a0a@dfw.dynamicsoft.com> <3C640AC6.96C6292A@cisco.com> <00aa01c1b20a$9c02f0a0$133fed0c@C1893415A> <3C687FB8.E80F6E6E@cisco.com> <00a501c1b467$5d40e610$133fed0c@C1893415A> Content-Type: text/plain; charset=us-ascii Content-Transfer-Encoding: 7bit Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit Dean Willis wrote: > Well, if 80% of the SIP clients are CDMA or W-CDMA (which by definitition > have synched clocks) that solves a large part of the problem. > > Of the otehr 20%, the 80% that know they have bad clocks could just ignore > the signing on RPIDS. Is this the SIP/SIPPING WG chair talking ? Are you seriously suggesting that a secure solution should only be applicable to some subset of the SIP clients ? I thought the whole idea with this discussion was to come up with a general solution. > > > Look, we're talking primarily about a PROXY MODERATED signature. If you're > willing to presume IPSEC and magic key management and all sorts of other > stuff on proxies, chances are the clock is synched up too . . . but at least > this approach lets MY operator proxy evaluate the signing by YOUR operator's > proxy. Well, this is different from what we have been talking about so far. We are concerned about user agents determining whether a given Remote-Party-ID is authentic or not, not proxies. > > > I don't understand the emphasis on the crippled RPID mechanism of the > privacy draft. Is there some sort of patent somebody is trying to submarine > into the works? Cheap and uncalled for Dean. The premise of the privacy-03 draft is a simple transitive trust model. Yourself and others desire a more general model and hence solution, which I have no problem with, *as long* as it then is a general model that is also usable in practice. You have suggested that we need to sign Remote-Party-Id's in this more general model, and a number of questions have then come up, several of which have yet to find a satisfactory answer. I'll reiterate the questions below: 1. Is signing of Remote-Party-ID a specific case of a more general problem of authenticating headers, and hence should be solved in a generic way (outside of the privacy draft) ? 2. Is it reasonable to require synchronized clocks between all SIP entities in order for signing of Remote-Party-Ids to be secure ? 3. Are people concerned about the computational overhead by always having to sign a Remote-Party-Id in a message (issue when record-routing for example) ? 4. If Remote-Party-Ids are signed (one way or the other), should they be protected against: a) cut-and-paste attacks ? b) replay attacks ? If not, i.e., it's not really secure after all, what are we gaining by having a signature scheme ? 5. If RPIDs can be signed by more than a small number of well-known parties (which I believe to be the case), how do you verify the signature ? a) In particular, how do you obtain the signer's certificate ? b) How do you handle revoked certificates ? I would appreciate input from more people on these - especially the security folks and of course 3GPP as well. Thanks Flemming -- Flemming Andreasen Cisco Systems _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Wed Feb 13 10:57:51 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id KAA08624 for ; Wed, 13 Feb 2002 10:57:51 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id KAA12079 for sipping-archive@odin.ietf.org; Wed, 13 Feb 2002 10:57:53 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id KAA10766; Wed, 13 Feb 2002 10:40:30 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id KAA10735 for ; Wed, 13 Feb 2002 10:40:28 -0500 (EST) Received: from mail.pingtel.com (IDENT:root@mail.pingtel.com [216.91.1.131]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id KAA07858 for ; Wed, 13 Feb 2002 10:40:21 -0500 (EST) Received: from pingtel.com (cdhcp163.pingtel.com [10.1.1.163]) by mail.pingtel.com (8.11.0/8.11.0) with ESMTP id g1DFguA08998; Wed, 13 Feb 2002 10:42:56 -0500 Message-ID: <3C6A88BB.4DB7ECD9@pingtel.com> Date: Wed, 13 Feb 2002 10:39:39 -0500 From: "Daniel G. Petrie" Organization: Pingtel Corp. http://www.pingtel.com X-Mailer: Mozilla 4.73 [en]C-{C-UDP; EBM-SONY1} (Windows NT 5.0; U) X-Accept-Language: en MIME-Version: 1.0 To: igor.miladinovic@tuwien.ac.at CC: "'Sean Olson'" , sipping@ietf.org, "'Johannes Stadler'" , Jonathan Rosenberg , Rohan Mahy Subject: Re: AW: AW: [Sipping] New draft about SIP conferencing avaiable References: <000601c1b47a$20c5d390$3b00a8c0@pioneer> Content-Type: text/plain; charset=us-ascii Content-Transfer-Encoding: 7bit Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit There is a generic way (i.e. independent of conference) to discover parties in a call of a conference already defined in: http://www.ietf.org/internet-drafts/draft-rosenberg-sip-call-package-00.txt What functionality is missing from the call-package draft? A draft also exists for providing operations to join the conference with a specifed media topology: http://search.ietf.org/internet-drafts/draft-mahy-sipping-join-and-fork-00.txt Igor Miladinovic wrote: > Hi, > > > It was my understanding that the CONF request > > was designed to be used to determine the participants > > of a conference call before joining that conference > > call. That is, at the point in time at which you > > construct the CONF request, you may not know the > > Call-ID for the conference. But more importantly, > > there probably will not be a unique Call-ID for > > the conference as in the case of a conference > > bridge. > > Actually, the purpose of the CONF request is to update participants lists in > a established conference. To determine the conference participants before > joining, we use the "participant" header in the INVITE request. > > Regards, > > Igor > > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP > Use sip-implementors@cs.columbia.edu for questions on current sip > Use sip@ietf.org for new developments of core SIP _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Wed Feb 13 11:09:19 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id LAA09229 for ; Wed, 13 Feb 2002 11:09:19 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id LAA13255 for sipping-archive@odin.ietf.org; Wed, 13 Feb 2002 11:09:22 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id KAA11806; Wed, 13 Feb 2002 10:53:56 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id KAA11774 for ; Wed, 13 Feb 2002 10:53:53 -0500 (EST) Received: from mail3.dynamicsoft.com ([63.113.44.69]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id KAA08507 for ; Wed, 13 Feb 2002 10:53:50 -0500 (EST) Received: from dynamicsoft.com ([63.113.46.42]) by mail3.dynamicsoft.com (8.12.1/8.12.1) with ESMTP id g1DFrl6Y029357; Wed, 13 Feb 2002 10:53:49 -0500 (EST) Message-ID: <3C6A8BDB.F5968548@dynamicsoft.com> Date: Wed, 13 Feb 2002 10:52:59 -0500 From: Jonathan Rosenberg Organization: dynamicsoft X-Mailer: Mozilla 4.75 [en] (Windows NT 5.0; U) X-Accept-Language: en MIME-Version: 1.0 To: "Daniel G. Petrie" CC: igor.miladinovic@tuwien.ac.at, "'Sean Olson'" , sipping@ietf.org, "'Johannes Stadler'" , Rohan Mahy Subject: Re: AW: AW: [Sipping] New draft about SIP conferencing avaiable References: <3C6A88BB.4DB7ECD9@pingtel.com> Content-Type: text/plain; charset=us-ascii Content-Transfer-Encoding: 7bit Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit I was about to chime in with the same note.... it seems clear to me that there are several independent drafts working on roughly the same problems. We also have some drafts on IM group conferencing: http://search.ietf.org/internet-drafts/draft-koskelainen-sip-group-00.txt (although this focuses on somewhat different problems) and also: http://search.ietf.org/internet-drafts/draft-khartabil-sip-conferencing-00.txt which does look at participant identification. I think a better approach is to understand the specific requirements, especially as they apply to voice and IM session conferencing, and also the IM page model, which I am beginning to think has subtlely different requirements. -Jonathan R. "Daniel G. Petrie" wrote: > > There is a generic way (i.e. independent of conference) to discover > parties in a call of a conference already defined in: > http://www.ietf.org/internet-drafts/draft-rosenberg-sip-call-package-00. > txt > What functionality is missing from the call-package draft? > > A draft also exists for providing operations to join the conference > with a specifed media topology: > http://search.ietf.org/internet-drafts/draft-mahy-sipping-join-and-fork- > 00.txt > > Igor Miladinovic wrote: > > > Hi, > > > > > It was my understanding that the CONF request > > > was designed to be used to determine the participants > > > of a conference call before joining that conference > > > call. That is, at the point in time at which you > > > construct the CONF request, you may not know the > > > Call-ID for the conference. But more importantly, > > > there probably will not be a unique Call-ID for > > > the conference as in the case of a conference > > > bridge. > > > > Actually, the purpose of the CONF request is to update participants > lists in > > a established conference. To determine the conference participants > before > > joining, we use the "participant" header in the INVITE request. > > > > Regards, > > > > Igor > > > > _______________________________________________ > > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > > This list is for NEW development of the application of SIP > > Use sip-implementors@cs.columbia.edu for questions on current sip > > Use sip@ietf.org for new developments of core SIP -- Jonathan D. Rosenberg, Ph.D. 72 Eagle Rock Avenue Chief Scientist First Floor dynamicsoft East Hanover, NJ 07936 jdrosen@dynamicsoft.com FAX: (973) 952-5050 http://www.jdrosen.net PH: (973) 952-5000 http://www.dynamicsoft.com _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Wed Feb 13 12:29:48 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id MAA14714 for ; Wed, 13 Feb 2002 12:29:46 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id MAA18676 for sipping-archive@odin.ietf.org; Wed, 13 Feb 2002 12:29:50 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id MAA17168; Wed, 13 Feb 2002 12:06:46 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id MAA17096 for ; Wed, 13 Feb 2002 12:06:41 -0500 (EST) Received: from pop.tuwien.ac.at (pop.tuwien.ac.at [128.130.2.60]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id MAA12979 for ; Wed, 13 Feb 2002 12:06:36 -0500 (EST) Received: from academia (chello212186107032.14.tuwien.teleweb.at [212.186.107.32]) by pop.tuwien.ac.at (8.12.0/8.12.0) with SMTP id g1DH6UWw022388; Wed, 13 Feb 2002 18:06:31 +0100 (MET) Message-ID: <002301c1b4b0$eb224520$206bbad4@academia> From: "Igor Miladinovic" To: "Sean Olson" , , Cc: References: <20020212171200.15879.qmail@web11605.mail.yahoo.com> Subject: Re: AW: [Sipping] New draft about SIP conferencing avaiable Date: Wed, 13 Feb 2002 18:07:27 +0100 MIME-Version: 1.0 Content-Type: text/plain; charset="iso-8859-1" Content-Transfer-Encoding: 7bit X-Priority: 3 X-MSMail-Priority: Normal X-Mailer: Microsoft Outlook Express 6.00.2600.0000 X-MimeOLE: Produced By Microsoft MimeOLE V6.00.2600.0000 X-Virus-Scanned: by amavisd-milter (http://amavis.org/) Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit Hi, > Hi Dirk, > > Sure. I don't think we should tackle all these > problems at the start. But perhaps we can come > up with a more generic way of approaching the > conference participant problem that will lend itself > to solving these other problems in the future. > > /sean I agree that we need a generic solution for the conference participant problem. Additional information (conference moderator, conference type etc.) could be placed in the body of the CONF request or as additional headers/parameters (e.g. conference moderator can be specified in the status parameter of the participant header). We tried to make the first step in the direction of a generic solution for discovering of participant identities. Of course, there are also some other conferencing problems, but we think that most of them could be solved by extending/adopting our approach. Regards, Igor > --- Dirk.Trossen@nokia.com wrote: > > Hi Sean, > > > > > On a different topic, it would be nice if there > > > was some generic request that could be sent to > > > obtain information about an ongoing call. The > > > participants of that call is just one such piece > > > of information. This kinda falls in line with the > > > call events package that was proposed to be used > > > with SUBSCRIBE/NOTIFY. By doing an immediate fetch > > > with that event package, you could get the > > > participants without inventing a new method. And > > > of course, the type of information you could > > retrieve > > > could be expanded over time to include things like > > > who is the moderator of this call, how long has > > this > > > call been going on, etc. > > > > This goes in the direction of course control of an > > ongoing > > call or conference. I do agree that this might be > > worthwhile to pursue. Looking into SIP events to > > realize > > (some of) this functionality is worthwhile to > > pursue, too. > > However, as a first step it should be clear what > > functionality > > is desired in this context. > > > > Obtaining the (public) part of a call or conference > > information, > > e.g., containing the current participants and call > > information, > > is only one part of the story. Finding out who is > > the moderator > > and being notified when he/she changed is another > > part. It might > > also be desired to provide generic functionality to > > realize moderator > > functionality, aka floor control. > > > > Regards, > > > > > > Dirk > > > ===== > Sean Olson > > __________________________________________________ > Do You Yahoo!? > Send FREE Valentine eCards with Yahoo! Greetings! > http://greetings.yahoo.com > > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP > Use sip-implementors@cs.columbia.edu for questions on current sip > Use sip@ietf.org for new developments of core SIP > _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Wed Feb 13 12:32:12 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id MAA14800 for ; Wed, 13 Feb 2002 12:32:12 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id MAA19182 for sipping-archive@odin.ietf.org; Wed, 13 Feb 2002 12:32:15 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id MAA17882; Wed, 13 Feb 2002 12:15:09 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id MAA17851 for ; Wed, 13 Feb 2002 12:15:06 -0500 (EST) Received: from pop.tuwien.ac.at (pop.tuwien.ac.at [128.130.2.60]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id MAA14269 for ; Wed, 13 Feb 2002 12:15:02 -0500 (EST) Received: from academia (chello212186107032.14.tuwien.teleweb.at [212.186.107.32]) by pop.tuwien.ac.at (8.12.0/8.12.0) with SMTP id g1DHEuWw023206; Wed, 13 Feb 2002 18:14:56 +0100 (MET) Message-ID: <002e01c1b4b2$17e27520$206bbad4@academia> From: "Igor Miladinovic" To: , References: <85256B5E.00646562.00@kanmta01.software.mitel.com> Subject: Re: [Sipping] Invite with Participant Header - Conferece Server Ebhavior Date: Wed, 13 Feb 2002 18:15:53 +0100 MIME-Version: 1.0 Content-Type: text/plain; charset="iso-8859-1" Content-Transfer-Encoding: 7bit X-Priority: 3 X-MSMail-Priority: Normal X-Mailer: Microsoft Outlook Express 6.00.2600.0000 X-MimeOLE: Produced By Microsoft MimeOLE V6.00.2600.0000 X-Virus-Scanned: by amavisd-milter (http://amavis.org/) Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit Hi, > From: Tom Gray@MITEL on 02/12/2002 01:16 PM > > In the SIP Extensions for Conferencing draft > (http://search.ietf.org/internet-drafts/draft-miladinovic-sip-multiparty-ext -00.txt), > > section 4.4 contains the text: > > 'When a dial-in conference server receives an INVITE request from a user that > wants to join in an existing conference and when the conference server allows > this user to join, it SHOULD check if there is a participant header field in > this request. If yes, the value of this header field is compared with the > participant list of the conference server. If there is no difference, the 200 > (OK) response is sent without the participant header field. If there is a > difference or if the participant header field is missing in the INVITE request, > the conference server SHOULD insert the participant header field with an actual > list of participants in the 200 (OK) response.' > > What is the point of the comparison of the participant list contained in the > INVITE with the list held at the conference server? This will add to the > overhead at the conference server for no reason that is obvious to me. This is > especially true given that the specification is listed with strength SHOULD and > the UAC cannot rely on this behavior being carried out. It would seem that if > the UAC wanted a comparison of this sort for some reason, it should just send > the INVITE without a participant header, receive the 200OK with the current list > of participants and then do the comparison itself - although I do not see why it > would want to do this. The intention was to make sure that the user wants to participate to a specified conference (the participants list in a INVITE request would be an additional criteria besides the requestURI). But because a reqeustURI is unique for a conference, it seems that it is not necessary and that it only produces overhead in the server. We will change this part in the next version of the draft. Thank you very much for your feedback. Regards, Igor _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Wed Feb 13 12:49:50 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id MAA15262 for ; Wed, 13 Feb 2002 12:49:50 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id MAA19644 for sipping-archive@odin.ietf.org; Wed, 13 Feb 2002 12:49:52 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id MAA18477; Wed, 13 Feb 2002 12:26:17 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id MAA18450 for ; Wed, 13 Feb 2002 12:26:15 -0500 (EST) Received: from radvpost.us.radvision.com ([12.44.63.6]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id MAA14569 for ; Wed, 13 Feb 2002 12:26:11 -0500 (EST) Received: by RADVPOST with Internet Mail Service (5.5.2650.21) id ; Wed, 13 Feb 2002 12:24:10 -0500 Message-ID: <0D5BBF5D638DD4119E3400508BD949459ED84C@RADVPOST> From: Orit Levin To: "'igor.miladinovic@tuwien.ac.at'" , "'Sean Olson'" , sipping@ietf.org Cc: "'Johannes Stadler'" Subject: RE: AW: [Sipping] New draft about SIP conferencing avaiable Date: Wed, 13 Feb 2002 12:24:09 -0500 MIME-Version: 1.0 X-Mailer: Internet Mail Service (5.5.2650.21) Content-Type: text/plain; charset="iso-8859-1" Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org A agree that standard way of participants' information distribution and floor control are essential for SIP Conferencing standard application. What bothers me with the current proposals is that they don't perceive a conference as a stand-on-its-own concept. That leads to controversial interpretations and would prevent from smooth service-interworking in future. In my opinion the very first requirements are to define: 1. "Conference Description": a way of specifying a desired (but unknown) conference in terms of capabilities, modes, ... 2. "Conference Identification": a global way of matching existing (and reserved) conferences. (I don't think that existing ids (such as Call-ID) is the right way of identifying a conference.) Those should be equally applicable for different conferencing models such as dial-in vs. dial-out and conferencing server vs. end user mixing. It is a very important but chronologically a second question how to signal these values: new methods, new headers and INVITE vs. Events model. My preference is by not overloading existing fields because of the same service-interworking issue. Orit Levin Chief Architect RADVISION Inc. TEL: +1.201.529.4300 x 230 FAX: +1.201.529.3516 -----Original Message----- From: Igor Miladinovic [mailto:igor.miladinovic@tuwien.ac.at] Sent: Wednesday, February 13, 2002 5:35 AM To: 'Sean Olson'; sipping@ietf.org Cc: 'Johannes Stadler' Subject: AW: AW: [Sipping] New draft about SIP conferencing avaiable Hi, > It was my understanding that the CONF request > was designed to be used to determine the participants > of a conference call before joining that conference > call. That is, at the point in time at which you > construct the CONF request, you may not know the > Call-ID for the conference. But more importantly, > there probably will not be a unique Call-ID for > the conference as in the case of a conference > bridge. Actually, the purpose of the CONF request is to update participants lists in a established conference. To determine the conference participants before joining, we use the "participant" header in the INVITE request. Regards, Igor _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Wed Feb 13 13:21:02 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id NAA16098 for ; Wed, 13 Feb 2002 13:21:01 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id NAA21177 for sipping-archive@odin.ietf.org; Wed, 13 Feb 2002 13:21:03 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id NAA20504; Wed, 13 Feb 2002 13:02:04 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id NAA20466 for ; Wed, 13 Feb 2002 13:02:01 -0500 (EST) Received: from gorilla.mchh.siemens.de (gorilla.mchh.siemens.de [194.138.158.18]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id NAA15634 for ; Wed, 13 Feb 2002 13:01:59 -0500 (EST) Received: from blues.mchh.siemens.de (mail2.mchh.siemens.de [194.138.158.227]) by gorilla.mchh.siemens.de (8.9.3/8.9.3) with ESMTP id TAA14664 for ; Wed, 13 Feb 2002 19:01:58 +0100 (MET) Received: from mchh247e.demchh201e.icn.siemens.de ([139.21.200.57]) by blues.mchh.siemens.de (8.9.1/8.9.1) with ESMTP id TAA22599 for ; Wed, 13 Feb 2002 19:01:52 +0100 (MET) Received: by MCHH247E with Internet Mail Service (5.5.2653.19) id <1J4QWHL5>; Wed, 13 Feb 2002 19:01:57 +0100 Message-ID: <45862F39CADAD511A92100500435EDA304F890@LNN202E> From: Samandi Sami To: sipping@ietf.org Date: Wed, 13 Feb 2002 19:00:00 +0100 MIME-Version: 1.0 X-Mailer: Internet Mail Service (5.5.2653.19) Content-Type: text/plain; charset="ISO-8859-1" Content-Transfer-Encoding: 8bit X-MIME-Autoconverted: from quoted-printable to 8bit by optimus.ietf.org id NAA20467 Subject: [Sipping] Status of draft "draft-rosenberg-sip-vxml-00" Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 8bit Dear all, Sorry if this question has been asked before. I wanted to know the status of draft "draft-rosenberg-sip-vxml-00". Are there any planned modifications or could we consider the draft stable enough to start building UA for VoiceXML dialog server based on its specification ? Thank you Sami Sami SAMANDI SIEMENS R閟eaux Informatiques et T閘閏ommunications 22300 LANNION Phone : + 33 2 96 48 73 99 Fax : + 33 2 96 48 74 73 Email : sami.samandi@srit.siemens.fr Web : http://intranet.srit.siemens.fr/department/se1/ _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Wed Feb 13 15:59:00 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id PAA20550 for ; Wed, 13 Feb 2002 15:59:00 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id PAA01020 for sipping-archive@odin.ietf.org; Wed, 13 Feb 2002 15:59:02 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id PAA29901; Wed, 13 Feb 2002 15:34:18 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id PAA29876 for ; Wed, 13 Feb 2002 15:34:16 -0500 (EST) Received: from mail3.dynamicsoft.com ([63.113.44.69]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id PAA20009 for ; Wed, 13 Feb 2002 15:34:07 -0500 (EST) Received: from dynamicsoft.com ([63.113.46.42]) by mail3.dynamicsoft.com (8.12.1/8.12.1) with ESMTP id g1DKYP6Y029820; Wed, 13 Feb 2002 15:34:26 -0500 (EST) Message-ID: <3C6ACDA0.38698B3F@dynamicsoft.com> Date: Wed, 13 Feb 2002 15:33:36 -0500 From: Jonathan Rosenberg Organization: dynamicsoft X-Mailer: Mozilla 4.75 [en] (Windows NT 5.0; U) X-Accept-Language: en MIME-Version: 1.0 To: Samandi Sami CC: sipping@ietf.org Subject: Re: [Sipping] Status of draft "draft-rosenberg-sip-vxml-00" References: <45862F39CADAD511A92100500435EDA304F890@LNN202E> Content-Type: text/plain; charset=us-ascii Content-Transfer-Encoding: 7bit Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit Samandi Sami wrote: > > Dear all, > > Sorry if this question has been asked before. > > I wanted to know the status of draft "draft-rosenberg-sip-vxml-00". > Are there any planned modifications or could we consider the draft > stable enough to start building UA for VoiceXML dialog server based on > its specification ? The draft is currently not a charter item for sipping. I know of a few people who have implemented it; the only bit which really requires any kind of formal standardization is the http URL in the r-uri. Most of the rest of it is largely implementation hints. -Jonathan R. -- Jonathan D. Rosenberg, Ph.D. 72 Eagle Rock Avenue Chief Scientist First Floor dynamicsoft East Hanover, NJ 07936 jdrosen@dynamicsoft.com FAX: (973) 952-5050 http://www.jdrosen.net PH: (973) 952-5000 http://www.dynamicsoft.com _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Wed Feb 13 17:00:46 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id RAA22464 for ; Wed, 13 Feb 2002 17:00:41 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id RAA04669 for sipping-archive@odin.ietf.org; Wed, 13 Feb 2002 17:00:44 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id QAA03366; Wed, 13 Feb 2002 16:38:23 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id QAA03310 for ; Wed, 13 Feb 2002 16:38:18 -0500 (EST) Received: from bdsl.66.12.12.130.gte.net (bdsl.66.12.12.130.gte.net [66.12.12.130]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id QAA22075 for ; Wed, 13 Feb 2002 16:38:16 -0500 (EST) Received: from dwillis (bdsl.greycouncil.com [127.0.0.1]) by bdsl.66.12.12.130.gte.net (8.11.6/8.11.6) with SMTP id g1DLbYw28031; Wed, 13 Feb 2002 15:37:34 -0600 Message-ID: <016801c1b4d6$d2325700$b3036e3f@dfw.dynamicsoft.com> From: "Dean Willis" To: "Flemming Andreasen" Cc: "Sipping \(E-mail\)" References: <3C60460A.1C8CCEA5@cisco.com> <000d01c1af71$e3658110$9219a8c0@dfw.dynamicsoft.com> <3C61EA80.2F209E5D@cisco.com> <02f301c1b014$80eb18d0$1df30a0a@dfw.dynamicsoft.com> <3C640AC6.96C6292A@cisco.com> <00aa01c1b20a$9c02f0a0$133fed0c@C1893415A> <3C687FB8.E80F6E6E@cisco.com> <00a501c1b467$5d40e610$133fed0c@C1893415A> <3C6A7F1A.84202868@cisco.com> Subject: Re: Signing RPIDS (Was: Re: [Sipping] calling number blocking at sip/isup gateway)] Date: Wed, 13 Feb 2002 15:38:47 -0600 MIME-Version: 1.0 Content-Type: text/plain; charset="iso-8859-1" Content-Transfer-Encoding: 7bit X-Priority: 3 X-MSMail-Priority: Normal X-Mailer: Microsoft Outlook Express 6.00.2600.0000 X-MIMEOLE: Produced By Microsoft MimeOLE V6.00.2600.0000 Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit ----- Original Message ----- From: "Flemming Andreasen" To: "Dean Willis" Cc: "Sipping (E-mail)" Sent: Wednesday, February 13, 2002 8:58 AM Subject: Re: Signing RPIDS (Was: Re: [Sipping] calling number blocking at sip/isup gateway)] > > > Dean Willis wrote: > > > Well, if 80% of the SIP clients are CDMA or W-CDMA (which by definitition > > have synched clocks) that solves a large part of the problem. > > > > Of the otehr 20%, the 80% that know they have bad clocks could just ignore > > the signing on RPIDS. > > Is this the SIP/SIPPING WG chair talking ? Are you seriously suggesting that a > secure solution should only be applicable to some subset of the SIP clients ? I > thought the whole idea with this discussion was to come up with a general > solution. What we're talking about here is providing additional information to help the human participants in a dialog evaluate the preseumed identity of the other participants. This is why I have clearly said that I don't believe RPID replaces any of the authentication techniques. It's a hint. The Privacy draft proposal for RPID provides a very weak indicator of trustworthinesss , basically on the order of "Some proxy somwhere in the network believed this RPID to be valid". All we're proposing doing is adding more indicators: 1) Which proxy(ies) or operator(s) belived this to be true, 2) When they believed it to be true, 3) For what other combinaion of call distinguishers (to, from, call-ID, etc) did they believe it to be true. The only "security weakness" you've suggested with the "signed RPID" approach is that we can't evaluate the reliability of the "when" unless we have a reasonable idea of what time it is. That's reducible to the simple reflexive "If we don't know what time it is, we don't know what time it is." So, even if my client has only a foggy notion of time, I still have a lot more to go on with a signed RPID than with a non-signed RPID. We're worried about replay detection, right. It might be very reasonable for me to look at an incoming INVITE and say "Well, this claims to be from Flemming, and we had a talk about this same subject an hour ago, which oddly enough used a call ID that looked very similar. And look here, the RPID was signed about an hour ago by Cisco's proxy, and my proxy checked that signature and found it to be valid. But INVITES usually don't hang around the network that long before getting to me. I bet this is spam with a stale RPID attached." That's about as much of an identity hint as we're going to be able to produce without a global PKI. . . . > 1. Is signing of Remote-Party-ID a specific case of a more general problem of > authenticating headers, and hence should be solved in a generic way (outside of > the privacy draft) ? Possible -- if there were a general way of authentiating headers, it could be re-used here. But the manner in which the authenticity of RPID is evaluated by humans is probably unique. > 2. Is it reasonable to require synchronized clocks between all SIP entities > in order for signing of Remote-Party-Ids to be secure ? The humans operating SIP entities with loosely synchronized clocks need to be aware of this and probably won't find the RPID as useful as it might be for people with good clocks. > 3. Are people concerned about the computational overhead by always having to > sign a Remote-Party-Id in a message (issue when record-routing for example) ? Then don't sign it. But people may be less willing to trust it if it isn't signed. > > 4. If Remote-Party-Ids are signed (one way or the other), should they be > protected against: > a) cut-and-paste attacks ? > b) replay attacks ? > If not, i.e., it's not really secure after all, what are we gaining by having a > signature scheme ? better hinting to the user than we have now. It's an improvement, not an absolute. > 5. If RPIDs can be signed by more than a small number of well-known parties > (which I believe to be the case), how do you verify the signature ? > a) In particular, how do you obtain the signer's certificate ? > b) How do you handle revoked certificates ? I'd suggest that the certificate model used for web browsing should be directly applicable. There are more than small number of TLS web servers out there, and it seems to work OK. -- Dean _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Wed Feb 13 17:55:11 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id RAA23503 for ; Wed, 13 Feb 2002 17:55:06 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id RAA07391 for sipping-archive@odin.ietf.org; Wed, 13 Feb 2002 17:55:09 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id RAA06664; Wed, 13 Feb 2002 17:39:40 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id RAA06634 for ; Wed, 13 Feb 2002 17:39:38 -0500 (EST) Received: from bdsl.66.12.12.130.gte.net (bdsl.66.12.12.130.gte.net [66.12.12.130]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id RAA23202 for ; Wed, 13 Feb 2002 17:39:34 -0500 (EST) Received: from dwillis (bdsl.greycouncil.com [127.0.0.1]) by bdsl.66.12.12.130.gte.net (8.11.6/8.11.6) with SMTP id g1DMcqw28504; Wed, 13 Feb 2002 16:38:52 -0600 Message-ID: <01b601c1b4df$61eb45c0$b3036e3f@dfw.dynamicsoft.com> From: "Dean Willis" To: "Flemming Andreasen" Cc: "Sipping \(E-mail\)" References: <3C60460A.1C8CCEA5@cisco.com> <000d01c1af71$e3658110$9219a8c0@dfw.dynamicsoft.com> <3C61EA80.2F209E5D@cisco.com> <02f301c1b014$80eb18d0$1df30a0a@dfw.dynamicsoft.com> <3C640AC6.96C6292A@cisco.com> <00aa01c1b20a$9c02f0a0$133fed0c@C1893415A> <3C687FB8.E80F6E6E@cisco.com> <00a501c1b467$5d40e610$133fed0c@C1893415A> <3C6A7F1A.84202868@cisco.com> Subject: Re: Signing RPIDS (Was: Re: [Sipping] calling number blocking at sip/isup gateway)] Date: Wed, 13 Feb 2002 16:40:04 -0600 MIME-Version: 1.0 Content-Type: text/plain; charset="iso-8859-1" Content-Transfer-Encoding: 7bit X-Priority: 3 X-MSMail-Priority: Normal X-Mailer: Microsoft Outlook Express 6.00.2600.0000 X-MIMEOLE: Produced By Microsoft MimeOLE V6.00.2600.0000 Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit ----- Original Message ----- From: "Flemming Andreasen" To: "Dean Willis" > Dean Willis wrote: . . . > > I don't understand the emphasis on the crippled RPID mechanism of the > > privacy draft. Is there some sort of patent somebody is trying to submarine > > into the works? > > Cheap and uncalled for Dean. I apologize for being paranoid. But just because I'm paranoid, it doesn't mean people aren't out to get me, or haven't done so already . . . To my knowledge, you personally have been forthright and helpful in all matters of intellectual property, and I apologize for any implication that you might be otherwise. However, that desirable state is by no means universal, and it is possible that any of us might be getting influenced by other parties who have non-technical interests of this sort. Almost everybody has a marketing department, or IPR office, or friend who's filed a patent application looking over their shoulder. -- Dean _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Wed Feb 13 18:19:21 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id SAA23995 for ; Wed, 13 Feb 2002 18:19:21 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id SAA08641 for sipping-archive@odin.ietf.org; Wed, 13 Feb 2002 18:19:24 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id SAA08101; Wed, 13 Feb 2002 18:03:57 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id SAA08068 for ; Wed, 13 Feb 2002 18:03:55 -0500 (EST) Received: from nisser.cisco.com (nisser.cisco.com [171.71.176.85]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id SAA23771 for ; Wed, 13 Feb 2002 18:03:52 -0500 (EST) Received: from cisco.com (ssh-sj1.cisco.com [171.68.225.134]) by nisser.cisco.com (8.8.6 (PHNE_14041)/CISCO.SERVER.1.2) with ESMTP id PAA21868; Wed, 13 Feb 2002 15:03:15 -0800 (PST) Message-ID: <3C6AF0AA.90B1F236@cisco.com> Date: Wed, 13 Feb 2002 18:03:06 -0500 From: Flemming Andreasen X-Mailer: Mozilla 4.79 [en]C-CCK-MCD (Windows NT 5.0; U) X-Accept-Language: en MIME-Version: 1.0 To: Dean Willis CC: "Sipping (E-mail)" Subject: Re: Signing RPIDS (Was: Re: [Sipping] calling number blocking at sip/isup gateway)] References: <3C60460A.1C8CCEA5@cisco.com> <000d01c1af71$e3658110$9219a8c0@dfw.dynamicsoft.com> <3C61EA80.2F209E5D@cisco.com> <02f301c1b014$80eb18d0$1df30a0a@dfw.dynamicsoft.com> <3C640AC6.96C6292A@cisco.com> <00aa01c1b20a$9c02f0a0$133fed0c@C1893415A> <3C687FB8.E80F6E6E@cisco.com> <00a501c1b467$5d40e610$133fed0c@C1893415A> <3C6A7F1A.84202868@cisco.com> <01b601c1b4df$61eb45c0$b3036e3f@dfw.dynamicsoft.com> Content-Type: text/plain; charset=us-ascii Content-Transfer-Encoding: 7bit Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit Dean Willis wrote: > ----- Original Message ----- > From: "Flemming Andreasen" > To: "Dean Willis" > > Dean Willis wrote: > . . . > > > I don't understand the emphasis on the crippled RPID mechanism of the > > > privacy draft. Is there some sort of patent somebody is trying to > submarine > > > into the works? > > > > Cheap and uncalled for Dean. > > I apologize for being paranoid. But just because I'm paranoid, it doesn't > mean people aren't out to get me, or haven't done so already . . . To my > knowledge, you personally have been forthright and helpful in all matters of > intellectual property, and I apologize for any implication that you might be > otherwise. No problem. -- Flemming > However, that desirable state is by no means universal, and it is > possible that any of us might be getting influenced by other parties who > have non-technical interests of this sort. Almost everybody has a marketing > department, or IPR office, or friend who's filed a patent application > looking over their shoulder. > > -- > Dean -- Flemming Andreasen Cisco Systems _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Thu Feb 14 08:49:51 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id IAA17965 for ; Thu, 14 Feb 2002 08:49:51 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id IAA28786 for sipping-archive@odin.ietf.org; Thu, 14 Feb 2002 08:49:54 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id IAA27444; Thu, 14 Feb 2002 08:17:57 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id IAA27415 for ; Thu, 14 Feb 2002 08:17:54 -0500 (EST) Received: from mr.tuwien.ac.at (mr.tuwien.ac.at [128.130.2.10]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id IAA17292 for ; Thu, 14 Feb 2002 08:17:51 -0500 (EST) Received: from eidolon (eidolon.ikn.tuwien.ac.at [128.131.88.99]) by mr.tuwien.ac.at (8.12.0/8.12.0) with SMTP id g1EDHaSq025232; Thu, 14 Feb 2002 14:17:37 +0100 (MET) From: "Igor Miladinovic" To: "Daniel G. Petrie" , Cc: "'Johannes Stadler'" Subject: RE: AW: AW: [Sipping] New draft about SIP conferencing avaiable Date: Thu, 14 Feb 2002 14:17:41 +0100 Message-ID: MIME-Version: 1.0 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: 7bit X-Priority: 3 (Normal) X-MSMail-Priority: Normal X-Mailer: Microsoft Outlook IMO, Build 9.0.2416 (9.0.2911.0) X-MimeOLE: Produced By Microsoft MimeOLE V5.50.4522.1200 Importance: Normal In-Reply-To: <3C6A88BB.4DB7ECD9@pingtel.com> X-Virus-Scanned: by amavisd-milter (http://amavis.org/) Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit Hi, > > There is a generic way (i.e. independent of conference) to discover > parties in a call of a conference already defined in: > http://www.ietf.org/internet-drafts/draft-rosenberg-sip-call-packa > ge-00.txt > What functionality is missing from the call-package draft? The conference-package can be used with SUBSCRIBE/NOTIFY instead of the CONF method. However, the body containing the conference-package could be used in a CONF request. This way, a UA that supports conferencing extension does not need to support SUB/NOT extension. Besides that, the conferencing extension provides the discovering of participant identities before the conference starts (this information is carried in the participant header of an INVITE request). We think that this draft can be combined with our draft in order to distribute not only participant identities, but also some additional information (floor control policies, the replace URLs...). > A draft also exists for providing operations to join the conference > with a specifed media topology: > http://search.ietf.org/internet-drafts/draft-mahy-sipping-join-and > -fork-00.txt This draft deals with another conferencing issue. The discovering of participants identities is not provided by this draft, but only how to join to an existing conference (with knowledge of the corresponding Call-ID). -- Igor > > Igor Miladinovic wrote: > > > Hi, > > > > > It was my understanding that the CONF request > > > was designed to be used to determine the participants > > > of a conference call before joining that conference > > > call. That is, at the point in time at which you > > > construct the CONF request, you may not know the > > > Call-ID for the conference. But more importantly, > > > there probably will not be a unique Call-ID for > > > the conference as in the case of a conference > > > bridge. > > > > Actually, the purpose of the CONF request is to update > participants lists in > > a established conference. To determine the conference > participants before > > joining, we use the "participant" header in the INVITE request. > > > > Regards, > > > > Igor > > > > _______________________________________________ > > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > > This list is for NEW development of the application of SIP > > Use sip-implementors@cs.columbia.edu for questions on current sip > > Use sip@ietf.org for new developments of core SIP > > > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP > Use sip-implementors@cs.columbia.edu for questions on current sip > Use sip@ietf.org for new developments of core SIP > _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Thu Feb 14 10:56:32 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id KAA25213 for ; Thu, 14 Feb 2002 10:56:32 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id KAA05788 for sipping-archive@odin.ietf.org; Thu, 14 Feb 2002 10:56:34 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id KAA04964; Thu, 14 Feb 2002 10:43:37 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id KAA04933 for ; Thu, 14 Feb 2002 10:43:35 -0500 (EST) Received: from gorilla.mchh.siemens.de (gorilla.mchh.siemens.de [194.138.158.18]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id KAA24928 for ; Thu, 14 Feb 2002 10:43:32 -0500 (EST) Received: from moody.mchh.siemens.de (mail2.mchh.siemens.de [194.138.158.226]) by gorilla.mchh.siemens.de (8.9.3/8.9.3) with ESMTP id QAA08487; Thu, 14 Feb 2002 16:35:52 +0100 (MET) Received: from icn.siemens.de ([139.21.148.41]) by moody.mchh.siemens.de (8.9.1/8.9.1) with ESMTP id QAA13569; Thu, 14 Feb 2002 16:35:50 +0100 (MET) Message-ID: <3C6BD959.93ADF9B7@icn.siemens.de> Date: Thu, 14 Feb 2002 16:35:53 +0100 From: Peter =?iso-8859-1?Q?P=E4ppinghaus?= Organization: Siemens AG X-Mailer: Mozilla 4.73 [en] (Windows NT 5.0; U) X-Accept-Language: en,de MIME-Version: 1.0 To: Jonathan Rosenberg CC: Sipping@ietf.org Subject: Questions on To tags (was Re: [Sipping] Comments on draft-ietf-sip-rfcbis-05.pdf) References: <3C4C2186.DF22B8F5@icn.siemens.de> <3C5470ED.548B8163@dynamicsoft.com> Content-Type: text/plain; charset=iso-8859-1 Content-Transfer-Encoding: 8bit Content-Transfer-Encoding: 8bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 8bit Thanks for your responses, more comments inline (note that I am now referencing bis-07). Jonathan Rosenberg wrote: > > Thanks for your comments. Responses inline. > > Peter P鋚pinghaus wrote: [snip] > > 2) When does a provisional response need to have a tag added to the To > > field by the agent generating this response? > > Always. From 8.2.7: > > If a request contained a To tag in the request, the To field in the > response MUST equal that of the request. However, if the To field in the > request did not contain a tag, the URI in the To field in the response > MUST 1112 > equal the URI in the To field in the request. Additionally, the UAS MUST > addatag to theTo field in the 1113 > response. This serves to identify the UAS that is responding, possibly > resulting in a component of a dialog 1114 > ID. The same tag MUST be used for all responses to that request, both > provisional and final. Procedures for 1115 > generation of tags are defined in Section 21.3. 1116 > > > > > Taking bis-05 literally, each provisional response needs to have a To > > tag. It seems like an oversight, that 100 Trying has not been exempted > > from this requirement. > > In the case of a UA, you are right. We have added text making the use of > a tag in 100 MAY. > > However, having a proxy insert tags in 100 responses isn't helpful. > INteresting, I didn't find any text in the spec that helps understand > how that 100 is generated; the server transaction section simply says > that it should generate a 100 response, but doesn't reference the UA > section for how to do that. > > I will add a reference there, indicating that the exception is that a > tag SHOULD NOT be present (as opposed to MAY), as this would create a > needless early dialog. > > > For example, in section 13.3.1.1, p. 49, it is > > said only of provisional responses between 101 and 199 that they > > establish early dialogs. But even this seems to lead to an unnecessary > > proliferation of early dialogs. To me it seems more reasonable to > > officially introduce the term of a "dialog establishing response" (see, > > e.g., bis-05, section 12.1, p. 41, l. 1536) and to stipulate that dialog > > establishing responses MUST have a To tag, while an intermediate proxy > > generating a provisional response may decide, whether it wants this > > response to be "early dialog establishing" or not. > > I see little harm in always having them established; this way, if the > UAC wants communications back towards the UAS, it can. For a UAC it's fine. However, I am involved in implementing a call stateful proxy. So what worries me is to avoid unnecessary load for a call stateful proxy. If there are many To tags around, a call stateful proxy will have to maintain state for many early dialogs. In particular, I have in mind the following situation. Up to the bis-05 draft it was my understanding that a SIP proxy may also act as follows: it forwards the request downstream AND generates a 183 (Session Progress) response and sends it upstream. In this case the proxy would not really be a communication end point for the signalling, and in this case I feel it makes not so much sense to insert a tag into the To header of this 183 response. This is the situation I am concerned about, because it might happen very frequently on the signalling path. Now the new wording (bis-07) in section 16.6 (Response Processing), step 6 (Choosing the best response) seems to imply that this situation is not supposed to occur. Quotation from lines 2811-2812 of bis-07: "Since a proxy may not insert a tag into the To header field of a 1xx response to a request that did not contain one, it cannot issue non-100 provisional responses on its own." The explanations following immediately (lines 2812-2816), however, seem to revoke lines 2811-2812 in a very subtle way. Does this mean that a proxy, if it wants to forward the request AND generate a 183 response to it, has to act as a forking proxy, forward the request to its destination AND to a "virtual UAS" within itself, including the insertion of a new Via header with branch parameter into both forked requests? This seems like a clumsy way to allow a proxy to send a provisional non-100 response upstream. > > 3) I have not found anywhere in bis-05 spelled out explicitly the "scope > > of uniqueness" of the tags in From and To headers inserted by a SIP > > entity. As I understand it, there should be one value for tags in From > > headers uniquely identifying the SIP entity when generating requests, > > and this value should be the same for all requests generated by it. And > > there should be one value (different from the previous one) for tags in > > To headers uniquely identifying the SIP entity when generating > > responses, and this value should be the same for all responses generated > > by it. > > The scope is broader; the from tag should be unique across all calls, > not just within the same UA. That is, two calls from the same UA would > have different from tags. I can see why that is not sufficiently clear. > I will clarify that. In bis-07, section 23.3, lines 4896-4897, now in fact say: "Similarly, two INVITEs for different calls will have different From tags." This really surprises me for two reasons. Firstly, if the From tag has to be unique across all calls, then it uniquely identifies a call, so what would the Call-ID be needed for? Secondly, if the tags were to uniquely identify a request-generating resp. a response-generating SIP entity, and the same tags were to be used for all requests resp. responses generated by the same SIP entity, this would give additional information (identification of requester resp. responder across calls). Regards, Peter P. -- Dr. Peter P鋚pinghaus Siemens AG | Phone: +49 89 - 722 40065 ICM N PG U ID A1 | Fax: +49 89 - 722 58726 Hofmannstr. 51 | Visitors: Building 1702 / Room 530 D - 81359 M黱chen | Email: peter.paeppinghaus@icn.siemens.de _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Thu Feb 14 16:56:26 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id QAA10940 for ; Thu, 14 Feb 2002 16:56:22 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id QAA29368 for sipping-archive@odin.ietf.org; Thu, 14 Feb 2002 16:56:25 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id QAA28827; Thu, 14 Feb 2002 16:42:23 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id QAA28800 for ; Thu, 14 Feb 2002 16:42:22 -0500 (EST) Received: from acmepacket.com (mail1.acmepacket.com [63.67.143.10]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id QAA10721 for ; Thu, 14 Feb 2002 16:42:17 -0500 (EST) Received: from rporter [63.67.143.2] by acmepacket.com (SMTPD32-7.05) id AF3C11830154; Thu, 14 Feb 2002 16:42:20 -0500 Message-ID: <02bd01c1b5a0$60202a00$2a00000a@acmepacket.com> From: "Rick W. Porter" To: Date: Thu, 14 Feb 2002 16:41:35 -0500 Organization: Acme Packet MIME-Version: 1.0 Content-Type: multipart/alternative; boundary="----=_NextPart_000_02BA_01C1B576.77395920" X-Priority: 3 X-MSMail-Priority: Normal X-Mailer: Microsoft Outlook Express 5.50.4522.1200 X-MimeOLE: Produced By Microsoft MimeOLE V5.50.4522.1200 Subject: [Sipping] (no subject) Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org This is a multi-part message in MIME format. ------=_NextPart_000_02BA_01C1B576.77395920 Content-Type: text/plain; charset="iso-8859-1" Content-Transfer-Encoding: quoted-printable Subscribe ------=_NextPart_000_02BA_01C1B576.77395920 Content-Type: text/html; charset="iso-8859-1" Content-Transfer-Encoding: quoted-printable
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------=_NextPart_000_02BA_01C1B576.77395920-- _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Fri Feb 15 07:35:12 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id HAA03461 for ; Fri, 15 Feb 2002 07:35:12 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id HAA17259 for sipping-archive@odin.ietf.org; Fri, 15 Feb 2002 07:35:13 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id HAA16287; Fri, 15 Feb 2002 07:18:47 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id HAA16246 for ; Fri, 15 Feb 2002 07:18:44 -0500 (EST) Received: from sjoki.uta.fi (sjoki.uta.fi [192.98.80.1]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id HAA03023 for ; Fri, 15 Feb 2002 07:18:42 -0500 (EST) Received: from localhost (localhost [127.0.0.1]) by sjoki.uta.fi (8.9.3/8.9.3) with ESMTP id OAA08933 for ; Fri, 15 Feb 2002 14:18:42 +0200 Date: Fri, 15 Feb 2002 14:18:42 +0200 (EET) From: Mika Mustikkamaki To: sipping@ietf.org Message-ID: MIME-Version: 1.0 Content-Type: TEXT/PLAIN; charset=US-ASCII Subject: [Sipping] SIP client for Linux Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Hi all, There's an updated version of Billy Biggs' KPhone available at http://www.wirlab.net/kphone/ Check out the changes-link (http://www.wirlab.net/kphone/changes.html) for the latest changes... at present KPhone supports all kinds of new features, including Digest-MD5 authentication, improved UI and overall improved functionality. regards, Mika Mustikkamaki Wirlab-project _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Fri Feb 15 10:26:23 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id KAA08080 for ; Fri, 15 Feb 2002 10:26:23 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id KAA25403 for sipping-archive@odin.ietf.org; Fri, 15 Feb 2002 10:26:24 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id KAA24105; Fri, 15 Feb 2002 10:04:08 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id KAA24043 for ; Fri, 15 Feb 2002 10:04:05 -0500 (EST) Received: from yesky.com ([211.167.73.210]) by ietf.org (8.9.1a/8.9.1a) with SMTP id KAA07384 for ; Fri, 15 Feb 2002 10:04:03 -0500 (EST) From: john@yesky.com Message-Id: <200202151504.KAA07384@ietf.org> Received: from world([211.144.74.232]) by yesky.com(JetMail 2.5.3.0) with SMTP id jm503c6d8bb5; Fri, 15 Feb 2002 14:58:36 -0000 To: sipping@ietf.org Content-Type: text/plain; Date: Fri, 15 Feb 2002 23:09:13 +0800 X-Priority: 3 X-Library: Indy 8.0.25 X-Mailer: Send Mail(http://www.co-top.com) Subject: [Sipping] =?GB2312?B?yrXP1sT6ILDZIM3yuLsgzsy1xMPOIM/r?= Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org 朋友: 您好吗?最近我很好!每天都收到很多很多钱,的确感觉好极了!! 魅力无限的internet现在有机会实现我们百 万富 翁的梦想!! 现在我向您介绍一个科学、先进、安全的赚钱系统网站“新 赢 利”。 “新 赢 利”:http://rich4all.yeah.net 好多人通过这种世界流行的“多层次网络营销 MLM” 成为了百万富 翁,这就是 Internet 的力量! 不要吝啬短短5分钟时间,几分钟往往会改变人一生的命运!赶快访问吧!不要犹豫: “新 赢 利”:http://rich4all.yeah.net 赶快加入吧!越早越好!恭喜发财! 您的朋友:天和 QQ:67262864 ★ 千万别错过机会!!!★ _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Sun Feb 17 00:53:16 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id AAA24033 for ; Sun, 17 Feb 2002 00:53:16 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id AAA07588 for sipping-archive@odin.ietf.org; Sun, 17 Feb 2002 00:53:18 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id AAA06348; Sun, 17 Feb 2002 00:21:48 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id AAA06316 for ; Sun, 17 Feb 2002 00:21:45 -0500 (EST) Received: from mail3.dynamicsoft.com ([63.113.44.69]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id AAA23750 for ; Sun, 17 Feb 2002 00:21:37 -0500 (EST) Received: from dynamicsoft.com ([63.113.46.69]) by mail3.dynamicsoft.com (8.12.1/8.12.1) with ESMTP id g1H5Lw6Y005009; Sun, 17 Feb 2002 00:21:58 -0500 (EST) Message-ID: <3C6F3DC4.F6AAB33F@dynamicsoft.com> Date: Sun, 17 Feb 2002 00:21:08 -0500 From: Jonathan Rosenberg Organization: dynamicsoft X-Mailer: Mozilla 4.75 [en] (Windows NT 5.0; U) X-Accept-Language: en MIME-Version: 1.0 To: Venkatesh Venkataramanan CC: "'sipping@ietf.org'" Subject: Re: [Sipping] UAS recovery mechanism from timeouts of responses that establish a dialog References: <79FEAA5FABA7D411BF580001023D1BBD9CFC29@mailserver.sylantro.com> Content-Type: text/plain; charset=us-ascii Content-Transfer-Encoding: 7bit Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit Venkatesh Venkataramanan wrote: > > Hi, > > I have a question regarding the same... > > Say, UA1 issues an INVITE to UA2. UA2 responds with a 100 Trying, > followed > by a 180 Ringing. On answer, it issues a 200 response. Say for some > reason > (short network outage in the subnet coresponding to UA2), the 200 OK > response timesout. Per bis07, UA2 SHOULD issue a BYE to terminate this > dialog. There is a possibility that this BYE reaches UA1, Not likely. The BYE will follow the route-set from the 180, and thus should go to UA2 unless something is misconfigured. -Jonathan R. -- Jonathan D. Rosenberg, Ph.D. 72 Eagle Rock Avenue Chief Scientist First Floor dynamicsoft East Hanover, NJ 07936 jdrosen@dynamicsoft.com FAX: (973) 952-5050 http://www.jdrosen.net PH: (973) 952-5000 http://www.dynamicsoft.com _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Sun Feb 17 00:53:45 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id AAA24052 for ; Sun, 17 Feb 2002 00:53:45 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id AAA07639 for sipping-archive@odin.ietf.org; Sun, 17 Feb 2002 00:53:48 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id AAA06215; Sun, 17 Feb 2002 00:17:06 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id AAA06186 for ; Sun, 17 Feb 2002 00:17:04 -0500 (EST) Received: from mail3.dynamicsoft.com ([63.113.44.69]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id AAA23718 for ; Sun, 17 Feb 2002 00:17:01 -0500 (EST) Received: from dynamicsoft.com ([63.113.46.69]) by mail3.dynamicsoft.com (8.12.1/8.12.1) with ESMTP id g1H5HJ6Y005006; Sun, 17 Feb 2002 00:17:20 -0500 (EST) Message-ID: <3C6F3CAD.275800F4@dynamicsoft.com> Date: Sun, 17 Feb 2002 00:16:29 -0500 From: Jonathan Rosenberg Organization: dynamicsoft X-Mailer: Mozilla 4.75 [en] (Windows NT 5.0; U) X-Accept-Language: en MIME-Version: 1.0 To: Peter =?iso-8859-1?Q?P=E4ppinghaus?= CC: Sipping@ietf.org Subject: Re: Questions on To tags (was Re: [Sipping] Comments on draft-ietf-sip-rfcbis-05.pdf) References: <3C6BD959.93ADF9B7@icn.siemens.de> Content-Type: text/plain; charset=iso-8859-1 Content-Transfer-Encoding: 8bit Content-Transfer-Encoding: 8bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 8bit Peter P鋚pinghaus wrote: > > > I see little harm in always having them established; this way, if the > > UAC wants communications back towards the UAS, it can. > > For a UAC it's fine. However, I am involved in implementing a call > stateful proxy. So what worries me is to avoid unnecessary load for a > call stateful proxy. If there are many To tags around, a call stateful > proxy will have to maintain state for many early dialogs. > > In particular, I have in mind the following situation. > > Up to the bis-05 draft it was my understanding that a SIP proxy may also > act as follows: it forwards the request downstream AND generates a > 183 (Session Progress) response and sends it upstream. In this case the > proxy would not really be a communication end point for the signalling, > and in this case I feel it makes not so much sense to insert a tag into > the To header of this 183 response. This is the situation I am concerned > about, because it might happen very frequently on the signalling path. Well, the problem is, there is no reason for the proxy to generate this provisional response. It can, but I would hardly think it a common occurrence. > > Now the new wording (bis-07) in section 16.6 (Response Processing), > step 6 (Choosing the best response) seems to imply that this situation > is not supposed to occur. Quotation from lines 2811-2812 of bis-07: > > "Since a proxy may not insert a tag into the To header field of a > 1xx response to a request that did not contain one, > it cannot issue non-100 provisional responses on its own." > > The explanations following immediately (lines 2812-2816), however, > seem to revoke lines 2811-2812 in a very subtle way. > > Does this mean that a proxy, if it wants to forward the request AND > generate a 183 response to it, has to act as a forking proxy, forward > the request to its destination AND to a "virtual UAS" within itself, > including the insertion of a new Via header with branch parameter > into both forked requests? > This seems like a clumsy way to allow a proxy to send a provisional > non-100 response upstream. The specification does not mandate an implementation architecture, it merely describes behavior. The point of this missive is to explain how a proxy, which is a LOGICAL element which does not participate in dialogs, would appear to issue a provisional response which establishes a dialog. I would be shocked if anyone actually implemented it in the way discussed above. > > The scope is broader; the from tag should be unique across all calls, > > not just within the same UA. That is, two calls from the same UA would > > have different from tags. I can see why that is not sufficiently > clear. > > I will clarify that. > > In bis-07, section 23.3, lines 4896-4897, now in fact say: > "Similarly, two INVITEs for different calls will have different From > tags." > > This really surprises me for two reasons. > > Firstly, if the From tag has to be unique across all calls, then it > uniquely identifies a call, so what would the Call-ID be needed for? If we were able to start SIP from scratch, I would probably agree that the Call-ID and from tag are redundant. But, we can't. So, the real question is this - given we have a unique call-id, and that people have been putting them in that way, and relying on their uniqueness, is there also a need for the from tag/to tag to be unique? There, there answer is in fact yes, but for subtle reasons. Certainly, the to tag in a response needs to be unique. What you really, really need is for both participants in the dialog to contribute unique identifiers, such that the combination of the two are the dialog ID. So, the to tag has to be unique, OK. Now, in requests in the reverse direction that the dialog was established, the To and From field values are reversed. This means that what was in the From in the initial request, is now in the To, and vice a versa. This means that elements which might like to take advantage of the fact that the to tag is unique, really cannot, since it wouldn't be in the reverse direction unless the from tag in the original request was also unique. And thus the requirement for unique from tags as well. So, simply put, the from tag is unique to provide symmetry wiht the to tag, which must be unique. > > Secondly, if the tags were to uniquely identify a request-generating > resp. a response-generating SIP entity, and the same tags were to be > used for all requests resp. responses generated by the same SIP entity, > this would give additional information (identification of requester > resp. responder across calls). I cannot parse this sentence. -Jonathan R. -- Jonathan D. Rosenberg, Ph.D. 72 Eagle Rock Avenue Chief Scientist First Floor dynamicsoft East Hanover, NJ 07936 jdrosen@dynamicsoft.com FAX: (973) 952-5050 http://www.jdrosen.net PH: (973) 952-5000 http://www.dynamicsoft.com _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Sun Feb 17 11:53:55 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id LAA08339 for ; Sun, 17 Feb 2002 11:53:55 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id LAA16116 for sipping-archive@odin.ietf.org; Sun, 17 Feb 2002 11:53:58 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id LAA14316; Sun, 17 Feb 2002 11:28:25 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id LAA14277 for ; Sun, 17 Feb 2002 11:28:22 -0500 (EST) Received: from maynard.mail.mindspring.net (maynard.mail.mindspring.net [207.69.200.243]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id LAA07946; Sun, 17 Feb 2002 11:28:18 -0500 (EST) Received: from user-2ivelt4.dialup.mindspring.com ([165.247.87.164] helo=dick.ix.netcom.com) by maynard.mail.mindspring.net with esmtp (Exim 3.33 #1) id 16cUAY-0004dR-00; Sun, 17 Feb 2002 11:28:07 -0500 Message-Id: <5.1.0.14.2.20020217111648.0234b0f8@popd.ix.netcom.com> X-Sender: rshockey@popd.ix.netcom.com X-Mailer: QUALCOMM Windows Eudora Version 5.1 Date: Sun, 17 Feb 2002 11:28:53 -0500 To: "'Patrik =?iso-8859-1?Q?F=E4ltstr=F6m=27?=" , "Peterson, Jon" , "'Kevin McCandless'" , "'enum@ietf.org'" From: Richard Shockey Cc: sipping@ietf.org, rohan@cisco.com, brian.rosen@marconi.com, dean.willis@softarmor.com In-Reply-To: <70565611B164D511957A001083FCDD56CAAF55@VA02> Mime-Version: 1.0 Content-Type: text/plain; charset="us-ascii"; format=flowed Subject: [Sipping] RE: [Enum] Using ENUM for SIP Applications Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org A >[snip] > > > > Should for example the flag in the NAPTR record differ between a SIP URI > > used for presence and one used for gaming and one used for voice calls? > > > > That MUST be resolved, I would say, before the registration document for > > SIP-related flags is finished. > >I agree totally. We present quite a few arguments for one perspective on >this matter in our document. While we don't go ahead and register any >SIP-related flags in the IANA considerations section, future versions of >this document are probably as good a place to register 'sip+E2U' as any. > >I'd be interested to know whether or not there is basically consensus, based >on the arguments that we've raised in sipping-e164-01, that we need no ENUM >service fields for SIP other than 'sip+E2U'. Maybe if we have time this is >something we could discuss in MN, at the discretion of the chairs. I think you can assume that there will be time on the ENUM WG agenda in MN for this document. There is obviously a lot of interest in this. We have a 2 1/2 hour slot .. and SIPPING has a lot on its plate. Now we have not cross posted these messages to SIPPING ...to avoid overburdening their totally overburdened list ... but we need to keep them in the loop and make sure they are aware of this discussion and that we probably will discuss the document in MN and that all SIPPING WG participants with an interest here might want to attend. I'm assuming this plan is OK with the SIPPING Chairs.... >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> Richard Shockey, Senior Manager, Strategic Technology Initiatives NeuStar Inc. 45980 Center Oak Plaza Bldg 8 Sterling, VA 20166 1120 Vermont Ave NW Suite 400 Washington DC 20005 Voice +1 571.434.5651 Cell : +1 314.503.0640, Fax: +1 815.333.1237 or <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Sun Feb 17 15:42:45 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id PAA11215 for ; Sun, 17 Feb 2002 15:42:44 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id PAA27159 for sipping-archive@odin.ietf.org; Sun, 17 Feb 2002 15:42:48 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id PAA25712; Sun, 17 Feb 2002 15:17:10 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id PAA25675 for ; Sun, 17 Feb 2002 15:17:08 -0500 (EST) Received: from cs.columbia.edu (cs.columbia.edu [128.59.16.20]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id PAA10843 for ; Sun, 17 Feb 2002 15:17:04 -0500 (EST) Received: from opus.cs.columbia.edu (opus.cs.columbia.edu [128.59.20.100]) by cs.columbia.edu (8.9.3/8.9.3) with ESMTP id PAA04697; Sun, 17 Feb 2002 15:17:06 -0500 (EST) Received: from cs.columbia.edu (cta.cs.columbia.edu [128.59.19.46]) (authenticated bits=0) by opus.cs.columbia.edu (8.12.1/8.12.1) with ESMTP id g1HKGtNB008548 (version=TLSv1/SSLv3 cipher=RC4-MD5 bits=128 verify=NOT); Sun, 17 Feb 2002 15:17:05 -0500 (EST) Message-ID: <3C700FB1.992AA7D3@cs.columbia.edu> Date: Sun, 17 Feb 2002 15:16:49 -0500 From: Henning Schulzrinne Organization: Columbia University X-Mailer: Mozilla 4.76 [en] (Windows NT 5.0; U) X-Accept-Language: en MIME-Version: 1.0 To: "Rick W. Porter" CC: sipping@ietf.org, Sankaran Narayanan , Anshul Kundaje References: <012f01c1b4a8$da106ee0$2a00000a@acmepacket.com> Content-Type: text/plain; charset=us-ascii Content-Transfer-Encoding: 7bit Content-Transfer-Encoding: 7bit Subject: [Sipping] Re: [Sip] comments on draft-schulzrinne-accounting-sip-00.txt Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit [Discussion moved to SIPPING.] Thanks for your comments. A new draft will appear shortly, renamed as draft-schulzrinne-sipping-radius-accounting-00, that tries to address these and other comments. The draft is at http://www.cs.columbia.edu/~hgs/sip/drafts/draft-schulzrinne-sipping-radius-accounting-00.txt until it appears in the I-D archives. "Rick W. Porter" wrote: > > My major concern is the draft does not address branching scenarios. The SIP > Call-Id will be the same for all branches, and there is nothing in the > RADIUS message to distinguish each branch. The back-end RADIUS processing > may break down if the fundamental assumption of one Start and one Stop per > session is violated. I see two possible solutions: include the Branch-ID for > all RADIUS messages, or make the SIP proxy stateful enough to generate > RADIUS messages for only chosen branch (or the whole session if all branches > fail). I've added a branch identifier, and noted that logging can be done either for one request/response or for each branch. > > My detailed concerns are listed by section comments below: > > Section 5.1 (User-name, p4) - > If the intent of this attribute is to establish responsibility for the > session, it does not work unless the SIP proxy is session stateful! The only > way to record responsibility is when the SIP proxy receives the initial > INVITE (or the first one that is Authenticated), because the SIP message > From, To, and Authentication username, will change based on who initiates > each SIP message, not who initiated the session. Therefore, I think the > attribute fails to meet the its intent. Furthermore, without Authentication > this attribute will be redundant--why include the information twice? Maybe > make this should be an optional RADIUS attribute that is include if > authentication is done? > Modified to reflect authentication user. This is not necessarily the party responsible for paying, say. It would be up to the consumer of the accounting data to determine, e.g., whether the one to send the first INVITE is responsible for the call. I don't see how we can say much more in SIP. Maybe in the future there will be explicit indications of this, but presumably outside core SIP. > Section 5.5 (Acct-Status-Type, p6) - > It appears that the draft suggests generating an Interim-Update to report > receiving all SIP messages other than initial INVITEs, BYEs, and failures > for initial INVITEs. This creates potential problems for branching > scenarios as discussed above. Clarified how this should work with call stateful, transaction stateful and stateless logging. > > Section 5.6 (Acct-Delay-Time, p7) - > By including this attribute, the RADUIS message must get a new ID and the > authenticator must be recalculated each time the message is retransmitted. > It would seem to me that it is more efficient to let messages age with the > "time sensitive" information included as part of the original message (i.e. > Event-timestamp). Or, do RADIUS servers do something useful with the > Acct-Delay-Time attribute? Agreed that this isn't all that helpful; dropped. > > Section 5.9 (Acct-Session-Time, p8) - > First, unless your SIP proxy is session stateful, where do you get the > information to calculate this? Second, since the amount of time from > receiving the INVITE to receiving the 200-OK is not deterministic, it seems > to me that this time should be 0 for all sessions that never received the > 200-OK; again, the SIP proxy needs to be session stateful to get this > information. You wouldn't want to get a phone bill for attempted calls where > the line was busy, so the reported time should be the time from the 200-OK > to the BYE. Noted that this is for call stateful only. Clarified 200 (OK) relationship. > > Section 5.10 (Acct-Terminate-Cause, p9) - > How do SIP events map to the RADIUS values? Some are intuitive (i.e. BYE = > User Request), but others are not (404, 482, 500, ...). This should only occur for call stateful logging. I've added some explanation. > > Section 5.14 (Sip-From, p12) and Section 5.15 (Sip-From, p12) - > Why not use the Calling-Station-Id and Called-Station-Id instead of defining > new attributes? Interesting idea. I was a little leery of this given the restrictive description of the field in RFC 2865, but this is probably close enough to be useable. > > Section 5.17 (Sip-Remote-IP-Address, p14) and Section 5.18 (Sip-Remote-Port, > p14) - > First, I would suggest a name change to Sip-Previous-IP-Address and > Sip-Previous-Port, since there may be a inbound and outbound Remote IP > Address/Port. Second, where does this info come from (Rec-Route header? Via > header? or socket?) There may be minor NAT issues that may be intertwined > with your choice. > I've changed this to Sip-Source*, reflecting the origin of the request. That seems to be the most interesting piece of information. Should the destination IP addresses also be logged? _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Sun Feb 17 16:39:13 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id QAA11813 for ; Sun, 17 Feb 2002 16:39:13 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id QAA29964 for sipping-archive@odin.ietf.org; Sun, 17 Feb 2002 16:39:17 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id QAA28427; Sun, 17 Feb 2002 16:14:09 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id QAA28364 for ; Sun, 17 Feb 2002 16:14:05 -0500 (EST) Received: from sj-msg-core-3.cisco.com (sj-msg-core-3.cisco.com [171.70.157.152]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id QAA11553; Sun, 17 Feb 2002 16:13:59 -0500 (EST) Received: from imop.cisco.com (imop.cisco.com [171.69.11.44]) by sj-msg-core-3.cisco.com (8.11.3/8.9.1) with ESMTP id g1HLDAZ25181; Sun, 17 Feb 2002 13:13:10 -0800 (PST) Received: from localhost (ssh-sj1.cisco.com [171.68.225.134]) by imop.cisco.com (Mirapoint) with ESMTP id ACM58632; Sun, 17 Feb 2002 13:13:16 -0800 (PST) Date: Sun, 17 Feb 2002 13:11:55 -0800 (Pacific Standard Time) From: Rohan Mahy To: Richard Shockey cc: "'Patrik =?iso-8859-1?Q?F=E4ltstr=F6m=27?=" , "Peterson, Jon" , "'Kevin McCandless'" , "'enum@ietf.org'" , , , In-Reply-To: <5.1.0.14.2.20020217111648.0234b0f8@popd.ix.netcom.com> Message-ID: X-X-Sender: rmahy@imop.cisco.com MIME-Version: 1.0 Content-Type: TEXT/PLAIN; charset=US-ASCII Subject: [Sipping] RE: [Enum] Using ENUM for SIP Applications Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org discussing this in ENUM is fine by me. we can make an announcement during our first session. thanks, -rohan On Sun, 17 Feb 2002, Richard Shockey wrote: > A > >[snip] > > > > > > Should for example the flag in the NAPTR record differ between a SIP URI > > > used for presence and one used for gaming and one used for voice calls? > > > > > > That MUST be resolved, I would say, before the registration document for > > > SIP-related flags is finished. > > > >I agree totally. We present quite a few arguments for one perspective on > >this matter in our document. While we don't go ahead and register any > >SIP-related flags in the IANA considerations section, future versions of > >this document are probably as good a place to register 'sip+E2U' as any. > > > >I'd be interested to know whether or not there is basically consensus, based > >on the arguments that we've raised in sipping-e164-01, that we need no ENUM > >service fields for SIP other than 'sip+E2U'. Maybe if we have time this is > >something we could discuss in MN, at the discretion of the chairs. > > I think you can assume that there will be time on the ENUM WG agenda in MN > for this document. There is obviously a lot of interest in this. We have a > 2 1/2 hour slot .. and SIPPING has a lot on its plate. > > Now we have not cross posted these messages to SIPPING ...to avoid > overburdening their totally overburdened list ... but we need to keep them > in the loop and make sure they are aware of this discussion and that we > probably will discuss the document in MN and that all SIPPING WG > participants with an interest here might want to attend. > > I'm assuming this plan is OK with the SIPPING Chairs.... > > > > > >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> > Richard Shockey, Senior Manager, Strategic Technology Initiatives > NeuStar Inc. > 45980 Center Oak Plaza Bldg 8 Sterling, VA 20166 > 1120 Vermont Ave NW Suite 400 Washington DC 20005 > Voice +1 571.434.5651 Cell : +1 314.503.0640, Fax: +1 815.333.1237 > or > > > > <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< > > _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Sun Feb 17 22:57:27 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id WAA17520 for ; Sun, 17 Feb 2002 22:57:27 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id WAA15415 for sipping-archive@odin.ietf.org; Sun, 17 Feb 2002 22:57:31 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id WAA14258; Sun, 17 Feb 2002 22:27:05 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id WAA14202 for ; Sun, 17 Feb 2002 22:27:02 -0500 (EST) Received: from mclean.mail.mindspring.net (mclean.mail.mindspring.net [207.69.200.57]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id WAA16200; Sun, 17 Feb 2002 22:26:57 -0500 (EST) Received: from user-2ivel9d.dialup.mindspring.com ([165.247.85.45] helo=dick.ix.netcom.com) by mclean.mail.mindspring.net with esmtp (Exim 3.33 #1) id 16ceRz-0001CZ-00; Sun, 17 Feb 2002 22:26:48 -0500 Message-Id: <5.1.0.14.2.20020217222000.044264f0@popd.ix.netcom.com> X-Sender: rshockey@popd.ix.netcom.com X-Mailer: QUALCOMM Windows Eudora Version 5.1 Date: Sun, 17 Feb 2002 22:28:59 -0500 To: Rohan Mahy From: Richard Shockey Cc: =?iso-8859-1?Q?=22=27Patrik_F=E4ltstr=F6m=27=22?= , "Peterson, Jon" , "'Kevin McCandless'" , "'enum@ietf.org'" , , , In-Reply-To: References: <5.1.0.14.2.20020217111648.0234b0f8@popd.ix.netcom.com> Mime-Version: 1.0 Content-Type: text/plain; charset="us-ascii"; format=flowed Subject: [Sipping] RE: [Enum] Using ENUM for SIP Applications Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org At 01:11 PM 2/17/2002 -0800, Rohan Mahy wrote: >discussing this in ENUM is fine by me. we can make an announcement during >our first session. I'll be happy to make the announcement myself it you think that will help...I do want this as a joint effort. >thanks, >-rohan Thanks again Rohan ..I thought that's how you might feel ..though I want to make sure that Brian and Dean concur. I'm deeply sympathetic to the work load the SIPPING WG has but I want to make sure your WG and ours is properly informed at all stages of this discussion and that when this document or one that is similar to it comes to last call it will be a joint action between both WG's. A URL for this Internet-Draft is: http://www.ietf.org/internet-drafts/draft-ietf-sipping-e164-01.txt BTW for those SIPPING folks who are scratching their heads wondering what this thread is all about . I'm the co-chair of the ENUM WG and if you need more info go to : http://www.ietf.org/html.charters/enum-charter.html >On Sun, 17 Feb 2002, Richard Shockey wrote: > > > A > > >[snip] > > > > > > > > Should for example the flag in the NAPTR record differ between a > SIP URI > > > > used for presence and one used for gaming and one used for voice calls? > > > > > > > > That MUST be resolved, I would say, before the registration > document for > > > > SIP-related flags is finished. > > > > > >I agree totally. We present quite a few arguments for one perspective on > > >this matter in our document. While we don't go ahead and register any > > >SIP-related flags in the IANA considerations section, future versions of > > >this document are probably as good a place to register 'sip+E2U' as any. > > > > > >I'd be interested to know whether or not there is basically consensus, > based > > >on the arguments that we've raised in sipping-e164-01, that we need no > ENUM > > >service fields for SIP other than 'sip+E2U'. Maybe if we have time this is > > >something we could discuss in MN, at the discretion of the chairs. > > > > I think you can assume that there will be time on the ENUM WG agenda in MN > > for this document. There is obviously a lot of interest in this. We have a > > 2 1/2 hour slot .. and SIPPING has a lot on its plate. > > > > Now we have not cross posted these messages to SIPPING ...to avoid > > overburdening their totally overburdened list ... but we need to keep them > > in the loop and make sure they are aware of this discussion and that we > > probably will discuss the document in MN and that all SIPPING WG > > participants with an interest here might want to attend. > > > > I'm assuming this plan is OK with the SIPPING Chairs.... > > > > >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> Richard Shockey, Senior Manager, Strategic Technology Initiatives NeuStar Inc. 45980 Center Oak Plaza Bldg 8 Sterling, VA 20166 1120 Vermont Ave NW Suite 400 Washington DC 20005 Voice +1 571.434.5651 Cell : +1 314.503.0640, Fax: +1 815.333.1237 or <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Mon Feb 18 15:40:25 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id PAA17369 for ; Mon, 18 Feb 2002 15:40:24 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id PAA11148 for sipping-archive@odin.ietf.org; Mon, 18 Feb 2002 15:40:27 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id PAA10054; Mon, 18 Feb 2002 15:24:21 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id PAA09935 for ; Mon, 18 Feb 2002 15:24:13 -0500 (EST) Received: from zrtps0kn.nortelnetworks.com (zrtps0kn.nortelnetworks.com [47.140.192.55]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id PAA16805 for ; Mon, 18 Feb 2002 15:24:08 -0500 (EST) Received: from zctfs063.nortelnetworks.com (zctfs063.europe.nortel.com [47.164.128.120]) by zrtps0kn.nortelnetworks.com (Switch-2.2.0/Switch-2.2.0) with ESMTP id g1IKNGG24842; Mon, 18 Feb 2002 15:23:17 -0500 (EST) Received: from znsgs01r.europe.nortel.com (znsgs01r.europe.nortel.com [47.137.129.92]) by zctfs063.nortelnetworks.com (Switch-2.2.0/Switch-2.2.0) with ESMTP id g1IKNDT25919; Mon, 18 Feb 2002 21:23:14 +0100 (MET) Received: from zwcwc012.europe.nortel.com (zwcwc012.europe.nortel.com [47.73.112.187]) by znsgs01r.europe.nortel.com (Switch-2.2.0/Switch-2.2.0) with ESMTP id g1IKMmT09192; Mon, 18 Feb 2002 20:22:48 GMT Received: by zwcwc012.europe.nortel.com with Internet Mail Service (5.5.2653.19) id <1XS8FBZ7>; Mon, 18 Feb 2002 20:23:11 -0000 Message-ID: From: "Mark Watson" To: Juha Heinanen , "Peterson, Jon" , Jonathan Rosenberg , "Tom-PT Taylor", "'Mpierce1@aol.com'" , sipping@ietf.org Subject: RE: [Sipping] RE: calling number blocking at sip/isup gateway Date: Mon, 18 Feb 2002 20:23:09 -0000 MIME-Version: 1.0 X-Mailer: Internet Mail Service (5.5.2653.19) Content-Type: multipart/alternative; boundary="----_=_NextPart_001_01C1B8BA.14CF43C0" Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org This message is in MIME format. Since your mail reader does not understand this format, some or all of this message may not be legible. ------_=_NextPart_001_01C1B8BA.14CF43C0 Content-Type: text/plain All, OK, so I'm a little bit behind (I have a good excuse!). I don't want to prolong this thread unnecessarily, and I don't see any reason to hold up the IETF SIP/ISUP drafts, but there are a couple of points worth making: 1) As to whether a 'restricted' indicator is *required* for PSTN interworking, yes there are protocols which interwork with the PSTN which do not support such an indication (e.g. DTMF signalling over an analogue loop), but at least in Europe, these have been kludged by defining prefix digit strings. You can get away with defaulting to 'restricted' if you don't know the users wishes, but not for very long: (i) users get unhappy about having no way around Anonymous Caller Rejection (ii) regulators will not be impressed that the feted new SIP domain doesn't support something they presently take for granted in the PSTN. Some PSTN networks support a 'restricted by network' indication which prevents display of the number, but does not imply that the user asked for the restriction, meaning that ACR should not reject the call. That the user should not need to reveal their identity to the called party is a fundamental requirement stemming from Data Protection legislation (at least in Europe). That the users identity will nevertheless need to be passed between Service Providers seems to me obvious. So, I think this *is* a requirement at least in the medium term. You will only suceed in pursuading regulators to ignore this whilst SIP is in its infancy. 2) We are hoping to take this into account in the ongoing ITU-T SIP/ISUP/SIP-T work, which will result in an ITU-T Recommendation. For this we would need IETF to define something in SIP - right now Remote-Party-ID looks sufficient. The IETF SIP/ISUP work (on which the ITU-T stuff is based) is only Informative, so I would expect gateway vendors to look to the ITU-T work when it is done. ...Mark > -----Original Message----- > From: Juha Heinanen [mailto:jh@lohi.eng.song.fi] > Sent: 07 February 2002 19:26 > To: Peterson, Jon > Cc: Jonathan Rosenberg; Taylor, Tom-PT [CAR:B800:EXCH]; > 'Mpierce1@aol.com'; sipping@ietf.org > Subject: RE: [Sipping] RE: calling number blocking at sip/isup gateway > > > Peterson, Jon writes: > > > I'll reiterate - the purpose of the SIP-ISUP mapping project (known > > collectively as SIP-T) is not to add new features to SIP - > it's to describe > > how SIP and SS7, without any modifications on either side, can be > > interworked. > > in that case, it might be fair to list in the i-d its applicability, > i.e., which ss7 capabilities are not supported by this i-d. > > > Anyway, the real problem is the want of a general privacy > mechanism in SIP - > > this really isn't draft-ietf-sipping-isup-00's problem. If > there weren't > > even an existing proposal for a presentation restriction > mechanism, would > > you still insist that the SIP-ISUP mapping draft should > block until someone > > proposes this feature for SIP? > > no, it would be ok to go forward as long as the things that can't be > mapped are listed. > > -- juha > > > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP > Use sip-implementors@cs.columbia.edu for questions on current sip > Use sip@ietf.org for new developments of core SIP > ------_=_NextPart_001_01C1B8BA.14CF43C0 Content-Type: text/html Content-Transfer-Encoding: quoted-printable RE: [Sipping] RE: calling number blocking at sip/isup = gateway

All,

OK, so I'm a little bit behind (I have a good = excuse!). I don't want to prolong this thread unnecessarily, and I = don't see any reason to hold up the IETF SIP/ISUP drafts, but there are = a couple of points worth making:

1) As to whether a 'restricted' indicator is = *required* for PSTN interworking, yes there are protocols which = interwork with the PSTN which do not support such an indication (e.g. = DTMF signalling over an analogue loop), but at least in Europe, these = have been kludged by defining prefix digit strings.

You can get away with defaulting to 'restricted' if = you don't know the users wishes, but not for very long: (i) users get = unhappy about having no way around Anonymous Caller Rejection (ii) = regulators will not be impressed that the feted new SIP domain doesn't = support something they presently take for granted in the = PSTN.

Some PSTN networks support a 'restricted by network' = indication which prevents display of the number, but does not imply = that the user asked for the restriction, meaning that ACR should not = reject the call.

That the user should not need to reveal their = identity to the called party is a fundamental requirement stemming from = Data Protection legislation (at least in Europe).

That the users identity will nevertheless need to be = passed between Service Providers seems to me obvious.

So, I think this *is* a requirement at least in the = medium term. You will only suceed in pursuading regulators to ignore = this whilst SIP is in its infancy.

2) We are hoping to take this into account in the = ongoing ITU-T SIP/ISUP/SIP-T work, which will result in an ITU-T = Recommendation. For this we would need IETF to define something in SIP = - right now Remote-Party-ID looks sufficient. The IETF SIP/ISUP work = (on which the ITU-T stuff is based) is only Informative, so I would = expect gateway vendors to look to the ITU-T work when it is = done.

...Mark

> -----Original Message-----
> From: Juha Heinanen [mailto:jh@lohi.eng.song.fi]
> Sent: 07 February 2002 19:26
> To: Peterson, Jon
> Cc: Jonathan Rosenberg; Taylor, Tom-PT = [CAR:B800:EXCH];
> 'Mpierce1@aol.com'; sipping@ietf.org
> Subject: RE: [Sipping] RE: calling number = blocking at sip/isup gateway
>
>
> Peterson, Jon writes:
>
>  > I'll reiterate - the purpose of the = SIP-ISUP mapping project (known
>  > collectively as SIP-T) is not to add = new features to SIP -
> it's to describe
>  > how SIP and SS7, without any = modifications on either side, can be
>  > interworked.
>
> in that case, it might be fair to list in the = i-d its applicability,
> i.e., which ss7 capabilities are not supported = by this i-d.
>
>  > Anyway, the real problem is the want = of a general privacy
> mechanism in SIP -
>  > this really isn't = draft-ietf-sipping-isup-00's problem. If
> there weren't
>  > even an existing proposal for a = presentation restriction
> mechanism, would
>  > you still insist that the SIP-ISUP = mapping draft should
> block until someone
>  > proposes this feature for = SIP?
>
> no, it would be ok to go forward as long as the = things that can't be
> mapped are listed.
>
> -- juha
>
>
> = _______________________________________________
> Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
> This list is for NEW development of the = application of SIP
> Use sip-implementors@cs.columbia.edu for = questions on current sip
> Use sip@ietf.org for new developments of core = SIP
>

------_=_NextPart_001_01C1B8BA.14CF43C0-- _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Mon Feb 18 15:40:26 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id PAA17386 for ; Mon, 18 Feb 2002 15:40:25 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id PAA11162 for sipping-archive@odin.ietf.org; Mon, 18 Feb 2002 15:40:28 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id PAA09970; Mon, 18 Feb 2002 15:24:15 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id PAA09927 for ; Mon, 18 Feb 2002 15:24:12 -0500 (EST) Received: from zrtps0kn.nortelnetworks.com (zrtps0kn.nortelnetworks.com [47.140.192.55]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id PAA16801 for ; Mon, 18 Feb 2002 15:24:07 -0500 (EST) Received: from zctfs063.nortelnetworks.com (zctfs063.europe.nortel.com [47.164.128.120]) by zrtps0kn.nortelnetworks.com (Switch-2.2.0/Switch-2.2.0) with ESMTP id g1IKN9G24826; Mon, 18 Feb 2002 15:23:09 -0500 (EST) Received: from znsgs01r.europe.nortel.com (znsgs01r.europe.nortel.com [47.137.129.92]) by zctfs063.nortelnetworks.com (Switch-2.2.0/Switch-2.2.0) with ESMTP id g1IKN5T25905; Mon, 18 Feb 2002 21:23:05 +0100 (MET) Received: from zwcwc012.europe.nortel.com (zwcwc012.europe.nortel.com [47.73.112.187]) by znsgs01r.europe.nortel.com (Switch-2.2.0/Switch-2.2.0) with ESMTP id g1IKMdT09180; Mon, 18 Feb 2002 20:22:39 GMT Received: by zwcwc012.europe.nortel.com with Internet Mail Service (5.5.2653.19) id <1XS8FBZX>; Mon, 18 Feb 2002 20:23:02 -0000 Message-ID: From: "Mark Watson" To: Christian Huitema , Juha Heinanen , Dean Willis Cc: sipping@ietf.org Subject: RE: [Sipping] calling number blocking at sip/isup gateway Date: Mon, 18 Feb 2002 20:23:01 -0000 MIME-Version: 1.0 X-Mailer: Internet Mail Service (5.5.2653.19) Content-Type: multipart/alternative; boundary="----_=_NextPart_001_01C1B8BA.0FD88340" Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org This message is in MIME format. Since your mail reader does not understand this format, some or all of this message may not be legible. ------_=_NextPart_001_01C1B8BA.0FD88340 Content-Type: text/plain; charset="iso-8859-1" Content-Transfer-Encoding: quoted-printable Christian, Data Protection legislation is not technology-specific or tied to = licencing. If you do operation X with the Personal Data of an individual, and X = falls within the scope of Data Protection, it does not matter whether you do = this in the phone network, the Internet or on a beach in Bali, you are = subject to the same rules. Now, in Europe, and perhaps elsewhere, our legislators have tried to be helpful by describing in detail the effect of these principles in = certain sectors of industry - for example in the European Data Protection in Telecommunications Directive (97/66/EC: http://europa.eu.int/eur-lex/en/lif/dat/1997/en_397L0066.html). The applicability of such statements to services offered over the = Internet can be debated but this does not affect the requirement to uphold the principles enshrined in the primary Data Protection legisation. If you = want to get privacy right, you need to understand these. Of course, what information two end-users do/don't exchange is entirely their own business, but where third parties (dare I say 'Service Providers') are involved they have certain obligations with respect to personal data of the end users. The situation may arise where the third party requires certain = information from the end user in order to provide a certain service, but where that information does not need to be passed on to other end users in order = for the service to operate sucessfully. In this case the end user has the = right to prevent their personal information being so transferred, and the = Service Provider needs the ability to ensure this, whilst also being able to = safely pass the information to other Service Providers to whom they may 'sub-contract' provision of aspects of the service. The above is completely technology neutral. ...Mark > -----Original Message----- > From: Christian Huitema [mailto:huitema@windows.microsoft.com] > Sent: 25 January 2002 17:07 > To: Watson, Mark [MDN05:EP10:EXCH]; Juha Heinanen; Dean Willis > Cc: sipping@ietf.org > Subject: RE: [Sipping] calling number blocking at sip/isup gateway >=20 >=20 > Judging me na=EFve, Mark Watson wrote: >=20 > > I have to say I am amazed too. But by the idea that the privacy=20 > > regulations we see in the PSTN/ISDN may not apply in the=20 > SIP world. These=20 > > stem from technology-neutral Data Protection legislation=20 > which applies as > much to SIP as to ISUP or TUP. >=20 > In fact, Mark's message contains a number of hidden=20 > assumptions that are well worth debunking. >=20 > A first assumption is that there is a "SIP world". With all=20 > due respect to this working group, this is by no means=20 > obvious. SIP is an Internet protocol that, in its simplest=20 > form, can be used directly between two Internet hosts,=20 > providing host names are registered in the DNS. The world=20 > there is the Internet, not SIP. >=20 > A second assumption is that the technology neutral doctrine=20 > applies, i.e. if I perform operation X over the Internet,=20 > then surely I am subject to the same rules as if I performed=20 > operation X over the telephone network. There are two=20 > problems with that rule, the fact that technology actually=20 > matters to legislation, and then the fact that that the=20 > regulation is generally tied to some form of public=20 > licensing, which actually varies with the technology. >=20 > If you thing that technology does not apply, then please=20 > consider the different regulations of postal services,=20 > telephony, broadcast TV, cable TV, press, and book=20 > publishing. SIP is in some aspects close to ISUP, since you=20 > can indeed gateway a phone call to the Internet using SIP;=20 > SIP is also in many ways closely related to e-mail, since a=20 > SIP header is pretty darn close to a mail header. Which=20 > regulation should we apply? That of telephone companies that=20 > are requested to provide law enforcement with facilities such=20 > as pen register or intercept, or that of postal services that=20 > definitely do not keep track of who sends mail to whom, and=20 > in which intercept is only practiced by dictatorships or=20 > war-time exceptions? >=20 > The laws regulating phone services are aimed at telephone=20 > service providers, not users. They do not apply, for example,=20 > to citizen band radios or private phone services. Assuming=20 > that the same laws apply is also assuming that the same=20 > structure applies, i.e. that the SIP calls are only possible=20 > through some kind of "SIP service" that would be provided by=20 > modern days equivalent of phone companies. This is not the=20 > way the Internet is organized today: SIP or VoIP packets are=20 > just some packets among many that are facilitated by the=20 > packet level service provided by an ISP. If a regulation=20 > targeted the third party, then it could indeed target those=20 > ASP that specifically set up to organize SIP proxies, but it=20 > would also have to target the generic ISPs that just carry IP=20 > traffic. This is going to be an interesting debate. >=20 > If we want to make anonymous SIP calls today, we must either=20 > use an "anonymizer" service, or use UA and network tricks.=20 > The cypherpunk and anon.fi experience leaves me quite=20 > skeptical with anonymizer relays: they tend to be abuse, or=20 > to become a magnet for law enforcement. If we really want=20 > anonymity, we should start at the UA, and allow it to make up=20 > some form of phony "From" line. We must also make sure that=20 > we use an anonymous IP address, possibly using the privacy=20 > extensions to IPv6. >=20 > -- Christian Huitema >=20 ------_=_NextPart_001_01C1B8BA.0FD88340 Content-Type: text/html; charset="iso-8859-1" Content-Transfer-Encoding: quoted-printable RE: [Sipping] calling number blocking at sip/isup = gateway

Christian,

Data Protection legislation is not = technology-specific or tied to licencing. If you do operation X with = the Personal Data of an individual, and X falls within the scope of = Data Protection, it does not matter whether you do this in the phone = network, the Internet or on a beach in Bali, you are subject to the = same rules.

Now, in Europe, and perhaps elsewhere, our = legislators have tried to be helpful by describing in detail the effect = of these principles in certain sectors of industry - for example in the = European Data Protection in Telecommunications Directive (97/66/EC: http://europa.eu.int/eur-lex/en/lif/dat/1997/en_397L00= 66.html).

The applicability of such statements to services = offered over the Internet can be debated but this does not affect the = requirement to uphold the principles enshrined in the primary Data = Protection legisation. If you want to get privacy right, you need to = understand these.

Of course, what information two end-users do/don't = exchange is entirely their own business, but where  third parties = (dare I say 'Service Providers') are involved they have certain = obligations with respect to personal data of the end users.

The situation may arise where the third party = requires certain information from the end user in order to provide a = certain service, but where that information does not need to be passed = on to other end users in order for the service to operate sucessfully. = In this case the end user has the right to prevent their personal = information being so transferred, and the Service Provider needs the = ability to ensure this, whilst also being able to safely pass the = information to other Service Providers to whom they may 'sub-contract' = provision of aspects of the service.

The above is completely technology neutral.

...Mark

> -----Original Message-----
> From: Christian Huitema [mailto:huitema@windows.mic= rosoft.com]
> Sent: 25 January 2002 17:07
> To: Watson, Mark [MDN05:EP10:EXCH]; Juha = Heinanen; Dean Willis
> Cc: sipping@ietf.org
> Subject: RE: [Sipping] calling number blocking = at sip/isup gateway
>
>
> Judging me na=EFve, Mark Watson wrote:
>
> > I have to say I am amazed too. But by the = idea that the privacy
> > regulations we see in the PSTN/ISDN may = not apply in the
> SIP world. These
> > stem from technology-neutral Data = Protection legislation
> which applies as > much to SIP as to ISUP or = TUP.
>
> In fact, Mark's message contains a number of = hidden
> assumptions that are well worth = debunking.
>
> A first assumption is that there is a "SIP = world". With all
> due respect to this working group, this is by = no means
> obvious. SIP is an Internet protocol that, in = its simplest
> form, can be used directly between two Internet = hosts,
> providing host names are registered in the DNS. = The world
> there is the Internet, not SIP.
>
> A second assumption is that the technology = neutral doctrine
> applies, i.e. if I perform operation X over the = Internet,
> then surely I am subject to the same rules as = if I performed
> operation X over the telephone network. There = are two
> problems with that rule, the fact that = technology actually
> matters to legislation, and then the fact that = that the
> regulation is generally tied to some form of = public
> licensing, which actually varies with the = technology.
>
> If you thing that technology does not apply, = then please
> consider the different regulations of postal = services,
> telephony, broadcast TV, cable TV, press, and = book
> publishing. SIP is in some aspects close to = ISUP, since you
> can indeed gateway a phone call to the Internet = using SIP;
> SIP is also in many ways closely related to = e-mail, since a
> SIP header is pretty darn close to a mail = header. Which
> regulation should we apply? That of telephone = companies that
> are requested to provide law enforcement with = facilities such
> as pen register or intercept, or that of postal = services that
> definitely do not keep track of who sends mail = to whom, and
> in which intercept is only practiced by = dictatorships or
> war-time exceptions?
>
> The laws regulating phone services are aimed at = telephone
> service providers, not users. They do not = apply, for example,
> to citizen band radios or private phone = services. Assuming
> that the same laws apply is also assuming that = the same
> structure applies, i.e. that the SIP calls are = only possible
> through some kind of "SIP service" = that would be provided by
> modern days equivalent of phone companies. This = is not the
> way the Internet is organized today: SIP or = VoIP packets are
> just some packets among many that are = facilitated by the
> packet level service provided by an ISP. If a = regulation
> targeted the third party, then it could indeed = target those
> ASP that specifically set up to organize SIP = proxies, but it
> would also have to target the generic ISPs that = just carry IP
> traffic. This is going to be an interesting = debate.
>
> If we want to make anonymous SIP calls today, = we must either
> use an "anonymizer" service, or use = UA and network tricks.
> The cypherpunk and anon.fi experience leaves me = quite
> skeptical with anonymizer relays: they tend to = be abuse, or
> to become a magnet for law enforcement. If we = really want
> anonymity, we should start at the UA, and allow = it to make up
> some form of phony "From" line. We = must also make sure that
> we use an anonymous IP address, possibly using = the privacy
> extensions to IPv6.
>
> -- Christian Huitema
>

------_=_NextPart_001_01C1B8BA.0FD88340-- _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Mon Feb 18 15:49:36 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id PAA17830 for ; Mon, 18 Feb 2002 15:49:32 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id PAA11472 for sipping-archive@odin.ietf.org; Mon, 18 Feb 2002 15:49:35 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id PAA10117; Mon, 18 Feb 2002 15:24:27 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id PAA09925 for ; Mon, 18 Feb 2002 15:24:12 -0500 (EST) Received: from zrtps0kn.nortelnetworks.com (zrtps0kn.nortelnetworks.com [47.140.192.55]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id PAA16802 for ; Mon, 18 Feb 2002 15:24:07 -0500 (EST) Received: from zctfs063.nortelnetworks.com (zctfs063.europe.nortel.com [47.164.128.120]) by zrtps0kn.nortelnetworks.com (Switch-2.2.0/Switch-2.2.0) with ESMTP id g1IKNDG24834; Mon, 18 Feb 2002 15:23:14 -0500 (EST) Received: from znsgs01r.europe.nortel.com (znsgs01r.europe.nortel.com [47.137.129.92]) by zctfs063.nortelnetworks.com (Switch-2.2.0/Switch-2.2.0) with ESMTP id g1IKNAT25911; Mon, 18 Feb 2002 21:23:11 +0100 (MET) Received: from zwcwc012.europe.nortel.com (zwcwc012.europe.nortel.com [47.73.112.187]) by znsgs01r.europe.nortel.com (Switch-2.2.0/Switch-2.2.0) with ESMTP id g1IKMjT09184; Mon, 18 Feb 2002 20:22:45 GMT Received: by zwcwc012.europe.nortel.com with Internet Mail Service (5.5.2653.19) id <1XS8FBZ5>; Mon, 18 Feb 2002 20:23:08 -0000 Message-ID: From: "Mark Watson" To: Dean Willis , Mpierce1@aol.com, jon.peterson@neustar.biz, sipping@ietf.org Date: Mon, 18 Feb 2002 20:23:05 -0000 MIME-Version: 1.0 X-Mailer: Internet Mail Service (5.5.2653.19) Content-Type: multipart/alternative; boundary="----_=_NextPart_001_01C1B8BA.126F5AC0" Subject: [Sipping] RE: [Sip] calling number blocking at sip/isup gateway Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org This message is in MIME format. Since your mail reader does not understand this format, some or all of this message may not be legible. ------_=_NextPart_001_01C1B8BA.126F5AC0 Content-Type: text/plain; charset="iso-8859-1" Dean, Your example below needs only a small extension to illustrate Mike's point and show why something like Remote-Party-ID as defined in the privacy draft is needed, and how it works despite the lack of a signature. Suppose I too run an IP telephony service and want to offer my subscribers the ability to call to the PSTN. I don't have a PSTN gateway of my own, so I make an agreement with you to allow my subscribers to use your gateway. You will need some way of authorising calls from my subscribers to be sure that you'll get revenue for the call. Let's assume you have some way of authenticating messages from my outgoing proxy, which all my subscribers use - you can't authenticate my subscribers directly as you don't know who they are. You will bill me for calls made by my subscribers, but you will need some kind of identity or token to tell me which subscriber it was, so that I can bill them. If the user has a PSTN number it would be desireable to to send it out to the PSTN (this would make the regulators happy from a Malicious Call perspective), but only if you can be sure it is valid for that user. Some PSTN networks support a second, user-chosen, CLI. For example Call Centers might provide their Freephone number. If the user wants such an identity, it would be good to know that too. This information could be carried in a Remote-Party-ID - there is no need to sign it because you are already sure that the message came from me, and we have a written agreement which can specify that the Remote-Party-ID values will be accurate for the subscriber who is calling - my proxy makes sure of that because it does authenticate the actual caller, and knows their PSTN number(s). Because I have to live up to my agreement with you, I ensure that my outgoing proxy does not route such messages via other entities which could modify them - I send them straight to you or via other Service Providers with whom I have similar agreements. Personally, I have trouble understanding how any system for a PSTN break-out service such as this could operate without such written agreements. If we pass this information between us, we do need to carry an indication of the subscribers privacy preference with respect to that. Regards...Mark > -----Original Message----- > From: Dean Willis [mailto:dean.willis@softarmor.com] > > I run a real IP telephone service. Unknown parties may place > calls over > the Internet to my subscribers. My subscribers's profiles may > result in > forwarding those calls to a PSTN gateway. PSTN charges > originating from > this operation are charged back to the subscriber (the called party), > not the calling party with whom I have no relationship > outside the SIP > used to set up the call. > > It really doesn't matter what the calling party put in their From: > field, I still need to complete the call. Now, if they were > kind enough > to put in a phone number, I can map that into the IAM. But if they > didn't, I just can't. Now, there's an interesting question on > "authenticating" the phone number that they might have included., but > that's a different problem. > > Even more interesting, I'm going to include any referenceable number > from the INVITE in the IAM, regardless of any privacy indicator > suggested by the SIP signaling, because that referencable > number would > have been included in the SIP signal delivered to my > subscriber anyhow > (The caller DID send it to him), and my job as a gateway > oeprator is to > get as much of the SIP experience as possible into the call being > forwarded to the PSTN. > > So, here we have a real phone service which conflicts with the > behavioral requirements suggested by mpierce's post, as well as a > whole can of privacy and transitive trust worms. > > -- > Dean > > _______________________________________________ > Sip mailing list https://www1.ietf.org/mailman/listinfo/sip > This list is for NEW development of the core SIP Protocol > Use sip-implementors@cs.columbia.edu for questions on current sip > Use sipping@ietf.org for new developments on the application of sip > ------_=_NextPart_001_01C1B8BA.126F5AC0 Content-Type: text/html; charset="iso-8859-1" Content-Transfer-Encoding: quoted-printable RE: [Sip] calling number blocking at sip/isup gateway

Dean,

Your example below needs only a small extension to = illustrate Mike's point and show why something like Remote-Party-ID as = defined in the privacy draft is needed, and how it works despite the = lack of a signature.

Suppose I too run an IP telephony service and want to = offer my subscribers the ability to call to the PSTN. I don't have a = PSTN gateway of my own, so I make an agreement with you to allow my = subscribers to use your gateway.

You will need some way of authorising calls from my = subscribers to be sure that you'll get revenue for the call. Let's = assume you have some way of authenticating messages from my outgoing = proxy, which all my subscribers use - you can't authenticate my = subscribers directly as you don't know who they are.

You will bill me for calls made by my subscribers, = but you will need some kind of identity or token to tell me which = subscriber it was, so that I can bill them.

If the user has a PSTN number it would be desireable = to to send it out to the PSTN (this would make the regulators happy = from a Malicious Call perspective), but only if you can be sure it is = valid for that user.

Some PSTN networks support a second, user-chosen, = CLI. For example Call Centers might provide their Freephone number. If = the user wants such an identity, it would be good to know that = too.

This information could be carried in a = Remote-Party-ID - there is no need to sign it because you are already = sure that the message came from me, and we have a written agreement = which can specify that the Remote-Party-ID values will be accurate for = the subscriber who is calling - my proxy makes sure of that because it = does authenticate the actual caller, and knows their PSTN = number(s).

Because I have to live up to my agreement with you, I = ensure that my outgoing proxy does not route such messages via other = entities which could modify them - I send them straight to you or via = other Service Providers with whom I have similar agreements.

Personally, I have trouble understanding how any = system for a PSTN break-out service such as this could operate without = such written agreements.

If we pass this information between us, we do need to = carry an indication of the subscribers privacy preference with respect = to that.

Regards...Mark

> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.c= om]

<snip>
>
> I run a real IP telephone service. Unknown = parties may place
> calls over
> the Internet to my subscribers. My = subscribers's profiles may
> result in
> forwarding those calls to a PSTN gateway. PSTN = charges
> originating from
> this operation are charged back to the = subscriber (the called party),
> not the calling party with whom I have no = relationship
> outside the SIP
> used to set up the call.
>
> It really doesn't matter what the calling party = put in their From:
> field, I still need to complete the call. Now, = if they were
> kind enough
> to put in a phone number, I can map that into = the IAM. But if they
> didn't, I just can't. Now, there's an = interesting question on
> "authenticating" the phone number = that they might have included., but
> that's a different problem.
>
> Even more interesting, I'm going to include any = referenceable number
> from the INVITE in the IAM, regardless of any = privacy indicator
> suggested by the SIP signaling, because that = referencable
> number would
> have been included in the SIP signal delivered = to my
> subscriber anyhow
> (The caller DID send it to him), and my job as = a gateway
> oeprator is to
> get as much of the SIP experience as possible = into the call being
> forwarded to the PSTN.
>
> So, here we have a real phone service which = conflicts with the
> behavioral requirements suggested by = mpierce's  post, as well as a
> whole can of privacy and transitive trust = worms.
>
> --
> Dean
>
> = _______________________________________________
> Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
> This list is for NEW development of the core = SIP Protocol
> Use sip-implementors@cs.columbia.edu for = questions on current sip
> Use sipping@ietf.org for new developments on = the application of sip
>

------_=_NextPart_001_01C1B8BA.126F5AC0-- _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@ns.ietf.org Mon Feb 18 18:25:28 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id SAA22915 for ; Mon, 18 Feb 2002 18:25:27 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id SAA19226 for sipping-archive@odin.ietf.org; Mon, 18 Feb 2002 18:25:32 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id SAA18313; Mon, 18 Feb 2002 18:01:42 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id RAA17541 for ; Mon, 18 Feb 2002 17:51:22 -0500 (EST) Received: from mailgate.pit.comms.marconi.com (mailgate.pit.comms.marconi.com [169.144.68.6]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id RAA21725; Mon, 18 Feb 2002 17:51:17 -0500 (EST) Received: from mailman.pit.comms.marconi.com (mailman.pit.comms.marconi.com [169.144.2.12]) by mailgate.pit.comms.marconi.com (8.9.3/8.9.3) with ESMTP id RAA16244; Mon, 18 Feb 2002 17:50:37 -0500 (EST) Received: from whq-msgrtr-01.pit.comms.marconi.com (whq-msgrtr-01.pit.comms.marconi.com [169.144.2.221]) by mailman.pit.comms.marconi.com (8.9.3/8.9.3) with ESMTP id RAA29346; Mon, 18 Feb 2002 17:50:38 -0500 (EST) Received: by whq-msgrtr-01.pit.comms.marconi.com with Internet Mail Service (5.5.2650.21) id ; Mon, 18 Feb 2002 17:50:37 -0500 Message-ID: <313680C9A886D511A06000204840E1CF57CA54@whq-msgusr-02.pit.comms.marconi.com> From: "Rosen, Brian" To: "'Richard Shockey'" , Rohan Mahy Cc: =?ISO-8859-1?Q?=22=27Patrik_F=E4ltstr=F6m=27=22?= , "Peterson, Jon" , "'Kevin McCandless'" , "'enum@ietf.org'" , sipping@ietf.org, "Rosen, Brian" , dean.willis@softarmor.com Date: Mon, 18 Feb 2002 17:50:36 -0500 MIME-Version: 1.0 X-Mailer: Internet Mail Service (5.5.2650.21) Content-Type: text/plain; charset="ISO-8859-1" Content-Transfer-Encoding: 8bit X-MIME-Autoconverted: from quoted-printable to 8bit by optimus.ietf.org id RAA17544 Subject: [Sipping] RE: [Enum] Using ENUM for SIP Applications Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 8bit I concur. Brian > -----Original Message----- > From: Richard Shockey [mailto:rshockey@ix.netcom.com] > Sent: Sunday, February 17, 2002 10:29 PM > To: Rohan Mahy > Cc: "'Patrik F鋖tstr鰉'"; Peterson, Jon; 'Kevin McCandless'; > 'enum@ietf.org'; sipping@ietf.org; brian.rosen@marconi.com; > dean.willis@softarmor.com > Subject: RE: [Enum] Using ENUM for SIP Applications > > > At 01:11 PM 2/17/2002 -0800, Rohan Mahy wrote: > >discussing this in ENUM is fine by me. we can make an > announcement during > >our first session. > > I'll be happy to make the announcement myself it you think that will > help...I do want this as a joint effort. > > > >thanks, > >-rohan > > Thanks again Rohan ..I thought that's how you might feel > ..though I want to > make sure that Brian and Dean concur. > > I'm deeply sympathetic to the work load the SIPPING WG has > but I want to > make sure your WG and ours is properly informed at all stages of this > discussion and that when this document or one that is similar > to it comes > to last call it will be a joint action between both WG's. > > A URL for this Internet-Draft is: > http://www.ietf.org/internet-drafts/draft-ietf-sipping-e164-01.txt > > BTW for those SIPPING folks who are scratching their heads > wondering what > this thread is all about . > > I'm the co-chair of the ENUM WG and if you need more info go to : > > http://www.ietf.org/html.charters/enum-charter.html > > > >On Sun, 17 Feb 2002, Richard Shockey wrote: > > > > > A > > > >[snip] > > > > > > > > > > Should for example the flag in the NAPTR record > differ between a > > SIP URI > > > > > used for presence and one used for gaming and one > used for voice calls? > > > > > > > > > > That MUST be resolved, I would say, before the registration > > document for > > > > > SIP-related flags is finished. > > > > > > > >I agree totally. We present quite a few arguments for > one perspective on > > > >this matter in our document. While we don't go ahead and > register any > > > >SIP-related flags in the IANA considerations section, > future versions of > > > >this document are probably as good a place to register > 'sip+E2U' as any. > > > > > > > >I'd be interested to know whether or not there is > basically consensus, > > based > > > >on the arguments that we've raised in sipping-e164-01, > that we need no > > ENUM > > > >service fields for SIP other than 'sip+E2U'. Maybe if we > have time this is > > > >something we could discuss in MN, at the discretion of > the chairs. > > > > > > I think you can assume that there will be time on the > ENUM WG agenda in MN > > > for this document. There is obviously a lot of interest > in this. We have a > > > 2 1/2 hour slot .. and SIPPING has a lot on its plate. > > > > > > Now we have not cross posted these messages to SIPPING ...to avoid > > > overburdening their totally overburdened list ... but we > need to keep them > > > in the loop and make sure they are aware of this > discussion and that we > > > probably will discuss the document in MN and that all SIPPING WG > > > participants with an interest here might want to attend. > > > > > > I'm assuming this plan is OK with the SIPPING Chairs.... > > > > > > > > >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> > Richard Shockey, Senior Manager, Strategic Technology Initiatives > NeuStar Inc. > 45980 Center Oak Plaza Bldg 8 Sterling, VA 20166 > 1120 Vermont Ave NW Suite 400 Washington DC 20005 > Voice +1 571.434.5651 Cell : +1 314.503.0640, Fax: +1 815.333.1237 > or > > > > <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< > This e-mail and any attachments are confidential. If you are not the intended recipient, please notify us immediately by reply e-mail and then delete this message from your system. Do not copy this e-mail or any attachment, use the contents for any purposes, or disclose the contents to any other person: to do so could be a breach of confidence. _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@ns.ietf.org Mon Feb 18 18:30:56 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id SAA23155 for ; Mon, 18 Feb 2002 18:30:56 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id SAA19772 for sipping-archive@odin.ietf.org; Mon, 18 Feb 2002 18:31:01 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id SAA18846; Mon, 18 Feb 2002 18:12:35 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id SAA18820 for ; Mon, 18 Feb 2002 18:12:33 -0500 (EST) Received: from bdsl.66.12.12.130.gte.net (bdsl.66.12.12.130.gte.net [66.12.12.130]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id SAA22615 for ; Mon, 18 Feb 2002 18:12:28 -0500 (EST) Received: from DWILLISS1 (bdsl.greycouncil.com [127.0.0.1]) by bdsl.66.12.12.130.gte.net (8.11.6/8.11.6) with SMTP id g1INBmB22579; Mon, 18 Feb 2002 17:11:48 -0600 Message-ID: <019f01c1b8d1$a05c3e40$f1029bd4@DWILLISS1> From: "Dean Willis" To: "Mark Watson" , , , References: Subject: Re: [Sipping] RE: [Sip] calling number blocking at sip/isup gateway Date: Mon, 18 Feb 2002 17:11:37 -0600 MIME-Version: 1.0 Content-Type: text/plain; charset="iso-8859-1" Content-Transfer-Encoding: 7bit X-Priority: 3 X-MSMail-Priority: Normal X-Mailer: Microsoft Outlook Express 6.00.2600.0000 X-MIMEOLE: Produced By Microsoft MimeOLE V6.00.2600.0000 Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit Mark said: > Your example below needs only a small extension to illustrate Mike's point > and show why something like Remote-Party-ID as defined in the privacy draft > is needed, and how it works despite the lack of a signature. Yep. And what happens when my misconfigured proxy forwards the call without paying attention to the Privacy header, because for some reason it's forwarding to the outside world instead of a PSTN gateway under my control? We get a privacy leak. But other than that, I'd say you've precisely described a "sunny day" use case for RPID. It still has a lot of security issues based on transitive trust, which make it a lot more palatable in a two-party case than in three or more party transitivity. I believe we're approaching a general consensus that the Privacy draft provides a reasonable approach to solving a specific class of problem wherein interoperator-transitive trust is couple to transport-layer message protection. That's not a general privacy mechanism, but it does seem to have enough constituency to justify publication as long as it has a REALLY serious applicability statement explaining the particular requirements and environments for which it is useful. How does that sound? -- Dean _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@ns.ietf.org Tue Feb 19 03:45:16 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id DAA12600 for ; Tue, 19 Feb 2002 03:45:15 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id DAA20989 for sipping-archive@odin.ietf.org; Tue, 19 Feb 2002 03:45:17 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id DAA20078; Tue, 19 Feb 2002 03:25:57 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id DAA20051 for ; Tue, 19 Feb 2002 03:25:55 -0500 (EST) Received: from gorilla.mchh.siemens.de (gorilla.mchh.siemens.de [194.138.158.18]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id DAA12418 for ; Tue, 19 Feb 2002 03:25:52 -0500 (EST) Received: from moody.mchh.siemens.de (mail2.mchh.siemens.de [194.138.158.226]) by gorilla.mchh.siemens.de (8.9.3/8.9.3) with ESMTP id JAA18444; Tue, 19 Feb 2002 09:25:49 +0100 (MET) Received: from icn.siemens.de ([139.21.148.41]) by moody.mchh.siemens.de (8.9.1/8.9.1) with ESMTP id JAA15642; Tue, 19 Feb 2002 09:25:46 +0100 (MET) Message-ID: <3C720C0F.2B29810E@icn.siemens.de> Date: Tue, 19 Feb 2002 09:25:51 +0100 From: Peter =?iso-8859-1?Q?P=E4ppinghaus?= Organization: Siemens AG X-Mailer: Mozilla 4.73 [en] (Windows NT 5.0; U) X-Accept-Language: en,de MIME-Version: 1.0 To: Jonathan Rosenberg CC: Sipping@ietf.org Subject: Re: Questions on To tags (was Re: [Sipping] Comments on draft-ietf-sip-rfcbis-05.pdf) References: <3C6BD959.93ADF9B7@icn.siemens.de> <3C6F3CAD.275800F4@dynamicsoft.com> Content-Type: text/plain; charset=iso-8859-1 Content-Transfer-Encoding: 8bit Content-Transfer-Encoding: 8bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 8bit I realize the incredible workload you are taking, and I really don't want to abuse your precious time. But could you help me to understand why I am mistaken in my inline comment below? Sorry for this. Peter P. Jonathan Rosenberg wrote: > > Peter P鋚pinghaus wrote: > > In bis-07, section 23.3, lines 4896-4897, now in fact say: > > "Similarly, two INVITEs for different calls will have different From > > tags." > > > > This really surprises me for two reasons. > > > > Firstly, if the From tag has to be unique across all calls, then it > > uniquely identifies a call, so what would the Call-ID be needed for? > > If we were able to start SIP from scratch, I would probably agree that > the Call-ID and from tag are redundant. But, we can't. So, the real > question is this - given we have a unique call-id, and that people have > been putting them in that way, and relying on their uniqueness, is there > also a need for the from tag/to tag to be unique? > > There, there answer is in fact yes, but for subtle reasons. Certainly, > the to tag in a response needs to be unique. What you really, really > need is for both participants in the dialog to contribute unique > identifiers, such that the combination of the two are the dialog ID. According to my understanding, what you really, really need is for both participants in the dialog to contribute unique identifiers, such that the combination of the two *together with the Call-ID* are the dialog ID. And if all of the following hold: 1) the From tag uniquely identifies a UAC (unique in space and role as a request-issuing entity), 2) the To tag uniquely identifies a UAS (unique in space and role as a response-issuing entity), 3) the Call-ID used for the request is unique "over time", i.e. as a minimal requirement: the same UAC ("same" as identified by the From tag) never issues another dialog-initiating request with the same Call-ID value, then the dialog ID consisting of Call-ID, From tag, and To tag, will uniquely identify a dialog, won't it? > So, the to tag has to be unique, OK. Now, in requests in the reverse > direction that the dialog was established, the To and From field values > are reversed. This means that what was in the From in the initial > request, is now in the To, and vice a versa. This means that elements > which might like to take advantage of the fact that the to tag is > unique, really cannot, since it wouldn't be in the reverse direction > unless the from tag in the original request was also unique. And thus > the requirement for unique from tags as well. > > So, simply put, the from tag is unique to provide symmetry wiht the to > tag, which must be unique. > > -Jonathan R. -- Dr. Peter P鋚pinghaus Siemens AG | Phone: +49 89 - 722 40065 ICM N PG U ID A1 | Fax: +49 89 - 722 58726 Hofmannstr. 51 | Visitors: Building 1702 / Room 530 D - 81359 M黱chen | Email: peter.paeppinghaus@icn.siemens.de _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Tue Feb 19 04:36:54 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id EAA13119 for ; Tue, 19 Feb 2002 04:36:54 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id EAA23379 for sipping-archive@odin.ietf.org; Tue, 19 Feb 2002 04:36:56 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id EAA22059; Tue, 19 Feb 2002 04:12:07 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id EAA22028 for ; Tue, 19 Feb 2002 04:12:04 -0500 (EST) Received: from sj-msg-core-2.cisco.com (sj-msg-core-2.cisco.com [171.69.24.11]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id EAA12891 for ; Tue, 19 Feb 2002 04:12:02 -0500 (EST) Received: from imop.cisco.com (imop.cisco.com [171.69.11.44]) by sj-msg-core-2.cisco.com (8.11.3/8.9.1) with ESMTP id g1J9BYE03657 for ; Tue, 19 Feb 2002 01:11:34 -0800 (PST) Received: from localhost (ssh-sj1.cisco.com [171.68.225.134]) by imop.cisco.com (Mirapoint) with ESMTP id ACM66651; Tue, 19 Feb 2002 01:11:33 -0800 (PST) Date: Tue, 19 Feb 2002 01:10:10 -0800 (Pacific Standard Time) From: Rohan Mahy To: sipping@ietf.org Message-ID: X-X-Sender: rmahy@imop.cisco.com MIME-Version: 1.0 Content-Type: TEXT/PLAIN; charset=US-ASCII Subject: [Sipping] SIPPING Agenda Slots in Minneapolis Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Hi, I'm collecting requests for discussion time at the SIPPING meeting. If you want discussion time for your topic, it must have received discussion on the list, and you must explain what kind of discussion this is (requirements, open issues, etc.). We will not have any tutorials this time. Please submit your requests to me. thanks, -rohan Rohan Mahy SIPPING co-chair rohan@cisco.com _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Tue Feb 19 08:25:43 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id IAA17564 for ; Tue, 19 Feb 2002 08:25:43 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id IAA04479 for sipping-archive@odin.ietf.org; Tue, 19 Feb 2002 08:25:44 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id IAA03749; Tue, 19 Feb 2002 08:03:36 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id IAA03709 for ; Tue, 19 Feb 2002 08:03:33 -0500 (EST) Received: from revere.sonusnet.com (mail.sonusnet.com [208.45.178.33]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id IAA16866 for ; Tue, 19 Feb 2002 08:03:31 -0500 (EST) Received: from sonusdc3.sonusnet.com (sonusdc3 [10.128.32.53]) by revere.sonusnet.com (Switch-2.1.0/Switch-2.1.0) with ESMTP id g1JD2YQ20298; Tue, 19 Feb 2002 08:02:34 -0500 (EST) Received: from matt.verizon.net (MATT [10.128.82.106]) by sonusdc3.sonusnet.com with SMTP (Microsoft Exchange Internet Mail Service Version 5.5.2653.13) id 1Z0J9RJ2; Tue, 19 Feb 2002 08:02:35 -0500 Message-Id: <5.1.0.14.2.20020219025040.00a51280@mail.verizon.net> X-Sender: res06gzk@mail.verizon.net (Unverified) X-Mailer: QUALCOMM Windows Eudora Version 5.1 Date: Tue, 19 Feb 2002 02:52:07 -0800 To: "Dean Willis" , "Mark Watson" , , , From: Matt Holdrege Subject: Re: [Sipping] RE: [Sip] calling number blocking at sip/isup gateway In-Reply-To: <019f01c1b8d1$a05c3e40$f1029bd4@DWILLISS1> References: Mime-Version: 1.0 Content-Type: text/plain; charset="us-ascii"; format=flowed Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org At 03:11 PM 2/18/2002, Dean Willis wrote: >I believe we're approaching a general consensus that the Privacy draft >provides a reasonable approach to solving a specific class of problem >wherein interoperator-transitive trust is couple to transport-layer message >protection. That's not a general privacy mechanism, but it does seem to have >enough constituency to justify publication as long as it has a REALLY >serious applicability statement explaining the particular requirements and >environments for which it is useful. > >How does that sound? Good compromise! That's why we pay you the big bucks to be a chair. _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Tue Feb 19 09:45:37 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id JAA21322 for ; Tue, 19 Feb 2002 09:45:36 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id JAA09546 for sipping-archive@odin.ietf.org; Tue, 19 Feb 2002 09:45:39 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id JAA07534; Tue, 19 Feb 2002 09:21:01 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id JAA07511 for ; Tue, 19 Feb 2002 09:20:59 -0500 (EST) Received: from zrtps0kn.nortelnetworks.com (zrtps0kn.nortelnetworks.com [47.140.192.55]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id JAA20199 for ; Tue, 19 Feb 2002 09:20:56 -0500 (EST) Received: from zctfs063.nortelnetworks.com (zctfs063.europe.nortel.com [47.164.128.120]) by zrtps0kn.nortelnetworks.com (Switch-2.2.0/Switch-2.2.0) with ESMTP id g1JEKCr24646; Tue, 19 Feb 2002 09:20:13 -0500 (EST) Received: from znsgs01r.europe.nortel.com (znsgs01r.europe.nortel.com [47.137.129.92]) by zctfs063.nortelnetworks.com (Switch-2.2.0/Switch-2.2.0) with ESMTP id g1JEKAj01401; Tue, 19 Feb 2002 15:20:10 +0100 (MET) Received: from zwcwc012.europe.nortel.com (zwcwc012.europe.nortel.com [47.73.112.187]) by znsgs01r.europe.nortel.com (Switch-2.2.0/Switch-2.2.0) with ESMTP id g1JEJhg14613; Tue, 19 Feb 2002 14:19:43 GMT Received: by zwcwc012.europe.nortel.com with Internet Mail Service (5.5.2653.19) id <1XS8GDSR>; Tue, 19 Feb 2002 14:20:10 -0000 Message-ID: From: "Mark Watson" To: Dean Willis , Mpierce1@aol.com, jon.peterson@neustar.biz, sipping@ietf.org Subject: RE: [Sipping] RE: [Sip] calling number blocking at sip/isup gatew ay Date: Tue, 19 Feb 2002 14:20:02 -0000 MIME-Version: 1.0 X-Mailer: Internet Mail Service (5.5.2653.19) Content-Type: multipart/alternative; boundary="----_=_NextPart_001_01C1B950.851CBD60" Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org This message is in MIME format. Since your mail reader does not understand this format, some or all of this message may not be legible. ------_=_NextPart_001_01C1B950.851CBD60 Content-Type: text/plain; charset="iso-8859-1" Dean, Sounds good to me. I expect that whatever we go forward with now for the limited context described below, can be extended later to meet the more general requirements. ...Mark > -----Original Message----- > From: Dean Willis [mailto:dean.willis@softarmor.com] > Sent: 18 February 2002 23:12 > To: Watson, Mark [MDN05:EP10:EXCH]; Mpierce1@aol.com; > jon.peterson@neustar.biz; sipping@ietf.org > Subject: Re: [Sipping] RE: [Sip] calling number blocking at sip/isup > gateway > > > Mark said: > > > Your example below needs only a small extension to > illustrate Mike's point > > and show why something like Remote-Party-ID as defined in > the privacy > draft > > is needed, and how it works despite the lack of a signature. > > Yep. And what happens when my misconfigured proxy forwards > the call without > paying attention to the Privacy header, because for some reason it's > forwarding to the outside world instead of a PSTN gateway > under my control? > We get a privacy leak. > > But other than that, I'd say you've precisely described a > "sunny day" use > case for RPID. > > It still has a lot of security issues based on transitive > trust, which make > it a lot more palatable in a two-party case than in three or > more party > transitivity. > > I believe we're approaching a general consensus that the Privacy draft > provides a reasonable approach to solving a specific class of problem > wherein interoperator-transitive trust is couple to > transport-layer message > protection. That's not a general privacy mechanism, but it > does seem to have > enough constituency to justify publication as long as it has a REALLY > serious applicability statement explaining the particular > requirements and > environments for which it is useful. > > How does that sound? > > -- > Dean > > ------_=_NextPart_001_01C1B950.851CBD60 Content-Type: text/html; charset="iso-8859-1" Content-Transfer-Encoding: quoted-printable RE: [Sipping] RE: [Sip] calling number blocking at sip/isup = gateway

Dean,

Sounds good to me.

I expect that whatever we go forward with now for the = limited context described below, can be extended later to meet the more = general requirements.

...Mark

> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.c= om]
> Sent: 18 February 2002 23:12
> To: Watson, Mark [MDN05:EP10:EXCH]; = Mpierce1@aol.com;
> jon.peterson@neustar.biz; = sipping@ietf.org
> Subject: Re: [Sipping] RE: [Sip] calling number = blocking at sip/isup
> gateway
>
>
> Mark said:
>
> > Your example below needs only a small = extension to
> illustrate Mike's point
> > and show why something like = Remote-Party-ID as defined in
> the privacy
> draft
> > is needed, and how it works despite the = lack of a signature.
>
> Yep. And what happens when my misconfigured = proxy forwards
> the call without
> paying attention to the Privacy header, because = for some reason it's
> forwarding to the outside world instead of a = PSTN gateway
> under my control?
> We get a privacy leak.
>
> But other than that, I'd say you've precisely = described a
> "sunny day" use
> case for RPID.
>
> It still has a lot of security issues based on = transitive
> trust, which make
> it a lot more palatable in a two-party case = than in three or
> more party
> transitivity.
>
> I believe we're approaching a general consensus = that the Privacy draft
> provides a reasonable approach to solving a = specific class of problem
> wherein interoperator-transitive trust is = couple to
> transport-layer message
> protection. That's not a general privacy = mechanism, but it
> does seem to have
> enough constituency to justify publication as = long as it has a REALLY
> serious applicability statement explaining the = particular
> requirements and
> environments for which it is useful.
>
> How does that sound?
>
> --
> Dean
>
>

------_=_NextPart_001_01C1B950.851CBD60-- _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Tue Feb 19 11:47:13 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id LAA26864 for ; Tue, 19 Feb 2002 11:47:13 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id LAA17784 for sipping-archive@odin.ietf.org; Tue, 19 Feb 2002 11:47:15 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id LAA16169; Tue, 19 Feb 2002 11:26:25 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id LAA16143 for ; Tue, 19 Feb 2002 11:26:23 -0500 (EST) Received: from nisser.cisco.com (nisser.cisco.com [171.71.176.85]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id LAA26244 for ; Tue, 19 Feb 2002 11:26:20 -0500 (EST) Received: from cisco.com (ssh-sj1.cisco.com [171.68.225.134]) by nisser.cisco.com (8.8.6 (PHNE_14041)/CISCO.SERVER.1.2) with ESMTP id IAA08389; Tue, 19 Feb 2002 08:25:17 -0800 (PST) Message-ID: <3C727C72.C3EEC300@cisco.com> Date: Tue, 19 Feb 2002 11:25:22 -0500 From: Flemming Andreasen X-Mailer: Mozilla 4.79 [en]C-CCK-MCD (Windows NT 5.0; U) X-Accept-Language: en MIME-Version: 1.0 To: Matt Holdrege CC: Dean Willis , Mark Watson , Mpierce1@aol.com, jon.peterson@neustar.biz, sipping@ietf.org Subject: Re: [Sipping] RE: [Sip] calling number blocking at sip/isupgateway References: <5.1.0.14.2.20020219025040.00a51280@mail.verizon.net> Content-Type: text/plain; charset=us-ascii Content-Transfer-Encoding: 7bit Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit I agree - sounds like a good and timely way forward. -- Flemming Matt Holdrege wrote: > At 03:11 PM 2/18/2002, Dean Willis wrote: > >I believe we're approaching a general consensus that the Privacy draft > >provides a reasonable approach to solving a specific class of problem > >wherein interoperator-transitive trust is couple to transport-layer message > >protection. That's not a general privacy mechanism, but it does seem to have > >enough constituency to justify publication as long as it has a REALLY > >serious applicability statement explaining the particular requirements and > >environments for which it is useful. > > > >How does that sound? > > Good compromise! That's why we pay you the big bucks to be a chair. > > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP > Use sip-implementors@cs.columbia.edu for questions on current sip > Use sip@ietf.org for new developments of core SIP -- Flemming Andreasen Cisco Systems _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Tue Feb 19 16:01:45 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id QAA05492 for ; Tue, 19 Feb 2002 16:01:44 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id QAA02982 for sipping-archive@odin.ietf.org; Tue, 19 Feb 2002 16:01:47 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id PAA00917; Tue, 19 Feb 2002 15:29:25 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id PAA00886 for ; Tue, 19 Feb 2002 15:29:23 -0500 (EST) Received: from zrtps0kn.nortelnetworks.com (zrtps0kn.nortelnetworks.com [47.140.192.55]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id PAA04464 for ; Tue, 19 Feb 2002 15:29:20 -0500 (EST) Received: from zctfs063.nortelnetworks.com (zctfs063.europe.nortel.com [47.164.128.120]) by zrtps0kn.nortelnetworks.com (Switch-2.2.0/Switch-2.2.0) with ESMTP id g1JKSor13705 for ; Tue, 19 Feb 2002 15:28:50 -0500 (EST) Received: from znsgs01r.europe.nortel.com (znsgs01r.europe.nortel.com [47.137.129.92]) by zctfs063.nortelnetworks.com (Switch-2.2.0/Switch-2.2.0) with ESMTP id g1JKSmj19087 for ; Tue, 19 Feb 2002 21:28:48 +0100 (MET) Received: from zwcwc012.europe.nortel.com (zwcwc012.europe.nortel.com [47.73.112.187]) by znsgs01r.europe.nortel.com (Switch-2.2.0/Switch-2.2.0) with ESMTP id g1JKSNg01681 for ; Tue, 19 Feb 2002 20:28:24 GMT Received: by zwcwc012.europe.nortel.com with Internet Mail Service (5.5.2653.19) id <1XS8GVQP>; Tue, 19 Feb 2002 20:28:50 -0000 Message-ID: From: "Mark Watson" To: sipping@ietf.org Date: Tue, 19 Feb 2002 20:28:46 -0000 MIME-Version: 1.0 X-Mailer: Internet Mail Service (5.5.2653.19) Content-Type: multipart/alternative; boundary="----_=_NextPart_001_01C1B984.08405520" Subject: [Sipping] Privacy comment (was Re: calling number blocking at sip/isup gate way) Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org This message is in MIME format. Since your mail reader does not understand this format, some or all of this message may not be legible. ------_=_NextPart_001_01C1B984.08405520 Content-Type: text/plain; charset="iso-8859-1" As presently described in privacy-03, the Remote-Party-ID header is only ever inserted by proxies, based on the proxy's knowledge of the identity of the UA (which it determines from some authentication mechanism out of scope of the privacy draft). Privacy-02 allowed the UA to insert Remote-Party-ID too. I think this should be reintroduced for the *specific case* of a UA with multiple identities, to allow the UA to choose which identity it wishes to be known by. The Proxy would still be responsible for ensuring that this identity was valid for the user and would still use some other authentication scheme to authenticate the user. The proxy would need knowledge of which identities were valid for which users. I believe this is a requirement for 3GPP, where a UA can have multiple 'public identities' and chooses which of these it wishes to be known by for each call. The proxy authenticates the UA by some other means and then checks the subscription data to ensure it is really allowed to use the public Identity it has included. I've suggested this a few times, and there has not been much comment for or against. It would be good if we could get a resolution before the next version of privacy is published. So, comments ? Regards...Mark > -----Original Message----- > From: Dean Willis [mailto:dean.willis@softarmor.com] > Sent: 18 February 2002 23:12 > To: Watson, Mark [MDN05:EP10:EXCH]; Mpierce1@aol.com; > jon.peterson@neustar.biz; sipping@ietf.org > Subject: Re: [Sipping] RE: [Sip] calling number blocking at sip/isup > gateway > > > Mark said: > > > Your example below needs only a small extension to > illustrate Mike's point > > and show why something like Remote-Party-ID as defined in > the privacy > draft > > is needed, and how it works despite the lack of a signature. > > Yep. And what happens when my misconfigured proxy forwards > the call without > paying attention to the Privacy header, because for some reason it's > forwarding to the outside world instead of a PSTN gateway > under my control? > We get a privacy leak. > > But other than that, I'd say you've precisely described a > "sunny day" use > case for RPID. > > It still has a lot of security issues based on transitive > trust, which make > it a lot more palatable in a two-party case than in three or > more party > transitivity. > > I believe we're approaching a general consensus that the Privacy draft > provides a reasonable approach to solving a specific class of problem > wherein interoperator-transitive trust is couple to > transport-layer message > protection. That's not a general privacy mechanism, but it > does seem to have > enough constituency to justify publication as long as it has a REALLY > serious applicability statement explaining the particular > requirements and > environments for which it is useful. > > How does that sound? > > -- > Dean > > ------_=_NextPart_001_01C1B984.08405520 Content-Type: text/html; charset="iso-8859-1" Content-Transfer-Encoding: quoted-printable Privacy comment (was Re: calling number blocking at sip/isup = gateway)

As presently described in privacy-03, the = Remote-Party-ID header is only ever inserted by proxies, based on the = proxy's knowledge of the identity of the UA (which it determines from = some authentication mechanism out of scope of the privacy = draft).

Privacy-02 allowed the UA to insert Remote-Party-ID = too. I think this should be reintroduced for the *specific case* of a = UA with multiple identities, to allow the UA to choose which identity = it wishes to be known by. The Proxy would still be responsible for = ensuring that this identity was valid for the user and would still use = some other authentication scheme to authenticate the user. The proxy = would need knowledge of which identities were valid for which = users.

I believe this is a requirement for 3GPP, where a UA = can have multiple 'public identities' and chooses which of these it = wishes to be known by for each call. The proxy authenticates the UA by = some other means and then checks the subscription data to ensure it is = really allowed to use the public Identity it has included.

I've suggested this a few times, and there has not = been much comment for or against. It would be good if we could get a = resolution before the next version of privacy is published.

So, comments ?

Regards...Mark





> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.c= om]
> Sent: 18 February 2002 23:12
> To: Watson, Mark [MDN05:EP10:EXCH]; = Mpierce1@aol.com;
> jon.peterson@neustar.biz; = sipping@ietf.org
> Subject: Re: [Sipping] RE: [Sip] calling number = blocking at sip/isup
> gateway
>
>
> Mark said:
>
> > Your example below needs only a small = extension to
> illustrate Mike's point
> > and show why something like = Remote-Party-ID as defined in
> the privacy
> draft
> > is needed, and how it works despite the = lack of a signature.
>
> Yep. And what happens when my misconfigured = proxy forwards
> the call without
> paying attention to the Privacy header, because = for some reason it's
> forwarding to the outside world instead of a = PSTN gateway
> under my control?
> We get a privacy leak.
>
> But other than that, I'd say you've precisely = described a
> "sunny day" use
> case for RPID.
>
> It still has a lot of security issues based on = transitive
> trust, which make
> it a lot more palatable in a two-party case = than in three or
> more party
> transitivity.
>
> I believe we're approaching a general consensus = that the Privacy draft
> provides a reasonable approach to solving a = specific class of problem
> wherein interoperator-transitive trust is = couple to
> transport-layer message
> protection. That's not a general privacy = mechanism, but it
> does seem to have
> enough constituency to justify publication as = long as it has a REALLY
> serious applicability statement explaining the = particular
> requirements and
> environments for which it is useful.
>
> How does that sound?
>
> --
> Dean
>
>

------_=_NextPart_001_01C1B984.08405520-- _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Tue Feb 19 19:55:55 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id TAA09561 for ; Tue, 19 Feb 2002 19:55:54 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id TAA12652 for sipping-archive@odin.ietf.org; Tue, 19 Feb 2002 19:55:55 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id TAA11662; Tue, 19 Feb 2002 19:27:47 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id TAA11631 for ; Tue, 19 Feb 2002 19:27:45 -0500 (EST) Received: from bdsl.66.12.12.130.gte.net (bdsl.66.12.12.130.gte.net [66.12.12.130]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id TAA09285 for ; Tue, 19 Feb 2002 19:27:44 -0500 (EST) Received: from DWILLISS1 (bdsl.greycouncil.com [127.0.0.1]) by bdsl.66.12.12.130.gte.net (8.11.6/8.11.6) with SMTP id g1K0R8B31181; Tue, 19 Feb 2002 18:27:09 -0600 Message-ID: <045101c1b9a5$4f5c3570$f1029bd4@DWILLISS1> From: "Dean Willis" To: "Mark Watson" , References: Subject: Re: [Sipping] Privacy comment (was Re: calling number blocking at sip/isup gate way) Date: Tue, 19 Feb 2002 18:26:55 -0600 MIME-Version: 1.0 Content-Type: text/plain; charset="iso-8859-1" Content-Transfer-Encoding: 7bit X-Priority: 3 X-MSMail-Priority: Normal X-Mailer: Microsoft Outlook Express 6.00.2600.0000 X-MIMEOLE: Produced By Microsoft MimeOLE V6.00.2600.0000 Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit I concur that your proposal would add to the utility of RPID. -- Dean ----- Original Message ----- From: "Mark Watson" To: Sent: Tuesday, February 19, 2002 2:28 PM Subject: [Sipping] Privacy comment (was Re: calling number blocking at sip/isup gate way) > As presently described in privacy-03, the Remote-Party-ID header is only > ever inserted by proxies, based on the proxy's knowledge of the identity of > the UA (which it determines from some authentication mechanism out of scope > of the privacy draft). > > Privacy-02 allowed the UA to insert Remote-Party-ID too. I think this should > be reintroduced for the *specific case* of a UA with multiple identities, to > allow the UA to choose which identity it wishes to be known by. The Proxy > would still be responsible for ensuring that this identity was valid for the > user and would still use some other authentication scheme to authenticate > the user. The proxy would need knowledge of which identities were valid for > which users. > > I believe this is a requirement for 3GPP, where a UA can have multiple > 'public identities' and chooses which of these it wishes to be known by for > each call. The proxy authenticates the UA by some other means and then > checks the subscription data to ensure it is really allowed to use the > public Identity it has included. > > I've suggested this a few times, and there has not been much comment for or > against. It would be good if we could get a resolution before the next > version of privacy is published. > > So, comments ? > > Regards...Mark > > > > > > > -----Original Message----- > > From: Dean Willis [mailto:dean.willis@softarmor.com] > > Sent: 18 February 2002 23:12 > > To: Watson, Mark [MDN05:EP10:EXCH]; Mpierce1@aol.com; > > jon.peterson@neustar.biz; sipping@ietf.org > > Subject: Re: [Sipping] RE: [Sip] calling number blocking at sip/isup > > gateway > > > > > > Mark said: > > > > > Your example below needs only a small extension to > > illustrate Mike's point > > > and show why something like Remote-Party-ID as defined in > > the privacy > > draft > > > is needed, and how it works despite the lack of a signature. > > > > Yep. And what happens when my misconfigured proxy forwards > > the call without > > paying attention to the Privacy header, because for some reason it's > > forwarding to the outside world instead of a PSTN gateway > > under my control? > > We get a privacy leak. > > > > But other than that, I'd say you've precisely described a > > "sunny day" use > > case for RPID. > > > > It still has a lot of security issues based on transitive > > trust, which make > > it a lot more palatable in a two-party case than in three or > > more party > > transitivity. > > > > I believe we're approaching a general consensus that the Privacy draft > > provides a reasonable approach to solving a specific class of problem > > wherein interoperator-transitive trust is couple to > > transport-layer message > > protection. That's not a general privacy mechanism, but it > > does seem to have > > enough constituency to justify publication as long as it has a REALLY > > serious applicability statement explaining the particular > > requirements and > > environments for which it is useful. > > > > How does that sound? > > > > -- > > Dean > > > > > _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Tue Feb 19 21:12:33 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id VAA10363 for ; Tue, 19 Feb 2002 21:12:32 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id VAA15890 for sipping-archive@odin.ietf.org; Tue, 19 Feb 2002 21:12:34 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id UAA14707; Tue, 19 Feb 2002 20:42:05 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id UAA14659 for ; Tue, 19 Feb 2002 20:42:02 -0500 (EST) Received: from sj-msg-core-1.cisco.com (sj-msg-core-1.cisco.com [171.71.163.11]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id UAA10074; Tue, 19 Feb 2002 20:42:00 -0500 (EST) Received: from zipper.cisco.com (zipper.cisco.com [171.69.25.142]) by sj-msg-core-1.cisco.com (8.11.3/8.9.1) with ESMTP id g1K1fVh17588; Tue, 19 Feb 2002 17:41:31 -0800 (PST) Date: Tue, 19 Feb 2002 17:41:31 -0800 (PST) From: Rohan Mahy To: , Message-ID: MIME-Version: 1.0 Content-Type: TEXT/PLAIN; charset=US-ASCII Subject: [Sipping] Draft Naming Conventions Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Hi Folks, In the next few days, many of you will submit Internet Drafts. Please remember that per the Transport Area SIP change policy all new proposed extensions, requirements, and usage drafts should go through the SIPPING WG. This means that if you have a proposed extension that is not already a SIP WG Milestone, it needs to be submitted with a file name like this draft-lastname-sipping-description-00.txt for example: draft-mahy-sipping-gaming-reqs-00.txt would be an appropriately named draft for internet gaming requirements on the SIP protocol. Please note that merely submitting an internet draft does not result in discussion time for your draft at a meeting. Meeting time in the SIP WG is for chartered work only, and meeting time in SIPPING is for discussion of work already discussed on the mailing list (and must be requested). Requests for Agenda time in SIPPING are being tracked on the SIPPING supplemental website: http://www.softarmor.com/sipping/meets/ietf53/requests.html many thanks, -rohan _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Wed Feb 20 05:15:32 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id FAA26300 for ; Wed, 20 Feb 2002 05:15:31 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id FAA14800 for sipping-archive@odin.ietf.org; Wed, 20 Feb 2002 05:15:34 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id EAA13642; Wed, 20 Feb 2002 04:55:10 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id EAA13613 for ; Wed, 20 Feb 2002 04:55:08 -0500 (EST) Received: from albatross.wise.edt.ericsson.se (albatross-ext.wise.edt.ericsson.se [193.180.251.36]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id EAA26111 for ; Wed, 20 Feb 2002 04:55:04 -0500 (EST) Received: from mbb5.ericsson.se (mbb5.ericsson.se [136.225.151.210]) by albatross.wise.edt.ericsson.se (8.12.1/8.12.1/WIREfire-1.4) with ESMTP id g1K9sZhQ027382 for ; Wed, 20 Feb 2002 10:54:36 +0100 (MET) Received: from esealnt462.al.sw.ericsson.se (ESEALNT462.al.sw.ericsson.se [153.88.251.62]) by mbb1.ericsson.se (PMDF V5.2-29 #39352) with SMTP id <0GRT00PPPQ3U4J@mbb1.ericsson.se> for sipping@ietf.org; Wed, 20 Feb 2002 09:55:06 +0100 (MET) Received: FROM esealnt400.al.sw.ericsson.se BY esealnt462.al.sw.ericsson.se ; Wed Feb 20 09:53:48 2002 +0100 Received: by esealnt400 with Internet Mail Service (5.5.2653.19) id ; Wed, 20 Feb 2002 09:52:59 +0100 Date: Wed, 20 Feb 2002 09:53:35 +0100 From: "Ilkka Uusitalo (LMF)" To: "'sipping@ietf.org'" Message-id: <0154633DAF7BD4119C360002A513AA0301326530@efijont102> MIME-version: 1.0 X-Mailer: Internet Mail Service (5.5.2653.19) Content-type: text/plain; charset=iso-8859-1 Subject: [Sipping] Three new requirement drafts for SIP available Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Hi, We have submitted three requirement drafts. These are "3GPP Requirements for SIP Authentication", "Requirements for SIP Security Mechanism Agreement" and "Requirements for Delegation of Message Protection for SIP". Most of the requirements have been discussed also in "3GPP requirements on SIP" (draft-garcia-sipping-3gpp-regs-02.txt). We separated the requirements into these shorter drafts because they are not all 3GPP specific. Draft "3GPP Requirements for SIP Authentication" discusses 3GPP requirements, the other two are more general. Also, we think they can be moved forward more easily as individual small requirements. The drafts can be found at http://search.ietf.org/internet-drafts/draft-uusitalo-sipping-authentication-00.txt http://search.ietf.org/internet-drafts/draft-uusitalo-sipping-algorithm-agreement-00.txt http://search.ietf.org/internet-drafts/draft-uusitalo-sipping-delegation-00.txt We hope to have consensus on the requirements soon, so comments are wellcome. We have already started working on the solution drafts and hope to submit them to SIP mailing list soon. Regards, Ilkka Uusitalo --------------------- Ilkka Uusitalo Research Scientist Oy L M Ericsson Ab Tel: +358 9 299 5141 Fax: +358 8 551 5105 Mobile: +358 40 724 5404 Address: Tutkijantie 2 C, FIN-90570 OULU Email: ilkka.uusitalo@ericsson.fi _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Wed Feb 20 12:14:50 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id MAA07320 for ; Wed, 20 Feb 2002 12:14:50 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id MAA07692 for sipping-archive@odin.ietf.org; Wed, 20 Feb 2002 12:14:53 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id LAA05029; Wed, 20 Feb 2002 11:57:02 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id LAA04998 for ; Wed, 20 Feb 2002 11:57:00 -0500 (EST) Received: from smtp2.libero.it (smtp2.libero.it [193.70.192.52]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id LAA06459 for ; Wed, 20 Feb 2002 11:56:56 -0500 (EST) Received: from libero.it (193.70.192.62) by smtp2.libero.it (6.0.040) id 3C58404400E03772 for sipping@ietf.org; Wed, 20 Feb 2002 17:56:25 +0100 Date: Wed, 20 Feb 2002 17:56:25 +0100 Message-Id: MIME-Version: 1.0 Content-Type: text/plain; charset=iso-8859-1 From: "=?utf-8?Q?vitiellopas@libero.it?=" To: sipping@ietf.org X-XaM3-API-Version: 2.5 X-type: 0 X-SenderIP: 193.205.164.86 Content-Transfer-Encoding: 8bit X-MIME-Autoconverted: from base64 to 8bit by optimus.ietf.org id LAA04999 Subject: [Sipping] =?iso-8859-1?Q?Forking_proxy?= Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 8bit Hi, I'm implementing a Stateful Proxy and I have a question. Forking proxies can perform parallel or sequential searches depending on their configurations. In a parallel search, when proxy receives a final response 2xx from one branch, must it send a CANCEL message on all other branch? If it receives more 2xx response, what does the proxy do? Regards _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Thu Feb 21 16:15:39 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id QAA04532 for ; Thu, 21 Feb 2002 16:15:39 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id QAA26728 for sipping-archive@odin.ietf.org; Thu, 21 Feb 2002 16:15:43 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id QAA26278; Thu, 21 Feb 2002 16:04:35 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id QAA26249 for ; Thu, 21 Feb 2002 16:04:33 -0500 (EST) Received: from ondar.cablelabs.com (ondar.cablelabs.com [192.160.73.61]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id QAA04163 for ; Thu, 21 Feb 2002 16:04:27 -0500 (EST) Received: from srvmail.cablelabs.com (srvmail.cablelabs.com [10.5.0.15]) by ondar.cablelabs.com (8.10.1/8.10.1) with ESMTP id g1LL3eb20608; Thu, 21 Feb 2002 14:03:40 -0700 (MST) Received: by srvmail.cablelabs.com with Internet Mail Service (5.5.2653.19) id <1FRA1HBN>; Thu, 21 Feb 2002 14:03:40 -0700 Message-ID: <4DF5E8A771ECD21187020008C7B1C5AF02E9D914@srvmail.cablelabs.com> From: Jean-Francois Mule To: "'Bill Sulzen'" , sipping@ietf.org Date: Thu, 21 Feb 2002 14:03:30 -0700 MIME-Version: 1.0 X-Mailer: Internet Mail Service (5.5.2653.19) Content-Type: text/plain; charset="iso-8859-1" X-Approved: ondar Subject: [Sipping] RE: Concerning draft-mule-sip-t38callflows-02 T.38 rejection - 60 6 and/or 488 Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Bill Sulzen wrote: > Section 6.2 of draft-mule-sip-t38callflows-02 describes how > a UA would > reject a T.38 INVITE. It indicates that the emitting gateway > rejects the re-INVITE with a "606 Not Acceptable" response. > Would a "488 Not Acceptable Media" not be at least equally, > if not more appropriate? after reading the bis-08 sections on 488/606, I see no objections for treating 488 and 606 as *equally* appropriate. The update will go into the upcoming draft-ietf-sipping-realtimefax-00. any other comments? Jean-Francois. > -----Original Message----- > From: Jean-Francois Mule > Sent: Thursday, February 21, 2002 1:56 PM > To: 'Bill Sulzen'; 'sip-implementors@cs.columbia.edu' > Subject: RE: Concerning draft-mule-sip-t38callflows-02 T.38 > rejection - > 606 and/or 488 > > > Bill & all: > > since draft-mule-sip-t38callflows-02 will morph into > draft-ietf-sipping-realtimefax-00, this discussion is now > happening on sipping. sorry for the delay in responsing to > your comment. > > Jean-Francois > > > >From: "Bill Sulzen" > > >To: > > >Cc: "Jean-Francois Mule" > > >Subject: Concerning draft-mule-sip-t38callflows-02 T.38 > > rejection - 606 or > > >488? > > >Date: Fri, 25 Jan 2002 15:01:55 -0500 > > > > > >Section 6.2 of draft-mule-sip-t38callflows-02 describes how > > a UA would > > >reject a T.38 INVITE. It indicates that the emitting gateway > > rejects the > > >re-INVITE with a "606 Not Acceptable" response. > > > > > >Would a "488 Not Acceptable Media" not be at least equally, > > if not more > > >appropriate? > > > > > >Thanks, > > >Bill Sulzen > _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Thu Feb 21 16:33:58 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id QAA05008 for ; Thu, 21 Feb 2002 16:33:58 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id QAA27933 for sipping-archive@odin.ietf.org; Thu, 21 Feb 2002 16:34:02 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id QAA27117; Thu, 21 Feb 2002 16:22:55 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id QAA27086 for ; Thu, 21 Feb 2002 16:22:53 -0500 (EST) Received: from ondar.cablelabs.com (ondar.cablelabs.com [192.160.73.61]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id QAA04736 for ; Thu, 21 Feb 2002 16:22:48 -0500 (EST) Received: from srvmail.cablelabs.com (srvmail.cablelabs.com [10.5.0.15]) by ondar.cablelabs.com (8.10.1/8.10.1) with ESMTP id g1LLLpb20970; Thu, 21 Feb 2002 14:21:51 -0700 (MST) Received: by srvmail.cablelabs.com with Internet Mail Service (5.5.2653.19) id <1FRA1HDN>; Thu, 21 Feb 2002 14:21:51 -0700 Message-ID: <4DF5E8A771ECD21187020008C7B1C5AF02E9D916@srvmail.cablelabs.com> From: Jean-Francois Mule To: "'Dean Willis'" , Mark Watson , Mpierce1@aol.com, jon.peterson@neustar.biz, sipping@ietf.org Subject: RE: [Sipping] RE: [Sip] calling number blocking at sip/isup gatew ay Date: Thu, 21 Feb 2002 14:21:41 -0700 MIME-Version: 1.0 X-Mailer: Internet Mail Service (5.5.2653.19) Content-Type: text/plain; charset="iso-8859-1" X-Approved: ondar Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org I agree with the approach too. Jean-Francois. Dean wrote: > I believe we're approaching a general consensus that the Privacy draft > provides a reasonable approach to solving a specific class of problem > wherein interoperator-transitive trust is couple to > transport-layer message > protection. That's not a general privacy mechanism, but it > does seem to have > enough constituency to justify publication as long as it has a REALLY > serious applicability statement explaining the particular > requirements and > environments for which it is useful. > > How does that sound? _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Fri Feb 22 10:22:31 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id KAA05536 for ; Fri, 22 Feb 2002 10:22:31 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id KAA28751 for sipping-archive@odin.ietf.org; Fri, 22 Feb 2002 10:22:35 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id KAA27925; Fri, 22 Feb 2002 10:03:41 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id KAA27866 for ; Fri, 22 Feb 2002 10:03:35 -0500 (EST) Received: from pmesmtp01.wcom.com (pmesmtp01.wcom.com [199.249.20.1]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id KAA04841; Fri, 22 Feb 2002 10:03:30 -0500 (EST) Received: from CONVERSION-DAEMON by firewall.wcom.com (PMDF V5.2-33 #47837) id <0GRX00F01WGZRX@firewall.wcom.com>; Fri, 22 Feb 2002 15:02:59 +0000 (GMT) Received: from pmismtp03.wcomnet.com ([166.38.62.38]) by firewall.wcom.com (PMDF V5.2-33 #47837) with ESMTP id <0GRX00F81WGY3S@firewall.wcom.com>; Fri, 22 Feb 2002 15:02:59 +0000 (GMT) Received: from pmismtp03.wcomnet.com by pmismtp03.wcomnet.com (PMDF V5.2-33 #42258) with SMTP id <0GRX00D01WGOPA@pmismtp03.wcomnet.com>; Fri, 22 Feb 2002 15:02:58 +0000 (GMT) Received: from hsinnreich2 ([166.35.224.250]) by pmismtp03.wcomnet.com (PMDF V5.2-33 #42258) with ESMTP id <0GRX00C7ZWGDBT@pmismtp03.wcomnet.com>; Fri, 22 Feb 2002 15:02:38 +0000 (GMT) Date: Fri, 22 Feb 2002 09:02:37 -0600 From: Henry Sinnreich To: sipping@ietf.org, sip@ietf.org Message-id: <000201c1bbb1$f7e0e0c0$fae023a6@hsinnreich2> Organization: WorldCom, Inc. MIME-version: 1.0 X-MIMEOLE: Produced By Microsoft MimeOLE V6.00.2600.0000 X-Mailer: Microsoft Outlook, Build 10.0.3416 Content-type: text/plain; charset=us-ascii Content-transfer-encoding: 7bit Importance: Normal X-Priority: 3 (Normal) X-MSMail-priority: Normal Content-Transfer-Encoding: 7bit Subject: [Sipping] Drafts on SIP Telephony Devices Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit Hi folks! Two related drafts on SIP devices have been submitted for the SIPPING and SIP WGs. They can be retrieved here until they will appear in the IETF archive. http://www.pingtel.com/docs/draft-sinnreich-sipping-device-requirements- 00.txt SIP Telephony Device Requirements Abstract The SIP Telephony Device Requirements in this document are aimed to enable users to procure SIP phones from various vendors and install them using a configuration process over the network, though still allowing manual configuration data input if desired. SIP telephony devices are also required to be updated up automatically, without the intervention of the end user. The requirements are aimed to allow service providers to support large numbers of SIP telephony devices from different vendors and SHOULD be used as a checklist for developers of SIP phones. SIP telephony devices may support basic telephony as well as advanced services such as PBX and Centrex-like telephony, third party call control, presence based services, roaming and other applications. The requirements aim to insure multi-vendor interoperability over the Internet and in private IP networks for such services. A Framework for SIP User Agent Configuration http://www.ietf.org/internet-drafts/draft-petrie-sip-config-framework-01 .txt Abstract This document defines the application of a set of protocols for configuring a SIP user agent. The SIP user agent must discover how and from where to retrieve its initial configuration and be notified of changes and updates which impact its configuration. The objective is to define a means for automatically configuring a user agent such that it can be functional without user or administrative intervention. The framework for discovery, delivery, notification and updates of user agent configuration is defined here. This framework is also intended to ease ongoing administration, configuration and upgrading of large scale deployments of SIP user agents. The contents and format of the configuration data to be defined is outside the scope of this document. It would be nice to discuss these two together. Comments and discussions are most welcome. Thanks, Henry Henry Sinnreich WorldCom 400 International Parkway Richardson, Texas 75081 USA _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Fri Feb 22 10:53:15 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id KAA07053 for ; Fri, 22 Feb 2002 10:53:15 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id KAA00939 for sipping-archive@odin.ietf.org; Fri, 22 Feb 2002 10:53:17 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id KAA29230; Fri, 22 Feb 2002 10:30:43 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id KAA29169 for ; Fri, 22 Feb 2002 10:30:38 -0500 (EST) Received: from zctfs063.nortelnetworks.com (h90s131a237n47.user.nortelnetworks.com [47.237.131.90]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id KAA05991 for ; Fri, 22 Feb 2002 10:30:33 -0500 (EST) Received: from znsgs01r.europe.nortel.com (znsgs01r.europe.nortel.com [47.137.129.92]) by zctfs063.nortelnetworks.com (Switch-2.2.0/Switch-2.2.0) with ESMTP id g1MFU3x21038 for ; Fri, 22 Feb 2002 16:30:04 +0100 (MET) Received: from zwcwc012.europe.nortel.com (zwcwc012.europe.nortel.com [47.73.112.187]) by znsgs01r.europe.nortel.com (Switch-2.2.0/Switch-2.2.0) with ESMTP id g1MFTcj17251 for ; Fri, 22 Feb 2002 15:29:38 GMT Received: by zwcwc012.europe.nortel.com with Internet Mail Service (5.5.2653.19) id ; Fri, 22 Feb 2002 15:30:10 -0000 Message-ID: From: "Mark Watson" To: sipping@ietf.org Subject: RE: [Sipping] Privacy comment (was Re: calling number blocking at sip/isup gate way) Date: Fri, 22 Feb 2002 15:30:12 -0000 MIME-Version: 1.0 X-Mailer: Internet Mail Service (5.5.2653.19) Content-Type: multipart/alternative; boundary="----_=_NextPart_001_01C1BBB5.D191EA00" Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org This message is in MIME format. Since your mail reader does not understand this format, some or all of this message may not be legible. ------_=_NextPart_001_01C1BBB5.D191EA00 Content-Type: text/plain; charset="iso-8859-1" So, in the absence of further comments, could the editor add this to the next version of the privacy draft ? (on the silence=agree/don't care principle) ...Mark > -----Original Message----- > From: Dean Willis [mailto:dean.willis@softarmor.com] > Sent: 20 February 2002 00:27 > To: Watson, Mark [MDN05:EP10:EXCH]; sipping@ietf.org > Subject: Re: [Sipping] Privacy comment (was Re: calling > number blocking > at sip/isup gate way) > > > I concur that your proposal would add to the utility of RPID. > > -- > Dean > > ----- Original Message ----- > From: "Mark Watson" > To: > Sent: Tuesday, February 19, 2002 2:28 PM > Subject: [Sipping] Privacy comment (was Re: calling number blocking at > sip/isup gate way) > > > > As presently described in privacy-03, the Remote-Party-ID > header is only > > ever inserted by proxies, based on the proxy's knowledge of > the identity > of > > the UA (which it determines from some authentication > mechanism out of > scope > > of the privacy draft). > > > > Privacy-02 allowed the UA to insert Remote-Party-ID too. I > think this > should > > be reintroduced for the *specific case* of a UA with > multiple identities, > to > > allow the UA to choose which identity it wishes to be known > by. The Proxy > > would still be responsible for ensuring that this identity > was valid for > the > > user and would still use some other authentication scheme > to authenticate > > the user. The proxy would need knowledge of which > identities were valid > for > > which users. > > > > I believe this is a requirement for 3GPP, where a UA can > have multiple > > 'public identities' and chooses which of these it wishes to > be known by > for > > each call. The proxy authenticates the UA by some other > means and then > > checks the subscription data to ensure it is really allowed > to use the > > public Identity it has included. > > > > I've suggested this a few times, and there has not been > much comment for > or > > against. It would be good if we could get a resolution > before the next > > version of privacy is published. > > > > So, comments ? > > > > Regards...Mark > > > > > > > > > > > > > -----Original Message----- > > > From: Dean Willis [mailto:dean.willis@softarmor.com] > > > Sent: 18 February 2002 23:12 > > > To: Watson, Mark [MDN05:EP10:EXCH]; Mpierce1@aol.com; > > > jon.peterson@neustar.biz; sipping@ietf.org > > > Subject: Re: [Sipping] RE: [Sip] calling number blocking > at sip/isup > > > gateway > > > > > > > > > Mark said: > > > > > > > Your example below needs only a small extension to > > > illustrate Mike's point > > > > and show why something like Remote-Party-ID as defined in > > > the privacy > > > draft > > > > is needed, and how it works despite the lack of a signature. > > > > > > Yep. And what happens when my misconfigured proxy forwards > > > the call without > > > paying attention to the Privacy header, because for some > reason it's > > > forwarding to the outside world instead of a PSTN gateway > > > under my control? > > > We get a privacy leak. > > > > > > But other than that, I'd say you've precisely described a > > > "sunny day" use > > > case for RPID. > > > > > > It still has a lot of security issues based on transitive > > > trust, which make > > > it a lot more palatable in a two-party case than in three or > > > more party > > > transitivity. > > > > > > I believe we're approaching a general consensus that the > Privacy draft > > > provides a reasonable approach to solving a specific > class of problem > > > wherein interoperator-transitive trust is couple to > > > transport-layer message > > > protection. That's not a general privacy mechanism, but it > > > does seem to have > > > enough constituency to justify publication as long as it > has a REALLY > > > serious applicability statement explaining the particular > > > requirements and > > > environments for which it is useful. > > > > > > How does that sound? > > > > > > -- > > > Dean > > > > > > > > > > ------_=_NextPart_001_01C1BBB5.D191EA00 Content-Type: text/html; charset="iso-8859-1" Content-Transfer-Encoding: quoted-printable RE: [Sipping] Privacy comment (was Re: calling number blocking = at sip/isup gate way)

So, in the absence of further comments, could the = editor add this to the next version of the privacy draft ? (on the = silence=3Dagree/don't care principle)

...Mark

> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.c= om]
> Sent: 20 February 2002 00:27
> To: Watson, Mark [MDN05:EP10:EXCH]; = sipping@ietf.org
> Subject: Re: [Sipping] Privacy comment (was Re: = calling
> number blocking
> at sip/isup gate way)
>
>
> I concur that your proposal would add to the = utility of RPID.
>
> --
> Dean
>
> ----- Original Message -----
> From: "Mark Watson" = <mwatson@nortelnetworks.com>
> To: <sipping@ietf.org>
> Sent: Tuesday, February 19, 2002 2:28 PM
> Subject: [Sipping] Privacy comment (was Re: = calling number blocking at
> sip/isup gate way)
>
>
> > As presently described in privacy-03, the = Remote-Party-ID
> header is only
> > ever inserted by proxies, based on the = proxy's knowledge of
> the identity
> of
> > the UA (which it determines from some = authentication
> mechanism out of
> scope
> > of the privacy draft).
> >
> > Privacy-02 allowed the UA to insert = Remote-Party-ID too. I
> think this
> should
> > be reintroduced for the *specific case* of = a UA with
> multiple identities,
> to
> > allow the UA to choose which identity it = wishes to be known
> by. The Proxy
> > would still be responsible for ensuring = that this identity
> was valid for
> the
> > user and would still use some other = authentication scheme
> to authenticate
> > the user. The proxy would need knowledge = of which
> identities were valid
> for
> > which users.
> >
> > I believe this is a requirement for 3GPP, = where a UA can
> have multiple
> > 'public identities' and chooses which of = these it wishes to
> be known by
> for
> > each call. The proxy authenticates the UA = by some other
> means and then
> > checks the subscription data to ensure it = is really allowed
> to use the
> > public Identity it has included.
> >
> > I've suggested this a few times, and there = has not been
> much comment for
> or
> > against. It would be good if we could get = a resolution
> before the next
> > version of privacy is published.
> >
> > So, comments ?
> >
> > Regards...Mark
> >
> >
> >
> >
> >
> > > -----Original Message-----
> > > From: Dean Willis [mailto:dean.willis@softarmor.c= om]
> > > Sent: 18 February 2002 23:12
> > > To: Watson, Mark [MDN05:EP10:EXCH]; = Mpierce1@aol.com;
> > > jon.peterson@neustar.biz; = sipping@ietf.org
> > > Subject: Re: [Sipping] RE: [Sip] = calling number blocking
> at sip/isup
> > > gateway
> > >
> > >
> > > Mark said:
> > >
> > > > Your example below needs only a = small extension to
> > > illustrate Mike's point
> > > > and show why something like = Remote-Party-ID as defined in
> > > the privacy
> > > draft
> > > > is needed, and how it works = despite the lack of a signature.
> > >
> > > Yep. And what happens when my = misconfigured proxy forwards
> > > the call without
> > > paying attention to the Privacy = header, because for some
> reason it's
> > > forwarding to the outside world = instead of a PSTN gateway
> > > under my control?
> > > We get a privacy leak.
> > >
> > > But other than that, I'd say you've = precisely described a
> > > "sunny day" use
> > > case for RPID.
> > >
> > > It still has a lot of security issues = based on transitive
> > > trust, which make
> > > it a lot more palatable in a = two-party case than in three or
> > > more party
> > > transitivity.
> > >
> > > I believe we're approaching a general = consensus that the
> Privacy draft
> > > provides a reasonable approach to = solving a specific
> class of problem
> > > wherein interoperator-transitive = trust is couple to
> > > transport-layer message
> > > protection. That's not a general = privacy mechanism, but it
> > > does seem to have
> > > enough constituency to justify = publication as long as it
> has a REALLY
> > > serious applicability statement = explaining the particular
> > > requirements and
> > > environments for which it is = useful.
> > >
> > > How does that sound?
> > >
> > > --
> > > Dean
> > >
> > >
> >
>
>

------_=_NextPart_001_01C1BBB5.D191EA00-- _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@ns.ietf.org Fri Feb 22 13:42:42 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id NAA15947 for ; Fri, 22 Feb 2002 13:42:37 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id NAA14256 for sipping-archive@odin.ietf.org; Fri, 22 Feb 2002 13:42:39 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id NAA12594; Fri, 22 Feb 2002 13:27:16 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id NAA12562 for ; Fri, 22 Feb 2002 13:27:11 -0500 (EST) Received: from pine.neustar.com (pine.neustar.com [209.173.57.70]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id NAA15080 for ; Fri, 22 Feb 2002 13:27:09 -0500 (EST) Received: from chiimc01.il.neustar.com (dmz1.il.neustar.com [209.173.57.65]) by pine.neustar.com (8.11.0/8.11.0) with ESMTP id g1MIQZp25540; Fri, 22 Feb 2002 12:26:35 -0600 Received: by chiimc01.il.neustar.com with Internet Mail Service (5.5.2653.19) id ; Fri, 22 Feb 2002 12:26:30 -0600 Message-ID: <70565611B164D511957A001083FCDD56018700EF@VA02> From: "Peterson, Jon" To: "'Mark Watson'" , sipping@ietf.org Subject: RE: [Sipping] Privacy comment (was Re: calling number blocking at sip/isup gate way) Date: Fri, 22 Feb 2002 12:26:28 -0600 MIME-Version: 1.0 X-Mailer: Internet Mail Service (5.5.2653.19) Content-Type: multipart/alternative; boundary="----_=_NextPart_001_01C1BBCE.71C50940" Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org This message is in MIME format. Since your mail reader does not understand this format, some or all of this message may not be legible. ------_=_NextPart_001_01C1BBCE.71C50940 Content-Type: text/plain; charset="iso-8859-1" Actually, I have a few slight confusions about the text below (it seems to use UA and user interchangeably a bit, which is worrisome when we're talking about proving identity). One question would probably clear things up: If the proxy still needs to authenticate the user (through whichever mechanism) to ensure that the identity that has been asserted in the RPID is valid for the user in question, why does the user need to supply an RPID at all? Why doesn't the user just authenticate itself to the proxy (however that happens) and let the proxy insert the RPID corresponding to the identity that the user has proven? This works fine for cases in which a single UA is used by multiple users, or when a single user has multiple provable identities. It assumes that you have a one-to-one mapping between credentials and identities, but that's usually par for the course. Again, if the proxy server holds a mapping between credentials and valid identities itself, as you suggest below, it doesn't seem at all unreasonable to adopt this model. Jon Peterson NeuStar, Inc. -----Original Message----- From: Mark Watson [mailto:mwatson@nortelnetworks.com] Sent: Tuesday, February 19, 2002 12:29 PM To: sipping@ietf.org Subject: [Sipping] Privacy comment (was Re: calling number blocking at sip/isup gate way) As presently described in privacy-03, the Remote-Party-ID header is only ever inserted by proxies, based on the proxy's knowledge of the identity of the UA (which it determines from some authentication mechanism out of scope of the privacy draft). Privacy-02 allowed the UA to insert Remote-Party-ID too. I think this should be reintroduced for the *specific case* of a UA with multiple identities, to allow the UA to choose which identity it wishes to be known by. The Proxy would still be responsible for ensuring that this identity was valid for the user and would still use some other authentication scheme to authenticate the user. The proxy would need knowledge of which identities were valid for which users. I believe this is a requirement for 3GPP, where a UA can have multiple 'public identities' and chooses which of these it wishes to be known by for each call. The proxy authenticates the UA by some other means and then checks the subscription data to ensure it is really allowed to use the public Identity it has included. I've suggested this a few times, and there has not been much comment for or against. It would be good if we could get a resolution before the next version of privacy is published. So, comments ? Regards...Mark > -----Original Message----- > From: Dean Willis [ mailto:dean.willis@softarmor.com ] > Sent: 18 February 2002 23:12 > To: Watson, Mark [MDN05:EP10:EXCH]; Mpierce1@aol.com; > jon.peterson@neustar.biz; sipping@ietf.org > Subject: Re: [Sipping] RE: [Sip] calling number blocking at sip/isup > gateway > > > Mark said: > > > Your example below needs only a small extension to > illustrate Mike's point > > and show why something like Remote-Party-ID as defined in > the privacy > draft > > is needed, and how it works despite the lack of a signature. > > Yep. And what happens when my misconfigured proxy forwards > the call without > paying attention to the Privacy header, because for some reason it's > forwarding to the outside world instead of a PSTN gateway > under my control? > We get a privacy leak. > > But other than that, I'd say you've precisely described a > "sunny day" use > case for RPID. > > It still has a lot of security issues based on transitive > trust, which make > it a lot more palatable in a two-party case than in three or > more party > transitivity. > > I believe we're approaching a general consensus that the Privacy draft > provides a reasonable approach to solving a specific class of problem > wherein interoperator-transitive trust is couple to > transport-layer message > protection. That's not a general privacy mechanism, but it > does seem to have > enough constituency to justify publication as long as it has a REALLY > serious applicability statement explaining the particular > requirements and > environments for which it is useful. > > How does that sound? > > -- > Dean > > ------_=_NextPart_001_01C1BBCE.71C50940 Content-Type: text/html; charset="iso-8859-1" Privacy comment (was Re: calling number blocking at sip/isup gateway)
Actually, I have a few slight confusions about the text below (it seems to use UA and user interchangeably a bit, which is worrisome when we're talking about proving identity). One question would probably clear things up:
 
If the proxy still needs to authenticate the user (through whichever mechanism) to ensure that the identity that has been asserted in the RPID is valid for the user in question, why does the user need to supply an RPID at all? Why doesn't the user just authenticate itself to the proxy (however that happens) and let the proxy insert the RPID corresponding to the identity that the user has proven? This works fine for cases in which a single UA is used by multiple users, or when a single user has multiple provable identities. It assumes that you have a one-to-one mapping between credentials and identities, but that's usually par for the course.
 
Again, if the proxy server holds a mapping between credentials and valid identities itself, as you suggest below, it doesn't seem at all unreasonable to adopt this model.
 
Jon Peterson
NeuStar, Inc.
-----Original Message-----
From: Mark Watson [mailto:mwatson@nortelnetworks.com]
Sent: Tuesday, February 19, 2002 12:29 PM
To: sipping@ietf.org
Subject: [Sipping] Privacy comment (was Re: calling number blocking at sip/isup gate way)

As presently described in privacy-03, the Remote-Party-ID header is only ever inserted by proxies, based on the proxy's knowledge of the identity of the UA (which it determines from some authentication mechanism out of scope of the privacy draft).

Privacy-02 allowed the UA to insert Remote-Party-ID too. I think this should be reintroduced for the *specific case* of a UA with multiple identities, to allow the UA to choose which identity it wishes to be known by. The Proxy would still be responsible for ensuring that this identity was valid for the user and would still use some other authentication scheme to authenticate the user. The proxy would need knowledge of which identities were valid for which users.

I believe this is a requirement for 3GPP, where a UA can have multiple 'public identities' and chooses which of these it wishes to be known by for each call. The proxy authenticates the UA by some other means and then checks the subscription data to ensure it is really allowed to use the public Identity it has included.

I've suggested this a few times, and there has not been much comment for or against. It would be good if we could get a resolution before the next version of privacy is published.

So, comments ?

Regards...Mark





> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.com]
> Sent: 18 February 2002 23:12
> To: Watson, Mark [MDN05:EP10:EXCH]; Mpierce1@aol.com;
> jon.peterson@neustar.biz; sipping@ietf.org
> Subject: Re: [Sipping] RE: [Sip] calling number blocking at sip/isup
> gateway
>
>
> Mark said:
>
> > Your example below needs only a small extension to
> illustrate Mike's point
> > and show why something like Remote-Party-ID as defined in
> the privacy
> draft
> > is needed, and how it works despite the lack of a signature.
>
> Yep. And what happens when my misconfigured proxy forwards
> the call without
> paying attention to the Privacy header, because for some reason it's
> forwarding to the outside world instead of a PSTN gateway
> under my control?
> We get a privacy leak.
>
> But other than that, I'd say you've precisely described a
> "sunny day" use
> case for RPID.
>
> It still has a lot of security issues based on transitive
> trust, which make
> it a lot more palatable in a two-party case than in three or
> more party
> transitivity.
>
> I believe we're approaching a general consensus that the Privacy draft
> provides a reasonable approach to solving a specific class of problem
> wherein interoperator-transitive trust is couple to
> transport-layer message
> protection. That's not a general privacy mechanism, but it
> does seem to have
> enough constituency to justify publication as long as it has a REALLY
> serious applicability statement explaining the particular
> requirements and
> environments for which it is useful.
>
> How does that sound?
>
> --
> Dean
>
>

------_=_NextPart_001_01C1BBCE.71C50940-- _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@ns.ietf.org Fri Feb 22 14:14:36 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id OAA17459 for ; Fri, 22 Feb 2002 14:14:36 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id OAA17308 for sipping-archive@odin.ietf.org; Fri, 22 Feb 2002 14:14:38 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id OAA16422; Fri, 22 Feb 2002 14:02:43 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id OAA16382 for ; Fri, 22 Feb 2002 14:02:41 -0500 (EST) Received: from lohi.eng.song.fi (lohi.eng.song.fi [195.10.149.18]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id OAA16986 for ; Fri, 22 Feb 2002 14:02:37 -0500 (EST) From: jh@lohi.eng.song.fi Received: from harjus.eng.song.fi ([195.10.149.20]) by lohi.eng.song.fi with esmtp (Exim 3.34 #1 (Debian)) id 16eKxm-00009A-00; Fri, 22 Feb 2002 21:02:34 +0200 MIME-Version: 1.0 Content-Type: text/plain; charset=us-ascii Content-Transfer-Encoding: 7bit Message-ID: <15478.38346.582278.804557@harjus.eng.song.fi> Date: Fri, 22 Feb 2002 21:02:34 +0200 To: "Peterson, Jon" Cc: "'Mark Watson'" , sipping@ietf.org Subject: RE: [Sipping] Privacy comment (was Re: calling number blocking at sip/isup gate way) In-Reply-To: <70565611B164D511957A001083FCDD56018700EF@VA02> References: <70565611B164D511957A001083FCDD56018700EF@VA02> X-Mailer: VM 7.01 under Emacs 21.1.1 Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit Peterson, Jon writes: > Why doesn't the user just authenticate itself to the proxy (however > that happens) and let the proxy insert the RPID corresponding to the > identity that the user has proven? this should be allowed, -- juha _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@ns.ietf.org Fri Feb 22 14:59:16 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id OAA19649 for ; Fri, 22 Feb 2002 14:59:14 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id OAA21548 for sipping-archive@odin.ietf.org; Fri, 22 Feb 2002 14:59:17 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id OAA20391; Fri, 22 Feb 2002 14:44:47 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id OAA20346 for ; Fri, 22 Feb 2002 14:44:44 -0500 (EST) Received: from zrc2s0jx.nortelnetworks.com (zrc2s0jx.nortelnetworks.com [47.103.122.112]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id OAA18853 for ; Fri, 22 Feb 2002 14:44:40 -0500 (EST) Received: from zrc2c011.us.nortel.com (zrc2c011.us.nortel.com [47.103.120.51]) by zrc2s0jx.nortelnetworks.com (Switch-2.2.0/Switch-2.2.0) with ESMTP id g1MJiBF17501 for ; Fri, 22 Feb 2002 13:44:11 -0600 (CST) Received: by zrc2c011.us.nortel.com with Internet Mail Service (5.5.2653.19) id <19H2KTLB>; Fri, 22 Feb 2002 13:44:10 -0600 Message-ID: <1B54FA3A2709D51195C800508BF9386A0313EE81@zrc2c000.us.nortel.com> From: "Mary Barnes" To: "'sipping@ietf.org'" Date: Fri, 22 Feb 2002 13:44:10 -0600 MIME-Version: 1.0 X-Mailer: Internet Mail Service (5.5.2653.19) Content-Type: multipart/alternative; boundary="----_=_NextPart_001_01C1BBD9.4C1C6070" Subject: [Sipping] New Requirements draft for Generic Request History Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org This message is in MIME format. Since your mail reader does not understand this format, some or all of this message may not be legible. ------_=_NextPart_001_01C1BBD9.4C1C6070 Content-Type: text/plain; charset="iso-8859-1" Hi all, We've submitted a draft capturing "Generic Request History" requirements. Until it appears in the repository, it's available at: http://www.obsidian97.com/draft-watson-sipping-req-history-00 Abstract: Many services that SIP is anticipated to support require the ability to determine why and how the call arrived at a specific application. Examples of such services include (but are not limited to) sessions initiated to call centers via "click to talk" SIP URLs on a web page, "call history/logging" style services within intelligent "call management" software for SIP UAs and calls to voicemail servers and call centers. While SIP implicitly provides the redirect/retarget capabilities that enable calls to be routed to chosen applications, there is currently no standard mechanism within SIP for communicating the history of such a request. This "request history" information allows the receiving application to determine hints about how and why the call arrived at the application/user. This draft discusses the motivations in support of a mechanism which records the "request history" and proposes detailed requirements for such a generic "request history" capability. There was a thread started on this topic following IETF-52 (http://www1.ietf.org/mail-archive/working-groups/sipping/current/msg00874.h tml ). We've attempted to capture the spirit of the requirements discussion from that thread which seems to have stopped around the end of January (right around the time bis-06 came out). We would like to get some additional discussion started to progress these requirements. Regards, Mary H. Barnes mbarnes@nortelnetworks.com 972-684-5432 Wireless 817-703-4806 ------_=_NextPart_001_01C1BBD9.4C1C6070 Content-Type: text/html; charset="iso-8859-1" Content-Transfer-Encoding: quoted-printable New Requirements draft for Generic Request History

Hi all,

We've submitted a draft capturing "Generic = Request History" requirements. Until it appears in the repository, = it's available at:

http://www.obsidian97.com/draft-watson-sipping-req-his= tory-00

Abstract:
   Many services that SIP is anticipated = to support require the
   ability to determine why and how the = call arrived at a specific
   application.  Examples of such = services include (but are not limited
   to) sessions initiated to call centers = via "click to talk" SIP URLs
   on a web page, "call = history/logging" style services within
   intelligent "call management" = software for SIP UAs and calls to
   voicemail servers and call = centers.  While SIP implicitly provides
   the redirect/retarget capabilities that = enable calls to be routed to
   chosen applications, there is currently = no standard mechanism within
   SIP for communicating the history of = such a request. This "request
   history" information allows the = receiving application to determine
   hints about how and why the call = arrived at the application/user.   

   This draft discusses the motivations in = support of a mechanism which
   records the "request history" and = proposes detailed requirements for
   such a generic "request = history" capability.


There was a thread started on this topic following = IETF-52 (http://www1.ietf.org/mail-archive/working-groups/sippi= ng/current/msg00874.html ).  We've attempted to capture the = spirit of the requirements discussion from that thread which seems to = have stopped around the end of January (right around the time bis-06 = came out). We would like to get some additional discussion started to = progress these requirements.

Regards,
Mary H. Barnes
mbarnes@nortelnetworks.com
972-684-5432
Wireless 817-703-4806

------_=_NextPart_001_01C1BBD9.4C1C6070-- _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@ns.ietf.org Fri Feb 22 16:23:06 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id QAA25908 for ; Fri, 22 Feb 2002 16:23:06 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id QAA27114 for sipping-archive@odin.ietf.org; Fri, 22 Feb 2002 16:23:09 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id QAA26385; Fri, 22 Feb 2002 16:06:02 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id QAA26357 for ; Fri, 22 Feb 2002 16:06:00 -0500 (EST) Received: from Mitel.COM ([216.191.234.70]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id QAA25238 for ; Fri, 22 Feb 2002 16:05:51 -0500 (EST) Received: from rndex50.software.mitel.com (rndex50 [134.199.17.160]) by Mitel.COM (V8/MAIL-RELAY-2.1) with ESMTP id QAA24390; Fri, 22 Feb 2002 16:00:16 -0500 (EST) Received: by rndex50.ottawa.mitel.com with Internet Mail Service (5.5.2653.19) id ; Fri, 22 Feb 2002 16:00:15 -0500 Message-ID: <29B5A21C13C8D51190F900805F65B4EC239E4E@rndex50.ottawa.mitel.com> From: "Blatherwick, Peter" To: sipping@ietf.org Cc: "'Henry Sinnreich'" , "'Dan Petrie (pingtel)'" Date: Fri, 22 Feb 2002 16:00:14 -0500 MIME-Version: 1.0 X-Mailer: Internet Mail Service (5.5.2653.19) Content-Type: text/plain; charset="iso-8859-1" Subject: [Sipping] RE: [Sip] Drafts on SIP Telephony Devices Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Thanks for these efforts Henry, Dan, all. As a member of a company that is building SIP phones, and plans to do lots more, I'd just like to state my strong support for these efforts. While SIP is an excellent toolkit to do all sorts of great things, it is also extremely important from an interoperability perspective to understand what the *minimal* requirements are for a device to be able to plug and play in a multi-vendor environment. This should not be taken as constraint on what can be designed, but a statement of "if you want to plug in anywhere, do at least this". Also allows us to say "if you want to do function X, of the N ways to do it, do at least this one". Same reasoning applies to both SIP protocol use and auto-config aspects. Both are big issues getting in the way of large scale deployment. -- Peter Blatherwick -----Original Message----- From: Henry Sinnreich [mailto:Henry.Sinnreich@wcom.com] Sent: February 22, 2002 10:03 To: sipping@ietf.org; sip@ietf.org Subject: [Sip] Drafts on SIP Telephony Devices Hi folks! Two related drafts on SIP devices have been submitted for the SIPPING and SIP WGs. They can be retrieved here until they will appear in the IETF archive. http://www.pingtel.com/docs/draft-sinnreich-sipping-device-requirements- 00.txt SIP Telephony Device Requirements Abstract The SIP Telephony Device Requirements in this document are aimed to enable users to procure SIP phones from various vendors and install them using a configuration process over the network, though still allowing manual configuration data input if desired. SIP telephony devices are also required to be updated up automatically, without the intervention of the end user. The requirements are aimed to allow service providers to support large numbers of SIP telephony devices from different vendors and SHOULD be used as a checklist for developers of SIP phones. SIP telephony devices may support basic telephony as well as advanced services such as PBX and Centrex-like telephony, third party call control, presence based services, roaming and other applications. The requirements aim to insure multi-vendor interoperability over the Internet and in private IP networks for such services. A Framework for SIP User Agent Configuration http://www.ietf.org/internet-drafts/draft-petrie-sip-config-framework-01 .txt Abstract This document defines the application of a set of protocols for configuring a SIP user agent. The SIP user agent must discover how and from where to retrieve its initial configuration and be notified of changes and updates which impact its configuration. The objective is to define a means for automatically configuring a user agent such that it can be functional without user or administrative intervention. The framework for discovery, delivery, notification and updates of user agent configuration is defined here. This framework is also intended to ease ongoing administration, configuration and upgrading of large scale deployments of SIP user agents. The contents and format of the configuration data to be defined is outside the scope of this document. It would be nice to discuss these two together. Comments and discussions are most welcome. Thanks, Henry Henry Sinnreich WorldCom 400 International Parkway Richardson, Texas 75081 USA _______________________________________________ Sip mailing list https://www1.ietf.org/mailman/listinfo/sip This list is for NEW development of the core SIP Protocol Use sip-implementors@cs.columbia.edu for questions on current sip Use sipping@ietf.org for new developments on the application of sip _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@ns.ietf.org Fri Feb 22 17:16:01 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id RAA28371 for ; Fri, 22 Feb 2002 17:16:01 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id RAA00624 for sipping-archive@odin.ietf.org; Fri, 22 Feb 2002 17:16:05 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id RAA29534; Fri, 22 Feb 2002 17:00:30 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id RAA29487 for ; Fri, 22 Feb 2002 17:00:27 -0500 (EST) Received: from sj-msg-core-3.cisco.com (sj-msg-core-3.cisco.com [171.70.157.152]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id RAA27748 for ; Fri, 22 Feb 2002 17:00:23 -0500 (EST) Received: from imop.cisco.com (imop.cisco.com [171.69.11.44]) by sj-msg-core-3.cisco.com (8.11.3/8.9.1) with ESMTP id g1MLxtw19594; Fri, 22 Feb 2002 13:59:55 -0800 (PST) Received: from localhost (ssh-sj1.cisco.com [171.68.225.134]) by imop.cisco.com (Mirapoint) with ESMTP id ACN12689; Fri, 22 Feb 2002 13:59:54 -0800 (PST) Date: Fri, 22 Feb 2002 13:58:24 -0800 (Pacific Standard Time) From: Rohan Mahy To: sipping@ietf.org cc: rohan@cisco.com Message-ID: X-X-Sender: rmahy@imop.cisco.com MIME-Version: 1.0 Content-Type: TEXT/PLAIN; charset=US-ASCII Subject: [Sipping] draft-ietf-sipping-cc-framework-00.txt Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Hello, The new multiparty application framework document has been submitted. Until it appears in the archives it is available at: ftp://ftpeng.cisco.com/rmahy/draft-ietf-sipping-cc-framework-00.txt Like all work from a design team, this is only *input* to the working group and can (and should) get ripped apart where appropriate, or thrown away if it is not useful (although I hope that is not the case). Unlike most work from a design team, most of the work in this document was summarizing or stitching together text and ideas from other drafts. I hope you like the result. Comments of all varieties are welcome. thanks, -rohan (as an individual contributor) _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Sun Feb 24 16:15:07 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id QAA16096 for ; Sun, 24 Feb 2002 16:15:06 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id QAA27792 for sipping-archive@odin.ietf.org; Sun, 24 Feb 2002 16:15:10 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id PAA27115; Sun, 24 Feb 2002 15:58:26 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id PAA27084 for ; Sun, 24 Feb 2002 15:58:24 -0500 (EST) Received: from bdsl.66.12.12.130.gte.net (bdsl.66.12.12.130.gte.net [66.12.12.130]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id PAA16014 for ; Sun, 24 Feb 2002 15:58:19 -0500 (EST) Received: from C1893415A (bdsl.greycouncil.com [127.0.0.1]) by bdsl.66.12.12.130.gte.net (8.11.6/8.11.6) with SMTP id g1OKvUB11166; Sun, 24 Feb 2002 14:57:32 -0600 Message-ID: <007101c1bd75$df34fe60$55fa403f@C1893415A> From: "Dean Willis" To: "Peterson, Jon" , "'Mark Watson'" , References: <70565611B164D511957A001083FCDD56018700EF@VA02> Subject: Re: [Sipping] Privacy comment (was Re: calling number blocking at sip/isup gate way) Date: Sun, 24 Feb 2002 14:57:26 -0600 MIME-Version: 1.0 Content-Type: multipart/alternative; boundary="----=_NextPart_000_006C_01C1BD43.92747A10" X-Priority: 3 X-MSMail-Priority: Normal X-Mailer: Microsoft Outlook Express 6.00.2600.0000 X-MimeOLE: Produced By Microsoft MimeOLE V6.00.2600.0000 Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org This is a multi-part message in MIME format. ------=_NextPart_000_006C_01C1BD43.92747A10 Content-Type: text/plain; charset="iso-8859-1" Content-Transfer-Encoding: quoted-printable Privacy comment (was Re: calling number blocking at sip/isup gateway) Your approach will generally work wherever the identity being used has = authentication credentials. However, 3GPP has posited (as usual) a slightly odd case. Each UA has a "private identity" which can authenticate using the = smart-card mechanisms in the SIM card. The home service proxy has a a table listing a number of public = identities that have been (via out-of-band mechanisms) associated with = the "private identity". The user may choose to use any of these associated public identities as = the originating identity for a call. However, proxy-authentication has = to be based on the private user identity, because we only have key = management for one identity in the SIM. So, the 3GPP proxy can accept a "hint" from the mobile supplied RPID as = to which of the known public identities the user ias asserting for this = specific dialog. I suppose that one could also do the same by appearing to authenticate = on the public identity, then using that to back-index into the private = identity for the actual authentication, but the mobile-supplied RPID = seems tidier to me. -- Dean ----- Original Message -----=20 From: Peterson, Jon=20 To: 'Mark Watson' ; sipping@ietf.org=20 Sent: Friday, February 22, 2002 12:26 PM Subject: RE: [Sipping] Privacy comment (was Re: calling number = blocking at sip/isup gate way) Actually, I have a few slight confusions about the text below (it = seems to use UA and user interchangeably a bit, which is worrisome when = we're talking about proving identity). One question would probably clear = things up: If the proxy still needs to authenticate the user (through whichever = mechanism) to ensure that the identity that has been asserted in the = RPID is valid for the user in question, why does the user need to supply = an RPID at all? Why doesn't the user just authenticate itself to the = proxy (however that happens) and let the proxy insert the RPID = corresponding to the identity that the user has proven? This works fine = for cases in which a single UA is used by multiple users, or when a = single user has multiple provable identities. It assumes that you have a = one-to-one mapping between credentials and identities, but that's = usually par for the course. Again, if the proxy server holds a mapping between credentials and = valid identities itself, as you suggest below, it doesn't seem at all = unreasonable to adopt this model. Jon Peterson NeuStar, Inc. -----Original Message----- From: Mark Watson [mailto:mwatson@nortelnetworks.com] Sent: Tuesday, February 19, 2002 12:29 PM To: sipping@ietf.org Subject: [Sipping] Privacy comment (was Re: calling number blocking = at sip/isup gate way) As presently described in privacy-03, the Remote-Party-ID header is = only ever inserted by proxies, based on the proxy's knowledge of the = identity of the UA (which it determines from some authentication = mechanism out of scope of the privacy draft). Privacy-02 allowed the UA to insert Remote-Party-ID too. I think = this should be reintroduced for the *specific case* of a UA with = multiple identities, to allow the UA to choose which identity it wishes = to be known by. The Proxy would still be responsible for ensuring that = this identity was valid for the user and would still use some other = authentication scheme to authenticate the user. The proxy would need = knowledge of which identities were valid for which users. I believe this is a requirement for 3GPP, where a UA can have = multiple 'public identities' and chooses which of these it wishes to be = known by for each call. The proxy authenticates the UA by some other = means and then checks the subscription data to ensure it is really = allowed to use the public Identity it has included.=20 I've suggested this a few times, and there has not been much comment = for or against. It would be good if we could get a resolution before the = next version of privacy is published. So, comments ?=20 Regards...Mark=20 > -----Original Message-----=20 > From: Dean Willis [mailto:dean.willis@softarmor.com]=20 > Sent: 18 February 2002 23:12=20 > To: Watson, Mark [MDN05:EP10:EXCH]; Mpierce1@aol.com;=20 > jon.peterson@neustar.biz; sipping@ietf.org=20 > Subject: Re: [Sipping] RE: [Sip] calling number blocking at = sip/isup=20 > gateway=20 >=20 >=20 > Mark said:=20 >=20 > > Your example below needs only a small extension to=20 > illustrate Mike's point=20 > > and show why something like Remote-Party-ID as defined in=20 > the privacy=20 > draft=20 > > is needed, and how it works despite the lack of a signature.=20 >=20 > Yep. And what happens when my misconfigured proxy forwards=20 > the call without=20 > paying attention to the Privacy header, because for some reason = it's=20 > forwarding to the outside world instead of a PSTN gateway=20 > under my control?=20 > We get a privacy leak.=20 >=20 > But other than that, I'd say you've precisely described a=20 > "sunny day" use=20 > case for RPID.=20 >=20 > It still has a lot of security issues based on transitive=20 > trust, which make=20 > it a lot more palatable in a two-party case than in three or=20 > more party=20 > transitivity.=20 >=20 > I believe we're approaching a general consensus that the Privacy = draft=20 > provides a reasonable approach to solving a specific class of = problem=20 > wherein interoperator-transitive trust is couple to=20 > transport-layer message=20 > protection. That's not a general privacy mechanism, but it=20 > does seem to have=20 > enough constituency to justify publication as long as it has a = REALLY=20 > serious applicability statement explaining the particular=20 > requirements and=20 > environments for which it is useful.=20 >=20 > How does that sound?=20 >=20 > --=20 > Dean=20 >=20 >=20 ------=_NextPart_000_006C_01C1BD43.92747A10 Content-Type: text/html; charset="iso-8859-1" Content-Transfer-Encoding: quoted-printable Privacy comment (was Re: calling number blocking at = sip/isup gateway)
 
Your approach will generally work = wherever the=20 identity being used has authentication credentials.
 
However, 3GPP has posited (as usual) a = slightly odd=20 case.
 
Each UA has a "private identity" which = can=20 authenticate using the smart-card mechanisms in the SIM = card.
 
The home service proxy has a a table = listing a=20 number of public identities that have been (via out-of-band mechanisms)=20 associated with the "private identity".
 
The user may choose to use any of these = associated=20 public identities as the originating identity for a call. However,=20 proxy-authentication has to be based on the private user identity, = because we=20 only have key management for one identity in the SIM.
 
So, the 3GPP proxy can accept a "hint" = from the=20 mobile supplied RPID as to which of the known public identities the user = ias=20 asserting for this specific dialog.
 
I suppose that one could also do the = same by=20 appearing to authenticate on the public identity, then using that to = back-index=20 into the private identity for the actual authentication, but the = mobile-supplied=20 RPID seems tidier to me.
 
--
Dean
 
 
 
----- Original Message -----
From:=20 Peterson, Jon
To: 'Mark Watson' ; sipping@ietf.org=20
Sent: Friday, February 22, 2002 = 12:26=20 PM
Subject: RE: [Sipping] Privacy = comment=20 (was Re: calling number blocking at sip/isup gate way)

Actually, I have a few slight confusions = about the=20 text below (it seems to use UA and user interchangeably a bit, which = is=20 worrisome when we're talking about proving identity). One = question would=20 probably clear things up:
 
If=20 the proxy still needs to authenticate the user (through whichever = mechanism)=20 to ensure that the identity that has been asserted in the RPID is = valid for=20 the user in question, why does the user need to supply an RPID at all? = Why=20 doesn't the user just authenticate itself to the proxy (however that = happens)=20 and let the proxy insert the RPID corresponding to the identity that = the user=20 has proven? This works fine for cases in which a single UA is used by = multiple=20 users, or when a single user has multiple provable identities. It = assumes that=20 you have a one-to-one mapping between credentials and identities, but = that's=20 usually par for the course.
 
Again, if the proxy server holds a mapping = between=20 credentials and valid identities itself, as you suggest below, it = doesn't seem=20 at all unreasonable to adopt this model.
 
Jon=20 Peterson
NeuStar, Inc.
-----Original Message-----
From: Mark Watson=20 [mailto:mwatson@nortelnetworks.com]
Sent: Tuesday, = February 19,=20 2002 12:29 PM
To: sipping@ietf.org
Subject: = [Sipping]=20 Privacy comment (was Re: calling number blocking at sip/isup gate=20 way)

As presently described in privacy-03, the = Remote-Party-ID=20 header is only ever inserted by proxies, based on the proxy's = knowledge of=20 the identity of the UA (which it determines from some authentication = mechanism out of scope of the privacy draft).

Privacy-02 allowed the UA to insert = Remote-Party-ID too. I=20 think this should be reintroduced for the *specific case* of a UA = with=20 multiple identities, to allow the UA to choose which identity it = wishes to=20 be known by. The Proxy would still be responsible for ensuring that = this=20 identity was valid for the user and would still use some other=20 authentication scheme to authenticate the user. The proxy would need = knowledge of which identities were valid for which users.

I believe this is a requirement for 3GPP, where a = UA can=20 have multiple 'public identities' and chooses which of these it = wishes to be=20 known by for each call. The proxy authenticates the UA by some other = means=20 and then checks the subscription data to ensure it is really allowed = to use=20 the public Identity it has included.

I've suggested this a few times, and there has not = been much=20 comment for or against. It would be good if we could get a = resolution before=20 the next version of privacy is published.

So, comments ?

Regards...Mark





> -----Original Message-----
>=20 From: Dean Willis [mailto:dean.willis@softarmor.co= m]=20
> Sent: 18 February 2002 23:12 =
> To: Watson, Mark [MDN05:EP10:EXCH]; = Mpierce1@aol.com;
=20
> jon.peterson@neustar.biz; = sipping@ietf.org=20
> Subject: Re: [Sipping] RE: [Sip] calling = number=20 blocking at sip/isup
> gateway =
>

>
> Mark=20 said:
>
> = > Your=20 example below needs only a small extension to
>=20 illustrate Mike's point
> > and show = why=20 something like Remote-Party-ID as defined in
>=20 the privacy
> draft
>=20 > is needed, and how it works despite the lack of a = signature.=20
>
> Yep. And what = happens=20 when my misconfigured proxy forwards
> = the call=20 without
> paying attention to the = Privacy header,=20 because for some reason it's
> = forwarding to the=20 outside world instead of a PSTN gateway
> under=20 my control?
> We get a privacy = leak.=20
>
> But other = than that, I'd=20 say you've precisely described a
> = "sunny day"=20 use
> case for RPID.
>=20
> It still has a lot of security issues = based on=20 transitive
> trust, which make =
> it a lot more palatable in a two-party case than in = three or=20
> more party
>=20 transitivity.
>
> I=20 believe we're approaching a general consensus that the Privacy = draft=20
> provides a reasonable approach to solving a = specific=20 class of problem
> wherein=20 interoperator-transitive trust is couple to
>=20 transport-layer message
> protection. = That's not=20 a general privacy mechanism, but it
> = does seem=20 to have
> enough constituency to = justify=20 publication as long as it has a REALLY
> serious=20 applicability statement explaining the particular
> requirements and
> = environments for=20 which it is useful.
>
> How does that sound?
>=20
> --
> = Dean=20
>
>=20

------=_NextPart_000_006C_01C1BD43.92747A10-- _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Mon Feb 25 06:35:39 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id GAA05710 for ; Mon, 25 Feb 2002 06:35:39 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id GAA13370 for sipping-archive@odin.ietf.org; Mon, 25 Feb 2002 06:35:41 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id GAA11882; Mon, 25 Feb 2002 06:22:25 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id GAA11852 for ; Mon, 25 Feb 2002 06:22:23 -0500 (EST) Received: from CNRI.Reston.VA.US (localhost [127.0.0.1]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id GAA04722; Mon, 25 Feb 2002 06:22:20 -0500 (EST) Message-Id: <200202251122.GAA04722@ietf.org> Mime-Version: 1.0 Content-Type: Multipart/Mixed; Boundary="NextPart" To: IETF-Announce: ; CC: sipping@ietf.org From: Internet-Drafts@ietf.org Reply-to: Internet-Drafts@ietf.org Date: Mon, 25 Feb 2002 06:22:20 -0500 Subject: [Sipping] I-D ACTION:draft-watson-sipping-req-history-00.txt Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org --NextPart A New Internet-Draft is available from the on-line Internet-Drafts directories. Title : Generic Request History Capability - Requirements Author(s) : M. Watson et al. Filename : draft-watson-sipping-req-history-00.txt Pages : 7 Date : 22-Feb-02 Many services that SIP is anticipated to support require the ability to determine why and how the call arrived at a specific application. Examples of such services include (but are not limited to) sessions initiated to call centers via 'click to talk' SIP URLs on a web page, 'call history/logging' style services within intelligent 'call management' software for SIP UAs and calls to voicemail servers and call centers. A URL for this Internet-Draft is: http://www.ietf.org/internet-drafts/draft-watson-sipping-req-history-00.txt To remove yourself from the IETF Announcement list, send a message to ietf-announce-request with the word unsubscribe in the body of the message. Internet-Drafts are also available by anonymous FTP. Login with the username "anonymous" and a password of your e-mail address. After logging in, type "cd internet-drafts" and then "get draft-watson-sipping-req-history-00.txt". A list of Internet-Drafts directories can be found in http://www.ietf.org/shadow.html or ftp://ftp.ietf.org/ietf/1shadow-sites.txt Internet-Drafts can also be obtained by e-mail. Send a message to: mailserv@ietf.org. In the body type: "FILE /internet-drafts/draft-watson-sipping-req-history-00.txt". NOTE: The mail server at ietf.org can return the document in MIME-encoded form by using the "mpack" utility. To use this feature, insert the command "ENCODING mime" before the "FILE" command. To decode the response(s), you will need "munpack" or a MIME-compliant mail reader. Different MIME-compliant mail readers exhibit different behavior, especially when dealing with "multipart" MIME messages (i.e. documents which have been split up into multiple messages), so check your local documentation on how to manipulate these messages. Below is the data which will enable a MIME compliant mail reader implementation to automatically retrieve the ASCII version of the Internet-Draft. --NextPart Content-Type: Multipart/Alternative; Boundary="OtherAccess" --OtherAccess Content-Type: Message/External-body; access-type="mail-server"; server="mailserv@ietf.org" Content-Type: text/plain Content-ID: <20020222110434.I-D@ietf.org> ENCODING mime FILE /internet-drafts/draft-watson-sipping-req-history-00.txt --OtherAccess Content-Type: Message/External-body; name="draft-watson-sipping-req-history-00.txt"; site="ftp.ietf.org"; access-type="anon-ftp"; directory="internet-drafts" Content-Type: text/plain Content-ID: <20020222110434.I-D@ietf.org> --OtherAccess-- --NextPart-- _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Mon Feb 25 06:36:23 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id GAA05738 for ; Mon, 25 Feb 2002 06:36:23 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id GAA13409 for sipping-archive@odin.ietf.org; Mon, 25 Feb 2002 06:36:25 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id GAA12149; Mon, 25 Feb 2002 06:24:36 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id GAA12122 for ; Mon, 25 Feb 2002 06:24:34 -0500 (EST) Received: from CNRI.Reston.VA.US (localhost [127.0.0.1]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id GAA05262; Mon, 25 Feb 2002 06:24:31 -0500 (EST) Message-Id: <200202251124.GAA05262@ietf.org> Mime-Version: 1.0 Content-Type: Multipart/Mixed; Boundary="NextPart" To: IETF-Announce: ; Cc: sipping@ietf.org From: Internet-Drafts@ietf.org Reply-to: Internet-Drafts@ietf.org Date: Mon, 25 Feb 2002 06:24:30 -0500 Subject: [Sipping] I-D ACTION:draft-ietf-sipping-deaf-req-00.txt Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org --NextPart A New Internet-Draft is available from the on-line Internet-Drafts directories. This draft is a work item of the Session Initiation Proposal Investigation Working Group of the IETF. Title : User Requirements for SIP in support of deaf, hard of hearing and speech-impaired individuals Author(s) : N. Charlton et al. Filename : draft-ietf-sipping-deaf-req-00.txt Pages : 11 Date : 22-Feb-02 This document aims to present a set of SIP user requirements that support communications for deaf, hard of hearing and speech-impaired individuals. These user requirements address the current difficulties of deaf, hard of hearing and speech-impaired individuals in using communications facilities, while acknowledging the multi-functional potential of SIP-based communications. A URL for this Internet-Draft is: http://www.ietf.org/internet-drafts/draft-ietf-sipping-deaf-req-00.txt To remove yourself from the IETF Announcement list, send a message to ietf-announce-request with the word unsubscribe in the body of the message. Internet-Drafts are also available by anonymous FTP. Login with the username "anonymous" and a password of your e-mail address. After logging in, type "cd internet-drafts" and then "get draft-ietf-sipping-deaf-req-00.txt". A list of Internet-Drafts directories can be found in http://www.ietf.org/shadow.html or ftp://ftp.ietf.org/ietf/1shadow-sites.txt Internet-Drafts can also be obtained by e-mail. Send a message to: mailserv@ietf.org. In the body type: "FILE /internet-drafts/draft-ietf-sipping-deaf-req-00.txt". NOTE: The mail server at ietf.org can return the document in MIME-encoded form by using the "mpack" utility. To use this feature, insert the command "ENCODING mime" before the "FILE" command. To decode the response(s), you will need "munpack" or a MIME-compliant mail reader. Different MIME-compliant mail readers exhibit different behavior, especially when dealing with "multipart" MIME messages (i.e. documents which have been split up into multiple messages), so check your local documentation on how to manipulate these messages. Below is the data which will enable a MIME compliant mail reader implementation to automatically retrieve the ASCII version of the Internet-Draft. --NextPart Content-Type: Multipart/Alternative; Boundary="OtherAccess" --OtherAccess Content-Type: Message/External-body; access-type="mail-server"; server="mailserv@ietf.org" Content-Type: text/plain Content-ID: <20020222110935.I-D@ietf.org> ENCODING mime FILE /internet-drafts/draft-ietf-sipping-deaf-req-00.txt --OtherAccess Content-Type: Message/External-body; name="draft-ietf-sipping-deaf-req-00.txt"; site="ftp.ietf.org"; access-type="anon-ftp"; directory="internet-drafts" Content-Type: text/plain Content-ID: <20020222110935.I-D@ietf.org> --OtherAccess-- --NextPart-- _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Mon Feb 25 08:04:04 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id IAA08371 for ; Mon, 25 Feb 2002 08:04:04 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id IAA18102 for sipping-archive@odin.ietf.org; Mon, 25 Feb 2002 08:04:07 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id HAA16698; Mon, 25 Feb 2002 07:47:02 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id HAA16590 for ; Mon, 25 Feb 2002 07:46:56 -0500 (EST) Received: from zctfs063.nortelnetworks.com (h90s131a237n47.user.nortelnetworks.com [47.237.131.90]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id HAA07545 for ; Mon, 25 Feb 2002 07:46:52 -0500 (EST) Received: from znsgs01r.europe.nortel.com (znsgs01r.europe.nortel.com [47.137.129.92]) by zctfs063.nortelnetworks.com (Switch-2.2.0/Switch-2.2.0) with ESMTP id g1PCkIN16245; Mon, 25 Feb 2002 13:46:18 +0100 (MET) Received: from zwcwc012.europe.nortel.com (zwcwc012.europe.nortel.com [47.73.112.187]) by znsgs01r.europe.nortel.com (Switch-2.2.0/Switch-2.2.0) with ESMTP id g1PCjp925620; Mon, 25 Feb 2002 12:45:52 GMT Received: by zwcwc012.europe.nortel.com with Internet Mail Service (5.5.2653.19) id ; Mon, 25 Feb 2002 12:46:24 -0000 Message-ID: From: "Mark Watson" To: "Peterson, Jon" , sipping@ietf.org Subject: RE: [Sipping] Privacy comment (was Re: calling number blocking at sip/isup gate way) Date: Mon, 25 Feb 2002 12:46:26 -0000 MIME-Version: 1.0 X-Mailer: Internet Mail Service (5.5.2653.19) Content-Type: multipart/alternative; boundary="----_=_NextPart_001_01C1BDFA.7011E860" Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org This message is in MIME format. Since your mail reader does not understand this format, some or all of this message may not be legible. ------_=_NextPart_001_01C1BDFA.7011E860 Content-Type: text/plain; charset="iso-8859-1" Jon wrote: > This works fine for cases in which a single UA is used by multiple users, or when a single user has multiple provable identities. > It assumes that you have a one-to-one mapping between credentials and identities, but that's usually par for the course. This assumption does not hold in the 3GPP case where authentication is based on a 'private identity' and associated keys in the SIM, but where there may be multiple 'public identies' associated with this single 'private identity'. In particular, this allows you easily to add/remove/manage the 'public identies' without getting involved in key management/distribution. All that is necessary to manage to public identities is to update the subscription information that the proxy uses to verify that a particular public identity is allowed for the particular private identity. In addition I can think of cases where the identity that the user may wish to have presented to the terminating UA, to the PSTN or other elements, may differ from the identity that they use for local authentication. For example, imagine an outbound call centre where all the agents use SIP terminals. For obvious reasons each agent must authenticate to the local proxy using a personal identity, but the identity they wish to be known by to the called party is the Freephone number of the call centre (or even a different, inbound, call centre). Cheers...Mark -----Original Message----- From: Peterson, Jon [mailto:jon.peterson@neustar.biz] Sent: 22 February 2002 18:26 To: Watson, Mark [MDN05:EP10:EXCH]; sipping@ietf.org Subject: RE: [Sipping] Privacy comment (was Re: calling number blocking at sip/isup gate way) Actually, I have a few slight confusions about the text below (it seems to use UA and user interchangeably a bit, which is worrisome when we're talking about proving identity). One question would probably clear things up: If the proxy still needs to authenticate the user (through whichever mechanism) to ensure that the identity that has been asserted in the RPID is valid for the user in question, why does the user need to supply an RPID at all? Why doesn't the user just authenticate itself to the proxy (however that happens) and let the proxy insert the RPID corresponding to the identity that the user has proven? This works fine for cases in which a single UA is used by multiple users, or when a single user has multiple provable identities. It assumes that you have a one-to-one mapping between credentials and identities, but that's usually par for the course. Again, if the proxy server holds a mapping between credentials and valid identities itself, as you suggest below, it doesn't seem at all unreasonable to adopt this model. Jon Peterson NeuStar, Inc. -----Original Message----- From: Mark Watson [mailto:mwatson@nortelnetworks.com] Sent: Tuesday, February 19, 2002 12:29 PM To: sipping@ietf.org Subject: [Sipping] Privacy comment (was Re: calling number blocking at sip/isup gate way) As presently described in privacy-03, the Remote-Party-ID header is only ever inserted by proxies, based on the proxy's knowledge of the identity of the UA (which it determines from some authentication mechanism out of scope of the privacy draft). Privacy-02 allowed the UA to insert Remote-Party-ID too. I think this should be reintroduced for the *specific case* of a UA with multiple identities, to allow the UA to choose which identity it wishes to be known by. The Proxy would still be responsible for ensuring that this identity was valid for the user and would still use some other authentication scheme to authenticate the user. The proxy would need knowledge of which identities were valid for which users. I believe this is a requirement for 3GPP, where a UA can have multiple 'public identities' and chooses which of these it wishes to be known by for each call. The proxy authenticates the UA by some other means and then checks the subscription data to ensure it is really allowed to use the public Identity it has included. I've suggested this a few times, and there has not been much comment for or against. It would be good if we could get a resolution before the next version of privacy is published. So, comments ? Regards...Mark > -----Original Message----- > From: Dean Willis [ mailto:dean.willis@softarmor.com ] > Sent: 18 February 2002 23:12 > To: Watson, Mark [MDN05:EP10:EXCH]; Mpierce1@aol.com; > jon.peterson@neustar.biz; sipping@ietf.org > Subject: Re: [Sipping] RE: [Sip] calling number blocking at sip/isup > gateway > > > Mark said: > > > Your example below needs only a small extension to > illustrate Mike's point > > and show why something like Remote-Party-ID as defined in > the privacy > draft > > is needed, and how it works despite the lack of a signature. > > Yep. And what happens when my misconfigured proxy forwards > the call without > paying attention to the Privacy header, because for some reason it's > forwarding to the outside world instead of a PSTN gateway > under my control? > We get a privacy leak. > > But other than that, I'd say you've precisely described a > "sunny day" use > case for RPID. > > It still has a lot of security issues based on transitive > trust, which make > it a lot more palatable in a two-party case than in three or > more party > transitivity. > > I believe we're approaching a general consensus that the Privacy draft > provides a reasonable approach to solving a specific class of problem > wherein interoperator-transitive trust is couple to > transport-layer message > protection. That's not a general privacy mechanism, but it > does seem to have > enough constituency to justify publication as long as it has a REALLY > serious applicability statement explaining the particular > requirements and > environments for which it is useful. > > How does that sound? > > -- > Dean > > ------_=_NextPart_001_01C1BDFA.7011E860 Content-Type: text/html; charset="iso-8859-1" Privacy comment (was Re: calling number blocking at sip/isup gateway)
Jon wrote:
 
> This works fine for cases in which a single UA is used by multiple users, or when a single user has multiple provable identities.
> It assumes that you have a one-to-one mapping between credentials and identities, but that's usually par for the course.
 
This assumption does not hold in the 3GPP case where authentication is based on a 'private identity' and associated keys in the SIM, but where there may be multiple 'public identies' associated with this single 'private identity'. In particular, this allows you easily to add/remove/manage the 'public identies' without getting involved in key management/distribution. All that is necessary to manage to public identities is to update the subscription information that the proxy uses to verify that a particular public identity is allowed for the particular private identity.
 
In addition I can think of cases where the identity that the user may wish to have presented to the terminating UA, to the PSTN or other elements, may differ from the identity that they use for local authentication. For example, imagine an outbound call centre where all the agents use SIP terminals. For obvious reasons each agent must authenticate to the local proxy using a personal identity, but the identity they wish to be known by to the called party is the Freephone number of the call centre (or even a different, inbound, call centre).
 
Cheers...Mark
 
 
 -----Original Message-----
From: Peterson, Jon [mailto:jon.peterson@neustar.biz]
Sent: 22 February 2002 18:26
To: Watson, Mark [MDN05:EP10:EXCH]; sipping@ietf.org
Subject: RE: [Sipping] Privacy comment (was Re: calling number blocking at sip/isup gate way)

Actually, I have a few slight confusions about the text below (it seems to use UA and user interchangeably a bit, which is worrisome when we're talking about proving identity). One question would probably clear things up:
 
If the proxy still needs to authenticate the user (through whichever mechanism) to ensure that the identity that has been asserted in the RPID is valid for the user in question, why does the user need to supply an RPID at all? Why doesn't the user just authenticate itself to the proxy (however that happens) and let the proxy insert the RPID corresponding to the identity that the user has proven? This works fine for cases in which a single UA is used by multiple users, or when a single user has multiple provable identities. It assumes that you have a one-to-one mapping between credentials and identities, but that's usually par for the course.
 
Again, if the proxy server holds a mapping between credentials and valid identities itself, as you suggest below, it doesn't seem at all unreasonable to adopt this model.
 
Jon Peterson
NeuStar, Inc.
-----Original Message-----
From: Mark Watson [mailto:mwatson@nortelnetworks.com]
Sent: Tuesday, February 19, 2002 12:29 PM
To: sipping@ietf.org
Subject: [Sipping] Privacy comment (was Re: calling number blocking at sip/isup gate way)

As presently described in privacy-03, the Remote-Party-ID header is only ever inserted by proxies, based on the proxy's knowledge of the identity of the UA (which it determines from some authentication mechanism out of scope of the privacy draft).

Privacy-02 allowed the UA to insert Remote-Party-ID too. I think this should be reintroduced for the *specific case* of a UA with multiple identities, to allow the UA to choose which identity it wishes to be known by. The Proxy would still be responsible for ensuring that this identity was valid for the user and would still use some other authentication scheme to authenticate the user. The proxy would need knowledge of which identities were valid for which users.

I believe this is a requirement for 3GPP, where a UA can have multiple 'public identities' and chooses which of these it wishes to be known by for each call. The proxy authenticates the UA by some other means and then checks the subscription data to ensure it is really allowed to use the public Identity it has included.

I've suggested this a few times, and there has not been much comment for or against. It would be good if we could get a resolution before the next version of privacy is published.

So, comments ?

Regards...Mark





> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.com]
> Sent: 18 February 2002 23:12
> To: Watson, Mark [MDN05:EP10:EXCH]; Mpierce1@aol.com;
> jon.peterson@neustar.biz; sipping@ietf.org
> Subject: Re: [Sipping] RE: [Sip] calling number blocking at sip/isup
> gateway
>
>
> Mark said:
>
> > Your example below needs only a small extension to
> illustrate Mike's point
> > and show why something like Remote-Party-ID as defined in
> the privacy
> draft
> > is needed, and how it works despite the lack of a signature.
>
> Yep. And what happens when my misconfigured proxy forwards
> the call without
> paying attention to the Privacy header, because for some reason it's
> forwarding to the outside world instead of a PSTN gateway
> under my control?
> We get a privacy leak.
>
> But other than that, I'd say you've precisely described a
> "sunny day" use
> case for RPID.
>
> It still has a lot of security issues based on transitive
> trust, which make
> it a lot more palatable in a two-party case than in three or
> more party
> transitivity.
>
> I believe we're approaching a general consensus that the Privacy draft
> provides a reasonable approach to solving a specific class of problem
> wherein interoperator-transitive trust is couple to
> transport-layer message
> protection. That's not a general privacy mechanism, but it
> does seem to have
> enough constituency to justify publication as long as it has a REALLY
> serious applicability statement explaining the particular
> requirements and
> environments for which it is useful.
>
> How does that sound?
>
> --
> Dean
>
>

------_=_NextPart_001_01C1BDFA.7011E860-- _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Mon Feb 25 11:02:28 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id LAA15264 for ; Mon, 25 Feb 2002 11:02:26 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id LAA28714 for sipping-archive@odin.ietf.org; Mon, 25 Feb 2002 11:02:29 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id KAA27531; Mon, 25 Feb 2002 10:47:41 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id KAA27506 for ; Mon, 25 Feb 2002 10:47:39 -0500 (EST) Received: from Mitel.COM ([216.191.234.70]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id KAA14497 for ; Mon, 25 Feb 2002 10:47:23 -0500 (EST) From: Tom_Gray@Mitel.COM Received: from kanmta01.software.mitel.com (kanmta01.kanata.mitel.com [134.199.37.58]) by Mitel.COM (V8/MAIL-RELAY-2.1) with SMTP id KAA18755 for ; Mon, 25 Feb 2002 10:44:54 -0500 (EST) Received: by kanmta01.software.mitel.com(Lotus SMTP MTA v4.6.7 (934.1 12-30-1999)) id 85256B6B.00567FD7 ; Mon, 25 Feb 2002 10:44:48 -0500 X-Lotus-FromDomain: MITEL To: sipping@ietf.org, Tom_Gray@Mitel.COM Message-ID: <85256B6B.00567EFF.00@kanmta01.software.mitel.com> Date: Mon, 25 Feb 2002 10:44:50 -0500 Subject: Re: [Sipping] I-D ACTION:draft-watson-sipping-req-history-00.txt Mime-Version: 1.0 Content-type: text/plain; charset=us-ascii Content-Disposition: inline Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org From: Tom Gray@MITEL on 02/25/2002 10:44 AM Looking at this document does not reveal a list of requirements but just a list of proposed capabilities that may or may not require additions to the SIP protocol. For this to be a useful requirement document, this list of capabilities must be linked to a set of problems that need to be solved. Indeed the list of problems would be the nub of the document. For example what problem is the 'capability-req' capability intended to solve. Just what is 'Issuer-req'? The description in the draft doesn't specify this beyond not identifying he reason fro its proposal. What trade-offs if any will be required for the solution to these problems? Internet-Drafts@ietf.org on 02/25/2002 06:22:20 AM Please respond to Internet-Drafts@ietf.org To: ietf.org cc: sipping@ietf.org (bcc: Tom Gray/Kan/Mitel) Subject: [Sipping] I-D ACTION:draft-watson-sipping-req-history-00.txt A New Internet-Draft is available from the on-line Internet-Drafts directories. Title : Generic Request History Capability - Requirements Author(s) : M. Watson et al. Filename : draft-watson-sipping-req-history-00.txt Pages : 7 Date : 22-Feb-02 Many services that SIP is anticipated to support require the ability to determine why and how the call arrived at a specific application. Examples of such services include (but are not limited to) sessions initiated to call centers via 'click to talk' SIP URLs on a web page, 'call history/logging' style services within intelligent 'call management' software for SIP UAs and calls to voicemail servers and call centers. A URL for this Internet-Draft is: http://www.ietf.org/internet-drafts/draft-watson-sipping-req-history-00.txt To remove yourself from the IETF Announcement list, send a message to ietf-announce-request with the word unsubscribe in the body of the message. Internet-Drafts are also available by anonymous FTP. Login with the username "anonymous" and a password of your e-mail address. After logging in, type "cd internet-drafts" and then "get draft-watson-sipping-req-history-00.txt". A list of Internet-Drafts directories can be found in http://www.ietf.org/shadow.html or ftp://ftp.ietf.org/ietf/1shadow-sites.txt Internet-Drafts can also be obtained by e-mail. Send a message to: mailserv@ietf.org. In the body type: "FILE /internet-drafts/draft-watson-sipping-req-history-00.txt". NOTE: The mail server at ietf.org can return the document in MIME-encoded form by using the "mpack" utility. To use this feature, insert the command "ENCODING mime" before the "FILE" command. To decode the response(s), you will need "munpack" or a MIME-compliant mail reader. Different MIME-compliant mail readers exhibit different behavior, especially when dealing with "multipart" MIME messages (i.e. documents which have been split up into multiple messages), so check your local documentation on how to manipulate these messages. Below is the data which will enable a MIME compliant mail reader implementation to automatically retrieve the ASCII version of the Internet-Draft. mailto:mailserv@ietf.org ftp://ftp.ietf.org/internet-drafts/draft-watson-sipping-req-history-00.txt _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Mon Feb 25 12:29:22 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id MAA20002 for ; Mon, 25 Feb 2002 12:29:22 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id MAA05498 for sipping-archive@odin.ietf.org; Mon, 25 Feb 2002 12:29:26 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id MAA04667; Mon, 25 Feb 2002 12:12:50 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id MAA04609 for ; Mon, 25 Feb 2002 12:12:46 -0500 (EST) Received: from t-6mnjnqs5jxuuc ([211.152.107.30]) by ietf.org (8.9.1a/8.9.1a) with SMTP id MAA19182 for ; Mon, 25 Feb 2002 12:12:40 -0500 (EST) Message-Id: <200202251712.MAA19182@ietf.org> From: "T&F" To: Mime-Version: 1.0 Content-Type: text/plain; charset="gb2312" Date: Tue, 26 Feb 2002 01:25:49 Subject: [Sipping] A professional company providing credit & status report of Chinese company Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org A professional company providing credit & status report of Chinese company. Dear Sirs, T&F is a senior investigative corporation providing credit & status report of Chinese companies for global clients. If you are in need of data from China, we are available to provide that information on consignment. We are an established authority in the field of research and information gathering in China. At the same time, we can also consign investigative missions to you when we need data from your country. In this manner we would hope to achieve a mutually beneficial arrangement exchanging needed information on a regular basis. 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Your name will be removed from our database upon request. 使用极星邮件群发,无须通过邮件服务器,直达对方邮箱,速度绝对一流! 下载网址:http://love2net.51.net/,更多免费的超酷软件等你来下…… ---------------------------------------------------- INFORMATION This message has been sent using a trial-run version of the TSmtpRelayServer Delphi Component. ---------------------------------------------------- _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Mon Feb 25 13:19:51 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id NAA22932 for ; Mon, 25 Feb 2002 13:19:51 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id NAA08250 for sipping-archive@odin.ietf.org; Mon, 25 Feb 2002 13:19:53 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id MAA06888; Mon, 25 Feb 2002 12:55:18 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id MAA06856 for ; Mon, 25 Feb 2002 12:55:15 -0500 (EST) Received: from zctfs063.nortelnetworks.com (h90s131a237n47.user.nortelnetworks.com [47.237.131.90]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id MAA21863 for ; Mon, 25 Feb 2002 12:55:11 -0500 (EST) Received: from zwcwc012.europe.nortel.com (zwcwc012.europe.nortel.com [47.73.112.187]) by zctfs063.nortelnetworks.com (Switch-2.2.0/Switch-2.2.0) with ESMTP id g1PHsZv02019; Mon, 25 Feb 2002 18:54:35 +0100 (MET) Received: by zwcwc012.europe.nortel.com with Internet Mail Service (5.5.2653.19) id ; Mon, 25 Feb 2002 17:54:35 -0000 Message-ID: From: "Mark Watson" To: Tom_Gray@Mitel.COM, sipping@ietf.org Subject: RE: [Sipping] I-D ACTION:draft-watson-sipping-req-history-00.txt Date: Mon, 25 Feb 2002 17:54:33 -0000 MIME-Version: 1.0 X-Mailer: Internet Mail Service (5.5.2653.19) Content-Type: multipart/alternative; boundary="----_=_NextPart_001_01C1BE25.7B9A4620" Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org This message is in MIME format. Since your mail reader does not understand this format, some or all of this message may not be legible. ------_=_NextPart_001_01C1BE25.7B9A4620 Content-Type: text/plain Tony, I think it depends what stage in the process you think we are at, and what you mean by requirements. 'Requirements' can be written at many levels, but I thought the stage we got to at the last IETF was a need to identify exactly a 'set of capabilities' that we wished to be able to provide with SIP. In terms of the overall problems that these capabilities solve: they enable certain services, some of which are discussed in the draft and some of which have been the motivation for the more specific mechanisms previously proposed and listed in the references. We don't in IETF define a set of services, and then design protocol to implement these, but rather identify generic capabilities which might be useful to service designers. This is where we were after IETF-52, with support for definition of a generic capability for request history. Once we have agreement on the details of what this capability does, *then* we can proceed to the next stage of identifying what technical problems need to be solved from a protocol perspective in order to provide achieve this. We thought this was the required process under sipchange - define the capability first in SIPPING and work on the solution later in SIP. Are you questioning the rationalle for the whole capability, or do you expect the requirements to go into more detail as to the effect on the protocol ? Regards, Mark > -----Original Message----- > From: Tom_Gray@Mitel.COM [mailto:Tom_Gray@Mitel.COM] > Sent: 25 February 2002 15:45 > To: sipping@ietf.org; Tom_Gray@Mitel.COM > Subject: Re: [Sipping] I-D > ACTION:draft-watson-sipping-req-history-00.txt > > > > > From: Tom Gray@MITEL on 02/25/2002 10:44 AM > > > Looking at this document does not reveal a list of > requirements but just a list > of proposed capabilities that may or may not require > additions to the SIP > protocol. For this to be a useful requirement document, this list of > capabilities must be linked to a set of problems that need to > be solved. Indeed > the list of problems would be the nub of the document. For > example what problem > is the 'capability-req' capability intended to solve. Just > what is 'Issuer-req'? > The description in the draft doesn't specify this beyond not > identifying he > reason fro its proposal. What trade-offs if any will be > required for the > solution to these problems? > > > > > > > Internet-Drafts@ietf.org on 02/25/2002 06:22:20 AM > > Please respond to Internet-Drafts@ietf.org > > To: ietf.org > cc: sipping@ietf.org (bcc: Tom Gray/Kan/Mitel) > > Subject: [Sipping] I-D ACTION:draft-watson-sipping-req-history-00.txt > > > > A New Internet-Draft is available from the on-line > Internet-Drafts directories. > > > Title : Generic Request History Capability - > Requirements > Author(s) : M. Watson et al. > Filename : draft-watson-sipping-req-history-00.txt > Pages : 7 > Date : 22-Feb-02 > > Many services that SIP is anticipated to support require the > ability to determine why and how the call arrived at a specific > application. Examples of such services include (but are not limited > to) sessions initiated to call centers via 'click to talk' SIP URLs > on a web page, 'call history/logging' style services within > intelligent 'call management' software for SIP UAs and calls to > voicemail servers and call centers. > > A URL for this Internet-Draft is: > http://www.ietf.org/internet-drafts/draft-watson-sipping-req-h istory-00.txt To remove yourself from the IETF Announcement list, send a message to ietf-announce-request with the word unsubscribe in the body of the message. Internet-Drafts are also available by anonymous FTP. Login with the username "anonymous" and a password of your e-mail address. After logging in, type "cd internet-drafts" and then "get draft-watson-sipping-req-history-00.txt". A list of Internet-Drafts directories can be found in http://www.ietf.org/shadow.html or ftp://ftp.ietf.org/ietf/1shadow-sites.txt Internet-Drafts can also be obtained by e-mail. Send a message to: mailserv@ietf.org. In the body type: "FILE /internet-drafts/draft-watson-sipping-req-history-00.txt". NOTE: The mail server at ietf.org can return the document in MIME-encoded form by using the "mpack" utility. To use this feature, insert the command "ENCODING mime" before the "FILE" command. To decode the response(s), you will need "munpack" or a MIME-compliant mail reader. Different MIME-compliant mail readers exhibit different behavior, especially when dealing with "multipart" MIME messages (i.e. documents which have been split up into multiple messages), so check your local documentation on how to manipulate these messages. Below is the data which will enable a MIME compliant mail reader implementation to automatically retrieve the ASCII version of the Internet-Draft. mailto:mailserv@ietf.org ftp://ftp.ietf.org/internet-drafts/draft-watson-sipping-req-history-00.txt _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP ------_=_NextPart_001_01C1BE25.7B9A4620 Content-Type: text/html Content-Transfer-Encoding: quoted-printable RE: [Sipping] I-D = ACTION:draft-watson-sipping-req-history-00.txt

Tony,

I think it depends what stage in the process you = think we are at, and what you mean by requirements. 'Requirements' can = be written at many levels, but I thought the stage we got to at the = last IETF was a need to identify exactly a 'set of capabilities' that = we wished to be able to provide with SIP.

In terms of the overall problems that these = capabilities solve: they enable certain services, some of which are = discussed in the draft and some of which have been the motivation for = the more specific mechanisms previously proposed and listed in the = references. We don't in IETF define a set of services, and then design = protocol to implement these, but rather identify generic capabilities = which might be useful to service designers.

This is where we were after IETF-52, with support for = definition of a generic capability for request history. Once we have = agreement on the details of what this capability does, *then* we can = proceed to the next stage of identifying what technical problems need = to be solved from a protocol perspective in order to provide achieve = this.

We thought this was the required process under = sipchange - define the capability first in SIPPING and work on the = solution later in SIP.

Are you questioning the rationalle for the whole = capability, or do you expect the requirements to go into more detail as = to the effect on the protocol ?

Regards,

Mark



> -----Original Message-----
> From: Tom_Gray@Mitel.COM [mailto:Tom_Gray@Mitel.COM]=
> Sent: 25 February 2002 15:45
> To: sipping@ietf.org; Tom_Gray@Mitel.COM
> Subject: Re: [Sipping] I-D
> = ACTION:draft-watson-sipping-req-history-00.txt
>
>
>
>
> From:  Tom Gray@MITEL on 02/25/2002 10:44 = AM
>
>
> Looking at this document does not reveal a list = of
> requirements but just a list
> of proposed capabilities that may or may not = require
> additions to the SIP
> protocol. For this to be a useful requirement = document, this list of
> capabilities must be linked to a set of = problems that need to
> be solved. Indeed
> the list of problems would be the nub of the = document. For
> example what problem
> is the 'capability-req' capability intended to = solve. Just
> what is 'Issuer-req'?
> The description in the draft doesn't specify = this beyond not
> identifying he
> reason fro its proposal. What trade-offs if any = will  be
> required for the
> solution to these problems?
>
>
>
>
>
>
> Internet-Drafts@ietf.org on 02/25/2002 06:22:20 = AM
>
> Please respond to = Internet-Drafts@ietf.org
>
> To:   ietf.org
> cc:   sipping@ietf.org (bcc: Tom = Gray/Kan/Mitel)
>
> Subject:  [Sipping] I-D = ACTION:draft-watson-sipping-req-history-00.txt
>
>
>
> A New Internet-Draft is available from the = on-line
> Internet-Drafts directories.
>
>
>      = Title          : Generic = Request History Capability -
> Requirements
>      Author(s) : M. = Watson et al.
>      Filename  : = draft-watson-sipping-req-history-00.txt
>      = Pages          : 7
>      = Date      : 22-Feb-02
>
> Many services that SIP is anticipated to = support require the
> ability to determine why and how the call = arrived at a specific
> application.  Examples of such services = include (but are not limited
> to) sessions initiated to call centers via = 'click to talk' SIP URLs
> on a web page, 'call history/logging' style = services within
> intelligent 'call management' software for SIP = UAs and calls to
> voicemail servers and call centers.
>
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-watson-sippi= ng-req-h
istory-00.txt

To remove yourself from the IETF Announcement list, = send a message to
ietf-announce-request with the word unsubscribe in = the body of the message.

Internet-Drafts are also available by anonymous FTP. = Login with the username
"anonymous" and a password of your e-mail = address. After logging in,
type "cd internet-drafts" and then
     "get = draft-watson-sipping-req-history-00.txt".

A list of Internet-Drafts directories can be found = in
http://www.ietf.org/shadow.html
or ftp://ftp.ietf.org/ietf/1shadow-sites.txt


Internet-Drafts can also be obtained by = e-mail.

Send a message to:
     mailserv@ietf.org.
In the body type:
     "FILE = /internet-drafts/draft-watson-sipping-req-history-00.txt".

NOTE:     The mail server at = ietf.org can return the document in
     MIME-encoded form by using = the "mpack" utility.  To use this
     feature, insert the command = "ENCODING mime" before the "FILE"
     command.  To decode = the response(s), you will need "munpack" or
     a MIME-compliant mail = reader.  Different MIME-compliant mail readers
     exhibit different behavior, = especially when dealing with
     "multipart" MIME = messages (i.e. documents which have been split
     up into multiple messages), = so check your local documentation on
     how to manipulate these = messages.


Below is the data which will enable a MIME compliant = mail reader
implementation to automatically retrieve the ASCII = version of the
Internet-Draft.

mailto:mailserv@ietf.org
ftp://ftp.ietf.org/internet-drafts/draft-watson-sippin= g-req-history-00.txt







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

------_=_NextPart_001_01C1BE25.7B9A4620-- _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Mon Feb 25 14:25:22 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id OAA26065 for ; Mon, 25 Feb 2002 14:25:22 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id OAA12473 for sipping-archive@odin.ietf.org; Mon, 25 Feb 2002 14:25:26 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id OAA12054; Mon, 25 Feb 2002 14:11:57 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id OAA11947 for ; Mon, 25 Feb 2002 14:11:51 -0500 (EST) Received: from Mitel.COM ([216.191.234.70]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id OAA25492 for ; Mon, 25 Feb 2002 14:11:33 -0500 (EST) From: Tom_Gray@Mitel.COM Received: from kanmta01.software.mitel.com (kanmta01.kanata.mitel.com [134.199.37.58]) by Mitel.COM (V8/MAIL-RELAY-2.1) with SMTP id OAA20168; Mon, 25 Feb 2002 14:10:13 -0500 (EST) Received: by kanmta01.software.mitel.com(Lotus SMTP MTA v4.6.7 (934.1 12-30-1999)) id 85256B6B.00694C47 ; Mon, 25 Feb 2002 14:10:08 -0500 X-Lotus-FromDomain: MITEL To: "Mark Watson" cc: sipping@ietf.org, Tom_Gray@Mitel.COM Message-ID: <85256B6B.00694A4E.00@kanmta01.software.mitel.com> Date: Mon, 25 Feb 2002 14:10:03 -0500 Subject: RE: [Sipping] I-D ACTION:draft-watson-sipping-req-history-00.txt Mime-Version: 1.0 Content-type: multipart/mixed; Boundary="0__=BM95s0xSWeivQ6XFdF93MUOAJogVpAeSzluawYg0hj5WfMCIoSbI1AIn" Content-Disposition: inline Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org --0__=BM95s0xSWeivQ6XFdF93MUOAJogVpAeSzluawYg0hj5WfMCIoSbI1AIn Content-type: text/plain; charset=us-ascii Content-Disposition: inline From: Tom Gray@MITEL on 02/25/2002 02:10 PM Reading this document, I do not see how anyone could assess its worth. What are the capabilities identified in this document supposed to accomplish? This understanding is what requirements documents are supposed to provide. I would recommend that the requirements be specific enough in their problem statement so that: a) it could be determined whether or not there is a problem that is of high enough priority that it needs to be solved. The outcome would naturally be an agreed upon problem statement. Intervenors could be invited to submit proposed solutions. b) the worth of proposed solutions could be assessed by comparing their ability to solve the agreed upon problems I see a very general description of a problem area in the document. However in my opinion this could be greatly improved by providing scenarios which illustrate specific deficiencies that are not currently solvable with the standard SIP protocol. Without this, I do not see how anyone can discuss the document. What basis would anyone have for accepting or rejecting it? "Mark Watson" on 02/25/2002 12:54:33 PM To: Tom Gray/Kan/Mitel@Mitel, sipping@ietf.org cc: Subject: RE: [Sipping] I-D ACTION:draft-watson-sipping-req-history-00.txt Tony, I think it depends what stage in the process you think we are at, and what you mean by requirements. 'Requirements' can be written at many levels, but I thought the stage we got to at the last IETF was a need to identify exactly a 'set of capabilities' that we wished to be able to provide with SIP. In terms of the overall problems that these capabilities solve: they enable certain services, some of which are discussed in the draft and some of which have been the motivation for the more specific mechanisms previously proposed and listed in the references. We don't in IETF define a set of services, and then design protocol to implement these, but rather identify generic capabilities which might be useful to service designers. This is where we were after IETF-52, with support for definition of a generic capability for request history. Once we have agreement on the details of what this capability does, *then* we can proceed to the next stage of identifying what technical problems need to be solved from a protocol perspective in order to provide achieve this. We thought this was the required process under sipchange - define the capability first in SIPPING and work on the solution later in SIP. Are you questioning the rationalle for the whole capability, or do you expect the requirements to go into more detail as to the effect on the protocol ? Regards, Mark > -----Original Message----- > From: Tom_Gray@Mitel.COM [mailto:Tom_Gray@Mitel.COM] > Sent: 25 February 2002 15:45 > To: sipping@ietf.org; Tom_Gray@Mitel.COM > Subject: Re: [Sipping] I-D > ACTION:draft-watson-sipping-req-history-00.txt > > > > > From: Tom Gray@MITEL on 02/25/2002 10:44 AM > > > Looking at this document does not reveal a list of > requirements but just a list > of proposed capabilities that may or may not require > additions to the SIP > protocol. For this to be a useful requirement document, this list of > capabilities must be linked to a set of problems that need to > be solved. Indeed > the list of problems would be the nub of the document. For > example what problem > is the 'capability-req' capability intended to solve. Just > what is 'Issuer-req'? > The description in the draft doesn't specify this beyond not > identifying he > reason fro its proposal. What trade-offs if any will be > required for the > solution to these problems? > > > > > > > Internet-Drafts@ietf.org on 02/25/2002 06:22:20 AM > > Please respond to Internet-Drafts@ietf.org > > To: ietf.org > cc: sipping@ietf.org (bcc: Tom Gray/Kan/Mitel) > > Subject: [Sipping] I-D ACTION:draft-watson-sipping-req-history-00.txt > > > > A New Internet-Draft is available from the on-line > Internet-Drafts directories. > > > Title : Generic Request History Capability - > Requirements > Author(s) : M. Watson et al. > Filename : draft-watson-sipping-req-history-00.txt > Pages : 7 > Date : 22-Feb-02 > > Many services that SIP is anticipated to support require the > ability to determine why and how the call arrived at a specific > application. Examples of such services include (but are not limited > to) sessions initiated to call centers via 'click to talk' SIP URLs > on a web page, 'call history/logging' style services within > intelligent 'call management' software for SIP UAs and calls to > voicemail servers and call centers. > > A URL for this Internet-Draft is: > http://www.ietf.org/internet-drafts/draft-watson-sipping-req-h istory-00.txt To remove yourself from the IETF Announcement list, send a message to ietf-announce-request with the word unsubscribe in the body of the message. Internet-Drafts are also available by anonymous FTP. Login with the username "anonymous" and a password of your e-mail address. After logging in, type "cd internet-drafts" and then "get draft-watson-sipping-req-history-00.txt". A list of Internet-Drafts directories can be found in http://www.ietf.org/shadow.html or ftp://ftp.ietf.org/ietf/1shadow-sites.txt Internet-Drafts can also be obtained by e-mail. Send a message to: mailserv@ietf.org. 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dGYub3JnIGZvciBuZXcgZGV2ZWxvcG1lbnRzIG9mIGNvcmUgU0lQPC9GT05UPg0KPC9QPg0KDQo8 L0JPRFk+DQo8L0hUTUw+DQo= --0__=BM95s0xSWeivQ6XFdF93MUOAJogVpAeSzluawYg0hj5WfMCIoSbI1AIn-- _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Mon Feb 25 15:13:33 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id PAA28087 for ; Mon, 25 Feb 2002 15:13:33 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id PAA15535 for sipping-archive@odin.ietf.org; Mon, 25 Feb 2002 15:13:36 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id OAA14249; Mon, 25 Feb 2002 14:56:05 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id OAA14222 for ; Mon, 25 Feb 2002 14:56:03 -0500 (EST) Received: from sj-msg-core-4.cisco.com (sj-msg-core-4.cisco.com [171.71.163.10]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id OAA27371 for ; Mon, 25 Feb 2002 14:55:58 -0500 (EST) Received: from nisser.cisco.com (nisser.cisco.com [171.71.176.85]) by sj-msg-core-4.cisco.com (8.11.3/8.9.1) with ESMTP id g1PJtUt19262; Mon, 25 Feb 2002 11:55:31 -0800 (PST) Received: from cisco.com (ssh-sj1.cisco.com [171.68.225.134]) by nisser.cisco.com (8.8.6 (PHNE_14041)/CISCO.SERVER.1.2) with ESMTP id LAA27503; Mon, 25 Feb 2002 11:55:27 -0800 (PST) Message-ID: <3C7A96B4.B97384D8@cisco.com> Date: Mon, 25 Feb 2002 14:55:32 -0500 From: Flemming Andreasen X-Mailer: Mozilla 4.79 [en]C-CCK-MCD (Windows NT 5.0; U) X-Accept-Language: en MIME-Version: 1.0 To: Mark Watson CC: sipping@ietf.org Subject: Re: [Sipping] Privacy comment (was Re: calling number blocking atsip/isup gate way) References: Content-Type: text/plain; charset=us-ascii Content-Transfer-Encoding: 7bit Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit Mark Watson wrote: > > > So, in the absence of further comments, could the editor add this to > the next version of the privacy draft ? (on the silence=agree/don't > care principle) Will do. -- Flemming > > > ...Mark > > > -----Original Message----- > > From: Dean Willis [mailto:dean.willis@softarmor.com] > > Sent: 20 February 2002 00:27 > > To: Watson, Mark [MDN05:EP10:EXCH]; sipping@ietf.org > > Subject: Re: [Sipping] Privacy comment (was Re: calling > > number blocking > > at sip/isup gate way) > > > > > > I concur that your proposal would add to the utility of RPID. > > > > -- > > Dean > > > > ----- Original Message ----- > > From: "Mark Watson" > > To: > > Sent: Tuesday, February 19, 2002 2:28 PM > > Subject: [Sipping] Privacy comment (was Re: calling number blocking > at > > sip/isup gate way) > > > > > > > As presently described in privacy-03, the Remote-Party-ID > > header is only > > > ever inserted by proxies, based on the proxy's knowledge of > > the identity > > of > > > the UA (which it determines from some authentication > > mechanism out of > > scope > > > of the privacy draft). > > > > > > Privacy-02 allowed the UA to insert Remote-Party-ID too. I > > think this > > should > > > be reintroduced for the *specific case* of a UA with > > multiple identities, > > to > > > allow the UA to choose which identity it wishes to be known > > by. The Proxy > > > would still be responsible for ensuring that this identity > > was valid for > > the > > > user and would still use some other authentication scheme > > to authenticate > > > the user. The proxy would need knowledge of which > > identities were valid > > for > > > which users. > > > > > > I believe this is a requirement for 3GPP, where a UA can > > have multiple > > > 'public identities' and chooses which of these it wishes to > > be known by > > for > > > each call. The proxy authenticates the UA by some other > > means and then > > > checks the subscription data to ensure it is really allowed > > to use the > > > public Identity it has included. > > > > > > I've suggested this a few times, and there has not been > > much comment for > > or > > > against. It would be good if we could get a resolution > > before the next > > > version of privacy is published. > > > > > > So, comments ? > > > > > > Regards...Mark > > > > > > > > > > > > > > > > > > > -----Original Message----- > > > > From: Dean Willis [mailto:dean.willis@softarmor.com] > > > > Sent: 18 February 2002 23:12 > > > > To: Watson, Mark [MDN05:EP10:EXCH]; Mpierce1@aol.com; > > > > jon.peterson@neustar.biz; sipping@ietf.org > > > > Subject: Re: [Sipping] RE: [Sip] calling number blocking > > at sip/isup > > > > gateway > > > > > > > > > > > > Mark said: > > > > > > > > > Your example below needs only a small extension to > > > > illustrate Mike's point > > > > > and show why something like Remote-Party-ID as defined in > > > > the privacy > > > > draft > > > > > is needed, and how it works despite the lack of a signature. > > > > > > > > Yep. And what happens when my misconfigured proxy forwards > > > > the call without > > > > paying attention to the Privacy header, because for some > > reason it's > > > > forwarding to the outside world instead of a PSTN gateway > > > > under my control? > > > > We get a privacy leak. > > > > > > > > But other than that, I'd say you've precisely described a > > > > "sunny day" use > > > > case for RPID. > > > > > > > > It still has a lot of security issues based on transitive > > > > trust, which make > > > > it a lot more palatable in a two-party case than in three or > > > > more party > > > > transitivity. > > > > > > > > I believe we're approaching a general consensus that the > > Privacy draft > > > > provides a reasonable approach to solving a specific > > class of problem > > > > wherein interoperator-transitive trust is couple to > > > > transport-layer message > > > > protection. That's not a general privacy mechanism, but it > > > > does seem to have > > > > enough constituency to justify publication as long as it > > has a REALLY > > > > serious applicability statement explaining the particular > > > > requirements and > > > > environments for which it is useful. > > > > > > > > How does that sound? > > > > > > > > -- > > > > Dean > > > > > > > > > > > > > > > -- Flemming Andreasen Cisco Systems _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Mon Feb 25 15:20:18 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id PAA28348 for ; Mon, 25 Feb 2002 15:20:14 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id PAA15951 for sipping-archive@odin.ietf.org; Mon, 25 Feb 2002 15:20:18 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id PAA15161; Mon, 25 Feb 2002 15:06:59 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id PAA15130 for ; Mon, 25 Feb 2002 15:06:57 -0500 (EST) Received: from zrc2s0jx.nortelnetworks.com (zrc2s0jx.nortelnetworks.com [47.103.122.112]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id PAA27838 for ; Mon, 25 Feb 2002 15:06:53 -0500 (EST) Received: from zrc2c011.us.nortel.com (zrc2c011.us.nortel.com [47.103.120.51]) by zrc2s0jx.nortelnetworks.com (Switch-2.2.0/Switch-2.2.0) with ESMTP id g1PK6Jb29063; Mon, 25 Feb 2002 14:06:19 -0600 (CST) Received: by zrc2c011.us.nortel.com with Internet Mail Service (5.5.2653.19) id ; Mon, 25 Feb 2002 14:06:19 -0600 Message-ID: <1B54FA3A2709D51195C800508BF9386A0313EE92@zrc2c000.us.nortel.com> From: "Mary Barnes" To: "'Tom_Gray@Mitel.COM'" , "Mark Watson" Cc: sipping@ietf.org Subject: RE: [Sipping] I-D ACTION:draft-watson-sipping-req-history-00.txt Date: Mon, 25 Feb 2002 14:06:16 -0600 MIME-Version: 1.0 X-Mailer: Internet Mail Service (5.5.2653.19) Content-Type: multipart/alternative; boundary="----_=_NextPart_001_01C1BE37.E1AC78E0" Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org This message is in MIME format. Since your mail reader does not understand this format, some or all of this message may not be legible. ------_=_NextPart_001_01C1BE37.E1AC78E0 Content-Type: text/plain; charset="iso-8859-1" Tom, As Mark states, this document wasn't intended to prescribe the requirements of a solution to solve specific problems or scenarios, but was intended to focus on the requirements of the capabilities required in SIP (protocol) to be able to use SIP as a solution for specific scenarios. As you say, we've been quite sketchy in describing the scenarios as we had assumed the ones that we did use were fairly well known scenarios to the audience. In a sense this document is a guinea pig for the new SIPPING WG process and will need refinement. We can certainly back up yet another step and add all the details for these scenarios and highlight how these scenarios would use the capabilities that have been specified. But, I would think that would be a rather lengthy document at a fairly high level of abstraction as to not be particularly useful either towards the end goal; with the end goal of this draft being an acknowledgement that there is not a "Request History" capability in SIP and that this is necessary in order to be able to use SIP for certain services. We do recognize that the level of abstraction of the current document is fairly high and we see value in taking it to the next step of evaluating different solution proposals applied against the required capabilities including specific scenarios to demonstrate the proposed solutions. Certainly, this step would be open to all "intervenors". I think the basic question being asked here is whether we want to add more to the requirements process by requiring a detailed description and agreement of a problem domain prior to proposing requirements for the capabilities required in the SIP protocol to solve a given problem? Do we have any other opinions, particularly from the WG chairs? Regards, Mary H. Barnes mbarnes@nortelnetworks.com -----Original Message----- From: Tom_Gray@Mitel.COM [mailto:Tom_Gray@Mitel.COM] Sent: Monday, February 25, 2002 1:10 PM To: Watson, Mark [MDN05:EP10:EXCH] Cc: sipping@ietf.org; Tom_Gray@Mitel.COM Subject: RE: [Sipping] I-D ACTION:draft-watson-sipping-req-history-00.txt From: Tom Gray@MITEL on 02/25/2002 02:10 PM Reading this document, I do not see how anyone could assess its worth. What are the capabilities identified in this document supposed to accomplish? This understanding is what requirements documents are supposed to provide. I would recommend that the requirements be specific enough in their problem statement so that: a) it could be determined whether or not there is a problem that is of high enough priority that it needs to be solved. The outcome would naturally be an agreed upon problem statement. Intervenors could be invited to submit proposed solutions. b) the worth of proposed solutions could be assessed by comparing their ability to solve the agreed upon problems I see a very general description of a problem area in the document. However in my opinion this could be greatly improved by providing scenarios which illustrate specific deficiencies that are not currently solvable with the standard SIP protocol. Without this, I do not see how anyone can discuss the document. What basis would anyone have for accepting or rejecting it? "Mark Watson" on 02/25/2002 12:54:33 PM To: Tom Gray/Kan/Mitel@Mitel, sipping@ietf.org cc: Subject: RE: [Sipping] I-D ACTION:draft-watson-sipping-req-history-00.txt Tony, I think it depends what stage in the process you think we are at, and what you mean by requirements. 'Requirements' can be written at many levels, but I thought the stage we got to at the last IETF was a need to identify exactly a 'set of capabilities' that we wished to be able to provide with SIP. In terms of the overall problems that these capabilities solve: they enable certain services, some of which are discussed in the draft and some of which have been the motivation for the more specific mechanisms previously proposed and listed in the references. We don't in IETF define a set of services, and then design protocol to implement these, but rather identify generic capabilities which might be useful to service designers. This is where we were after IETF-52, with support for definition of a generic capability for request history. Once we have agreement on the details of what this capability does, *then* we can proceed to the next stage of identifying what technical problems need to be solved from a protocol perspective in order to provide achieve this. We thought this was the required process under sipchange - define the capability first in SIPPING and work on the solution later in SIP. Are you questioning the rationalle for the whole capability, or do you expect the requirements to go into more detail as to the effect on the protocol ? Regards, Mark > -----Original Message----- > From: Tom_Gray@Mitel.COM [mailto:Tom_Gray@Mitel.COM] > Sent: 25 February 2002 15:45 > To: sipping@ietf.org; Tom_Gray@Mitel.COM > Subject: Re: [Sipping] I-D > ACTION:draft-watson-sipping-req-history-00.txt > > > > > From: Tom Gray@MITEL on 02/25/2002 10:44 AM > > > Looking at this document does not reveal a list of > requirements but just a list > of proposed capabilities that may or may not require > additions to the SIP > protocol. For this to be a useful requirement document, this list of > capabilities must be linked to a set of problems that need to > be solved. Indeed > the list of problems would be the nub of the document. For > example what problem > is the 'capability-req' capability intended to solve. Just > what is 'Issuer-req'? > The description in the draft doesn't specify this beyond not > identifying he > reason fro its proposal. What trade-offs if any will be > required for the > solution to these problems? > > > > > > > Internet-Drafts@ietf.org on 02/25/2002 06:22:20 AM > > Please respond to Internet-Drafts@ietf.org > > To: ietf.org > cc: sipping@ietf.org (bcc: Tom Gray/Kan/Mitel) > > Subject: [Sipping] I-D ACTION:draft-watson-sipping-req-history-00.txt > > > > A New Internet-Draft is available from the on-line > Internet-Drafts directories. > > > Title : Generic Request History Capability - > Requirements > Author(s) : M. Watson et al. > Filename : draft-watson-sipping-req-history-00.txt > Pages : 7 > Date : 22-Feb-02 > > Many services that SIP is anticipated to support require the > ability to determine why and how the call arrived at a specific > application. Examples of such services include (but are not limited > to) sessions initiated to call centers via 'click to talk' SIP URLs > on a web page, 'call history/logging' style services within > intelligent 'call management' software for SIP UAs and calls to > voicemail servers and call centers. > > A URL for this Internet-Draft is: > http://www.ietf.org/internet-drafts/draft-watson-sipping-req-h istory-00.txt To remove yourself from the IETF Announcement list, send a message to ietf-announce-request with the word unsubscribe in the body of the message. Internet-Drafts are also available by anonymous FTP. Login with the username "anonymous" and a password of your e-mail address. After logging in, type "cd internet-drafts" and then "get draft-watson-sipping-req-history-00.txt". A list of Internet-Drafts directories can be found in http://www.ietf.org/shadow.html or ftp://ftp.ietf.org/ietf/1shadow-sites.txt Internet-Drafts can also be obtained by e-mail. Send a message to: mailserv@ietf.org. In the body type: "FILE /internet-drafts/draft-watson-sipping-req-history-00.txt". NOTE: The mail server at ietf.org can return the document in MIME-encoded form by using the "mpack" utility. To use this feature, insert the command "ENCODING mime" before the "FILE" command. To decode the response(s), you will need "munpack" or a MIME-compliant mail reader. Different MIME-compliant mail readers exhibit different behavior, especially when dealing with "multipart" MIME messages (i.e. documents which have been split up into multiple messages), so check your local documentation on how to manipulate these messages. Below is the data which will enable a MIME compliant mail reader implementation to automatically retrieve the ASCII version of the Internet-Draft. mailto:mailserv@ietf.org ftp://ftp.ietf.org/internet-drafts/draft-watson-sipping-req-history-00.txt _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP ------_=_NextPart_001_01C1BE37.E1AC78E0 Content-Type: text/html; charset="iso-8859-1" Content-Transfer-Encoding: quoted-printable RE: [Sipping] I-D = ACTION:draft-watson-sipping-req-history-00.txt

Tom,

As Mark states, this document wasn't intended to = prescribe the requirements of a solution to solve specific problems or = scenarios, but was intended to focus on the requirements of the = capabilities required in SIP (protocol) to be able to use SIP as a = solution for specific scenarios. 

As you say, we've been quite sketchy in describing = the scenarios as we had assumed the ones that we did use were fairly = well known scenarios to the audience.  In a sense this document is = a guinea pig for the new SIPPING WG process and will need = refinement.  We can certainly back up yet another step and add all = the details for these scenarios and highlight how these scenarios would = use the capabilities that have been specified.  But, I would think = that would be a rather lengthy document at a fairly high level of = abstraction as to not be particularly useful either towards the end = goal; with the end goal of this draft being an acknowledgement that = there is not a "Request History" capability in SIP and that = this is necessary in order to be able to use SIP for certain = services.  We do recognize that the level of abstraction of the = current document is fairly high and we see value in taking it to the = next step of evaluating different solution proposals applied against = the required capabilities including specific scenarios to demonstrate = the proposed solutions. Certainly, this step would be open to all = "intervenors".

I think the basic question being asked here is = whether we want to add more to the requirements process by requiring a = detailed description and agreement of a problem domain prior to = proposing requirements for the capabilities required in the SIP = protocol to solve a given problem? Do we have any other opinions, = particularly from the WG chairs?

Regards,
Mary H. Barnes
mbarnes@nortelnetworks.com


-----Original Message-----
From: Tom_Gray@Mitel.COM [mailto:Tom_Gray@Mitel.COM]=
Sent: Monday, February 25, 2002 1:10 PM
To: Watson, Mark [MDN05:EP10:EXCH]
Cc: sipping@ietf.org; Tom_Gray@Mitel.COM
Subject: RE: [Sipping] I-D
ACTION:draft-watson-sipping-req-history-00.txt




From:  Tom Gray@MITEL on 02/25/2002 02:10 = PM

Reading this document, I do not see how anyone could = assess its worth. What are
the capabilities identified in this document = supposed to accomplish? This
understanding is what requirements documents are = supposed to provide. I would
recommend that the requirements be specific  = enough in their problem statement
so that:

a) it could be determined whether or not there is a = problem that is of high
enough priority that it needs to be solved. The = outcome would naturally be an
agreed upon problem statement. Intervenors could be = invited to submit proposed
solutions.

b) the worth of proposed solutions could be assessed = by comparing their ability
to solve the agreed upon problems

I see a very general description of a problem area in = the document. However in
my opinion this could  be greatly improved by = providing scenarios which
illustrate specific deficiencies that are not = currently solvable with the
standard SIP protocol. Without this, I do not see = how anyone can discuss the
document. What basis would anyone have for accepting = or rejecting it?






"Mark Watson" = <mwatson@nortelnetworks.com> on 02/25/2002 12:54:33 PM

To:   Tom Gray/Kan/Mitel@Mitel, = sipping@ietf.org
cc:

Subject:  RE: [Sipping] I-D = ACTION:draft-watson-sipping-req-history-00.txt



Tony,

I think it depends what stage in the process you = think we are at, and what
you mean by requirements. 'Requirements' can be = written at many levels, but
I thought the stage we got to at the last IETF was a = need to identify
exactly a 'set of capabilities' that we wished to be = able to provide with
SIP.

In terms of the overall problems that these = capabilities solve: they enable
certain services, some of which are discussed in the = draft and some of which
have been the motivation for the more specific = mechanisms previously
proposed and listed in the references. We don't in = IETF define a set of
services, and then design protocol to implement = these, but rather identify
generic capabilities which might be useful to = service designers.

This is where we were after IETF-52, with support for = definition of a
generic capability for request history. Once we have = agreement on the
details of what this capability does, *then* we can = proceed to the next
stage of identifying what technical problems need to = be solved from a
protocol perspective in order to provide achieve = this.

We thought this was the required process under = sipchange - define the
capability first in SIPPING and work on the solution = later in SIP.

Are you questioning the rationalle for the whole = capability, or do you
expect the requirements to go into more detail as to = the effect on the
protocol ?

Regards,

Mark



> -----Original Message-----
> From: Tom_Gray@Mitel.COM [mailto:Tom_Gray@Mitel.COM]=
> Sent: 25 February 2002 15:45
> To: sipping@ietf.org; Tom_Gray@Mitel.COM
> Subject: Re: [Sipping] I-D
> = ACTION:draft-watson-sipping-req-history-00.txt
>
>
>
>
> From:  Tom Gray@MITEL on 02/25/2002 10:44 = AM
>
>
> Looking at this document does not reveal a list = of
> requirements but just a list
> of proposed capabilities that may or may not = require
> additions to the SIP
> protocol. For this to be a useful requirement = document, this list of
> capabilities must be linked to a set of = problems that need to
> be solved. Indeed
> the list of problems would be the nub of the = document. For
> example what problem
> is the 'capability-req' capability intended to = solve. Just
> what is 'Issuer-req'?
> The description in the draft doesn't specify = this beyond not
> identifying he
> reason fro its proposal. What trade-offs if any = will  be
> required for the
> solution to these problems?
>
>
>
>
>
>
> Internet-Drafts@ietf.org on 02/25/2002 06:22:20 = AM
>
> Please respond to = Internet-Drafts@ietf.org
>
> To:   ietf.org
> cc:   sipping@ietf.org (bcc: Tom = Gray/Kan/Mitel)
>
> Subject:  [Sipping] I-D = ACTION:draft-watson-sipping-req-history-00.txt
>
>
>
> A New Internet-Draft is available from the = on-line
> Internet-Drafts directories.
>
>
>      = Title          : Generic = Request History Capability -
> Requirements
>      Author(s) : M. = Watson et al.
>      Filename  : = draft-watson-sipping-req-history-00.txt
>      = Pages          : 7
>      = Date      : 22-Feb-02
>
> Many services that SIP is anticipated to = support require the
> ability to determine why and how the call = arrived at a specific
> application.  Examples of such services = include (but are not limited
> to) sessions initiated to call centers via = 'click to talk' SIP URLs
> on a web page, 'call history/logging' style = services within
> intelligent 'call management' software for SIP = UAs and calls to
> voicemail servers and call centers.
>
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-watson-sippi= ng-req-h
istory-00.txt

To remove yourself from the IETF Announcement list, = send a message to
ietf-announce-request with the word unsubscribe in = the body of the message.

Internet-Drafts are also available by anonymous FTP. = Login with the username
"anonymous" and a password of your e-mail = address. After logging in,
type "cd internet-drafts" and then
     "get = draft-watson-sipping-req-history-00.txt".

A list of Internet-Drafts directories can be found = in
http://www.ietf.org/shadow.html
or ftp://ftp.ietf.org/ietf/1shadow-sites.txt


Internet-Drafts can also be obtained by = e-mail.

Send a message to:
     mailserv@ietf.org.
In the body type:
     "FILE = /internet-drafts/draft-watson-sipping-req-history-00.txt".

NOTE:     The mail server at = ietf.org can return the document in
     MIME-encoded form by using = the "mpack" utility.  To use this
     feature, insert the command = "ENCODING mime" before the "FILE"
     command.  To decode = the response(s), you will need "munpack" or
     a MIME-compliant mail = reader.  Different MIME-compliant mail readers
     exhibit different behavior, = especially when dealing with
     "multipart" MIME = messages (i.e. documents which have been split
     up into multiple messages), = so check your local documentation on
     how to manipulate these = messages.


Below is the data which will enable a MIME compliant = mail reader
implementation to automatically retrieve the ASCII = version of the
Internet-Draft.

mailto:mailserv@ietf.org
ftp://ftp.ietf.org/internet-drafts/draft-watson-sippin= g-req-history-00.txt







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

------_=_NextPart_001_01C1BE37.E1AC78E0-- _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Mon Feb 25 15:46:20 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id PAA29306 for ; Mon, 25 Feb 2002 15:46:16 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id PAA17315 for sipping-archive@odin.ietf.org; Mon, 25 Feb 2002 15:46:18 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id PAA16802; Mon, 25 Feb 2002 15:33:40 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id PAA16701 for ; Mon, 25 Feb 2002 15:33:34 -0500 (EST) Received: from Mitel.COM ([216.191.234.70]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id PAA28920 for ; Mon, 25 Feb 2002 15:33:28 -0500 (EST) From: Tom_Gray@Mitel.COM Received: from kanmta01.software.mitel.com (kanmta01.kanata.mitel.com [134.199.37.58]) by Mitel.COM (V8/MAIL-RELAY-2.1) with SMTP id PAA01236; Mon, 25 Feb 2002 15:31:19 -0500 (EST) Received: by kanmta01.software.mitel.com(Lotus SMTP MTA v4.6.7 (934.1 12-30-1999)) id 85256B6B.0070B52F ; Mon, 25 Feb 2002 15:31:04 -0500 X-Lotus-FromDomain: MITEL To: "Mark Watson" , sipping@ietf.org, Tom_Gray@Mitel.COM Message-ID: <85256B6B.0070B44B.00@kanmta01.software.mitel.com> Date: Mon, 25 Feb 2002 15:31:04 -0500 Subject: RE: [Sipping] I-D ACTION:draft-watson-sipping-req-history-00.txt Mime-Version: 1.0 Content-type: multipart/mixed; Boundary="0__=EulXsDROgeRypKz0P9VFvS91ZKYukU1LdddejxHroEspMFAYUpAsbo0O" Content-Disposition: inline Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org --0__=EulXsDROgeRypKz0P9VFvS91ZKYukU1LdddejxHroEspMFAYUpAsbo0O Content-type: text/plain; charset=us-ascii Content-Disposition: inline From: Tom Gray@MITEL on 02/25/2002 03:31 PM It is my understanding that this document is about providing a traditional telephony call redirection service within SIP. If that is so, I would just like a requirements document for this to say that. If not, it would be nice to know what is the purpose of the capabilities described. The capabilites are describing a proposed solution to an unnamed problem Calling them requirements does not make them so. Given a derscription of the problem that is being addressed, there could be a determination then of the information that would be need to be carried by SIP for this service and whether or not this was a problem of broad enough significance that it would be addressed in the general protocol. "Mary Barnes" on 02/25/2002 03:06:16 PM To: Tom Gray/Kan/Mitel@Mitel, "Mark Watson" cc: sipping@ietf.org Subject: RE: [Sipping] I-D ACTION:draft-watson-sipping-req-history-00.txt Tom, As Mark states, this document wasn't intended to prescribe the requirements of a solution to solve specific problems or scenarios, but was intended to focus on the requirements of the capabilities required in SIP (protocol) to be able to use SIP as a solution for specific scenarios. As you say, we've been quite sketchy in describing the scenarios as we had assumed the ones that we did use were fairly well known scenarios to the audience. In a sense this document is a guinea pig for the new SIPPING WG process and will need refinement. We can certainly back up yet another step and add all the details for these scenarios and highlight how these scenarios would use the capabilities that have been specified. But, I would think that would be a rather lengthy document at a fairly high level of abstraction as to not be particularly useful either towards the end goal; with the end goal of this draft being an acknowledgement that there is not a "Request History" capability in SIP and that this is necessary in order to be able to use SIP for certain services. We do recognize that the level of abstraction of the current document is fairly high and we see value in taking it to the next step of evaluating different solution proposals applied against the required capabilities including specific scenarios to demonstrate the proposed solutions. Certainly, this step would be open to all "intervenors". I think the basic question being asked here is whether we want to add more to the requirements process by requiring a detailed description and agreement of a problem domain prior to proposing requirements for the capabilities required in the SIP protocol to solve a given problem? Do we have any other opinions, particularly from the WG chairs? Regards, Mary H. Barnes mbarnes@nortelnetworks.com -----Original Message----- From: Tom_Gray@Mitel.COM [mailto:Tom_Gray@Mitel.COM] Sent: Monday, February 25, 2002 1:10 PM To: Watson, Mark [MDN05:EP10:EXCH] Cc: sipping@ietf.org; Tom_Gray@Mitel.COM Subject: RE: [Sipping] I-D ACTION:draft-watson-sipping-req-history-00.txt From: Tom Gray@MITEL on 02/25/2002 02:10 PM Reading this document, I do not see how anyone could assess its worth. What are the capabilities identified in this document supposed to accomplish? This understanding is what requirements documents are supposed to provide. I would recommend that the requirements be specific enough in their problem statement so that: a) it could be determined whether or not there is a problem that is of high enough priority that it needs to be solved. The outcome would naturally be an agreed upon problem statement. Intervenors could be invited to submit proposed solutions. b) the worth of proposed solutions could be assessed by comparing their ability to solve the agreed upon problems I see a very general description of a problem area in the document. However in my opinion this could be greatly improved by providing scenarios which illustrate specific deficiencies that are not currently solvable with the standard SIP protocol. Without this, I do not see how anyone can discuss the document. What basis would anyone have for accepting or rejecting it? "Mark Watson" on 02/25/2002 12:54:33 PM To: Tom Gray/Kan/Mitel@Mitel, sipping@ietf.org cc: Subject: RE: [Sipping] I-D ACTION:draft-watson-sipping-req-history-00.txt Tony, I think it depends what stage in the process you think we are at, and what you mean by requirements. 'Requirements' can be written at many levels, but I thought the stage we got to at the last IETF was a need to identify exactly a 'set of capabilities' that we wished to be able to provide with SIP. In terms of the overall problems that these capabilities solve: they enable certain services, some of which are discussed in the draft and some of which have been the motivation for the more specific mechanisms previously proposed and listed in the references. We don't in IETF define a set of services, and then design protocol to implement these, but rather identify generic capabilities which might be useful to service designers. This is where we were after IETF-52, with support for definition of a generic capability for request history. Once we have agreement on the details of what this capability does, *then* we can proceed to the next stage of identifying what technical problems need to be solved from a protocol perspective in order to provide achieve this. We thought this was the required process under sipchange - define the capability first in SIPPING and work on the solution later in SIP. Are you questioning the rationalle for the whole capability, or do you expect the requirements to go into more detail as to the effect on the protocol ? Regards, Mark > -----Original Message----- > From: Tom_Gray@Mitel.COM [mailto:Tom_Gray@Mitel.COM] > Sent: 25 February 2002 15:45 > To: sipping@ietf.org; Tom_Gray@Mitel.COM > Subject: Re: [Sipping] I-D > ACTION:draft-watson-sipping-req-history-00.txt > > > > > From: Tom Gray@MITEL on 02/25/2002 10:44 AM > > > Looking at this document does not reveal a list of > requirements but just a list > of proposed capabilities that may or may not require > additions to the SIP > protocol. For this to be a useful requirement document, this list of > capabilities must be linked to a set of problems that need to > be solved. Indeed > the list of problems would be the nub of the document. For > example what problem > is the 'capability-req' capability intended to solve. Just > what is 'Issuer-req'? > The description in the draft doesn't specify this beyond not > identifying he > reason fro its proposal. What trade-offs if any will be > required for the > solution to these problems? > > > > > > > Internet-Drafts@ietf.org on 02/25/2002 06:22:20 AM > > Please respond to Internet-Drafts@ietf.org > > To: ietf.org > cc: sipping@ietf.org (bcc: Tom Gray/Kan/Mitel) > > Subject: [Sipping] I-D ACTION:draft-watson-sipping-req-history-00.txt > > > > A New Internet-Draft is available from the on-line > Internet-Drafts directories. > > > Title : Generic Request History Capability - > Requirements > Author(s) : M. Watson et al. > Filename : draft-watson-sipping-req-history-00.txt > Pages : 7 > Date : 22-Feb-02 > > Many services that SIP is anticipated to support require the > ability to determine why and how the call arrived at a specific > application. Examples of such services include (but are not limited > to) sessions initiated to call centers via 'click to talk' SIP URLs > on a web page, 'call history/logging' style services within > intelligent 'call management' software for SIP UAs and calls to > voicemail servers and call centers. > > A URL for this Internet-Draft is: > http://www.ietf.org/internet-drafts/draft-watson-sipping-req-h istory-00.txt To remove yourself from the IETF Announcement list, send a message to ietf-announce-request with the word unsubscribe in the body of the message. Internet-Drafts are also available by anonymous FTP. Login with the username "anonymous" and a password of your e-mail address. After logging in, type "cd internet-drafts" and then "get draft-watson-sipping-req-history-00.txt". A list of Internet-Drafts directories can be found in http://www.ietf.org/shadow.html or ftp://ftp.ietf.org/ietf/1shadow-sites.txt Internet-Drafts can also be obtained by e-mail. Send a message to: mailserv@ietf.org. 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sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Mon Feb 25 15:58:44 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id PAA29754 for ; Mon, 25 Feb 2002 15:58:39 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id PAA17804 for sipping-archive@odin.ietf.org; Mon, 25 Feb 2002 15:58:42 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id PAA17243; Mon, 25 Feb 2002 15:45:04 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id PAA17135 for ; Mon, 25 Feb 2002 15:44:58 -0500 (EST) Received: from zrc2s0jx.nortelnetworks.com (zrc2s0jx.nortelnetworks.com [47.103.122.112]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id PAA29275 for ; Mon, 25 Feb 2002 15:44:55 -0500 (EST) Received: from zrc2c011.us.nortel.com (zrc2c011.us.nortel.com [47.103.120.51]) by zrc2s0jx.nortelnetworks.com (Switch-2.2.0/Switch-2.2.0) with ESMTP id g1PKiGb08362; Mon, 25 Feb 2002 14:44:16 -0600 (CST) Received: by zrc2c011.us.nortel.com with Internet Mail Service (5.5.2653.19) id ; Mon, 25 Feb 2002 14:44:16 -0600 Message-ID: <1B54FA3A2709D51195C800508BF9386A0313EE96@zrc2c000.us.nortel.com> From: "Mary Barnes" To: "'Tom_Gray@Mitel.COM'" , "Mark Watson", sipping@ietf.org Subject: RE: [Sipping] I-D ACTION:draft-watson-sipping-req-history-00.txt Date: Mon, 25 Feb 2002 14:44:14 -0600 MIME-Version: 1.0 X-Mailer: Internet Mail Service (5.5.2653.19) Content-Type: multipart/alternative; boundary="----_=_NextPart_001_01C1BE3D.2FA50170" Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org This message is in MIME format. Since your mail reader does not understand this format, some or all of this message may not be legible. ------_=_NextPart_001_01C1BE3D.2FA50170 Content-Type: text/plain; charset="iso-8859-1" Tom, The intent of this document is not to provide a traditional telephony call redirection service within SIP. That might be just one use. Quite clearly we saw at IETF-52 that it was indeed a mistake to provide that as an example. The goal is to ensure that what gets progressed is a general purpose Request History mechanism which could be used for a variety of services. I do understand your concerns about this being too fuzzy and unspecific at this time as I myself have struggled with that lack of concreteness and the level of abstraction of the current draft. If you have specific suggestions on how to get more concrete on characterizing the nature of problems to which these requirements apply without going either into the solution domain or prescribing requirements for a narrow problem domain, that would be much appreciated. Thanks, Mary. -----Original Message----- From: Tom_Gray@Mitel.COM [mailto:Tom_Gray@Mitel.COM] Sent: Monday, February 25, 2002 2:31 PM To: Watson, Mark [MDN05:EP10:EXCH]; sipping@ietf.org; Tom_Gray@Mitel.COM Subject: RE: [Sipping] I-D ACTION:draft-watson-sipping-req-history-00.txt From: Tom Gray@MITEL on 02/25/2002 03:31 PM It is my understanding that this document is about providing a traditional telephony call redirection service within SIP. If that is so, I would just like a requirements document for this to say that. If not, it would be nice to know what is the purpose of the capabilities described. The capabilites are describing a proposed solution to an unnamed problem Calling them requirements does not make them so. Given a derscription of the problem that is being addressed, there could be a determination then of the information that would be need to be carried by SIP for this service and whether or not this was a problem of broad enough significance that it would be addressed in the general protocol. "Mary Barnes" on 02/25/2002 03:06:16 PM To: Tom Gray/Kan/Mitel@Mitel, "Mark Watson" cc: sipping@ietf.org Subject: RE: [Sipping] I-D ACTION:draft-watson-sipping-req-history-00.txt Tom, As Mark states, this document wasn't intended to prescribe the requirements of a solution to solve specific problems or scenarios, but was intended to focus on the requirements of the capabilities required in SIP (protocol) to be able to use SIP as a solution for specific scenarios. As you say, we've been quite sketchy in describing the scenarios as we had assumed the ones that we did use were fairly well known scenarios to the audience. In a sense this document is a guinea pig for the new SIPPING WG process and will need refinement. We can certainly back up yet another step and add all the details for these scenarios and highlight how these scenarios would use the capabilities that have been specified. But, I would think that would be a rather lengthy document at a fairly high level of abstraction as to not be particularly useful either towards the end goal; with the end goal of this draft being an acknowledgement that there is not a "Request History" capability in SIP and that this is necessary in order to be able to use SIP for certain services. We do recognize that the level of abstraction of the current document is fairly high and we see value in taking it to the next step of evaluating different solution proposals applied against the required capabilities including specific scenarios to demonstrate the proposed solutions. Certainly, this step would be open to all "intervenors". I think the basic question being asked here is whether we want to add more to the requirements process by requiring a detailed description and agreement of a problem domain prior to proposing requirements for the capabilities required in the SIP protocol to solve a given problem? Do we have any other opinions, particularly from the WG chairs? Regards, Mary H. Barnes mbarnes@nortelnetworks.com -----Original Message----- From: Tom_Gray@Mitel.COM [mailto:Tom_Gray@Mitel.COM] Sent: Monday, February 25, 2002 1:10 PM To: Watson, Mark [MDN05:EP10:EXCH] Cc: sipping@ietf.org; Tom_Gray@Mitel.COM Subject: RE: [Sipping] I-D ACTION:draft-watson-sipping-req-history-00.txt From: Tom Gray@MITEL on 02/25/2002 02:10 PM Reading this document, I do not see how anyone could assess its worth. What are the capabilities identified in this document supposed to accomplish? This understanding is what requirements documents are supposed to provide. I would recommend that the requirements be specific enough in their problem statement so that: a) it could be determined whether or not there is a problem that is of high enough priority that it needs to be solved. The outcome would naturally be an agreed upon problem statement. Intervenors could be invited to submit proposed solutions. b) the worth of proposed solutions could be assessed by comparing their ability to solve the agreed upon problems I see a very general description of a problem area in the document. However in my opinion this could be greatly improved by providing scenarios which illustrate specific deficiencies that are not currently solvable with the standard SIP protocol. Without this, I do not see how anyone can discuss the document. What basis would anyone have for accepting or rejecting it? "Mark Watson" on 02/25/2002 12:54:33 PM To: Tom Gray/Kan/Mitel@Mitel, sipping@ietf.org cc: Subject: RE: [Sipping] I-D ACTION:draft-watson-sipping-req-history-00.txt Tony, I think it depends what stage in the process you think we are at, and what you mean by requirements. 'Requirements' can be written at many levels, but I thought the stage we got to at the last IETF was a need to identify exactly a 'set of capabilities' that we wished to be able to provide with SIP. In terms of the overall problems that these capabilities solve: they enable certain services, some of which are discussed in the draft and some of which have been the motivation for the more specific mechanisms previously proposed and listed in the references. We don't in IETF define a set of services, and then design protocol to implement these, but rather identify generic capabilities which might be useful to service designers. This is where we were after IETF-52, with support for definition of a generic capability for request history. Once we have agreement on the details of what this capability does, *then* we can proceed to the next stage of identifying what technical problems need to be solved from a protocol perspective in order to provide achieve this. We thought this was the required process under sipchange - define the capability first in SIPPING and work on the solution later in SIP. Are you questioning the rationalle for the whole capability, or do you expect the requirements to go into more detail as to the effect on the protocol ? Regards, Mark > -----Original Message----- > From: Tom_Gray@Mitel.COM [mailto:Tom_Gray@Mitel.COM] > Sent: 25 February 2002 15:45 > To: sipping@ietf.org; Tom_Gray@Mitel.COM > Subject: Re: [Sipping] I-D > ACTION:draft-watson-sipping-req-history-00.txt > > > > > From: Tom Gray@MITEL on 02/25/2002 10:44 AM > > > Looking at this document does not reveal a list of > requirements but just a list > of proposed capabilities that may or may not require > additions to the SIP > protocol. For this to be a useful requirement document, this list of > capabilities must be linked to a set of problems that need to > be solved. Indeed > the list of problems would be the nub of the document. For > example what problem > is the 'capability-req' capability intended to solve. Just > what is 'Issuer-req'? > The description in the draft doesn't specify this beyond not > identifying he > reason fro its proposal. What trade-offs if any will be > required for the > solution to these problems? > > > > > > > Internet-Drafts@ietf.org on 02/25/2002 06:22:20 AM > > Please respond to Internet-Drafts@ietf.org > > To: ietf.org > cc: sipping@ietf.org (bcc: Tom Gray/Kan/Mitel) > > Subject: [Sipping] I-D ACTION:draft-watson-sipping-req-history-00.txt > > > > A New Internet-Draft is available from the on-line > Internet-Drafts directories. > > > Title : Generic Request History Capability - > Requirements > Author(s) : M. Watson et al. > Filename : draft-watson-sipping-req-history-00.txt > Pages : 7 > Date : 22-Feb-02 > > Many services that SIP is anticipated to support require the > ability to determine why and how the call arrived at a specific > application. Examples of such services include (but are not limited > to) sessions initiated to call centers via 'click to talk' SIP URLs > on a web page, 'call history/logging' style services within > intelligent 'call management' software for SIP UAs and calls to > voicemail servers and call centers. > > A URL for this Internet-Draft is: > http://www.ietf.org/internet-drafts/draft-watson-sipping-req-h istory-00.txt To remove yourself from the IETF Announcement list, send a message to ietf-announce-request with the word unsubscribe in the body of the message. Internet-Drafts are also available by anonymous FTP. Login with the username "anonymous" and a password of your e-mail address. After logging in, type "cd internet-drafts" and then "get draft-watson-sipping-req-history-00.txt". A list of Internet-Drafts directories can be found in http://www.ietf.org/shadow.html or ftp://ftp.ietf.org/ietf/1shadow-sites.txt Internet-Drafts can also be obtained by e-mail. Send a message to: mailserv@ietf.org. In the body type: "FILE /internet-drafts/draft-watson-sipping-req-history-00.txt". NOTE: The mail server at ietf.org can return the document in MIME-encoded form by using the "mpack" utility. To use this feature, insert the command "ENCODING mime" before the "FILE" command. To decode the response(s), you will need "munpack" or a MIME-compliant mail reader. Different MIME-compliant mail readers exhibit different behavior, especially when dealing with "multipart" MIME messages (i.e. documents which have been split up into multiple messages), so check your local documentation on how to manipulate these messages. Below is the data which will enable a MIME compliant mail reader implementation to automatically retrieve the ASCII version of the Internet-Draft. mailto:mailserv@ietf.org ftp://ftp.ietf.org/internet-drafts/draft-watson-sipping-req-history-00.txt _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP ------_=_NextPart_001_01C1BE3D.2FA50170 Content-Type: text/html; charset="iso-8859-1" Content-Transfer-Encoding: quoted-printable RE: [Sipping] I-D = ACTION:draft-watson-sipping-req-history-00.txt

Tom,

The intent of this document is not to provide a = traditional telephony call redirection service within SIP. That might = be just one use.  Quite clearly we saw at IETF-52 that it was = indeed a mistake to provide that as an example.  The goal is to = ensure that what gets progressed is a general purpose Request History = mechanism which could be used for a variety of services.

I do understand your concerns about this being too = fuzzy and unspecific at this time as I myself have struggled with that = lack of concreteness and the level of abstraction of the current = draft.  If you have specific suggestions on how to get more = concrete on characterizing the nature of problems to which these = requirements apply without going either into the solution domain or = prescribing requirements for a narrow problem domain, that would be = much appreciated.

Thanks,
Mary.

-----Original Message-----
From: Tom_Gray@Mitel.COM [mailto:Tom_Gray@Mitel.COM]=
Sent: Monday, February 25, 2002 2:31 PM
To: Watson, Mark [MDN05:EP10:EXCH]; = sipping@ietf.org; Tom_Gray@Mitel.COM
Subject: RE: [Sipping] I-D
ACTION:draft-watson-sipping-req-history-00.txt




From:  Tom Gray@MITEL on 02/25/2002 03:31 = PM

It is my understanding that this document is about = providing a traditional
telephony call redirection service within SIP. If = that is so, I would just like
a requirements document for this to say that.  = If not, it would be nice to know
what is the purpose of the capabilities = described.

The capabilites are describing a proposed solution to = an unnamed problem Calling
them requirements does not make them so.  Given = a derscription of the problem
that is being addressed,  there could be a = determination then of the information
that would be need to be carried by SIP for this = service and whether or not this
was a problem of broad enough significance that it = would be addressed in the
general protocol.




"Mary Barnes" = <mbarnes@nortelnetworks.com> on 02/25/2002 03:06:16 PM

To:   Tom Gray/Kan/Mitel@Mitel, "Mark = Watson" <mwatson@nortelnetworks.com>
cc:   sipping@ietf.org

Subject:  RE: [Sipping] I-D = ACTION:draft-watson-sipping-req-history-00.txt



Tom,

As Mark states, this document wasn't intended to = prescribe the requirements
of a solution to solve specific problems or = scenarios, but was intended to
focus on the requirements of the capabilities = required in SIP (protocol) to
be able to use SIP as a solution for specific = scenarios.

As you say, we've been quite sketchy in describing = the scenarios as we had
assumed the ones that we did use were fairly well = known scenarios to the
audience.  In a sense this document is a guinea = pig for the new SIPPING WG
process and will need refinement.  We can = certainly back up yet another step
and add all the details for these scenarios and = highlight how these
scenarios would use the capabilities that have been = specified.  But, I would
think that would be a rather lengthy document at a = fairly high level of
abstraction as to not be particularly useful either = towards the end goal;
with the end goal of this draft being an = acknowledgement that there is not a
"Request History" capability in SIP and = that this is necessary in order to
be able to use SIP for certain services.  We do = recognize that the level of
abstraction of the current document is fairly high = and we see value in
taking it to the next step of evaluating different = solution proposals
applied against the required capabilities including = specific scenarios to
demonstrate the proposed solutions. Certainly, this = step would be open to
all "intervenors".

I think the basic question being asked here is = whether we want to add more
to the requirements process by requiring a detailed = description and
agreement of a problem domain prior to proposing = requirements for the
capabilities required in the SIP protocol to solve a = given problem? Do we
have any other opinions, particularly from the WG = chairs?

Regards,
Mary H. Barnes
mbarnes@nortelnetworks.com


-----Original Message-----
From: Tom_Gray@Mitel.COM [mailto:Tom_Gray@Mitel.COM]=
Sent: Monday, February 25, 2002 1:10 PM
To: Watson, Mark [MDN05:EP10:EXCH]
Cc: sipping@ietf.org; Tom_Gray@Mitel.COM
Subject: RE: [Sipping] I-D
ACTION:draft-watson-sipping-req-history-00.txt




From:  Tom Gray@MITEL on 02/25/2002 02:10 = PM

Reading this document, I do not see how anyone could = assess its worth. What
are
the capabilities identified in this document = supposed to accomplish? This
understanding is what requirements documents are = supposed to provide. I
would
recommend that the requirements be specific  = enough in their problem
statement
so that:

a) it could be determined whether or not there is a = problem that is of high
enough priority that it needs to be solved. The = outcome would naturally be
an
agreed upon problem statement. Intervenors could be = invited to submit
proposed
solutions.

b) the worth of proposed solutions could be assessed = by comparing their
ability
to solve the agreed upon problems

I see a very general description of a problem area in = the document. However
in
my opinion this could  be greatly improved by = providing scenarios which
illustrate specific deficiencies that are not = currently solvable with the
standard SIP protocol. Without this, I do not see = how anyone can discuss the
document. What basis would anyone have for accepting = or rejecting it?






"Mark Watson" = <mwatson@nortelnetworks.com> on 02/25/2002 12:54:33 PM

To:   Tom Gray/Kan/Mitel@Mitel, = sipping@ietf.org
cc:

Subject:  RE: [Sipping] I-D = ACTION:draft-watson-sipping-req-history-00.txt



Tony,

I think it depends what stage in the process you = think we are at, and what
you mean by requirements. 'Requirements' can be = written at many levels, but
I thought the stage we got to at the last IETF was a = need to identify
exactly a 'set of capabilities' that we wished to be = able to provide with
SIP.

In terms of the overall problems that these = capabilities solve: they enable
certain services, some of which are discussed in the = draft and some of which
have been the motivation for the more specific = mechanisms previously
proposed and listed in the references. We don't in = IETF define a set of
services, and then design protocol to implement = these, but rather identify
generic capabilities which might be useful to = service designers.

This is where we were after IETF-52, with support for = definition of a
generic capability for request history. Once we have = agreement on the
details of what this capability does, *then* we can = proceed to the next
stage of identifying what technical problems need to = be solved from a
protocol perspective in order to provide achieve = this.

We thought this was the required process under = sipchange - define the
capability first in SIPPING and work on the solution = later in SIP.

Are you questioning the rationalle for the whole = capability, or do you
expect the requirements to go into more detail as to = the effect on the
protocol ?

Regards,

Mark



> -----Original Message-----
> From: Tom_Gray@Mitel.COM [mailto:Tom_Gray@Mitel.COM]=
> Sent: 25 February 2002 15:45
> To: sipping@ietf.org; Tom_Gray@Mitel.COM
> Subject: Re: [Sipping] I-D
> = ACTION:draft-watson-sipping-req-history-00.txt
>
>
>
>
> From:  Tom Gray@MITEL on 02/25/2002 10:44 = AM
>
>
> Looking at this document does not reveal a list = of
> requirements but just a list
> of proposed capabilities that may or may not = require
> additions to the SIP
> protocol. For this to be a useful requirement = document, this list of
> capabilities must be linked to a set of = problems that need to
> be solved. Indeed
> the list of problems would be the nub of the = document. For
> example what problem
> is the 'capability-req' capability intended to = solve. Just
> what is 'Issuer-req'?
> The description in the draft doesn't specify = this beyond not
> identifying he
> reason fro its proposal. What trade-offs if any = will  be
> required for the
> solution to these problems?
>
>
>
>
>
>
> Internet-Drafts@ietf.org on 02/25/2002 06:22:20 = AM
>
> Please respond to = Internet-Drafts@ietf.org
>
> To:   ietf.org
> cc:   sipping@ietf.org (bcc: Tom = Gray/Kan/Mitel)
>
> Subject:  [Sipping] I-D = ACTION:draft-watson-sipping-req-history-00.txt
>
>
>
> A New Internet-Draft is available from the = on-line
> Internet-Drafts directories.
>
>
>      = Title          : Generic = Request History Capability -
> Requirements
>      Author(s) : M. = Watson et al.
>      Filename  : = draft-watson-sipping-req-history-00.txt
>      = Pages          : 7
>      = Date      : 22-Feb-02
>
> Many services that SIP is anticipated to = support require the
> ability to determine why and how the call = arrived at a specific
> application.  Examples of such services = include (but are not limited
> to) sessions initiated to call centers via = 'click to talk' SIP URLs
> on a web page, 'call history/logging' style = services within
> intelligent 'call management' software for SIP = UAs and calls to
> voicemail servers and call centers.
>
> A URL for this Internet-Draft is:
> http://www.ietf.org/internet-drafts/draft-watson-sippi= ng-req-h
istory-00.txt

To remove yourself from the IETF Announcement list, = send a message to
ietf-announce-request with the word unsubscribe in = the body of the message.

Internet-Drafts are also available by anonymous FTP. = Login with the username
"anonymous" and a password of your e-mail = address. After logging in,
type "cd internet-drafts" and then
     "get = draft-watson-sipping-req-history-00.txt".

A list of Internet-Drafts directories can be found = in
http://www.ietf.org/shadow.html
or ftp://ftp.ietf.org/ietf/1shadow-sites.txt


Internet-Drafts can also be obtained by = e-mail.

Send a message to:
     mailserv@ietf.org.
In the body type:
     "FILE = /internet-drafts/draft-watson-sipping-req-history-00.txt".

NOTE:     The mail server at = ietf.org can return the document in
     MIME-encoded form by using = the "mpack" utility.  To use this
     feature, insert the command = "ENCODING mime" before the "FILE"
     command.  To decode = the response(s), you will need "munpack" or
     a MIME-compliant mail = reader.  Different MIME-compliant mail readers
     exhibit different behavior, = especially when dealing with
     "multipart" MIME = messages (i.e. documents which have been split
     up into multiple messages), = so check your local documentation on
     how to manipulate these = messages.


Below is the data which will enable a MIME compliant = mail reader
implementation to automatically retrieve the ASCII = version of the
Internet-Draft.

mailto:mailserv@ietf.org
ftp://ftp.ietf.org/internet-drafts/draft-watson-sippin= g-req-history-00.txt







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


------_=_NextPart_001_01C1BE3D.2FA50170-- _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Mon Feb 25 17:16:36 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id RAA01866 for ; Mon, 25 Feb 2002 17:16:36 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id RAA21623 for sipping-archive@odin.ietf.org; Mon, 25 Feb 2002 17:16:39 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id RAA21117; Mon, 25 Feb 2002 17:02:29 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id RAA21086 for ; Mon, 25 Feb 2002 17:02:27 -0500 (EST) Received: from sj-msg-core-4.cisco.com (sj-msg-core-4.cisco.com [171.71.163.10]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id RAA01597 for ; Mon, 25 Feb 2002 17:02:20 -0500 (EST) Received: from nisser.cisco.com (nisser.cisco.com [171.71.176.85]) by sj-msg-core-4.cisco.com (8.11.3/8.9.1) with ESMTP id g1PM1qt23782 for ; Mon, 25 Feb 2002 14:01:52 -0800 (PST) Received: from cisco.com (ssh-sj1.cisco.com [171.68.225.134]) by nisser.cisco.com (8.8.6 (PHNE_14041)/CISCO.SERVER.1.2) with ESMTP id OAA01446 for ; Mon, 25 Feb 2002 14:01:45 -0800 (PST) Message-ID: <3C7AB446.DCD84E32@cisco.com> Date: Mon, 25 Feb 2002 17:01:43 -0500 From: Flemming Andreasen X-Mailer: Mozilla 4.79 [en]C-CCK-MCD (Windows NT 5.0; U) X-Accept-Language: en MIME-Version: 1.0 To: "sipping@ietf.org" Content-Type: text/plain; charset=us-ascii Content-Transfer-Encoding: 7bit Content-Transfer-Encoding: 7bit Subject: [Sipping] privacy-04 Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit Greetings Over the last month or so, we've had some good discussions about the privacy-03 draft (draft-ietf-sip-privacy-03.txt). Based on the discussions, I have noted consensus on the following: - Untrusted UAs may provide a Remote-Party-Id header to their proxy in order to indicate a desired Remote-Party-Id they wish to be known under. This is needed for the case where there may be multiple Remote-Party-Ids associated with the UA (or the user using the UA) and the UA (or user) has a preference as to which one is to be used. Separate UA (or user) authentication, which is outside the scope of the privacy draft, will still be required. - The Anonymity header will be removed in anticipation of a more general routing/service invocation solution to this problem. The issue of IP address will still be described though. - The transitive trust model currently described in privacy-03 will remain unmodified. Furthermore, the Remote-Party-Id header will remain unsigned. An applicability statement explaining the environments in which the solution can be used and the limitations associated with it will be included. - Grammar fixes: a) Remote-Party-ID and RPID-Privacy need to be on list form b) RPID-Privacy values should not start with semi-colon. If you believe the above does not reflect consensus, or if I omitted something in the above list, please let me know as soon as possible. Otherwise, the above will be reflected in privacy-04. Thanks Flemming -- Flemming Andreasen Cisco Systems _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Tue Feb 26 00:52:28 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id AAA10989 for ; Tue, 26 Feb 2002 00:52:28 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id AAA10502 for sipping-archive@odin.ietf.org; Tue, 26 Feb 2002 00:52:29 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id AAA10202; Tue, 26 Feb 2002 00:36:50 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id AAA10171 for ; Tue, 26 Feb 2002 00:36:47 -0500 (EST) Received: from oak.neustar.com (oak.neustar.com [209.173.53.70]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id AAA10879 for ; Tue, 26 Feb 2002 00:36:45 -0500 (EST) Received: from chiimc01.il.neustar.com (dmz1.va.neustar.com [209.173.53.65]) by oak.neustar.com (8.11.0/8.11.0) with ESMTP id g1Q5aD800965; Tue, 26 Feb 2002 00:36:13 -0500 Received: by chiimc01.il.neustar.com with Internet Mail Service (5.5.2653.19) id ; Mon, 25 Feb 2002 23:36:07 -0600 Message-ID: <70565611B164D511957A001083FCDD5601870105@VA02> From: "Peterson, Jon" To: "'Mark Watson'" , sipping@ietf.org Subject: RE: [Sipping] Privacy comment (was Re: calling number blocking at sip/isup gate way) Date: Mon, 25 Feb 2002 23:36:06 -0600 MIME-Version: 1.0 X-Mailer: Internet Mail Service (5.5.2653.19) Content-Type: multipart/alternative; boundary="----_=_NextPart_001_01C1BE87.7CEEC610" Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org This message is in MIME format. Since your mail reader does not understand this format, some or all of this message may not be legible. ------_=_NextPart_001_01C1BE87.7CEEC610 Content-Type: text/plain; charset="iso-8859-1" Although I appreciate the explanations that you and Dean have given, I don't think I really understand what these public and private identities signify. I do believe that the scope of the RPID seems much more comprehensible to me if it is restricted to network-asserted identity. Again, there are already a number of fields in SIP where the originating user, and user agent, can represent themselves with URIs. The user can populate the From and Reply-To (not to mention Authorization) header fields with AoR identities representing themselves, and the user agent can populate the Contact header with a contact address representing itself. Clearly, there is no requirement that these all say the same thing, and I'd be surprised to learn that in combination they can't do more or less what you'd need. I'm not sure, anyway, why the solution would be to throw more headers at the problem. RPID makes sense to me scoped as an identity that is asserted by network operators based on their assessment of the user's identity, possibly resulting from the evaluation of the headers mentioned above (or even lower-layer or out-of-band security functions). I think your call center architecture below is permitted today by the existing baseline authentication system. There is no necessary correspondence between the username of the URI given in the From field in a request and the username/password that is provided by a user agent in response to a challenge. While many proxy servers would legitimately require that the two be the same, clearly your call center's proxy would be happy to allow the From field to represent the call center's AoR (an 800-number, say), while the username/password provides credentials that identify the user. An even better scenario might involve the Reply-To header as well, providing a means for the callee to reach the same call center agent when appropriate. I don't think the requirements of this architecture, at any rate, give rise to a need for the RPID, let alone for a user to provide an RPID. Jon Peterson NeuStar, Inc. -----Original Message----- From: Mark Watson [mailto:mwatson@nortelnetworks.com] Sent: Monday, February 25, 2002 4:46 AM To: Peterson, Jon; sipping@ietf.org Subject: RE: [Sipping] Privacy comment (was Re: calling number blocking at sip/isup gate way) Jon wrote: > This works fine for cases in which a single UA is used by multiple users, or when a single user has multiple provable identities. > It assumes that you have a one-to-one mapping between credentials and identities, but that's usually par for the course. This assumption does not hold in the 3GPP case where authentication is based on a 'private identity' and associated keys in the SIM, but where there may be multiple 'public identies' associated with this single 'private identity'. In particular, this allows you easily to add/remove/manage the 'public identies' without getting involved in key management/distribution. All that is necessary to manage to public identities is to update the subscription information that the proxy uses to verify that a particular public identity is allowed for the particular private identity. In addition I can think of cases where the identity that the user may wish to have presented to the terminating UA, to the PSTN or other elements, may differ from the identity that they use for local authentication. For example, imagine an outbound call centre where all the agents use SIP terminals. For obvious reasons each agent must authenticate to the local proxy using a personal identity, but the identity they wish to be known by to the called party is the Freephone number of the call centre (or even a different, inbound, call centre). Cheers...Mark -----Original Message----- From: Peterson, Jon [mailto:jon.peterson@neustar.biz] Sent: 22 February 2002 18:26 To: Watson, Mark [MDN05:EP10:EXCH]; sipping@ietf.org Subject: RE: [Sipping] Privacy comment (was Re: calling number blocking at sip/isup gate way) Actually, I have a few slight confusions about the text below (it seems to use UA and user interchangeably a bit, which is worrisome when we're talking about proving identity). One question would probably clear things up: If the proxy still needs to authenticate the user (through whichever mechanism) to ensure that the identity that has been asserted in the RPID is valid for the user in question, why does the user need to supply an RPID at all? Why doesn't the user just authenticate itself to the proxy (however that happens) and let the proxy insert the RPID corresponding to the identity that the user has proven? This works fine for cases in which a single UA is used by multiple users, or when a single user has multiple provable identities. It assumes that you have a one-to-one mapping between credentials and identities, but that's usually par for the course. Again, if the proxy server holds a mapping between credentials and valid identities itself, as you suggest below, it doesn't seem at all unreasonable to adopt this model. Jon Peterson NeuStar, Inc. -----Original Message----- From: Mark Watson [mailto:mwatson@nortelnetworks.com] Sent: Tuesday, February 19, 2002 12:29 PM To: sipping@ietf.org Subject: [Sipping] Privacy comment (was Re: calling number blocking at sip/isup gate way) As presently described in privacy-03, the Remote-Party-ID header is only ever inserted by proxies, based on the proxy's knowledge of the identity of the UA (which it determines from some authentication mechanism out of scope of the privacy draft). Privacy-02 allowed the UA to insert Remote-Party-ID too. I think this should be reintroduced for the *specific case* of a UA with multiple identities, to allow the UA to choose which identity it wishes to be known by. The Proxy would still be responsible for ensuring that this identity was valid for the user and would still use some other authentication scheme to authenticate the user. The proxy would need knowledge of which identities were valid for which users. I believe this is a requirement for 3GPP, where a UA can have multiple 'public identities' and chooses which of these it wishes to be known by for each call. The proxy authenticates the UA by some other means and then checks the subscription data to ensure it is really allowed to use the public Identity it has included. I've suggested this a few times, and there has not been much comment for or against. It would be good if we could get a resolution before the next version of privacy is published. So, comments ? Regards...Mark > -----Original Message----- > From: Dean Willis [ mailto:dean.willis@softarmor.com ] > Sent: 18 February 2002 23:12 > To: Watson, Mark [MDN05:EP10:EXCH]; Mpierce1@aol.com; > jon.peterson@neustar.biz; sipping@ietf.org > Subject: Re: [Sipping] RE: [Sip] calling number blocking at sip/isup > gateway > > > Mark said: > > > Your example below needs only a small extension to > illustrate Mike's point > > and show why something like Remote-Party-ID as defined in > the privacy > draft > > is needed, and how it works despite the lack of a signature. > > Yep. And what happens when my misconfigured proxy forwards > the call without > paying attention to the Privacy header, because for some reason it's > forwarding to the outside world instead of a PSTN gateway > under my control? > We get a privacy leak. > > But other than that, I'd say you've precisely described a > "sunny day" use > case for RPID. > > It still has a lot of security issues based on transitive > trust, which make > it a lot more palatable in a two-party case than in three or > more party > transitivity. > > I believe we're approaching a general consensus that the Privacy draft > provides a reasonable approach to solving a specific class of problem > wherein interoperator-transitive trust is couple to > transport-layer message > protection. That's not a general privacy mechanism, but it > does seem to have > enough constituency to justify publication as long as it has a REALLY > serious applicability statement explaining the particular > requirements and > environments for which it is useful. > > How does that sound? > > -- > Dean > > ------_=_NextPart_001_01C1BE87.7CEEC610 Content-Type: text/html; charset="iso-8859-1" Privacy comment (was Re: calling number blocking at sip/isup gateway)
Although I appreciate the explanations that you and Dean have given, I don't think I really understand what these public and private identities signify. I do believe that the scope of the RPID seems much more comprehensible to me if it is restricted to network-asserted identity. Again, there are already a number of fields in SIP where the originating user, and user agent, can represent themselves with URIs. The user can populate the From and Reply-To (not to mention Authorization) header fields with AoR identities representing themselves, and the user agent can populate the Contact header with a contact address representing itself. Clearly, there is no requirement that these all say the same thing, and I'd be surprised to learn that in combination they can't do more or less what you'd need. I'm not sure, anyway, why the solution would be to throw more headers at the problem. RPID makes sense to me scoped as an identity that is asserted by network operators based on their assessment of the user's identity, possibly resulting from the evaluation of the headers mentioned above (or even lower-layer or out-of-band security functions).
 
I think your call center architecture below is permitted today by the existing baseline authentication system. There is no necessary correspondence between the username of the URI given in the From field in a request and the username/password that is provided by a user agent in response to a challenge. While many proxy servers would legitimately require that the two be the same, clearly your call center's proxy would be happy to allow the From field to represent the call center's AoR (an 800-number, say), while the username/password provides credentials that identify the user. An even better scenario might involve the Reply-To header as well, providing a means for the callee to reach the same call center agent when appropriate. I don't think the requirements of this architecture, at any rate, give rise to a need for the RPID, let alone for a user to provide an RPID.
 
Jon Peterson
NeuStar, Inc.
-----Original Message-----
From: Mark Watson [mailto:mwatson@nortelnetworks.com]
Sent: Monday, February 25, 2002 4:46 AM
To: Peterson, Jon; sipping@ietf.org
Subject: RE: [Sipping] Privacy comment (was Re: calling number blocking at sip/isup gate way)

Jon wrote:
 
> This works fine for cases in which a single UA is used by multiple users, or when a single user has multiple provable identities.
> It assumes that you have a one-to-one mapping between credentials and identities, but that's usually par for the course.
 
This assumption does not hold in the 3GPP case where authentication is based on a 'private identity' and associated keys in the SIM, but where there may be multiple 'public identies' associated with this single 'private identity'. In particular, this allows you easily to add/remove/manage the 'public identies' without getting involved in key management/distribution. All that is necessary to manage to public identities is to update the subscription information that the proxy uses to verify that a particular public identity is allowed for the particular private identity.
 
In addition I can think of cases where the identity that the user may wish to have presented to the terminating UA, to the PSTN or other elements, may differ from the identity that they use for local authentication. For example, imagine an outbound call centre where all the agents use SIP terminals. For obvious reasons each agent must authenticate to the local proxy using a personal identity, but the identity they wish to be known by to the called party is the Freephone number of the call centre (or even a different, inbound, call centre).
 
Cheers...Mark
 
 
 -----Original Message-----
From: Peterson, Jon [mailto:jon.peterson@neustar.biz]
Sent: 22 February 2002 18:26
To: Watson, Mark [MDN05:EP10:EXCH]; sipping@ietf.org
Subject: RE: [Sipping] Privacy comment (was Re: calling number blocking at sip/isup gate way)

Actually, I have a few slight confusions about the text below (it seems to use UA and user interchangeably a bit, which is worrisome when we're talking about proving identity). One question would probably clear things up:
 
If the proxy still needs to authenticate the user (through whichever mechanism) to ensure that the identity that has been asserted in the RPID is valid for the user in question, why does the user need to supply an RPID at all? Why doesn't the user just authenticate itself to the proxy (however that happens) and let the proxy insert the RPID corresponding to the identity that the user has proven? This works fine for cases in which a single UA is used by multiple users, or when a single user has multiple provable identities. It assumes that you have a one-to-one mapping between credentials and identities, but that's usually par for the course.
 
Again, if the proxy server holds a mapping between credentials and valid identities itself, as you suggest below, it doesn't seem at all unreasonable to adopt this model.
 
Jon Peterson
NeuStar, Inc.
-----Original Message-----
From: Mark Watson [mailto:mwatson@nortelnetworks.com]
Sent: Tuesday, February 19, 2002 12:29 PM
To: sipping@ietf.org
Subject: [Sipping] Privacy comment (was Re: calling number blocking at sip/isup gate way)

As presently described in privacy-03, the Remote-Party-ID header is only ever inserted by proxies, based on the proxy's knowledge of the identity of the UA (which it determines from some authentication mechanism out of scope of the privacy draft).

Privacy-02 allowed the UA to insert Remote-Party-ID too. I think this should be reintroduced for the *specific case* of a UA with multiple identities, to allow the UA to choose which identity it wishes to be known by. The Proxy would still be responsible for ensuring that this identity was valid for the user and would still use some other authentication scheme to authenticate the user. The proxy would need knowledge of which identities were valid for which users.

I believe this is a requirement for 3GPP, where a UA can have multiple 'public identities' and chooses which of these it wishes to be known by for each call. The proxy authenticates the UA by some other means and then checks the subscription data to ensure it is really allowed to use the public Identity it has included.

I've suggested this a few times, and there has not been much comment for or against. It would be good if we could get a resolution before the next version of privacy is published.

So, comments ?

Regards...Mark





> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.com]
> Sent: 18 February 2002 23:12
> To: Watson, Mark [MDN05:EP10:EXCH]; Mpierce1@aol.com;
> jon.peterson@neustar.biz; sipping@ietf.org
> Subject: Re: [Sipping] RE: [Sip] calling number blocking at sip/isup
> gateway
>
>
> Mark said:
>
> > Your example below needs only a small extension to
> illustrate Mike's point
> > and show why something like Remote-Party-ID as defined in
> the privacy
> draft
> > is needed, and how it works despite the lack of a signature.
>
> Yep. And what happens when my misconfigured proxy forwards
> the call without
> paying attention to the Privacy header, because for some reason it's
> forwarding to the outside world instead of a PSTN gateway
> under my control?
> We get a privacy leak.
>
> But other than that, I'd say you've precisely described a
> "sunny day" use
> case for RPID.
>
> It still has a lot of security issues based on transitive
> trust, which make
> it a lot more palatable in a two-party case than in three or
> more party
> transitivity.
>
> I believe we're approaching a general consensus that the Privacy draft
> provides a reasonable approach to solving a specific class of problem
> wherein interoperator-transitive trust is couple to
> transport-layer message
> protection. That's not a general privacy mechanism, but it
> does seem to have
> enough constituency to justify publication as long as it has a REALLY
> serious applicability statement explaining the particular
> requirements and
> environments for which it is useful.
>
> How does that sound?
>
> --
> Dean
>
>

------_=_NextPart_001_01C1BE87.7CEEC610-- _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Tue Feb 26 08:29:13 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id IAA28234 for ; Tue, 26 Feb 2002 08:29:04 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id IAA12090 for sipping-archive@odin.ietf.org; Tue, 26 Feb 2002 08:29:07 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id IAA11539; Tue, 26 Feb 2002 08:11:19 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id IAA11424 for ; Tue, 26 Feb 2002 08:11:13 -0500 (EST) Received: from Mitel.COM ([216.191.234.70]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id IAA27470 for ; Tue, 26 Feb 2002 08:11:09 -0500 (EST) From: Tom_Gray@Mitel.COM Received: from kanmta01.software.mitel.com (kanmta01.kanata.mitel.com [134.199.37.58]) by Mitel.COM (V8/MAIL-RELAY-2.1) with SMTP id IAA28388; Tue, 26 Feb 2002 08:08:48 -0500 (EST) Received: by kanmta01.software.mitel.com(Lotus SMTP MTA v4.6.7 (934.1 12-30-1999)) id 85256B6C.00483469 ; Tue, 26 Feb 2002 08:08:40 -0500 X-Lotus-FromDomain: MITEL To: "Mary Barnes" , Tom_Gray@Mitel.COM cc: "Mark Watson" , sipping@ietf.org Message-ID: <85256B6C.00483366.00@kanmta01.software.mitel.com> Date: Tue, 26 Feb 2002 08:08:37 -0500 Subject: RE: [Sipping] I-D ACTION:draft-watson-sipping-req-history-00.txt Mime-Version: 1.0 Content-type: multipart/mixed; Boundary="0__=Lzg3SGiAqCnRp3O8me8r9uAdGrE5cqKSieJ5mF1fSGxYaskdB6ySYEdU" Content-Disposition: inline Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org --0__=Lzg3SGiAqCnRp3O8me8r9uAdGrE5cqKSieJ5mF1fSGxYaskdB6ySYEdU Content-type: text/plain; charset=us-ascii Content-Disposition: inline From: Tom Gray@MITEL on 02/26/2002 08:08 AM As a final comment on this, a disembodied set of 'requirements' that bear no relation to any palpable function simply cannot function. Consider what would happen is someone were to comment on this draft by advocating that the 'BACKWARDS-req' be removed as being unneeded. Others would presumably disagree and intervene to say that its removal would damage some valuable scenario. I These scenarios, if they are valuable, should be identified in the requirements document. Failing to do this will result in sterile open-ended debate since their can be no limitation on its scope. No set of problems has been defined to limit the scope of the debate and so any topic must be germane to it. Having made my opinion known on this, I'll leave it to the management of the working group to decide if this draft holds a valid set of requirements that can be used to define and limit discussion or not. "Mary Barnes" on 02/25/2002 03:44:14 PM To: Tom Gray/Kan/Mitel@Mitel, "Mark Watson" , sipping@ietf.org cc: Subject: RE: [Sipping] I-D ACTION:draft-watson-sipping-req-history-00.txt Tom, The intent of this document is not to provide a traditional telephony call redirection service within SIP. That might be just one use. Quite clearly we saw at IETF-52 that it was indeed a mistake to provide that as an example. The goal is to ensure that what gets progressed is a general purpose Request History mechanism which could be used for a variety of services. I do understand your concerns about this being too fuzzy and unspecific at this time as I myself have struggled with that lack of concreteness and the level of abstraction of the current draft. If you have specific suggestions on how to get more concrete on characterizing the nature of problems to which these requirements apply without going either into the solution domain or prescribing requirements for a narrow problem domain, that would be much appreciated. Thanks, Mary. -----Original Message----- From: Tom_Gray@Mitel.COM [mailto:Tom_Gray@Mitel.COM] Sent: Monday, February 25, 2002 2:31 PM To: Watson, Mark [MDN05:EP10:EXCH]; sipping@ietf.org; Tom_Gray@Mitel.COM Subject: RE: [Sipping] I-D ACTION:draft-watson-sipping-req-history-00.txt From: Tom Gray@MITEL on 02/25/2002 03:31 PM It is my understanding that this document is about providing a traditional telephony call redirection service within SIP. If that is so, I would just like a requirements document for this to say that. If not, it would be nice to know what is the purpose of the capabilities described. The capabilites are describing a proposed solution to an unnamed problem Calling them requirements does not make them so. Given a derscription of the problem that is being addressed, there could be a determination then of the information that would be need to be carried by SIP for this service and whether or not this was a problem of broad enough significance that it would be addressed in the general protocol. "Mary Barnes" on 02/25/2002 03:06:16 PM To: Tom Gray/Kan/Mitel@Mitel, "Mark Watson" cc: sipping@ietf.org Subject: RE: [Sipping] I-D ACTION:draft-watson-sipping-req-history-00.txt Tom, As Mark states, this document wasn't intended to prescribe the requirements of a solution to solve specific problems or scenarios, but was intended to focus on the requirements of the capabilities required in SIP (protocol) to be able to use SIP as a solution for specific scenarios. As you say, we've been quite sketchy in describing the scenarios as we had assumed the ones that we did use were fairly well known scenarios to the audience. In a sense this document is a guinea pig for the new SIPPING WG process and will need refinement. We can certainly back up yet another step and add all the details for these scenarios and highlight how these scenarios would use the capabilities that have been specified. But, I would think that would be a rather lengthy document at a fairly high level of abstraction as to not be particularly useful either towards the end goal; with the end goal of this draft being an acknowledgement that there is not a "Request History" capability in SIP and that this is necessary in order to be able to use SIP for certain services. We do recognize that the level of abstraction of the current document is fairly high and we see value in taking it to the next step of evaluating different solution proposals applied against the required capabilities including specific scenarios to demonstrate the proposed solutions. Certainly, this step would be open to all "intervenors". I think the basic question being asked here is whether we want to add more to the requirements process by requiring a detailed description and agreement of a problem domain prior to proposing requirements for the capabilities required in the SIP protocol to solve a given problem? Do we have any other opinions, particularly from the WG chairs? Regards, Mary H. Barnes mbarnes@nortelnetworks.com -----Original Message----- From: Tom_Gray@Mitel.COM [mailto:Tom_Gray@Mitel.COM] Sent: Monday, February 25, 2002 1:10 PM To: Watson, Mark [MDN05:EP10:EXCH] Cc: sipping@ietf.org; Tom_Gray@Mitel.COM Subject: RE: [Sipping] I-D ACTION:draft-watson-sipping-req-history-00.txt From: Tom Gray@MITEL on 02/25/2002 02:10 PM Reading this document, I do not see how anyone could assess its worth. What are the capabilities identified in this document supposed to accomplish? This understanding is what requirements documents are supposed to provide. I would recommend that the requirements be specific enough in their problem statement so that: a) it could be determined whether or not there is a problem that is of high enough priority that it needs to be solved. The outcome would naturally be an agreed upon problem statement. Intervenors could be invited to submit proposed solutions. b) the worth of proposed solutions could be assessed by comparing their ability to solve the agreed upon problems I see a very general description of a problem area in the document. However in my opinion this could be greatly improved by providing scenarios which illustrate specific deficiencies that are not currently solvable with the standard SIP protocol. Without this, I do not see how anyone can discuss the document. What basis would anyone have for accepting or rejecting it? "Mark Watson" on 02/25/2002 12:54:33 PM To: Tom Gray/Kan/Mitel@Mitel, sipping@ietf.org cc: Subject: RE: [Sipping] I-D ACTION:draft-watson-sipping-req-history-00.txt Tony, I think it depends what stage in the process you think we are at, and what you mean by requirements. 'Requirements' can be written at many levels, but I thought the stage we got to at the last IETF was a need to identify exactly a 'set of capabilities' that we wished to be able to provide with SIP. In terms of the overall problems that these capabilities solve: they enable certain services, some of which are discussed in the draft and some of which have been the motivation for the more specific mechanisms previously proposed and listed in the references. We don't in IETF define a set of services, and then design protocol to implement these, but rather identify generic capabilities which might be useful to service designers. This is where we were after IETF-52, with support for definition of a generic capability for request history. Once we have agreement on the details of what this capability does, *then* we can proceed to the next stage of identifying what technical problems need to be solved from a protocol perspective in order to provide achieve this. We thought this was the required process under sipchange - define the capability first in SIPPING and work on the solution later in SIP. Are you questioning the rationalle for the whole capability, or do you expect the requirements to go into more detail as to the effect on the protocol ? Regards, Mark > -----Original Message----- > From: Tom_Gray@Mitel.COM [mailto:Tom_Gray@Mitel.COM] > Sent: 25 February 2002 15:45 > To: sipping@ietf.org; Tom_Gray@Mitel.COM > Subject: Re: [Sipping] I-D > ACTION:draft-watson-sipping-req-history-00.txt > > > > > From: Tom Gray@MITEL on 02/25/2002 10:44 AM > > > Looking at this document does not reveal a list of > requirements but just a list > of proposed capabilities that may or may not require > additions to the SIP > protocol. For this to be a useful requirement document, this list of > capabilities must be linked to a set of problems that need to > be solved. Indeed > the list of problems would be the nub of the document. For > example what problem > is the 'capability-req' capability intended to solve. Just > what is 'Issuer-req'? > The description in the draft doesn't specify this beyond not > identifying he > reason fro its proposal. What trade-offs if any will be > required for the > solution to these problems? > > > > > > > Internet-Drafts@ietf.org on 02/25/2002 06:22:20 AM > > Please respond to Internet-Drafts@ietf.org > > To: ietf.org > cc: sipping@ietf.org (bcc: Tom Gray/Kan/Mitel) > > Subject: [Sipping] I-D ACTION:draft-watson-sipping-req-history-00.txt > > > > A New Internet-Draft is available from the on-line > Internet-Drafts directories. > > > Title : Generic Request History Capability - > Requirements > Author(s) : M. Watson et al. > Filename : draft-watson-sipping-req-history-00.txt > Pages : 7 > Date : 22-Feb-02 > > Many services that SIP is anticipated to support require the > ability to determine why and how the call arrived at a specific > application. Examples of such services include (but are not limited > to) sessions initiated to call centers via 'click to talk' SIP URLs > on a web page, 'call history/logging' style services within > intelligent 'call management' software for SIP UAs and calls to > voicemail servers and call centers. > > A URL for this Internet-Draft is: > http://www.ietf.org/internet-drafts/draft-watson-sipping-req-h istory-00.txt To remove yourself from the IETF Announcement list, send a message to ietf-announce-request with the word unsubscribe in the body of the message. Internet-Drafts are also available by anonymous FTP. Login with the username "anonymous" and a password of your e-mail address. After logging in, type "cd internet-drafts" and then "get draft-watson-sipping-req-history-00.txt". A list of Internet-Drafts directories can be found in http://www.ietf.org/shadow.html or ftp://ftp.ietf.org/ietf/1shadow-sites.txt Internet-Drafts can also be obtained by e-mail. Send a message to: mailserv@ietf.org. 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development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Tue Feb 26 08:30:58 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id IAA28395 for ; Tue, 26 Feb 2002 08:30:58 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id IAA12582 for sipping-archive@odin.ietf.org; Tue, 26 Feb 2002 08:31:01 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id IAA11724; Tue, 26 Feb 2002 08:16:51 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id IAA11696 for ; Tue, 26 Feb 2002 08:16:49 -0500 (EST) Received: from smtp2.gtsgroup.com (smtp2.gtsgroup.com [195.158.230.80]) by ietf.org (8.9.1a/8.9.1a) with SMTP id IAA27598 for ; Tue, 26 Feb 2002 08:16:45 -0500 (EST) Received: from 207.77.231.73 by smtp2.gtsgroup.com (InterScan E-Mail VirusWall NT); Tue, 26 Feb 2002 13:16:47 -0000 Received: by brubhdpnt01.gtsgroup.com with Internet Mail Service (5.5.2653.19) id <10WWJ9C2>; Tue, 26 Feb 2002 14:16:47 +0100 Message-ID: From: "Agboh, Charles" To: "'sipping@ietf.org'" Date: Tue, 26 Feb 2002 14:15:46 +0100 MIME-Version: 1.0 X-Mailer: Internet Mail Service (5.5.2653.19) Content-Type: text/plain; charset="iso-8859-1" Subject: [Sipping] Billing and Settlement Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Hi all; Could someone please provide a pointer to a draft(s), if any, addressing billing and settlement. Regards, Charles _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Tue Feb 26 14:54:25 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id OAA19151 for ; Tue, 26 Feb 2002 14:54:24 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id OAA15894 for sipping-archive@odin.ietf.org; Tue, 26 Feb 2002 14:54:27 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id OAA15081; Tue, 26 Feb 2002 14:40:56 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id OAA14972 for ; Tue, 26 Feb 2002 14:40:49 -0500 (EST) Received: from zctfs063.nortelnetworks.com (h90s131a237n47.user.nortelnetworks.com [47.237.131.90]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id OAA18514 for ; Tue, 26 Feb 2002 14:40:45 -0500 (EST) Received: from znsgs01r.europe.nortel.com (znsgs01r.europe.nortel.com [47.137.129.92]) by zctfs063.nortelnetworks.com (Switch-2.2.0/Switch-2.2.0) with ESMTP id g1QJeDL10188; Tue, 26 Feb 2002 20:40:13 +0100 (MET) Received: from zwcwc012.europe.nortel.com (zwcwc012.europe.nortel.com [47.73.112.187]) by znsgs01r.europe.nortel.com (Switch-2.2.0/Switch-2.2.0) with ESMTP id g1QJdle12638; Tue, 26 Feb 2002 19:39:47 GMT Received: by zwcwc012.europe.nortel.com with Internet Mail Service (5.5.2653.19) id ; Tue, 26 Feb 2002 19:40:16 -0000 Message-ID: From: "Mark Watson" To: Flemming Andreasen Cc: sipping@ietf.org Subject: RE: [Sipping] Privacy comment (was Re: calling number blocking a tsip/isup gate way) Date: Tue, 26 Feb 2002 19:40:10 -0000 MIME-Version: 1.0 X-Mailer: Internet Mail Service (5.5.2653.19) Content-Type: multipart/alternative; boundary="----_=_NextPart_001_01C1BEFD.66CEB270" Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org This message is in MIME format. Since your mail reader does not understand this format, some or all of this message may not be legible. ------_=_NextPart_001_01C1BEFD.66CEB270 Content-Type: text/plain A corollory of this is that we don't really need RPID-Privacy any more. This was introduced in privacy-03 on the basis that the ability for UAs to include RPID was removed, but UAs still needed to indicate their privacy requirements in respect of RPIDs to be added by proxies. IF UA-inserted RPID is re-introduced, we can take RPID-Privacy back out. UAs wishing to indicate privacy requirements for RPIDs to be inserted by Proxies, without the UA itself indicating an identity, can include an RPID with the privacy & type values set appropriatly, but no actual identity (as was the original intention). Regards...Mark > -----Original Message----- > From: Flemming Andreasen [mailto:fandreas@cisco.com] > Sent: 25 February 2002 19:56 > To: Watson, Mark [MDN05:EP10:EXCH] > Cc: sipping@ietf.org > Subject: Re: [Sipping] Privacy comment (was Re: calling > number blocking > atsip/isup gate way) > > > > > Mark Watson wrote: > > > > > > > So, in the absence of further comments, could the editor add this to > > the next version of the privacy draft ? (on the silence=agree/don't > > care principle) > > Will do. > > -- Flemming > > > > > > > > ...Mark > > > > > -----Original Message----- > > > From: Dean Willis [mailto:dean.willis@softarmor.com] > > > Sent: 20 February 2002 00:27 > > > To: Watson, Mark [MDN05:EP10:EXCH]; sipping@ietf.org > > > Subject: Re: [Sipping] Privacy comment (was Re: calling > > > number blocking > > > at sip/isup gate way) > > > > > > > > > I concur that your proposal would add to the utility of RPID. > > > > > > -- > > > Dean > > > > > > ----- Original Message ----- > > > From: "Mark Watson" > > > To: > > > Sent: Tuesday, February 19, 2002 2:28 PM > > > Subject: [Sipping] Privacy comment (was Re: calling > number blocking > > at > > > sip/isup gate way) > > > > > > > > > > As presently described in privacy-03, the Remote-Party-ID > > > header is only > > > > ever inserted by proxies, based on the proxy's knowledge of > > > the identity > > > of > > > > the UA (which it determines from some authentication > > > mechanism out of > > > scope > > > > of the privacy draft). > > > > > > > > Privacy-02 allowed the UA to insert Remote-Party-ID too. I > > > think this > > > should > > > > be reintroduced for the *specific case* of a UA with > > > multiple identities, > > > to > > > > allow the UA to choose which identity it wishes to be known > > > by. The Proxy > > > > would still be responsible for ensuring that this identity > > > was valid for > > > the > > > > user and would still use some other authentication scheme > > > to authenticate > > > > the user. The proxy would need knowledge of which > > > identities were valid > > > for > > > > which users. > > > > > > > > I believe this is a requirement for 3GPP, where a UA can > > > have multiple > > > > 'public identities' and chooses which of these it wishes to > > > be known by > > > for > > > > each call. The proxy authenticates the UA by some other > > > means and then > > > > checks the subscription data to ensure it is really allowed > > > to use the > > > > public Identity it has included. > > > > > > > > I've suggested this a few times, and there has not been > > > much comment for > > > or > > > > against. It would be good if we could get a resolution > > > before the next > > > > version of privacy is published. > > > > > > > > So, comments ? > > > > > > > > Regards...Mark > > > > > > > > > > > > > > > > > > > > > > > > > -----Original Message----- > > > > > From: Dean Willis [mailto:dean.willis@softarmor.com] > > > > > Sent: 18 February 2002 23:12 > > > > > To: Watson, Mark [MDN05:EP10:EXCH]; Mpierce1@aol.com; > > > > > jon.peterson@neustar.biz; sipping@ietf.org > > > > > Subject: Re: [Sipping] RE: [Sip] calling number blocking > > > at sip/isup > > > > > gateway > > > > > > > > > > > > > > > Mark said: > > > > > > > > > > > Your example below needs only a small extension to > > > > > illustrate Mike's point > > > > > > and show why something like Remote-Party-ID as defined in > > > > > the privacy > > > > > draft > > > > > > is needed, and how it works despite the lack of a signature. > > > > > > > > > > Yep. And what happens when my misconfigured proxy forwards > > > > > the call without > > > > > paying attention to the Privacy header, because for some > > > reason it's > > > > > forwarding to the outside world instead of a PSTN gateway > > > > > under my control? > > > > > We get a privacy leak. > > > > > > > > > > But other than that, I'd say you've precisely described a > > > > > "sunny day" use > > > > > case for RPID. > > > > > > > > > > It still has a lot of security issues based on transitive > > > > > trust, which make > > > > > it a lot more palatable in a two-party case than in three or > > > > > more party > > > > > transitivity. > > > > > > > > > > I believe we're approaching a general consensus that the > > > Privacy draft > > > > > provides a reasonable approach to solving a specific > > > class of problem > > > > > wherein interoperator-transitive trust is couple to > > > > > transport-layer message > > > > > protection. That's not a general privacy mechanism, but it > > > > > does seem to have > > > > > enough constituency to justify publication as long as it > > > has a REALLY > > > > > serious applicability statement explaining the particular > > > > > requirements and > > > > > environments for which it is useful. > > > > > > > > > > How does that sound? > > > > > > > > > > -- > > > > > Dean > > > > > > > > > > > > > > > > > > > > > > -- > Flemming Andreasen > Cisco Systems > > > ------_=_NextPart_001_01C1BEFD.66CEB270 Content-Type: text/html Content-Transfer-Encoding: quoted-printable RE: [Sipping] Privacy comment (was Re: calling number blocking = atsip/isup gate way)

A corollory of this is that we don't really need = RPID-Privacy any more. This was introduced in privacy-03 on the basis = that the ability for UAs to include RPID was removed, but UAs still = needed to indicate their privacy requirements in respect of RPIDs to be = added by proxies.

IF UA-inserted RPID is re-introduced, we can take = RPID-Privacy back out.

UAs wishing to indicate privacy requirements for = RPIDs to be inserted by Proxies, without the UA itself indicating an = identity, can include an RPID with the privacy & type values set = appropriatly, but no actual identity (as was the original = intention).

Regards...Mark

> -----Original Message-----
> From: Flemming Andreasen [mailto:fandreas@cisco.com]=
> Sent: 25 February 2002 19:56
> To: Watson, Mark [MDN05:EP10:EXCH]
> Cc: sipping@ietf.org
> Subject: Re: [Sipping] Privacy comment (was Re: = calling
> number blocking
> atsip/isup gate way)
>
>
>
>
> Mark Watson wrote:
>
> >
> >
> > So, in the absence of further comments, = could the editor add this to
> > the next version of the privacy draft ? = (on the silence=3Dagree/don't
> > care principle)
>
> Will do.
>
> -- Flemming
>
>
> >
> >
> > ...Mark
> >
> > > -----Original Message-----
> > > From: Dean Willis [mailto:dean.willis@softarmor.c= om]
> > > Sent: 20 February 2002 00:27
> > > To: Watson, Mark [MDN05:EP10:EXCH]; = sipping@ietf.org
> > > Subject: Re: [Sipping] Privacy = comment (was Re: calling
> > > number blocking
> > > at sip/isup gate way)
> > >
> > >
> > > I concur that your proposal would add = to the utility of RPID.
> > >
> > > --
> > > Dean
> > >
> > > ----- Original Message -----
> > > From: "Mark Watson" = <mwatson@nortelnetworks.com>
> > > To: <sipping@ietf.org>
> > > Sent: Tuesday, February 19, 2002 2:28 = PM
> > > Subject: [Sipping] Privacy comment = (was Re: calling
> number blocking
> > at
> > > sip/isup gate way)
> > >
> > >
> > > > As presently described in = privacy-03, the Remote-Party-ID
> > > header is only
> > > > ever inserted by proxies, based = on the proxy's knowledge of
> > > the identity
> > > of
> > > > the UA (which it determines from = some authentication
> > > mechanism out of
> > > scope
> > > > of the privacy draft).
> > > >
> > > > Privacy-02 allowed the UA to = insert Remote-Party-ID too. I
> > > think this
> > > should
> > > > be reintroduced for the = *specific case* of a UA with
> > > multiple identities,
> > > to
> > > > allow the UA to choose which = identity it wishes to be known
> > > by. The Proxy
> > > > would still be responsible for = ensuring that this identity
> > > was valid for
> > > the
> > > > user and would still use some = other authentication scheme
> > > to authenticate
> > > > the user. The proxy would need = knowledge of which
> > > identities were valid
> > > for
> > > > which users.
> > > >
> > > > I believe this is a requirement = for 3GPP, where a UA can
> > > have multiple
> > > > 'public identities' and chooses = which of these it wishes to
> > > be known by
> > > for
> > > > each call. The proxy = authenticates the UA by some other
> > > means and then
> > > > checks the subscription data to = ensure it is really allowed
> > > to use the
> > > > public Identity it has = included.
> > > >
> > > > I've suggested this a few times, = and there has not been
> > > much comment for
> > > or
> > > > against. It would be good if we = could get a resolution
> > > before the next
> > > > version of privacy is = published.
> > > >
> > > > So, comments ?
> > > >
> > > > Regards...Mark
> > > >
> > > >
> > > >
> > > >
> > > >
> > > > > -----Original = Message-----
> > > > > From: Dean Willis [mailto:dean.willis@softarmor.c= om]
> > > > > Sent: 18 February 2002 = 23:12
> > > > > To: Watson, Mark = [MDN05:EP10:EXCH]; Mpierce1@aol.com;
> > > > > jon.peterson@neustar.biz; = sipping@ietf.org
> > > > > Subject: Re: [Sipping] RE: = [Sip] calling number blocking
> > > at sip/isup
> > > > > gateway
> > > > >
> > > > >
> > > > > Mark said:
> > > > >
> > > > > > Your example below = needs only a small extension to
> > > > > illustrate Mike's = point
> > > > > > and show why something = like Remote-Party-ID as defined in
> > > > > the privacy
> > > > > draft
> > > > > > is needed, and how it = works despite the lack of a signature.
> > > > >
> > > > > Yep. And what happens when = my misconfigured proxy forwards
> > > > > the call without
> > > > > paying attention to the = Privacy header, because for some
> > > reason it's
> > > > > forwarding to the outside = world instead of a PSTN gateway
> > > > > under my control?
> > > > > We get a privacy = leak.
> > > > >
> > > > > But other than that, I'd = say you've precisely described a
> > > > > "sunny day" = use
> > > > > case for RPID.
> > > > >
> > > > > It still has a lot of = security issues based on transitive
> > > > > trust, which make
> > > > > it a lot more palatable in = a two-party case than in three or
> > > > > more party
> > > > > transitivity.
> > > > >
> > > > > I believe we're approaching = a general consensus that the
> > > Privacy draft
> > > > > provides a reasonable = approach to solving a specific
> > > class of problem
> > > > > wherein = interoperator-transitive trust is couple to
> > > > > transport-layer = message
> > > > > protection. That's not a = general privacy mechanism, but it
> > > > > does seem to have
> > > > > enough constituency to = justify publication as long as it
> > > has a REALLY
> > > > > serious applicability = statement explaining the particular
> > > > > requirements and
> > > > > environments for which it = is useful.
> > > > >
> > > > > How does that sound?
> > > > >
> > > > > --
> > > > > Dean
> > > > >
> > > > >
> > > >
> > >
> > >
>
> --
> Flemming Andreasen
> Cisco Systems
>
>
>

------_=_NextPart_001_01C1BEFD.66CEB270-- _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Tue Feb 26 14:58:03 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id OAA19324 for ; Tue, 26 Feb 2002 14:57:59 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id OAA16056 for sipping-archive@odin.ietf.org; Tue, 26 Feb 2002 14:58:03 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id OAA15136; Tue, 26 Feb 2002 14:40:59 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id OAA14974 for ; Tue, 26 Feb 2002 14:40:49 -0500 (EST) Received: from zctfs063.nortelnetworks.com (h90s131a237n47.user.nortelnetworks.com [47.237.131.90]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id OAA18513 for ; Tue, 26 Feb 2002 14:40:45 -0500 (EST) Received: from znsgs01r.europe.nortel.com (znsgs01r.europe.nortel.com [47.137.129.92]) by zctfs063.nortelnetworks.com (Switch-2.2.0/Switch-2.2.0) with ESMTP id g1QJeAL10180; Tue, 26 Feb 2002 20:40:11 +0100 (MET) Received: from zwcwc012.europe.nortel.com (zwcwc012.europe.nortel.com [47.73.112.187]) by znsgs01r.europe.nortel.com (Switch-2.2.0/Switch-2.2.0) with ESMTP id g1QJdie12634; Tue, 26 Feb 2002 19:39:45 GMT Received: by zwcwc012.europe.nortel.com with Internet Mail Service (5.5.2653.19) id ; Tue, 26 Feb 2002 19:40:13 -0000 Message-ID: From: "Mark Watson" To: "Peterson, Jon" , sipping@ietf.org Subject: RE: [Sipping] Privacy comment (was Re: calling number blocking at sip/isup gate way) Date: Tue, 26 Feb 2002 19:40:08 -0000 MIME-Version: 1.0 X-Mailer: Internet Mail Service (5.5.2653.19) Content-Type: multipart/alternative; boundary="----_=_NextPart_001_01C1BEFD.65C13A10" Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org This message is in MIME format. Since your mail reader does not understand this format, some or all of this message may not be legible. ------_=_NextPart_001_01C1BEFD.65C13A10 Content-Type: text/plain; charset="iso-8859-1" The difference between RPID and the existing headers is that RPID includes support for Privacy. The UA cannot include anything in any of the existing headers that the user is not happy to have presented to other users. However, if the UA trusts the proxy, it can insert such information in the RPID. If you accept that an RPID inserted by a proxy might need to be based on 'evaluation of the [existing] headers ...', then you accept that it is not sufficient to have it based only on the authentication result. But if you only allow the existing headers to be evaluated, you limit the information the UA can provide for this purpose to 'public' information. In the 3GPP case, your proposal would equate to placing the 'public identity' in the From, Reply-To or Contact field: 'From' cannot be used (in all cases) because the UA cannot be required to reveal the public identity to other users (due to Data Protection), and the proxy cannot modify the From field. The From field therefore contains a crypto-random hash or something similar. Both Reply-To and Contact have meanings of their own, which may result in a need to include in them a different address from the 'public identity' making the call. As with From, Contact is limited to containing information which can be revealed to other users and although Reply-To could be policed out by a proxy before forwarding to an untrusted element, there is no way for the UA to indicate to the Proxy that it should do this (although I guess this could be added). The UA is required to declare the public identity that is making the call to the network, though. I agree with your concern about proliferation of 'identity' headers, and I did advocate implementing Reply-To using RPID, which would have reduced the count by one. However, headers without support for privacy are all useless in an environment where the UA is required to reveal certain information to the network (proxies) to access a service, but where that information does not _need to_ be passed to other users in order for the service to operate. In these circumstances the calling user has a right to keep such information private from other users. So the UA needs some way of ensuring that the proxy knows this information must not be so revealed. Encryption is always a (perhaps cleaner) solution, but has its own problems, and as this scenario never exists without a trust relationship between UA and proxy, RPID does the job. Regards...Mark -----Original Message----- From: Peterson, Jon [mailto:jon.peterson@neustar.biz] Sent: 26 February 2002 05:36 To: Watson, Mark [MDN05:EP10:EXCH]; sipping@ietf.org Subject: RE: [Sipping] Privacy comment (was Re: calling number blocking at sip/isup gate way) Although I appreciate the explanations that you and Dean have given, I don't think I really understand what these public and private identities signify. I do believe that the scope of the RPID seems much more comprehensible to me if it is restricted to network-asserted identity. Again, there are already a number of fields in SIP where the originating user, and user agent, can represent themselves with URIs. The user can populate the From and Reply-To (not to mention Authorization) header fields with AoR identities representing themselves, and the user agent can populate the Contact header with a contact address representing itself. Clearly, there is no requirement that these all say the same thing, and I'd be surprised to learn that in combination they can't do more or less what you'd need. I'm not sure, anyway, why the solution would be to throw more headers at the problem. RPID makes sense to me scoped as an identity that is asserted by network operators based on their assessment of the user's identity, possibly resulting from the evaluation of the headers mentioned above (or even lower-layer or out-of-band security functions). I think your call center architecture below is permitted today by the existing baseline authentication system. There is no necessary correspondence between the username of the URI given in the From field in a request and the username/password that is provided by a user agent in response to a challenge. While many proxy servers would legitimately require that the two be the same, clearly your call center's proxy would be happy to allow the From field to represent the call center's AoR (an 800-number, say), while the username/password provides credentials that identify the user. An even better scenario might involve the Reply-To header as well, providing a means for the callee to reach the same call center agent when appropriate. I don't think the requirements of this architecture, at any rate, give rise to a need for the RPID, let alone for a user to provide an RPID. Jon Peter so n NeuStar, Inc. -----Original Message----- From: Mark Watson [mailto:mwatson@nortelnetworks.com] Sent: Monday, February 25, 2002 4:46 AM To: Peterson, Jon; sipping@ietf.org Subject: RE: [Sipping] Privacy comment (was Re: calling number blocking at sip/isup gate way) Jon wrote: > This works fine for cases in which a single UA is used by multiple users, or when a single user has multiple provable identities. > It assumes that you have a one-to-one mapping between credentials and identities, but that's usually par for the course. This assumption does not hold in the 3GPP case where authentication is based on a 'private identity' and associated keys in the SIM, but where there may be multiple 'public identies' associated with this single 'private identity'. In particular, this allows you easily to add/remove/manage the 'public identies' without getting involved in key management/distribution. All that is necessary to manage to public identities is to update the subscription information that the proxy uses to verify that a particular public identity is allowed for the particular private identity. In addition I can think of cases where the identity that the user may wish to have presented to the terminating UA, to the PSTN or other elements, may differ from the identity that they use for local authentication. For example, imagine an outbound call centre where all the agents use SIP terminals. For obvious reasons each agent must authenticate to the local proxy using a personal identity, but the identity they wish to be known by to the called party is the Freephone number of the call centre (or even a different, inbound, call centre). Cheers...Mark -----Original Message----- From: Peterson, Jon [mailto:jon.peterson@neustar.biz] Sent: 22 February 2002 18:26 To: Watson, Mark [MDN05:EP10:EXCH]; sipping@ietf.org Subject: RE: [Sipping] Privacy comment (was Re: calling number blocking at sip/isup gate way) Actually, I have a few slight confusions about the text below (it seems to use UA and user interchangeably a bit, which is worrisome when we're talking about proving identity). One question would probably clear things up: If the proxy still needs to authenticate the user (through whichever mechanism) to ensure that the identity that has been asserted in the RPID is valid for the user in question, why does the user need to supply an RPID at all? Why doesn't the user just authenticate itself to the proxy (however that happens) and let the proxy insert the RPID corresponding to the identity that the user has proven? This works fine for cases in which a single UA is used by multiple users, or when a single user has multiple provable identities. It assumes that you have a one-to-one mapping between credentials and identities, but that's usually par for the course. Again, if the proxy server holds a mapping between credentials and valid identities itself, as you suggest below, it doesn't seem at all unreasonable to adopt this model. Jon Peterson NeuStar, Inc. -----Original Message----- From: Mark Watson [mailto:mwatson@nortelnetworks.com] Sent: Tuesday, February 19, 2002 12:29 PM To: sipping@ietf.org Subject: [Sipping] Privacy comment (was Re: calling number blocking at sip/isup gate way) As presently described in privacy-03, the Remote-Party-ID header is only ever inserted by proxies, based on the proxy's knowledge of the identity of the UA (which it determines from some authentication mechanism out of scope of the privacy draft). Privacy-02 allowed the UA to insert Remote-Party-ID too. I think this should be reintroduced for the *specific case* of a UA with multiple identities, to allow the UA to choose which identity it wishes to be known by. The Proxy would still be responsible for ensuring that this identity was valid for the user and would still use some other authentication scheme to authenticate the user. The proxy would need knowledge of which identities were valid for which users. I believe this is a requirement for 3GPP, where a UA can have multiple 'public identities' and chooses which of these it wishes to be known by for each call. The proxy authenticates the UA by some other means and then checks the subscription data to ensure it is really allowed to use the public Identity it has included. I've suggested this a few times, and there has not been much comment for or against. It would be good if we could get a resolution before the next version of privacy is published. So, comments ? Regards...Mark > -----Original Message----- > From: Dean Willis [ mailto:dean.willis@softarmor.com ] > Sent: 18 February 2002 23:12 > To: Watson, Mark [MDN05:EP10:EXCH]; Mpierce1@aol.com; > jon.peterson@neustar.biz; sipping@ietf.org > Subject: Re: [Sipping] RE: [Sip] calling number blocking at sip/isup > gateway > > > Mark said: > > > Your example below needs only a small extension to > illustrate Mike's point > > and show why something like Remote-Party-ID as defined in > the privacy > draft > > is needed, and how it works despite the lack of a signature. > > Yep. And what happens when my misconfigured proxy forwards > the call without > paying attention to the Privacy header, because for some reason it's > forwarding to the outside world instead of a PSTN gateway > under my control? > We get a privacy leak. > > But other than that, I'd say you've precisely described a > "sunny day" use > case for RPID. > > It still has a lot of security issues based on transitive > trust, which make > it a lot more palatable in a two-party case than in three or > more party > transitivity. > > I believe we're approaching a general consensus that the Privacy draft > provides a reasonable approach to solving a specific class of problem > wherein interoperator-transitive trust is couple to > transport-layer message > protection. That's not a general privacy mechanism, but it > does seem to have > enough constituency to justify publication as long as it has a REALLY > serious applicability statement explaining the particular > requirements and > environments for which it is useful. > > How does that sound? > > -- > Dean > > ------_=_NextPart_001_01C1BEFD.65C13A10 Content-Type: text/html; charset="iso-8859-1" Privacy comment (was Re: calling number blocking at sip/isup gateway)
The difference between RPID and the existing headers is that RPID includes support for Privacy. The UA cannot include anything in any of the existing headers that the user is not happy to have presented to other users. However, if the UA trusts the proxy, it can insert such information in the RPID.
 
If you accept that an RPID inserted by a proxy might need to be based on 'evaluation of the [existing] headers ...', then you accept that it is not sufficient to have it based only on the authentication result. But if you only allow the existing headers to be evaluated, you limit the information the UA can provide for this purpose to 'public' information.
 
In the 3GPP case, your proposal would equate to placing the 'public identity' in the From, Reply-To or Contact field:
 
'From' cannot be used (in all cases) because the UA cannot be required to reveal the public identity to other users (due to Data Protection), and the proxy cannot modify the From field. The From field therefore contains a crypto-random hash or something similar.
 
Both Reply-To and Contact have meanings of their own, which may result in a need to include in them a different address from the 'public identity' making the call. As with From, Contact is limited to containing information which can be revealed to other users and although Reply-To could be policed out by a proxy before forwarding to an untrusted element, there is no way for the UA to indicate to the Proxy that it should do this (although I guess this could be added).
 
The UA is required to declare the public identity that is making the call to the network, though.
 
I agree with your concern about proliferation of 'identity' headers, and I did advocate implementing Reply-To using RPID, which would have reduced the count by one. However, headers without support for privacy are all useless in an environment where the UA is required to reveal certain information to the network (proxies) to access a service, but where that information does not _need to_ be passed to other users in order for the service to operate. In these circumstances the calling user has a right to keep such information private from other users. So the UA needs some way of ensuring that the proxy knows this information must not be so revealed. Encryption is always a (perhaps cleaner) solution, but has its own problems, and as this scenario never exists without a trust relationship between UA and proxy, RPID does the job.
 
Regards...Mark
 
 
 -----Original Message-----
From: Peterson, Jon [mailto:jon.peterson@neustar.biz]
Sent: 26 February 2002 05:36
To: Watson, Mark [MDN05:EP10:EXCH]; sipping@ietf.org
Subject: RE: [Sipping] Privacy comment (was Re: calling number blocking at sip/isup gate way)

Although I appreciate the explanations that you and Dean have given, I don't think I really understand what these public and private identities signify. I do believe that the scope of the RPID seems much more comprehensible to me if it is restricted to network-asserted identity. Again, there are already a number of fields in SIP where the originating user, and user agent, can represent themselves with URIs. The user can populate the From and Reply-To (not to mention Authorization) header fields with AoR identities representing themselves, and the user agent can populate the Contact header with a contact address representing itself. Clearly, there is no requirement that these all say the same thing, and I'd be surprised to learn that in combination they can't do more or less what you'd need. I'm not sure, anyway, why the solution would be to throw more headers at the problem. RPID makes sense to me scoped as an identity that is asserted by network operators based on their assessment of the user's identity, possibly resulting from the evaluation of the headers mentioned above (or even lower-layer or out-of-band security functions).
 
I think your call center architecture below is permitted today by the existing baseline authentication system. There is no necessary correspondence between the username of the URI given in the From field in a request and the username/password that is provided by a user agent in response to a challenge. While many proxy servers would legitimately require that the two be the same, clearly your call center's proxy would be happy to allow the From field to represent the call center's AoR (an 800-number, say), while the username/password provides credentials that identify the user. An even better scenario might involve the Reply-To header as well, providing a means for the callee to reach the same call center agent when appropriate. I don't think the requirements of this architecture, at any rate, give rise to a need for the RPID, let alone for a user to provide an RPID.
 
Jon Peter so n
NeuStar, Inc.
-----Original Message-----
From: Mark Watson [mailto:mwatson@nortelnetworks.com]
Sent: Monday, February 25, 2002 4:46 AM
To: Peterson, Jon; sipping@ietf.org
Subject: RE: [Sipping] Privacy comment (was Re: calling number blocking at sip/isup gate way)

Jon wrote:
 
> This works fine for cases in which a single UA is used by multiple users, or when a single user has multiple provable identities.
> It assumes that you have a one-to-one mapping between credentials and identities, but that's usually par for the course.
 
This assumption does not hold in the 3GPP case where authentication is based on a 'private identity' and associated keys in the SIM, but where there may be multiple 'public identies' associated with this single 'private identity'. In particular, this allows you easily to add/remove/manage the 'public identies' without getting involved in key management/distribution. All that is necessary to manage to public identities is to update the subscription information that the proxy uses to verify that a particular public identity is allowed for the particular private identity.
 
In addition I can think of cases where the identity that the user may wish to have presented to the terminating UA, to the PSTN or other elements, may differ from the identity that they use for local authentication. For example, imagine an outbound call centre where all the agents use SIP terminals. For obvious reasons each agent must authenticate to the local proxy using a personal identity, but the identity they wish to be known by to the called party is the Freephone number of the call centre (or even a different, inbound, call centre).
 
Cheers...Mark
 
 
 -----Original Message-----
From: Peterson, Jon [mailto:jon.peterson@neustar.biz]
Sent: 22 February 2002 18:26
To: Watson, Mark [MDN05:EP10:EXCH]; sipping@ietf.org
Subject: RE: [Sipping] Privacy comment (was Re: calling number blocking at sip/isup gate way)

Actually, I have a few slight confusions about the text below (it seems to use UA and user interchangeably a bit, which is worrisome when we're talking about proving identity). One question would probably clear things up:
 
If the proxy still needs to authenticate the user (through whichever mechanism) to ensure that the identity that has been asserted in the RPID is valid for the user in question, why does the user need to supply an RPID at all? Why doesn't the user just authenticate itself to the proxy (however that happens) and let the proxy insert the RPID corresponding to the identity that the user has proven? This works fine for cases in which a single UA is used by multiple users, or when a single user has multiple provable identities. It assumes that you have a one-to-one mapping between credentials and identities, but that's usually par for the course.
 
Again, if the proxy server holds a mapping between credentials and valid identities itself, as you suggest below, it doesn't seem at all unreasonable to adopt this model.
 
Jon Peterson
NeuStar, Inc.
-----Original Message-----
From: Mark Watson [mailto:mwatson@nortelnetworks.com]
Sent: Tuesday, February 19, 2002 12:29 PM
To: sipping@ietf.org
Subject: [Sipping] Privacy comment (was Re: calling number blocking at sip/isup gate way)

As presently described in privacy-03, the Remote-Party-ID header is only ever inserted by proxies, based on the proxy's knowledge of the identity of the UA (which it determines from some authentication mechanism out of scope of the privacy draft).

Privacy-02 allowed the UA to insert Remote-Party-ID too. I think this should be reintroduced for the *specific case* of a UA with multiple identities, to allow the UA to choose which identity it wishes to be known by. The Proxy would still be responsible for ensuring that this identity was valid for the user and would still use some other authentication scheme to authenticate the user. The proxy would need knowledge of which identities were valid for which users.

I believe this is a requirement for 3GPP, where a UA can have multiple 'public identities' and chooses which of these it wishes to be known by for each call. The proxy authenticates the UA by some other means and then checks the subscription data to ensure it is really allowed to use the public Identity it has included.

I've suggested this a few times, and there has not been much comment for or against. It would be good if we could get a resolution before the next version of privacy is published.

So, comments ?

Regards...Mark





> -----Original Message-----
> From: Dean Willis [mailto:dean.willis@softarmor.com]
> Sent: 18 February 2002 23:12
> To: Watson, Mark [MDN05:EP10:EXCH]; Mpierce1@aol.com;
> jon.peterson@neustar.biz; sipping@ietf.org
> Subject: Re: [Sipping] RE: [Sip] calling number blocking at sip/isup
> gateway
>
>
> Mark said:
>
> > Your example below needs only a small extension to
> illustrate Mike's point
> > and show why something like Remote-Party-ID as defined in
> the privacy
> draft
> > is needed, and how it works despite the lack of a signature.
>
> Yep. And what happens when my misconfigured proxy forwards
> the call without
> paying attention to the Privacy header, because for some reason it's
> forwarding to the outside world instead of a PSTN gateway
> under my control?
> We get a privacy leak.
>
> But other than that, I'd say you've precisely described a
> "sunny day" use
> case for RPID.
>
> It still has a lot of security issues based on transitive
> trust, which make
> it a lot more palatable in a two-party case than in three or
> more party
> transitivity.
>
> I believe we're approaching a general consensus that the Privacy draft
> provides a reasonable approach to solving a specific class of problem
> wherein interoperator-transitive trust is couple to
> transport-layer message
> protection. That's not a general privacy mechanism, but it
> does seem to have
> enough constituency to justify publication as long as it has a REALLY
> serious applicability statement explaining the particular
> requirements and
> environments for which it is useful.
>
> How does that sound?
>
> --
> Dean
>
>

------_=_NextPart_001_01C1BEFD.65C13A10-- _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Tue Feb 26 16:32:50 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id QAA22299 for ; Tue, 26 Feb 2002 16:32:50 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id QAA22362 for sipping-archive@odin.ietf.org; Tue, 26 Feb 2002 16:32:54 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id QAA21263; Tue, 26 Feb 2002 16:15:42 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id QAA21232 for ; Tue, 26 Feb 2002 16:15:39 -0500 (EST) Received: from imo-r06.mx.aol.com (imo-r06.mx.aol.com [152.163.225.102]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id QAA21922 for ; Tue, 26 Feb 2002 16:15:35 -0500 (EST) From: Mpierce1@aol.com Received: from Mpierce1@aol.com by imo-r06.mx.aol.com (mail_out_v32.5.) id l.193.2eb2892 (661) for ; Tue, 26 Feb 2002 16:15:02 -0500 (EST) Message-ID: <193.2eb2892.29ad54d5@aol.com> Date: Tue, 26 Feb 2002 16:15:01 EST To: sipping@ietf.org MIME-Version: 1.0 Content-Type: text/plain; charset="US-ASCII" Content-Transfer-Encoding: 7bit X-Mailer: AOL 5.0 for Windows sub 138 Content-Transfer-Encoding: 7bit Subject: [Sipping] New Assured Service draft Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit To all, A revision to the ID for Assured Service definition in SIP has been submitted which provides changes based on those comments received. This ID defines the requirements for Assured Service in SIP to provide prioritized handling for certain sessions established using SIP. It also discusses some of the factors to be taken into account when planning protocol revisions which may be required. The draft can be found at: http://ietf.org/internet-drafts/draft-pierce-sipping-assured-service-01.txt Mike Pierce _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Tue Feb 26 16:47:09 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id QAA22801 for ; Tue, 26 Feb 2002 16:47:08 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id QAA23010 for sipping-archive@odin.ietf.org; Tue, 26 Feb 2002 16:47:13 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id QAA22290; Tue, 26 Feb 2002 16:31:09 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id QAA22259 for ; Tue, 26 Feb 2002 16:31:06 -0500 (EST) Received: from imo-r07.mx.aol.com (imo-r07.mx.aol.com [152.163.225.103]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id QAA22284 for ; Tue, 26 Feb 2002 16:31:01 -0500 (EST) From: Mpierce1@aol.com Received: from Mpierce1@aol.com by imo-r07.mx.aol.com (mail_out_v32.5.) id 7.171.9778f2b (4207); Tue, 26 Feb 2002 16:29:52 -0500 (EST) Message-ID: <171.9778f2b.29ad5850@aol.com> Date: Tue, 26 Feb 2002 16:29:52 EST To: schulzrinne@cs.columbia.edu, jmpolk@cisco.com, sipping@ietf.org MIME-Version: 1.0 Content-Type: text/plain; charset="US-ASCII" Content-Transfer-Encoding: 7bit X-Mailer: AOL 5.0 for Windows sub 138 Content-Transfer-Encoding: 7bit Subject: [Sipping] Draft resource priority Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit Minor comment: In the new Annex B, it states: with "0" being the lowest value and the "4" the highest. Current Q.735 defines "0" as the highest and "4" as the lowest. Although this seems non-intuitive, this definition should be retained there. The idea is for SIP proxies to not have to do anything with this field - just copy it directly from/to the Q.735 message (except maybe check that it is in the range 0-4). Mike _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Tue Feb 26 19:23:49 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id TAA26174 for ; Tue, 26 Feb 2002 19:23:49 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id TAA01197 for sipping-archive@odin.ietf.org; Tue, 26 Feb 2002 19:23:51 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id TAA00299; Tue, 26 Feb 2002 19:02:17 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id TAA00262 for ; Tue, 26 Feb 2002 19:02:13 -0500 (EST) Received: from oak.neustar.com (oak.neustar.com [209.173.53.70]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id TAA25666 for ; Tue, 26 Feb 2002 19:02:08 -0500 (EST) Received: from chiimc01.il.neustar.com (dmz1.va.neustar.com [209.173.53.65]) by oak.neustar.com (8.11.0/8.11.0) with ESMTP id g1R01c822830; Tue, 26 Feb 2002 19:01:38 -0500 Received: by chiimc01.il.neustar.com with Internet Mail Service (5.5.2653.19) id ; Tue, 26 Feb 2002 18:01:32 -0600 Message-ID: <70565611B164D511957A001083FCDD5601870109@VA02> From: "Peterson, Jon" To: "'Mark Watson'" , sipping@ietf.org Subject: RE: [Sipping] Privacy comment (was Re: calling number blocking at sip/isup gate way) Date: Tue, 26 Feb 2002 18:01:31 -0600 MIME-Version: 1.0 X-Mailer: Internet Mail Service (5.5.2653.19) Content-Type: text/plain; charset="iso-8859-1" Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Well, I mean, it's three days to IETF cut-off and sip-privacy needs to pass. There's only so much we're going to be able to agree to in this time frame. I continue to feel strongly that we should restrict the scope of RPID to network-asserted identity. If it is in this scope, then I for one wouldn't oppose the passage of the draft. Just to be clear, what I object to here is the use of sip-privacy-03 for user-asserted identity, not for user-requested privacy. I can live with allowing UAs to add RPID-Privacy to requests; once the user then proves the identity that will be asserted by the network in the RPID, the network can include the privacy preferences expressed in the RPID-P - although of course IMHO having separate headers for either user-asserted identity or privacy is cumbersome. Below I don't believe that you substantiate any new need for user-asserted identity. To the 'evaluation of the existing headers' point, Proxy-Authorization is such an existing header - so in fact I'm not arguing that the result of authentication is insufficient; I was just speaking to the fact that a 'public' identity sounds like a name for the identity that a user would like other users to see (is there a name for that identity in your scheme?). I would also note that there are numerous elements of RPID in sip-privacy-03 (including the screening indicator, the presence of a "privacy" parameter in RPID separate from the RPID-Privacy header) that suggest to me that RPID should not be added by untrusted entities (i.e. users) - eliminating or reinterpreting these would seem to change the sense of RPID a bit. To go into this much further will probably raise more serious questions about the draft overall. I can't say I would recommend that at this juncture. Suffice it to say that I hope you agree that Basic Privacy (sip-privacy-03 7.1) is not going to be regulatory-grade in most countries. Jon Peterson NeuStar, Inc. -----Original Message----- From: Mark Watson [mailto:mwatson@nortelnetworks.com] Sent: Tuesday, February 26, 2002 11:40 AM To: Peterson, Jon; sipping@ietf.org Subject: RE: [Sipping] Privacy comment (was Re: calling number blocking at sip/isup gate way) The difference between RPID and the existing headers is that RPID includes support for Privacy. The UA cannot include anything in any of the existing headers that the user is not happy to have presented to other users. However, if the UA trusts the proxy, it can insert such information in the RPID. If you accept that an RPID inserted by a proxy might need to be based on 'evaluation of the [existing] headers ...', then you accept that it is not sufficient to have it based only on the authentication result. But if you only allow the existing headers to be evaluated, you limit the information the UA can provide for this purpose to 'public' information. In the 3GPP case, your proposal would equate to placing the 'public identity' in the From, Reply-To or Contact field: 'From' cannot be used (in all cases) because the UA cannot be required to reveal the public identity to other users (due to Data Protection), and the proxy cannot modify the From field. The From field therefore contains a crypto-random hash or something similar. Both Reply-To and Contact have meanings of their own, which may result in a need to include in them a different address from the 'public identity' making the call. As with From, Contact is limited to containing information which can be revealed to other users and although Reply-To could be policed out by a proxy before forwarding to an untrusted element, there is no way for the UA to indicate to the Proxy that it should do this (although I guess this could be added). The UA is required to declare the public identity that is making the call to the network, though. I agree with your concern about proliferation of 'identity' headers, and I did advocate implementing Reply-To using RPID, which would have reduced the count by one. However, headers without support for privacy are all useless in an environment where the UA is required to reveal certain information to the network (proxies) to access a service, but where that information does not _need to_ be passed to other users in order for the service to operate. In these circumstances the calling user has a right to keep such information private from other users. So the UA needs some way of ensuring that the proxy knows this information must not be so revealed. Encryption is always a (perhaps cleaner) solution, but has its own problems, and as this scenario never exists without a trust relationship between UA and proxy, RPID does the job. Regards...Mark -----Original Message----- From: Peterson, Jon [mailto:jon.peterson@neustar.biz] Sent: 26 February 2002 05:36 To: Watson, Mark [MDN05:EP10:EXCH]; sipping@ietf.org Subject: RE: [Sipping] Privacy comment (was Re: calling number blocking at sip/isup gate way) Although I appreciate the explanations that you and Dean have given, I don't think I really understand what these public and private identities signify. I do believe that the scope of the RPID seems much more comprehensible to me if it is restricted to network-asserted identity. Again, there are already a number of fields in SIP where the originating user, and user agent, can represent themselves with URIs. The user can populate the From and Reply-To (not to mention Authorization) header fields with AoR identities representing themselves, and the user agent can populate the Contact header with a contact address representing itself. Clearly, there is no requirement that these all say the same thing, and I'd be surprised to learn that in combination they can't do more or less what you'd need. I'm not sure, anyway, why the solution would be to throw more headers at the problem. RPID makes sense to me scoped as an identity that is asserted by network operators based on their assessment of the user's identity, possibly resulting from the evaluation of the headers mentioned above (or even lower-layer or out-of-band security functions). I think your call center architecture below is permitted today by the existing baseline authentication system. There is no necessary correspondence between the username of the URI given in the From field in a request and the username/password that is provided by a user agent in response to a challenge. While many proxy servers would legitimately require that the two be the same, clearly your call center's proxy would be happy to allow the From field to represent the call center's AoR (an 800-number, say), while the username/password provides credentials that identify the user. An even better scenario might involve the Reply-To header as well, providing a means for the callee to reach the same call center agent when appropriate. I don't think the requirements of this architecture, at any rate, give rise to a need for the RPID, let alone for a user to provide an RPID. Jon Peter so n NeuStar, Inc. -----Original Message----- From: Mark Watson [mailto:mwatson@nortelnetworks.com] Sent: Monday, February 25, 2002 4:46 AM To: Peterson, Jon; sipping@ietf.org Subject: RE: [Sipping] Privacy comment (was Re: calling number blocking at sip/isup gate way) Jon wrote: > This works fine for cases in which a single UA is used by multiple users, or when a single user has multiple provable identities. > It assumes that you have a one-to-one mapping between credentials and identities, but that's usually par for the course. This assumption does not hold in the 3GPP case where authentication is based on a 'private identity' and associated keys in the SIM, but where there may be multiple 'public identies' associated with this single 'private identity'. In particular, this allows you easily to add/remove/manage the 'public identies' without getting involved in key management/distribution. All that is necessary to manage to public identities is to update the subscription information that the proxy uses to verify that a particular public identity is allowed for the particular private identity. In addition I can think of cases where the identity that the user may wish to have presented to the terminating UA, to the PSTN or other elements, may differ from the identity that they use for local authentication. For example, imagine an outbound call centre where all the agents use SIP terminals. For obvious reasons each agent must authenticate to the local proxy using a personal identity, but the identity they wish to be known by to the called party is the Freephone number of the call centre (or even a different, inbound, call centre). Cheers...Mark -----Original Message----- From: Peterson, Jon [mailto:jon.peterson@neustar.biz] Sent: 22 February 2002 18:26 To: Watson, Mark [MDN05:EP10:EXCH]; sipping@ietf.org Subject: RE: [Sipping] Privacy comment (was Re: calling number blocking at sip/isup gate way) Actually, I have a few slight confusions about the text below (it seems to use UA and user interchangeably a bit, which is worrisome when we're talking about proving identity). One question would probably clear things up: If the proxy still needs to authenticate the user (through whichever mechanism) to ensure that the identity that has been asserted in the RPID is valid for the user in question, why does the user need to supply an RPID at all? Why doesn't the user just authenticate itself to the proxy (however that happens) and let the proxy insert the RPID corresponding to the identity that the user has proven? This works fine for cases in which a single UA is used by multiple users, or when a single user has multiple provable identities. It assumes that you have a one-to-one mapping between credentials and identities, but that's usually par for the course. Again, if the proxy server holds a mapping between credentials and valid identities itself, as you suggest below, it doesn't seem at all unreasonable to adopt this model. Jon Peterson NeuStar, Inc. -----Original Message----- From: Mark Watson [mailto:mwatson@nortelnetworks.com] Sent: Tuesday, February 19, 2002 12:29 PM To: sipping@ietf.org Subject: [Sipping] Privacy comment (was Re: calling number blocking at sip/isup gate way) As presently described in privacy-03, the Remote-Party-ID header is only ever inserted by proxies, based on the proxy's knowledge of the identity of the UA (which it determines from some authentication mechanism out of scope of the privacy draft). Privacy-02 allowed the UA to insert Remote-Party-ID too. I think this should be reintroduced for the *specific case* of a UA with multiple identities, to allow the UA to choose which identity it wishes to be known by. The Proxy would still be responsible for ensuring that this identity was valid for the user and would still use some other authentication scheme to authenticate the user. The proxy would need knowledge of which identities were valid for which users. I believe this is a requirement for 3GPP, where a UA can have multiple 'public identities' and chooses which of these it wishes to be known by for each call. The proxy authenticates the UA by some other means and then checks the subscription data to ensure it is really allowed to use the public Identity it has included. I've suggested this a few times, and there has not been much comment for or against. It would be good if we could get a resolution before the next version of privacy is published. So, comments ? Regards...Mark > -----Original Message----- > From: Dean Willis [mailto:dean.willis@softarmor.com] > Sent: 18 February 2002 23:12 > To: Watson, Mark [MDN05:EP10:EXCH]; Mpierce1@aol.com; > jon.peterson@neustar.biz; sipping@ietf.org > Subject: Re: [Sipping] RE: [Sip] calling number blocking at sip/isup > gateway > > > Mark said: > > > Your example below needs only a small extension to > illustrate Mike's point > > and show why something like Remote-Party-ID as defined in > the privacy > draft > > is needed, and how it works despite the lack of a signature. > > Yep. And what happens when my misconfigured proxy forwards > the call without > paying attention to the Privacy header, because for some reason it's > forwarding to the outside world instead of a PSTN gateway > under my control? > We get a privacy leak. > > But other than that, I'd say you've precisely described a > "sunny day" use > case for RPID. > > It still has a lot of security issues based on transitive > trust, which make > it a lot more palatable in a two-party case than in three or > more party > transitivity. > > I believe we're approaching a general consensus that the Privacy draft > provides a reasonable approach to solving a specific class of problem > wherein interoperator-transitive trust is couple to > transport-layer message > protection. That's not a general privacy mechanism, but it > does seem to have > enough constituency to justify publication as long as it has a REALLY > serious applicability statement explaining the particular > requirements and > environments for which it is useful. > > How does that sound? > > -- > Dean > > _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Tue Feb 26 19:36:29 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id TAA26393 for ; Tue, 26 Feb 2002 19:36:29 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id TAA01895 for sipping-archive@odin.ietf.org; Tue, 26 Feb 2002 19:36:31 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id TAA00483; Tue, 26 Feb 2002 19:09:07 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id TAA00452 for ; Tue, 26 Feb 2002 19:09:05 -0500 (EST) Received: from imr2.ericy.com (imr2.ericy.com [198.24.6.3]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id TAA25844 for ; Tue, 26 Feb 2002 19:09:02 -0500 (EST) Received: from mr5.exu.ericsson.se (mr5att.ericy.com [138.85.224.141]) by imr2.ericy.com (8.11.3/8.11.3) with ESMTP id g1R08PS01137; Tue, 26 Feb 2002 18:08:25 -0600 (CST) Received: from newman.exu.ericsson.se (newman.exu.ericsson.se [138.85.75.179]) by mr5.exu.ericsson.se (8.11.3/8.11.3) with ESMTP id g1R08PW25564; Tue, 26 Feb 2002 18:08:25 -0600 (CST) Received: from lmf.ericsson.se (rmt160217.am.ericsson.se [138.85.160.217]) by newman.exu.ericsson.se (8.7.5/8.7.3) with ESMTP id SAA27898; Tue, 26 Feb 2002 18:08:23 -0600 (CST) Message-ID: <3C7C2395.DDD29A34@lmf.ericsson.se> Date: Wed, 27 Feb 2002 02:08:53 +0200 From: Gonzalo Camarillo X-Mailer: Mozilla 4.79 [en] (WinNT; U) X-Accept-Language: en MIME-Version: 1.0 To: Mark Watson CC: "Peterson, Jon" , sipping@ietf.org Subject: Re: [Sipping] Privacy comment (was Re: calling number blocking atsip/isup gate way) References: Content-Type: text/plain; charset=us-ascii Content-Transfer-Encoding: 7bit Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit Mark writes: >'From' cannot be used (in all cases) because the UA cannot be required > to reveal the public identity to other users (due to Data >Protection), and the proxy cannot modify the From field. The From field > therefore contains a crypto-random hash or something >similar. Now a proxy can modify the From field. Dialog identification relies solely on tags. Gonzalo _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Tue Feb 26 20:43:04 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id UAA27323 for ; Tue, 26 Feb 2002 20:43:04 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id UAA05008 for sipping-archive@odin.ietf.org; Tue, 26 Feb 2002 20:43:06 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id UAA03992; Tue, 26 Feb 2002 20:25:49 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id UAA03961 for ; Tue, 26 Feb 2002 20:25:47 -0500 (EST) Received: from sj-msg-core-1.cisco.com (sj-msg-core-1.cisco.com [171.71.163.11]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id UAA27103 for ; Tue, 26 Feb 2002 20:25:43 -0500 (EST) Received: from nisser.cisco.com (nisser.cisco.com [171.71.176.85]) by sj-msg-core-1.cisco.com (8.11.3/8.9.1) with ESMTP id g1R1PCh15644; Tue, 26 Feb 2002 17:25:12 -0800 (PST) Received: from cisco.com (ssh-sj1.cisco.com [171.68.225.134]) by nisser.cisco.com (8.8.6 (PHNE_14041)/CISCO.SERVER.1.2) with ESMTP id RAA13294; Tue, 26 Feb 2002 17:25:10 -0800 (PST) Message-ID: <3C7C3578.CC39280B@cisco.com> Date: Tue, 26 Feb 2002 20:25:12 -0500 From: Flemming Andreasen X-Mailer: Mozilla 4.79 [en]C-CCK-MCD (Windows NT 5.0; U) X-Accept-Language: en MIME-Version: 1.0 To: "Peterson, Jon" CC: "'Mark Watson'" , sipping@ietf.org Subject: Re: [Sipping] Privacy comment (was Re: calling number blocking atsip/isup gate way) References: <70565611B164D511957A001083FCDD5601870109@VA02> Content-Type: text/plain; charset=us-ascii Content-Transfer-Encoding: 7bit Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit I personally have little preference one way or the other here, but at this point I need consensus. The current privacy-03 draft specifically does not talk about untrusted UAs inserting Remote-Party-Id headers, however it defines what a proxy does when it receives a message with a Remote-Party-Id from either a trusted or untrusted entity (which of course can be either a proxy or a UA). I propose to just leave it at that, which hopefully will give everybody enough of what they want. If you have a major problem with this, speak up now or forever hold your peace Thanks Flemming "Peterson, Jon" wrote: > Well, I mean, it's three days to IETF cut-off and sip-privacy needs to pass. > There's only so much we're going to be able to agree to in this time frame. > I continue to feel strongly that we should restrict the scope of RPID to > network-asserted identity. If it is in this scope, then I for one wouldn't > oppose the passage of the draft. > > Just to be clear, what I object to here is the use of sip-privacy-03 for > user-asserted identity, not for user-requested privacy. I can live with > allowing UAs to add RPID-Privacy to requests; once the user then proves the > identity that will be asserted by the network in the RPID, the network can > include the privacy preferences expressed in the RPID-P - although of > course IMHO having separate headers for either user-asserted identity or > privacy is cumbersome. > > Below I don't believe that you substantiate any new need for user-asserted > identity. > > To the 'evaluation of the existing headers' point, Proxy-Authorization is > such an existing header - so in fact I'm not arguing that the result of > authentication is insufficient; I was just speaking to the fact that a > 'public' identity sounds like a name for the identity that a user would like > other users to see (is there a name for that identity in your scheme?). I > would also note that there are numerous elements of RPID in sip-privacy-03 > (including the screening indicator, the presence of a "privacy" parameter in > RPID separate from the RPID-Privacy header) that suggest to me that RPID > should not be added by untrusted entities (i.e. users) - eliminating or > reinterpreting these would seem to change the sense of RPID a bit. > > To go into this much further will probably raise more serious questions > about the draft overall. I can't say I would recommend that at this > juncture. Suffice it to say that I hope you agree that Basic Privacy > (sip-privacy-03 7.1) is not going to be regulatory-grade in most countries. > > Jon Peterson > NeuStar, Inc. > > -----Original Message----- > From: Mark Watson [mailto:mwatson@nortelnetworks.com] > Sent: Tuesday, February 26, 2002 11:40 AM > To: Peterson, Jon; sipping@ietf.org > Subject: RE: [Sipping] Privacy comment (was Re: calling number blocking at > sip/isup gate way) > > The difference between RPID and the existing headers is that RPID includes > support for Privacy. The UA cannot include anything in any of the existing > headers that the user is not happy to have presented to other users. > However, if the UA trusts the proxy, it can insert such information in the > RPID. > > If you accept that an RPID inserted by a proxy might need to be based on > 'evaluation of the [existing] headers ...', then you accept that it is not > sufficient to have it based only on the authentication result. But if you > only allow the existing headers to be evaluated, you limit the information > the UA can provide for this purpose to 'public' information. > > In the 3GPP case, your proposal would equate to placing the 'public > identity' in the From, Reply-To or Contact field: > > 'From' cannot be used (in all cases) because the UA cannot be required to > reveal the public identity to other users (due to Data Protection), and the > proxy cannot modify the From field. The From field therefore contains a > crypto-random hash or something similar. > > Both Reply-To and Contact have meanings of their own, which may result in a > need to include in them a different address from the 'public identity' > making the call. As with From, Contact is limited to containing information > which can be revealed to other users and although Reply-To could be policed > out by a proxy before forwarding to an untrusted element, there is no way > for the UA to indicate to the Proxy that it should do this (although I guess > this could be added). > > The UA is required to declare the public identity that is making the call to > the network, though. > > I agree with your concern about proliferation of 'identity' headers, and I > did advocate implementing Reply-To using RPID, which would have reduced the > count by one. However, headers without support for privacy are all useless > in an environment where the UA is required to reveal certain information to > the network (proxies) to access a service, but where that information does > not _need to_ be passed to other users in order for the service to operate. > In these circumstances the calling user has a right to keep such information > private from other users. So the UA needs some way of ensuring that the > proxy knows this information must not be so revealed. Encryption is always a > (perhaps cleaner) solution, but has its own problems, and as this scenario > never exists without a trust relationship between UA and proxy, RPID does > the job. > > Regards...Mark > -- Flemming Andreasen Cisco Systems _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Tue Feb 26 21:00:42 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id VAA27615 for ; Tue, 26 Feb 2002 21:00:42 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id VAA05633 for sipping-archive@odin.ietf.org; Tue, 26 Feb 2002 21:00:15 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id UAA05170; Tue, 26 Feb 2002 20:45:37 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id UAA05124 for ; Tue, 26 Feb 2002 20:45:34 -0500 (EST) Received: from wells.cisco.com (wells.cisco.com [171.71.177.223]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id UAA27379 for ; Tue, 26 Feb 2002 20:45:30 -0500 (EST) Received: from JMPOLK-W2K (ssh-sj1.cisco.com [171.68.225.134]) by wells.cisco.com (8.8.6 (PHNE_14041)/CISCO.SERVER.1.2) with SMTP id RAA04333; Tue, 26 Feb 2002 17:44:30 -0800 (PST) Message-Id: <4.1.20020226194230.02bdceb8@localhost> X-Sender: jmpolk@localhost X-Mailer: QUALCOMM Windows Eudora Pro Version 4.1 Date: Tue, 26 Feb 2002 19:44:05 -0600 To: Mpierce1@aol.com, schulzrinne@cs.columbia.edu, sipping@ietf.org From: "James M. Polk" In-Reply-To: <171.9778f2b.29ad5850@aol.com> Mime-Version: 1.0 Content-Type: multipart/alternative; boundary="=====================_1114894323==_.ALT" Subject: [Sipping] Re: Draft resource priority Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org --=====================_1114894323==_.ALT Content-Type: text/plain; charset="us-ascii" Mike I saw that inaccuracy, but didn't somehow correct for it. It will be corrected in the v -01 of this ID. Thank you for the heads-up At 04:29 PM 2/26/2002 -0500, Mpierce1@aol.com wrote: >Minor comment: > >In the new Annex B, it states: > >with "0" being the lowest value and the "4" the highest. > >Current Q.735 defines "0" as the highest and "4" as the lowest. Although this >seems non-intuitive, this definition should be retained there. The idea is >for SIP proxies to not have to do anything with this field - just copy it >directly from/to the Q.735 message (except maybe check that it is in the >range 0-4). > >Mike ************************************* "People generally demand more respect for their own rights than they are willing to allow for others" James M. Polk Consulting Engineer Office of the CTO Cisco Systems 2200 East President George Bush Turnpike Richardson, TX 75082 USA w) 972.813.5208 f) 972.813.5280 www.cisco.com --=====================_1114894323==_.ALT Content-Type: text/html; charset="us-ascii"
Mike

I saw that inaccuracy, but didn't somehow correct for it. It will be corrected in the v -01 of this ID.

Thank you for the heads-up

At 04:29 PM 2/26/2002 -0500, Mpierce1@aol.com wrote:
>Minor comment:
>
>In the new Annex B, it states:
>
>with "0" being the lowest value and the "4" the highest.
>
>Current Q.735 defines "0" as the highest and "4" as the lowest. Although this
>seems non-intuitive, this definition should be retained there. The idea is
>for SIP proxies to not have to do anything with this field - just copy it
>directly from/to the Q.735 message (except maybe check that it is in the
>range 0-4).
>
>Mike

*************************************
"People generally demand more respect for their own rights than they are willing to allow for others"

James M. Polk
Consulting Engineer
Office of the CTO

Cisco Systems
2200 East President George Bush Turnpike
Richardson, TX  75082 USA
w) 972.813.5208
f)  972.813.5280
www.cisco.com --=====================_1114894323==_.ALT-- _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Tue Feb 26 23:00:10 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id XAA01390 for ; Tue, 26 Feb 2002 23:00:10 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id XAA10802 for sipping-archive@odin.ietf.org; Tue, 26 Feb 2002 23:00:14 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id WAA10105; Tue, 26 Feb 2002 22:42:14 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id WAA10078 for ; Tue, 26 Feb 2002 22:42:12 -0500 (EST) Received: from bdsl.66.12.12.130.gte.net (bdsl.66.12.12.130.gte.net [66.12.12.130]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id WAA01045 for ; Tue, 26 Feb 2002 22:42:07 -0500 (EST) Received: from C1893415A (bdsl.greycouncil.com [127.0.0.1]) by bdsl.66.12.12.130.gte.net (8.11.6/8.11.6) with SMTP id g1R3fXB31511; Tue, 26 Feb 2002 21:41:33 -0600 Message-ID: <002501c1bf40$a0e74dd0$133fed0c@C1893415A> From: "Dean Willis" To: "Mark Watson" , "Gonzalo Camarillo" Cc: , "Peterson, Jon" References: <3C7C2395.DDD29A34@lmf.ericsson.se> Subject: Re: [Sipping] Privacy comment (was Re: calling number blocking atsip/isup gate way) Date: Tue, 26 Feb 2002 21:41:22 -0600 MIME-Version: 1.0 Content-Type: text/plain; charset="iso-8859-1" Content-Transfer-Encoding: 7bit X-Priority: 3 X-MSMail-Priority: Normal X-Mailer: Microsoft Outlook Express 6.00.2600.0000 X-MimeOLE: Produced By Microsoft MimeOLE V6.00.2600.0000 Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit Ah, sort of. Modifying a To or From field will still break 2543 systems, so it's not backward compatible, and there's no "test" for this to allow it to be detected, so modifying them in a proxy is still a bad idea. Also, I think there are some issues with enhanced digest here . . . -- Dean ----- Original Message ----- From: "Gonzalo Camarillo" To: "Mark Watson" Cc: "Peterson, Jon" ; Sent: Tuesday, February 26, 2002 6:08 PM Subject: Re: [Sipping] Privacy comment (was Re: calling number blocking atsip/isup gate way) > > Mark writes: > >'From' cannot be used (in all cases) because the UA cannot be required > > to reveal the public identity to other users (due to Data > >Protection), and the proxy cannot modify the From field. The From field > > therefore contains a crypto-random hash or something > >similar. > > Now a proxy can modify the From field. Dialog identification relies > solely on tags. > > Gonzalo > > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP > Use sip-implementors@cs.columbia.edu for questions on current sip > Use sip@ietf.org for new developments of core SIP > > _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Wed Feb 27 01:04:22 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id BAA03737 for ; Wed, 27 Feb 2002 01:04:21 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id BAA16196 for sipping-archive@odin.ietf.org; Wed, 27 Feb 2002 01:04:22 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id AAA15318; Wed, 27 Feb 2002 00:44:46 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id AAA15287 for ; Wed, 27 Feb 2002 00:44:43 -0500 (EST) Received: from albatross.wise.edt.ericsson.se (albatross-ext.wise.edt.ericsson.se [193.180.251.46]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id AAA03545 for ; Wed, 27 Feb 2002 00:44:42 -0500 (EST) Received: from fogerty.lmf.ericsson.se (fogerty.lmf.ericsson.se [131.160.11.6]) by albatross.wise.edt.ericsson.se (8.12.1/8.12.1/WIREfire-1.4) with ESMTP id g1R5iZZc012745; Wed, 27 Feb 2002 06:44:35 +0100 (MET) Received: from lmf.ericsson.se (3OQK900K04BAB3K.lmf.ericsson.se [131.160.30.135]) by fogerty.lmf.ericsson.se (8.12.1/8.12.1/lmf.8.12.1.jcs) with ESMTP id g1R5iYmK023484; Wed, 27 Feb 2002 07:44:34 +0200 (EET) Message-ID: <3C7C727D.30119BF9@lmf.ericsson.se> Date: Wed, 27 Feb 2002 07:45:33 +0200 From: Christer Holmberg X-Mailer: Mozilla 4.74 [en] (Windows NT 5.0; U) X-Accept-Language: en MIME-Version: 1.0 To: Dean Willis CC: Mark Watson , Gonzalo Camarillo , sipping@ietf.org, "Peterson, Jon" Subject: Re: [Sipping] Privacy comment (was Re: calling number blocking atsip/isup gate way) References: <3C7C2395.DDD29A34@lmf.ericsson.se> <002501c1bf40$a0e74dd0$133fed0c@C1893415A> Content-Type: text/plain; charset=us-ascii Content-Transfer-Encoding: 7bit Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit Hi, > Ah, sort of. Modifying a To or From field will still break 2543 systems, so > it's not backward compatible, and there's no "test" for this to allow it to > be detected, so modifying them in a proxy is still a bad idea. > > Also, I think there are some issues with enhanced digest here . . . It also depends on what "modify" really means. Are we, for example, allowed to "switch" between different URI schemes? And, IF a proxy is allowed to change the field, is there a reason why the end-user should not be allowed to do so? etc etc Regards, Christer Holmberg Ericsson Finland > > > -- > Dean > > ----- Original Message ----- > From: "Gonzalo Camarillo" > To: "Mark Watson" > Cc: "Peterson, Jon" ; > Sent: Tuesday, February 26, 2002 6:08 PM > Subject: Re: [Sipping] Privacy comment (was Re: calling number blocking > atsip/isup gate way) > > > > > Mark writes: > > >'From' cannot be used (in all cases) because the UA cannot be required > > > to reveal the public identity to other users (due to Data > > >Protection), and the proxy cannot modify the From field. The From field > > > therefore contains a crypto-random hash or something > > >similar. > > > > Now a proxy can modify the From field. Dialog identification relies > > solely on tags. > > > > Gonzalo > > > > _______________________________________________ > > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > > This list is for NEW development of the application of SIP > > Use sip-implementors@cs.columbia.edu for questions on current sip > > Use sip@ietf.org for new developments of core SIP > > > > > > _______________________________________________ > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > This list is for NEW development of the application of SIP > Use sip-implementors@cs.columbia.edu for questions on current sip > Use sip@ietf.org for new developments of core SIP _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Wed Feb 27 01:27:22 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id BAA04035 for ; Wed, 27 Feb 2002 01:27:21 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id BAA26688 for sipping-archive@odin.ietf.org; Wed, 27 Feb 2002 01:27:22 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id BAA20167; Wed, 27 Feb 2002 01:08:30 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id BAA20042 for ; Wed, 27 Feb 2002 01:08:23 -0500 (EST) Received: from auds953.usa.alcatel.com (auds953.usa.alcatel.com [143.209.238.6] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id BAA03784 for ; Wed, 27 Feb 2002 01:08:21 -0500 (EST) Received: from axsppc65 (localhost [127.0.0.1]) by auds953.usa.alcatel.com (8.10.2/8.10.2) with SMTP id g1R67pl20833 for ; Wed, 27 Feb 2002 00:07:52 -0600 (CST) Message-ID: <003401c1bf56$1784db00$7301ff0a@axsppc65> From: "Jagannathan R" To: Date: Wed, 27 Feb 2002 11:44:59 +0530 MIME-Version: 1.0 Content-Type: multipart/alternative; boundary="----=_NextPart_000_0031_01C1BF84.2F868D60" X-Priority: 3 X-MSMail-Priority: Normal X-Mailer: Microsoft Outlook Express 5.00.2919.6700 X-MimeOLE: Produced By Microsoft MimeOLE V5.00.2919.6700 Subject: [Sipping] SIP-PSTN Interconnection Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org This is a multi-part message in MIME format. ------=_NextPart_000_0031_01C1BF84.2F868D60 Content-Type: text/plain; charset="iso-8859-1" Content-Transfer-Encoding: quoted-printable Hi all, I have a some basic doubts regarding the SIP-PSTN Interworking, this is = more on SIP-T. Can any one clarify these doubts please. 1. Proxy server is connected to MGC (point of SIP-PSTN Interconnection), = 1a. Is there a way proxy server registering itself with MGC? =20 This is required at MGC to know which are the SIP users this proxy = can service,=20 so that the PSTN originated call can be correctly terminated to = that Proxy.=20 1b. How the MGC knows the status of Proxy (In-Service or Out-of-Service) = in order to terminate the call. 1c. How many Proxy servers with different/same set of SIP users can be = linked to one MGC. 2. When SIP call arrives at MGC, MGC requires NOA (Nature of Address) in = order to translate the called number and route the call to the correct = PSTN Trunk Group. How the NOA address (like 0+IDDD, IDDD, DDD calls) can = be represented in the SIP messages. 3. For SIP-PSTN call, how different digit sequences like ANI, = Authorization code, Account code (these are required for billing at MGC = and to have Dialing plan) can be sent to MGC in SIP Messages.=20 =20 Thanks in advance, =20 jagannathan ------=_NextPart_000_0031_01C1BF84.2F868D60 Content-Type: text/html; charset="iso-8859-1" Content-Transfer-Encoding: quoted-printable
Hi all,
 
I have a some basic doubts regarding = the SIP-PSTN=20 Interworking, this is more on SIP-T. Can any one clarify these doubts=20 please.
 
1. Proxy server is connected to MGC = (point of=20 SIP-PSTN Interconnection),
 
1a. Is there a way proxy server = registering itself=20 with MGC?  
      This is = required at=20 MGC to know which are the SIP users this proxy  can service, =
      so = that the=20 PSTN originated call can be correctly terminated to that Proxy. =
 
1b. How the MGC knows the = status of Proxy=20 (In-Service or Out-of-Service) in order to terminate the = call.
 
1c. How many Proxy servers with = different/same set=20 of SIP users can be linked to one MGC.
 
2. When SIP call arrives at MGC, MGC = requires NOA=20 (Nature of Address) in order to translate the called number and route = the call=20 to the correct PSTN Trunk Group. How the NOA address (like 0+IDDD, = IDDD,=20 DDD calls) can be represented in the SIP messages.
 
3. For SIP-PSTN call, how different = digit sequences=20 like ANI, Authorization code, Account code (these are required for = billing at=20 MGC and to have Dialing plan) can be sent to MGC in SIP Messages. =
 
 
Thanks in advance,
 
jagannathan
 
------=_NextPart_000_0031_01C1BF84.2F868D60-- _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Wed Feb 27 05:07:05 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id FAA15447 for ; Wed, 27 Feb 2002 05:07:05 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id FAA08013 for sipping-archive@odin.ietf.org; Wed, 27 Feb 2002 05:07:08 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id EAA06454; Wed, 27 Feb 2002 04:37:39 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id EAA06420 for ; Wed, 27 Feb 2002 04:37:36 -0500 (EST) Received: from t-6mnjnqs5jxuuc ([211.152.107.30]) by ietf.org (8.9.1a/8.9.1a) with SMTP id EAA14894 for ; Wed, 27 Feb 2002 04:37:31 -0500 (EST) Message-Id: <200202270937.EAA14894@ietf.org> From: "T&F" To: Mime-Version: 1.0 Content-Type: text/plain; charset="gb2312" Date: Wed, 27 Feb 2002 17:50:44 Subject: [Sipping] A professional company providing credit & status report of Chinese company Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org A professional company providing credit & status report of Chinese company. 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Your name will be removed from our database upon request. 使用极星邮件群发,无须通过邮件服务器,直达对方邮箱,速度绝对一流! 下载网址:http://love2net.51.net/,更多免费的超酷软件等你来下…… ---------------------------------------------------- INFORMATION This message has been sent using a trial-run version of the TSmtpRelayServer Delphi Component. ---------------------------------------------------- _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Wed Feb 27 07:43:02 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id HAA19556 for ; Wed, 27 Feb 2002 07:43:01 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id HAA17100 for sipping-archive@odin.ietf.org; Wed, 27 Feb 2002 07:43:04 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id HAA15598; Wed, 27 Feb 2002 07:27:26 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id HAA15561 for ; Wed, 27 Feb 2002 07:27:22 -0500 (EST) Received: from CNRI.Reston.VA.US (localhost [127.0.0.1]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id HAA18804; Wed, 27 Feb 2002 07:27:19 -0500 (EST) Message-Id: <200202271227.HAA18804@ietf.org> Mime-Version: 1.0 Content-Type: Multipart/Mixed; Boundary="NextPart" To: IETF-Announce: ; Cc: sipping@ietf.org From: Internet-Drafts@ietf.org Reply-to: Internet-Drafts@ietf.org Date: Wed, 27 Feb 2002 07:27:18 -0500 Subject: [Sipping] I-D ACTION:draft-ietf-sipping-service-examples-00.txt Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org --NextPart A New Internet-Draft is available from the on-line Internet-Drafts directories. This draft is a work item of the Session Initiation Proposal Investigation Working Group of the IETF. Title : SIP Service Examples Author(s) : A. Johnston et al. Filename : draft-ietf-sipping-service-examples-00.txt Pages : 113 Date : 26-Feb-02 This informational document gives examples of SIP (Session Initiation Protocol) services. This covers most features offered in so-called Centrex offerings from local exchange carriers and PBX (Private Branch Exchange) features. Most of the services shown in this document are implemented in the SIP User Agents, although some require the assistance of a SIP Proxy. Some require some extensions to SIP including the REFER method and Replaces header. These features are not intended to be an exhaustive set, but rather show implementations of common features likely to be implemented on SIP IP telephones in a business environment. A URL for this Internet-Draft is: http://www.ietf.org/internet-drafts/draft-ietf-sipping-service-examples-00.txt To remove yourself from the IETF Announcement list, send a message to ietf-announce-request with the word unsubscribe in the body of the message. Internet-Drafts are also available by anonymous FTP. Login with the username "anonymous" and a password of your e-mail address. After logging in, type "cd internet-drafts" and then "get draft-ietf-sipping-service-examples-00.txt". A list of Internet-Drafts directories can be found in http://www.ietf.org/shadow.html or ftp://ftp.ietf.org/ietf/1shadow-sites.txt Internet-Drafts can also be obtained by e-mail. Send a message to: mailserv@ietf.org. In the body type: "FILE /internet-drafts/draft-ietf-sipping-service-examples-00.txt". NOTE: The mail server at ietf.org can return the document in MIME-encoded form by using the "mpack" utility. To use this feature, insert the command "ENCODING mime" before the "FILE" command. To decode the response(s), you will need "munpack" or a MIME-compliant mail reader. Different MIME-compliant mail readers exhibit different behavior, especially when dealing with "multipart" MIME messages (i.e. documents which have been split up into multiple messages), so check your local documentation on how to manipulate these messages. Below is the data which will enable a MIME compliant mail reader implementation to automatically retrieve the ASCII version of the Internet-Draft. --NextPart Content-Type: Multipart/Alternative; Boundary="OtherAccess" --OtherAccess Content-Type: Message/External-body; access-type="mail-server"; server="mailserv@ietf.org" Content-Type: text/plain Content-ID: <20020226131313.I-D@ietf.org> ENCODING mime FILE /internet-drafts/draft-ietf-sipping-service-examples-00.txt --OtherAccess Content-Type: Message/External-body; name="draft-ietf-sipping-service-examples-00.txt"; site="ftp.ietf.org"; access-type="anon-ftp"; directory="internet-drafts" Content-Type: text/plain Content-ID: <20020226131313.I-D@ietf.org> --OtherAccess-- --NextPart-- _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Wed Feb 27 07:45:23 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id HAA19618 for ; Wed, 27 Feb 2002 07:45:19 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id HAA17232 for sipping-archive@odin.ietf.org; Wed, 27 Feb 2002 07:45:22 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id HAA15555; Wed, 27 Feb 2002 07:27:21 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id HAA15524 for ; Wed, 27 Feb 2002 07:27:18 -0500 (EST) Received: from CNRI.Reston.VA.US (localhost [127.0.0.1]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id HAA18784; Wed, 27 Feb 2002 07:27:14 -0500 (EST) Message-Id: <200202271227.HAA18784@ietf.org> Mime-Version: 1.0 Content-Type: Multipart/Mixed; Boundary="NextPart" To: IETF-Announce: ; Cc: sipping@ietf.org From: Internet-Drafts@ietf.org Reply-to: Internet-Drafts@ietf.org Date: Wed, 27 Feb 2002 07:27:14 -0500 Subject: [Sipping] I-D ACTION:draft-ietf-sipping-call-flows-00.txt Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org --NextPart A New Internet-Draft is available from the on-line Internet-Drafts directories. This draft is a work item of the Session Initiation Proposal Investigation Working Group of the IETF. Title : SIP Call Flow Examples Author(s) : A. Johnston et al. Filename : draft-ietf-sipping-call-flows-00.txt Pages : 229 Date : 26-Feb-02 This informational document gives examples of SIP (Session Initiation Protocol) call flows for IP telephony. Elements in these call flows include SIP User Agents and Clients, SIP Proxy and Redirect Servers, and Gateways to the PSTN (Public Switch Telephone Network). A URL for this Internet-Draft is: http://www.ietf.org/internet-drafts/draft-ietf-sipping-call-flows-00.txt To remove yourself from the IETF Announcement list, send a message to ietf-announce-request with the word unsubscribe in the body of the message. Internet-Drafts are also available by anonymous FTP. Login with the username "anonymous" and a password of your e-mail address. After logging in, type "cd internet-drafts" and then "get draft-ietf-sipping-call-flows-00.txt". A list of Internet-Drafts directories can be found in http://www.ietf.org/shadow.html or ftp://ftp.ietf.org/ietf/1shadow-sites.txt Internet-Drafts can also be obtained by e-mail. Send a message to: mailserv@ietf.org. In the body type: "FILE /internet-drafts/draft-ietf-sipping-call-flows-00.txt". NOTE: The mail server at ietf.org can return the document in MIME-encoded form by using the "mpack" utility. To use this feature, insert the command "ENCODING mime" before the "FILE" command. To decode the response(s), you will need "munpack" or a MIME-compliant mail reader. Different MIME-compliant mail readers exhibit different behavior, especially when dealing with "multipart" MIME messages (i.e. documents which have been split up into multiple messages), so check your local documentation on how to manipulate these messages. Below is the data which will enable a MIME compliant mail reader implementation to automatically retrieve the ASCII version of the Internet-Draft. --NextPart Content-Type: Multipart/Alternative; Boundary="OtherAccess" --OtherAccess Content-Type: Message/External-body; access-type="mail-server"; server="mailserv@ietf.org" Content-Type: text/plain Content-ID: <20020226131301.I-D@ietf.org> ENCODING mime FILE /internet-drafts/draft-ietf-sipping-call-flows-00.txt --OtherAccess Content-Type: Message/External-body; name="draft-ietf-sipping-call-flows-00.txt"; site="ftp.ietf.org"; access-type="anon-ftp"; directory="internet-drafts" Content-Type: text/plain Content-ID: <20020226131301.I-D@ietf.org> --OtherAccess-- --NextPart-- _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Wed Feb 27 11:19:32 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id LAA00038 for ; Wed, 27 Feb 2002 11:19:32 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id LAA29411 for sipping-archive@odin.ietf.org; Wed, 27 Feb 2002 11:19:35 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id LAA28310; Wed, 27 Feb 2002 11:02:04 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id LAA28279 for ; Wed, 27 Feb 2002 11:02:02 -0500 (EST) Received: from dgesmtp01.wcom.com (dgesmtp01.wcom.com [199.249.16.16]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id LAA29171 for ; Wed, 27 Feb 2002 11:01:58 -0500 (EST) Received: from CONVERSION-DAEMON by firewall.wcom.com (PMDF V5.2-33 #42260) id <0GS700K018I897@firewall.wcom.com> for sipping@ietf.org; Wed, 27 Feb 2002 16:01:20 +0000 (GMT) Received: from dgismtp02.wcomnet.com ([166.38.58.142]) by firewall.wcom.com (PMDF V5.2-33 #42260) with ESMTP id <0GS700J4O8I8WI@firewall.wcom.com>; Wed, 27 Feb 2002 16:01:20 +0000 (GMT) Received: from dgismtp02.wcomnet.com by dgismtp02.wcomnet.com (PMDF V5.2-33 #42263) with SMTP id <0GS700J018I58Y@dgismtp02.wcomnet.com>; Wed, 27 Feb 2002 16:01:19 +0000 (GMT) Received: from hsinnreich2 ([166.35.224.250]) by dgismtp02.wcomnet.com (PMDF V5.2-33 #42263) with ESMTP id <0GS700GLK8HQB7@dgismtp02.wcomnet.com>; Wed, 27 Feb 2002 16:01:03 +0000 (GMT) Date: Wed, 27 Feb 2002 10:01:02 -0600 From: Henry Sinnreich Subject: RE: [Sipping] New Assured Service draft In-reply-to: <193.2eb2892.29ad54d5@aol.com> To: Mpierce1@aol.com, sipping@ietf.org Message-id: <002201c1bfa7$f4d44440$fae023a6@hsinnreich2> Organization: WorldCom, Inc. MIME-version: 1.0 X-MIMEOLE: Produced By Microsoft MimeOLE V6.00.2600.0000 X-Mailer: Microsoft Outlook, Build 10.0.3416 Content-type: text/plain; charset=us-ascii Content-transfer-encoding: 7bit Importance: Normal X-Priority: 3 (Normal) X-MSMail-priority: Normal Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit While the draft provides good background on circuit-style approaches, such as PSTN, Intserv/RSVP, MPLS, one has to wonder if it makes sense to emulate telephony approaches and lay countless paths and priorities over the Internet and in private IP networks. Daily use of SIP phones for international calls over the public Internet and in enterprise networks certainly show that voice is doing just OK, or better than OK without all these mechanisms. If you are bandwidth-starved (bad decision), it is still better to get an important/emergency call across without QoS than not to communicate at all. My two cents. Henry > -----Original Message----- > From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] > On Behalf Of Mpierce1@aol.com > Sent: Tuesday, February 26, 2002 3:15 PM > To: sipping@ietf.org > Subject: [Sipping] New Assured Service draft > > > To all, > > A revision to the ID for Assured Service definition in SIP > has been submitted > which provides changes based on those comments received. This > ID defines the > requirements for Assured Service in SIP to provide > prioritized handling for > certain sessions established using SIP. It also discusses > some of the factors > to be taken into account when planning protocol revisions > which may be > required. > > The draft can be found at: > http://ietf.org/internet-drafts/draft-pierce-sipping-assured-s ervice-01.txt Mike Pierce _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Wed Feb 27 12:27:15 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id MAA04617 for ; Wed, 27 Feb 2002 12:27:15 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id MAA04328 for sipping-archive@odin.ietf.org; Wed, 27 Feb 2002 12:27:19 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id MAA03344; Wed, 27 Feb 2002 12:05:09 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id MAA03307 for ; Wed, 27 Feb 2002 12:05:04 -0500 (EST) Received: from sj-msg-core-3.cisco.com (sj-msg-core-3.cisco.com [171.70.157.152]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id MAA03406 for ; Wed, 27 Feb 2002 12:04:59 -0500 (EST) Received: from nisser.cisco.com (nisser.cisco.com [171.71.176.85]) by sj-msg-core-3.cisco.com (8.11.3/8.9.1) with ESMTP id g1RH4Vw16509; Wed, 27 Feb 2002 09:04:31 -0800 (PST) Received: from cisco.com (ssh-sj1.cisco.com [171.68.225.134]) by nisser.cisco.com (8.8.6 (PHNE_14041)/CISCO.SERVER.1.2) with ESMTP id JAA27903; Wed, 27 Feb 2002 09:04:31 -0800 (PST) Message-ID: <3C7D119F.361B2CDB@cisco.com> Date: Wed, 27 Feb 2002 12:04:31 -0500 From: Flemming Andreasen X-Mailer: Mozilla 4.79 [en]C-CCK-MCD (Windows NT 5.0; U) X-Accept-Language: en MIME-Version: 1.0 To: Mark Watson CC: sipping@ietf.org Subject: Re: [Sipping] Privacy comment (was Re: calling number blocking atsip/isup gate way) References: Content-Type: text/plain; charset=us-ascii Content-Transfer-Encoding: 7bit Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit Just to be clear, since untrusted UAs inserting Remote-Party-Ids is not the norm in the privacy draft, and downstream proxies may be adding Remote-Party-Ids, I'm not planning on removing the RPID-Privacy header. -- Flemming Mark Watson wrote: > > > A corollory of this is that we don't really need RPID-Privacy any > more. This was introduced in privacy-03 on the basis that the ability > for UAs to include RPID was removed, but UAs still needed to indicate > their privacy requirements in respect of RPIDs to be added by proxies. > > IF UA-inserted RPID is re-introduced, we can take RPID-Privacy back > out. > > UAs wishing to indicate privacy requirements for RPIDs to be inserted > by Proxies, without the UA itself indicating an identity, can include > an RPID with the privacy & type values set appropriatly, but no actual > identity (as was the original intention). > > Regards...Mark > > > -----Original Message----- > > From: Flemming Andreasen [mailto:fandreas@cisco.com] > > Sent: 25 February 2002 19:56 > > To: Watson, Mark [MDN05:EP10:EXCH] > > Cc: sipping@ietf.org > > Subject: Re: [Sipping] Privacy comment (was Re: calling > > number blocking > > atsip/isup gate way) > > > > > > > > > > Mark Watson wrote: > > > > > > > > > > > So, in the absence of further comments, could the editor add this > to > > > the next version of the privacy draft ? (on the > silence=agree/don't > > > care principle) > > > > Will do. > > > > -- Flemming > > > > > > > > > > > > > ...Mark > > > > > > > -----Original Message----- > > > > From: Dean Willis [mailto:dean.willis@softarmor.com] > > > > Sent: 20 February 2002 00:27 > > > > To: Watson, Mark [MDN05:EP10:EXCH]; sipping@ietf.org > > > > Subject: Re: [Sipping] Privacy comment (was Re: calling > > > > number blocking > > > > at sip/isup gate way) > > > > > > > > > > > > I concur that your proposal would add to the utility of RPID. > > > > > > > > -- > > > > Dean > > > > > > > > ----- Original Message ----- > > > > From: "Mark Watson" > > > > To: > > > > Sent: Tuesday, February 19, 2002 2:28 PM > > > > Subject: [Sipping] Privacy comment (was Re: calling > > number blocking > > > at > > > > sip/isup gate way) > > > > > > > > > > > > > As presently described in privacy-03, the Remote-Party-ID > > > > header is only > > > > > ever inserted by proxies, based on the proxy's knowledge of > > > > the identity > > > > of > > > > > the UA (which it determines from some authentication > > > > mechanism out of > > > > scope > > > > > of the privacy draft). > > > > > > > > > > Privacy-02 allowed the UA to insert Remote-Party-ID too. I > > > > think this > > > > should > > > > > be reintroduced for the *specific case* of a UA with > > > > multiple identities, > > > > to > > > > > allow the UA to choose which identity it wishes to be known > > > > by. The Proxy > > > > > would still be responsible for ensuring that this identity > > > > was valid for > > > > the > > > > > user and would still use some other authentication scheme > > > > to authenticate > > > > > the user. The proxy would need knowledge of which > > > > identities were valid > > > > for > > > > > which users. > > > > > > > > > > I believe this is a requirement for 3GPP, where a UA can > > > > have multiple > > > > > 'public identities' and chooses which of these it wishes to > > > > be known by > > > > for > > > > > each call. The proxy authenticates the UA by some other > > > > means and then > > > > > checks the subscription data to ensure it is really allowed > > > > to use the > > > > > public Identity it has included. > > > > > > > > > > I've suggested this a few times, and there has not been > > > > much comment for > > > > or > > > > > against. It would be good if we could get a resolution > > > > before the next > > > > > version of privacy is published. > > > > > > > > > > So, comments ? > > > > > > > > > > Regards...Mark > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > -----Original Message----- > > > > > > From: Dean Willis [mailto:dean.willis@softarmor.com] > > > > > > Sent: 18 February 2002 23:12 > > > > > > To: Watson, Mark [MDN05:EP10:EXCH]; Mpierce1@aol.com; > > > > > > jon.peterson@neustar.biz; sipping@ietf.org > > > > > > Subject: Re: [Sipping] RE: [Sip] calling number blocking > > > > at sip/isup > > > > > > gateway > > > > > > > > > > > > > > > > > > Mark said: > > > > > > > > > > > > > Your example below needs only a small extension to > > > > > > illustrate Mike's point > > > > > > > and show why something like Remote-Party-ID as defined in > > > > > > the privacy > > > > > > draft > > > > > > > is needed, and how it works despite the lack of a > signature. > > > > > > > > > > > > Yep. And what happens when my misconfigured proxy forwards > > > > > > the call without > > > > > > paying attention to the Privacy header, because for some > > > > reason it's > > > > > > forwarding to the outside world instead of a PSTN gateway > > > > > > under my control? > > > > > > We get a privacy leak. > > > > > > > > > > > > But other than that, I'd say you've precisely described a > > > > > > "sunny day" use > > > > > > case for RPID. > > > > > > > > > > > > It still has a lot of security issues based on transitive > > > > > > trust, which make > > > > > > it a lot more palatable in a two-party case than in three or > > > > > > > more party > > > > > > transitivity. > > > > > > > > > > > > I believe we're approaching a general consensus that the > > > > Privacy draft > > > > > > provides a reasonable approach to solving a specific > > > > class of problem > > > > > > wherein interoperator-transitive trust is couple to > > > > > > transport-layer message > > > > > > protection. That's not a general privacy mechanism, but it > > > > > > does seem to have > > > > > > enough constituency to justify publication as long as it > > > > has a REALLY > > > > > > serious applicability statement explaining the particular > > > > > > requirements and > > > > > > environments for which it is useful. > > > > > > > > > > > > How does that sound? > > > > > > > > > > > > -- > > > > > > Dean > > > > > > > > > > > > > > > > > > > > > > > > > > > > > -- > > Flemming Andreasen > > Cisco Systems > > > > > > -- Flemming Andreasen Cisco Systems _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@ns.ietf.org Wed Feb 27 22:10:16 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id WAA00493 for ; Wed, 27 Feb 2002 22:10:16 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id WAA19339 for sipping-archive@odin.ietf.org; Wed, 27 Feb 2002 22:10:20 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id VAA18204; Wed, 27 Feb 2002 21:44:19 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id VAA18176 for ; Wed, 27 Feb 2002 21:44:17 -0500 (EST) Received: from sj-msg-core-3.cisco.com (sj-msg-core-3.cisco.com [171.70.157.152]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id VAA00113 for ; Wed, 27 Feb 2002 21:44:12 -0500 (EST) Received: from nisser.cisco.com (nisser.cisco.com [171.71.176.85]) by sj-msg-core-3.cisco.com (8.11.3/8.9.1) with ESMTP id g1S2hhw09903 for ; Wed, 27 Feb 2002 18:43:43 -0800 (PST) Received: from cisco.com (ssh-sj1.cisco.com [171.68.225.134]) by nisser.cisco.com (8.8.6 (PHNE_14041)/CISCO.SERVER.1.2) with ESMTP id SAA22665 for ; Wed, 27 Feb 2002 18:43:43 -0800 (PST) Message-ID: <3C7D995E.842F3768@cisco.com> Date: Wed, 27 Feb 2002 21:43:42 -0500 From: Flemming Andreasen X-Mailer: Mozilla 4.79 [en]C-CCK-MCD (Windows NT 5.0; U) X-Accept-Language: en MIME-Version: 1.0 To: sipping@ietf.org Content-Type: text/plain; charset=us-ascii Content-Transfer-Encoding: 7bit Content-Transfer-Encoding: 7bit Subject: [Sipping] privacy-04 discussion on sip list Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit Most of the recent discussion on the privacy draft has taken place on the SIPPING list, however privacy is a SIP working group item and hence should be discussed on the SIP list. FYI, I just posted a privacy-04 related mail to the SIP list. Please continue the privacy discussions on the SIP list. Thanks Flemming -- Flemming Andreasen Cisco Systems _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@ns.ietf.org Thu Feb 28 04:22:17 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id EAA15273 for ; Thu, 28 Feb 2002 04:22:17 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id EAA15735 for sipping-archive@odin.ietf.org; Thu, 28 Feb 2002 04:22:20 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id DAA14188; Thu, 28 Feb 2002 03:52:30 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id DAA14159 for ; Thu, 28 Feb 2002 03:52:27 -0500 (EST) Received: from gorilla.mchh.siemens.de (gorilla.mchh.siemens.de [194.138.158.18]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id DAA14742 for ; Thu, 28 Feb 2002 03:52:23 -0500 (EST) Received: from moody.mchh.siemens.de (mail2.mchh.siemens.de [194.138.158.226]) by gorilla.mchh.siemens.de (8.9.3/8.9.3) with ESMTP id JAA22832; Thu, 28 Feb 2002 09:52:20 +0100 (MET) Received: from mchh246e.demchh201e.icn.siemens.de ([139.21.200.56]) by moody.mchh.siemens.de (8.9.1/8.9.1) with ESMTP id JAA13517; Thu, 28 Feb 2002 09:52:11 +0100 (MET) Received: by MCHH246E with Internet Mail Service (5.5.2653.19) id ; Thu, 28 Feb 2002 09:52:22 +0100 Message-ID: From: Lueken Joachim To: "'Jean-Francois Mule'" Cc: sipping@ietf.org, "'Bill Sulzen'" Subject: AW: [Sipping] RE: Concerning draft-mule-sip-t38callflows-02 T.38 rejection - 60 6 and/or 488 Date: Thu, 28 Feb 2002 09:52:19 +0100 MIME-Version: 1.0 X-Mailer: Internet Mail Service (5.5.2653.19) Content-Type: text/plain; charset="iso-8859-1" Content-Transfer-Encoding: 8bit X-MIME-Autoconverted: from quoted-printable to 8bit by optimus.ietf.org id DAA14160 Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 8bit Hi Jean-Francois, in the fax pass-thru mode (re-INVITE after 606) new attributes a=ecan: fb on a=silenceSupp: off are used. Are these attributes registered with IANA? a:silenceSupp: off is also needed when the V.25 ANS modem tone (similar to CED) is detected. Therefore in my opinion the gateway should switch to a Modem Transparency mode (this is a new RTP payload with G.711, silence suppression=off, max length for jitter buffer) when either CED or V.25 ANS modem tone is detected. Signalling of 'a=ecan: fb on' is not necessary since this is controlled by CED w/ or w/o phase reversal. Mit freundlichen Gruessen / kind regards Joachim Lueken ---------------------------------------- SIEMENS Dept. ICN WN CC NA A17 Hofmannstr. 51, 81359 Munich, Germany Tel: +49 89 722-63251 Fax: +49 89 722-28770 eMail: joachim.lueken@icn.siemens.de ---------------------------------------- -----Urspr黱gliche Nachricht----- Von: Jean-Francois Mule [mailto:jf.mule@cablelabs.com] Gesendet: Donnerstag, 21. Februar 2002 22:04 An: 'Bill Sulzen'; sipping@ietf.org Betreff: [Sipping] RE: Concerning draft-mule-sip-t38callflows-02 T.38 rejection - 60 6 and/or 488 Bill Sulzen wrote: > Section 6.2 of draft-mule-sip-t38callflows-02 describes how > a UA would > reject a T.38 INVITE. It indicates that the emitting gateway > rejects the re-INVITE with a "606 Not Acceptable" response. > Would a "488 Not Acceptable Media" not be at least equally, > if not more appropriate? after reading the bis-08 sections on 488/606, I see no objections for treating 488 and 606 as *equally* appropriate. The update will go into the upcoming draft-ietf-sipping-realtimefax-00. any other comments? Jean-Francois. > -----Original Message----- > From: Jean-Francois Mule > Sent: Thursday, February 21, 2002 1:56 PM > To: 'Bill Sulzen'; 'sip-implementors@cs.columbia.edu' > Subject: RE: Concerning draft-mule-sip-t38callflows-02 T.38 > rejection - > 606 and/or 488 > > > Bill & all: > > since draft-mule-sip-t38callflows-02 will morph into > draft-ietf-sipping-realtimefax-00, this discussion is now > happening on sipping. sorry for the delay in responsing to > your comment. > > Jean-Francois > > > >From: "Bill Sulzen" > > >To: > > >Cc: "Jean-Francois Mule" > > >Subject: Concerning draft-mule-sip-t38callflows-02 T.38 > > rejection - 606 or > > >488? > > >Date: Fri, 25 Jan 2002 15:01:55 -0500 > > > > > >Section 6.2 of draft-mule-sip-t38callflows-02 describes how > > a UA would > > >reject a T.38 INVITE. It indicates that the emitting gateway > > rejects the > > >re-INVITE with a "606 Not Acceptable" response. > > > > > >Would a "488 Not Acceptable Media" not be at least equally, > > if not more > > >appropriate? > > > > > >Thanks, > > >Bill Sulzen > _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@ns.ietf.org Thu Feb 28 05:22:59 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id FAA16186 for ; Thu, 28 Feb 2002 05:22:58 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id FAA18938 for sipping-archive@odin.ietf.org; Thu, 28 Feb 2002 05:23:02 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id FAA18385; Thu, 28 Feb 2002 05:07:42 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id FAA18356 for ; Thu, 28 Feb 2002 05:07:39 -0500 (EST) Received: from mgw-x2.nokia.com (mgw-x2.nokia.com [131.228.20.22]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id FAA15954 for ; Thu, 28 Feb 2002 05:07:35 -0500 (EST) Received: from esvir03nok.nokia.com (esvir03nokt.ntc.nokia.com [172.21.143.35]) by mgw-x2.nokia.com (Switch-2.1.0/Switch-2.1.0) with ESMTP id g1SA7kZ02845 for ; Thu, 28 Feb 2002 12:07:46 +0200 (EET) Received: from esebh003.NOE.Nokia.com (unverified) by esvir03nok.nokia.com (Content Technologies SMTPRS 4.2.5) with ESMTP id ; Thu, 28 Feb 2002 12:07:37 +0200 Received: from nokia.com ([172.21.149.100]) by esebh003.NOE.Nokia.com with Microsoft SMTPSVC(5.0.2195.3779); Thu, 28 Feb 2002 12:07:37 +0200 Message-ID: <3C7E0F6C.2040900@nokia.com> Date: Thu, 28 Feb 2002 13:07:24 +0200 From: Aki Petteri Niemi User-Agent: Mozilla/5.0 (X11; U; Linux i686; en-US; rv:0.9.8) Gecko/20020212 X-Accept-Language: en-us MIME-Version: 1.0 To: sipping@ietf.org CC: "Vesa Torvinen (LMF)" , "Jari Arkko (LMF)" References: <4.1.20020226194230.02bdceb8@localhost> Content-Type: text/plain; charset=us-ascii; format=flowed Content-Transfer-Encoding: 7bit X-OriginalArrivalTime: 28 Feb 2002 10:07:37.0467 (UTC) FILETIME=[BFB1FCB0:01C1C03F] Content-Transfer-Encoding: 7bit Subject: [Sipping] New I-D on Digest AKA authenticaton Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit Hi All, We have written an extension to Digest authentication enabling its use in 3GPP IMS. The draft defines a mechanism, in which the UMTS Authentication and Key Agreement (AKA) protocol is used for generating Digest passwords. Overall, Digest operation has been preserved, with some added semantics and parameters. The I-D can be fetched from: http://www.ietf.org/internet-drafts/draft-niemi-sipping-digest-aka-00.txt Requirements for the work are discussed in: http://www.ietf.org/internet-drafts/draft-uusitalo-sipping-authentication-00.txt http://www.ietf.org/internet-drafts/draft-garcia-sipping-3gpp-reqs-02.txt The aim with this document is to have a SIP WG work item for it, and progress the document to standards track RFC. Note, that this work is in some sense continuing the SIP EAP work, but with a more well defined scope based on the feedback from the Transport AD's and various other experts in the field. Comments are appreciated. Thanks, Aki --- Abstract The Hypertext Transfer Protocol (HTTP) Authentication Framework includes two authentication schemes: Basic and Digest. Both schemes employ a shared secret based mechanism for access authentication. The Authentication and Key Agreement (AKA) mechanism performs user authentication and session key distribution in Universal Mobile Telecommunications System (UMTS) networks. AKA is a challenge- response based mechanism that uses symmetric cryptography. This memo specifies an AKA based password generation mechanism for HTTP Digest access authentication. _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Thu Feb 28 10:17:15 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id KAA27663 for ; Thu, 28 Feb 2002 10:17:15 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id KAA05728 for sipping-archive@odin.ietf.org; Thu, 28 Feb 2002 10:17:18 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id JAA02700; Thu, 28 Feb 2002 09:32:30 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id JAA02671 for ; Thu, 28 Feb 2002 09:32:28 -0500 (EST) Received: from pmesmtp01.wcom.com (pmesmtp01.wcom.com [199.249.20.1]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id JAA24890 for ; Thu, 28 Feb 2002 09:32:25 -0500 (EST) Received: from CONVERSION-DAEMON by firewall.wcom.com (PMDF V5.2-33 #47837) id <0GS800D01Z14GF@firewall.wcom.com> for sipping@ietf.org; Thu, 28 Feb 2002 14:31:52 +0000 (GMT) Received: from pmismtp04.wcomnet.com ([166.38.62.39]) by firewall.wcom.com (PMDF V5.2-33 #47837) with ESMTP id <0GS800BO5Z136U@firewall.wcom.com> for sipping@ietf.org; Thu, 28 Feb 2002 14:31:52 +0000 (GMT) Received: from pmismtp04.wcomnet.com by pmismtp04.wcomnet.com (PMDF V5.2-33 #42258) with SMTP id <0GS800E01Z1388@pmismtp04.wcomnet.com> for sipping@ietf.org; Thu, 28 Feb 2002 14:31:51 +0000 (GMT) Received: from ajohnston ([166.42.33.131]) by pmismtp04.wcomnet.com (PMDF V5.2-33 #42258) with ESMTP id <0GS800GM6Z0WFD@pmismtp04.wcomnet.com> for sipping@ietf.org; Thu, 28 Feb 2002 14:31:45 +0000 (GMT) Date: Thu, 28 Feb 2002 08:31:20 -0600 From: Alan Johnston Subject: RE: [Sipping] I-D ACTION:draft-ietf-sipping-call-flows-00.txt In-reply-to: <200202271227.HAA18784@ietf.org> To: sipping@ietf.org Message-id: <000401c1c064$97f36d60$83212aa6@ajohnston> MIME-version: 1.0 X-MIMEOLE: Produced By Microsoft MimeOLE V6.00.2600.0000 X-Mailer: Microsoft Outlook, Build 10.0.3416 Content-type: text/plain; charset=us-ascii Content-transfer-encoding: 7bit Importance: Normal X-Priority: 3 (Normal) X-MSMail-priority: Normal Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit All, This draft updates and replaces draft-ietf-sip-call-flows-05.txt. The major changes are as follows: - Changed references from RFC2543 to RFC2543bis and offer/answer. - Added mandatory branch tag for all Via headers. - Added From tags. - Changed Record-Route and Route behavior to show bis loose routing. - Proxies now keep To tag the same when proxying non-2xx responses. - Added Max-Forwards header. - Changes to Digest: removed domain in challenges, fixed uri in responses, and changed realm to be a domain name. - Renamed Firewall Proxy to Application Layer Gateway (ALG) - Fixed SDP in 183 responses. - Made other changes to bring SIP/ISUP interworking in line with SIP/ISUP document. - Added Error-Info example The big To Do left is to make bis changes to the Test Messages in Section 7. Besides the test messages, the call flows should be in line with bis - send me any corrections. Thanks, Alan. sipping@ietf.org > -----Original Message----- > From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] > On Behalf Of Internet-Drafts@ietf.org > Sent: Wednesday, February 27, 2002 6:27 AM > To: IETF-Announce: > Cc: sipping@ietf.org > Subject: [Sipping] I-D ACTION:draft-ietf-sipping-call-flows-00.txt > > > A New Internet-Draft is available from the on-line > Internet-Drafts directories. This draft is a work item of the > Session Initiation Proposal Investigation Working Group of the IETF. > > Title : SIP Call Flow Examples > Author(s) : A. Johnston et al. > Filename : draft-ietf-sipping-call-flows-00.txt > Pages : 229 > Date : 26-Feb-02 > > This informational document gives examples of SIP (Session Initiation > Protocol) call flows for IP telephony. Elements in these call flows > include SIP User Agents and Clients, SIP Proxy and Redirect Servers, > and Gateways to the PSTN (Public Switch Telephone Network). > > A URL for this Internet-Draft is: http://www.ietf.org/internet-drafts/draft-ietf-sipping-call-flows-00.txt To remove yourself from the IETF Announcement list, send a message to ietf-announce-request with the word unsubscribe in the body of the message. Internet-Drafts are also available by anonymous FTP. Login with the username "anonymous" and a password of your e-mail address. After logging in, type "cd internet-drafts" and then "get draft-ietf-sipping-call-flows-00.txt". A list of Internet-Drafts directories can be found in http://www.ietf.org/shadow.html or ftp://ftp.ietf.org/ietf/1shadow-sites.txt Internet-Drafts can also be obtained by e-mail. Send a message to: mailserv@ietf.org. In the body type: "FILE /internet-drafts/draft-ietf-sipping-call-flows-00.txt". NOTE: The mail server at ietf.org can return the document in MIME-encoded form by using the "mpack" utility. To use this feature, insert the command "ENCODING mime" before the "FILE" command. To decode the response(s), you will need "munpack" or a MIME-compliant mail reader. Different MIME-compliant mail readers exhibit different behavior, especially when dealing with "multipart" MIME messages (i.e. documents which have been split up into multiple messages), so check your local documentation on how to manipulate these messages. Below is the data which will enable a MIME compliant mail reader implementation to automatically retrieve the ASCII version of the Internet-Draft. _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Thu Feb 28 10:19:27 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id KAA27841 for ; Thu, 28 Feb 2002 10:19:26 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id KAA05871 for sipping-archive@odin.ietf.org; Thu, 28 Feb 2002 10:19:29 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id JAA03348; Thu, 28 Feb 2002 09:42:20 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id JAA03303 for ; Thu, 28 Feb 2002 09:42:14 -0500 (EST) Received: from dgesmtp02.wcom.com (dgesmtp02.wcom.com [199.249.16.17]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id JAA25452 for ; Thu, 28 Feb 2002 09:42:11 -0500 (EST) Received: from CONVERSION-DAEMON by firewall.wcom.com (PMDF V5.2-33 #42261) id <0GS800F01ZGPGS@firewall.wcom.com> for sipping@ietf.org; Thu, 28 Feb 2002 14:41:13 +0000 (GMT) Received: from pmismtp03.wcomnet.com ([166.38.62.38]) by firewall.wcom.com (PMDF V5.2-33 #42261) with ESMTP id <0GS800F1YZGPA1@firewall.wcom.com> for sipping@ietf.org; Thu, 28 Feb 2002 14:41:13 +0000 (GMT) Received: from pmismtp03.wcomnet.com by pmismtp03.wcomnet.com (PMDF V5.2-33 #42258) with SMTP id <0GS800101ZG7WM@pmismtp03.wcomnet.com> for sipping@ietf.org; Thu, 28 Feb 2002 14:41:12 +0000 (GMT) Received: from ajohnston ([166.42.33.131]) by pmismtp03.wcomnet.com (PMDF V5.2-33 #42258) with ESMTP id <0GS800LNNZFS99@pmismtp03.wcomnet.com> for sipping@ietf.org; Thu, 28 Feb 2002 14:40:41 +0000 (GMT) Date: Thu, 28 Feb 2002 08:40:14 -0600 From: Alan Johnston Subject: RE: [Sipping] I-D ACTION:draft-ietf-sipping-service-examples-00.txt In-reply-to: <200202271227.HAA18804@ietf.org> To: sipping@ietf.org Message-id: <000501c1c065$d78bae50$83212aa6@ajohnston> MIME-version: 1.0 X-MIMEOLE: Produced By Microsoft MimeOLE V6.00.2600.0000 X-Mailer: Microsoft Outlook, Build 10.0.3416 Content-type: text/plain; charset=us-ascii Content-transfer-encoding: 7bit Importance: Normal X-Priority: 3 (Normal) X-MSMail-priority: Normal Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit All, This document updates and replaces draft-ietf-sip-service-examples-03.txt The major changes are as follows: - Added Max-Forwards - Added branch parm and received param to Vias - Changed Record-Route and Route behavior to show loose routing. - Added Record-Route and Contact to 18x responses - Changed Call Pickup to show Replaces with an early dialog - Changed description of Single Line Extension service - Changed Call Park and Pickup flow to show a different user picking up the call. - Added Allow and Supported headers to REFER and replaces flows. These changes should bring the document in line with the latest bis draft. The biggest hole in the document is a method for a UA to discover dialog information so that it can pickup a parked call, or pick a ringing phone. While this can be done outside of SIP, SUBSCRIBE/NOTIFY combined with the Call Event Package (draft-rosenberg-sip-call-package-00.txt) is a good method that will be incorporated in future drafts. As always, comments and corrections are greatly appreciated. Thanks, Alan Johnston WorldCom sip:alan@siptest.wcom.com > -----Original Message----- > From: sipping-admin@ietf.org [mailto:sipping-admin@ietf.org] > On Behalf Of Internet-Drafts@ietf.org > Sent: Wednesday, February 27, 2002 6:27 AM > To: IETF-Announce: > Cc: sipping@ietf.org > Subject: [Sipping] I-D > ACTION:draft-ietf-sipping-service-examples-00.txt > > > A New Internet-Draft is available from the on-line > Internet-Drafts directories. This draft is a work item of the > Session Initiation Proposal Investigation Working Group of the IETF. > > Title : SIP Service Examples > Author(s) : A. Johnston et al. > Filename : draft-ietf-sipping-service-examples-00.txt > Pages : 113 > Date : 26-Feb-02 > > This informational document gives examples of SIP (Session Initiation > Protocol) services. This covers most features offered in so-called > Centrex offerings from local exchange carriers and PBX (Private > Branch Exchange) features. Most of the services shown in this > document are implemented in the SIP User Agents, although some > require the assistance of a SIP Proxy. Some require some extensions > to SIP including the REFER method and Replaces header. These features > are not intended to be an exhaustive set, but rather show > implementations of common features likely to be implemented on SIP IP > telephones in a business environment. > > A URL for this Internet-Draft is: http://www.ietf.org/internet-drafts/draft-ietf-sipping-service-examples- 00.txt To remove yourself from the IETF Announcement list, send a message to ietf-announce-request with the word unsubscribe in the body of the message. Internet-Drafts are also available by anonymous FTP. Login with the username "anonymous" and a password of your e-mail address. After logging in, type "cd internet-drafts" and then "get draft-ietf-sipping-service-examples-00.txt". A list of Internet-Drafts directories can be found in http://www.ietf.org/shadow.html or ftp://ftp.ietf.org/ietf/1shadow-sites.txt Internet-Drafts can also be obtained by e-mail. Send a message to: mailserv@ietf.org. In the body type: "FILE /internet-drafts/draft-ietf-sipping-service-examples-00.txt". NOTE: The mail server at ietf.org can return the document in MIME-encoded form by using the "mpack" utility. To use this feature, insert the command "ENCODING mime" before the "FILE" command. To decode the response(s), you will need "munpack" or a MIME-compliant mail reader. Different MIME-compliant mail readers exhibit different behavior, especially when dealing with "multipart" MIME messages (i.e. documents which have been split up into multiple messages), so check your local documentation on how to manipulate these messages. Below is the data which will enable a MIME compliant mail reader implementation to automatically retrieve the ASCII version of the Internet-Draft. _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Thu Feb 28 12:26:14 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id MAA06397 for ; Thu, 28 Feb 2002 12:26:14 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id MAA16555 for sipping-archive@odin.ietf.org; Thu, 28 Feb 2002 12:26:18 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id MAA14924; Thu, 28 Feb 2002 12:07:40 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id MAA14897 for ; Thu, 28 Feb 2002 12:07:37 -0500 (EST) Received: from imo-r10.mx.aol.com (imo-r10.mx.aol.com [152.163.225.106]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id MAA04848 for ; Thu, 28 Feb 2002 12:07:33 -0500 (EST) From: Mpierce1@aol.com Received: from Mpierce1@aol.com by imo-r10.mx.aol.com (mail_out_v32.5.) id l.60.1bb8016f (25511); Thu, 28 Feb 2002 12:06:54 -0500 (EST) Message-ID: <60.1bb8016f.29afbdae@aol.com> Date: Thu, 28 Feb 2002 12:06:54 EST To: sipping@ietf.org CC: schulzrinne@cs.columbia.edu, jmpolk@cisco.com MIME-Version: 1.0 Content-Type: text/plain; charset="US-ASCII" Content-Transfer-Encoding: 7bit X-Mailer: AOL 5.0 for Windows sub 138 Content-Transfer-Encoding: 7bit Subject: [Sipping] Draft resource priority Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit Comments on draft resource priority: 4 IANA There should not be a total restriction against adding new priorities within a namespace between others. The DSN can't be prevented from ever expanding its set of priorities. It already states (in Annex A, but should be moved) that unrecognized values are treated as Routine (the lowest). Granted, the network would have to ensure upgrade of all equipment before making full use of a new value, but it should be possible to do this. At most, the text should recommend (with "SHOULD") that new values not be added, as in the previous draft. Or a statement like: New priority values MAY be added to a namespace, however, it SHOULD be understood that any new value would continue to be treated the same as the lowest assigned value by equipment which has not been upgraded to take into account the newly registered value. Annex A "critic-ecp" should be added to the list in the first paragraph. The statement about what is highest and lowest can be deleted, since it is stated in the third paragraph. Since this is a "well-known" set of priorities, and very short, it is useful to include its definition here. The statement in third paragraph about the treatment of additional values should be moved to the main body of the draft to apply to all cases, not just the DSN namespace. It should refer to "priority value", not "extension parameter", since no place else in the draft refers to it as "extension parameter" (unless that should clearly be understood to be the part after the "dot"). The default (when the value is unrecognized) should be the lowest value assigned (in the general case) rather than "routine" (only for DSN). Mike _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Thu Feb 28 12:26:15 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id MAA06413 for ; Thu, 28 Feb 2002 12:26:15 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id MAA16569 for sipping-archive@odin.ietf.org; Thu, 28 Feb 2002 12:26:19 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id MAA14968; Thu, 28 Feb 2002 12:07:49 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id MAA14936 for ; Thu, 28 Feb 2002 12:07:47 -0500 (EST) Received: from imo-r08.mx.aol.com (imo-r08.mx.aol.com [152.163.225.104]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id MAA04878 for ; Thu, 28 Feb 2002 12:07:42 -0500 (EST) From: Mpierce1@aol.com Received: from Mpierce1@aol.com by imo-r08.mx.aol.com (mail_out_v32.5.) id d.109.e32b474 (25511); Thu, 28 Feb 2002 12:06:51 -0500 (EST) Message-ID: <109.e32b474.29afbdaa@aol.com> Date: Thu, 28 Feb 2002 12:06:50 EST Subject: Re: [Sipping] New Assured Service draft To: Henry.Sinnreich@wcom.com, sipping@ietf.org MIME-Version: 1.0 Content-Type: text/plain; charset="US-ASCII" Content-Transfer-Encoding: 7bit X-Mailer: AOL 5.0 for Windows sub 138 Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit In a message dated 2/27/02 11:01:47 AM Eastern Standard Time, Henry.Sinnreich@wcom.com writes: > While the draft provides good background on circuit-style approaches, > such as PSTN, Intserv/RSVP, MPLS, one has to wonder if it makes sense to > emulate telephony approaches and lay countless paths and priorities over > the Internet and in private IP networks. Daily use of SIP phones for > international calls over the public Internet and in enterprise networks > certainly show that voice is doing just OK, or better than OK without > all these mechanisms. > > If you are bandwidth-starved (bad decision), it is still better to get > an important/emergency call across without QoS than not to communicate > at all. > > My two cents. > > Henry Henry, Your're right that we need to get away from the circuit-mode approaches, which is why I don't think call (connection) preemption is the solution. The problem is that I am doing this work to support military networks which always seem to be bandwidth starved, especially in case of damage to large parts of the infrastructure. We know that there is no problem demonstrated with today's IP network and the level of voice calls using it (especially since the vast majority of the bandwidth is used for data traffic that can be "pushed aside" to let the voice get through). But we're addressing the possibility of the completely non-normal condition, including low-bandwidth applications being used mostly for voice traffic, as well as non-voice traffic that also needs such Assured Service considerations. Yes, it is important to get the emergency call through, even if it has really bad QoS (lost packets, etc.), so high priority doesn't require high quality. But there needs to be mechanisms to give the high priority calls better quality than the lower priority calls are getting at the same time. Thanks for your comments. Mike _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Thu Feb 28 13:57:36 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id NAA12767 for ; Thu, 28 Feb 2002 13:57:36 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id NAA21827 for sipping-archive@odin.ietf.org; Thu, 28 Feb 2002 13:57:39 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id NAA21198; Thu, 28 Feb 2002 13:44:45 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id NAA21153 for ; Thu, 28 Feb 2002 13:44:42 -0500 (EST) Received: from revere.sonusnet.com (mail.sonusnet.com [208.45.178.33]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id NAA11790 for ; Thu, 28 Feb 2002 13:44:38 -0500 (EST) Received: from sonusdc3.sonusnet.com (sonusdc3 [10.128.32.53]) by revere.sonusnet.com (Switch-2.1.0/Switch-2.1.0) with ESMTP id g1SIhVI18635; Thu, 28 Feb 2002 13:43:31 -0500 (EST) Received: by sonusdc3.sonusnet.com with Internet Mail Service (5.5.2653.19) id <1Z0KBV7Z>; Thu, 28 Feb 2002 13:43:39 -0500 Message-ID: <4CF48720B6E9D4118D040060CF2074E904A143FF@sonusdc3.sonusnet.com> From: "Farahmand, Fardad" To: "'Henning Schulzrinne'" , sipping@ietf.org Subject: [Sipping] sip-911, Preventing dialog termination Date: Thu, 28 Feb 2002 13:43:37 -0500 MIME-Version: 1.0 X-Mailer: Internet Mail Service (5.5.2653.19) Content-Type: text/plain; charset="iso-8859-1" Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org In draft-schulzrinne-sip-911-01 there is a brief mention of some existing E911 features for disabling call-waiting, three-party-calls and not allowing the caller to hang up the call during an E911 call. But the document has no recommendation of how to do this with SIP. In particular, consider the scenario where someone calls an emergency number and then hangs up. In PSTN E911 case, the operator is notified that the caller has hung up but the call is not released. The operator has the option of (1) initiate a request by which the caller's device starts ringing or (2) release the call. Is there any plan to cover recommendations for this feature in this (or another) draft? Thanks, Fardad Farahmand _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Thu Feb 28 16:13:07 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id QAA19959 for ; Thu, 28 Feb 2002 16:13:07 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id QAA00909 for sipping-archive@odin.ietf.org; Thu, 28 Feb 2002 16:13:12 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id PAA29236; Thu, 28 Feb 2002 15:48:10 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id PAA29209 for ; Thu, 28 Feb 2002 15:48:08 -0500 (EST) Received: from cs.columbia.edu (cs.columbia.edu [128.59.16.20]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id PAA18942 for ; Thu, 28 Feb 2002 15:48:03 -0500 (EST) Received: from orion.cs.columbia.edu (orion.cs.columbia.edu [128.59.16.21]) by cs.columbia.edu (8.9.3/8.9.3) with ESMTP id PAA05206 for ; Thu, 28 Feb 2002 15:48:06 -0500 (EST) Received: from ind.cs.columbia.edu (ind.cs.columbia.edu [128.59.19.27]) by orion.cs.columbia.edu (8.12.1/8.12.1) with ESMTP id g1SKm4Yr006474 for ; Thu, 28 Feb 2002 15:48:04 -0500 (EST) Date: Thu, 28 Feb 2002 15:48:04 -0500 (EST) From: Xiaotao Wu To: Message-ID: MIME-Version: 1.0 Content-Type: TEXT/PLAIN; charset=US-ASCII Subject: [Sipping] New I-D on conference floor control Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Hi, All, We have written an I-D for conference floor control. Floor control is used to manage who can input for the media applications during a conference. The draft defines a mechanism of using SIP event notification architecture and SOAP to perform the floor control functions. A new event package named floor-control is defined in the draft. The draft is at: http://www.ietf.org/internet-drafts/draft-wu-sipping-floor-control-00.txt Thanks! -Xiaotao ========================================================= Name : Xiaotao Wu Email : xiaotaow@cs.columbia.edu, xw71@columbia.edu URL : http://www.cs.columbia.edu/~xiaotaow Phone : (212)939-7020, (212)939-7133, Fax: (801)751-0217 SIP : sip:xiaotaow@conductor.cs.columbia.edu Office: Room 463, Mudd building, West 120th ========================================================= _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Thu Feb 28 16:17:12 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id QAA20142 for ; Thu, 28 Feb 2002 16:17:12 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id QAA01266 for sipping-archive@odin.ietf.org; Thu, 28 Feb 2002 16:17:16 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id PAA29716; Thu, 28 Feb 2002 15:53:41 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id PAA29686 for ; Thu, 28 Feb 2002 15:53:39 -0500 (EST) Received: from revere.sonusnet.com (mail.sonusnet.com [208.45.178.33]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id PAA19223 for ; Thu, 28 Feb 2002 15:53:34 -0500 (EST) Received: from sonusdc3.sonusnet.com (sonusdc3 [10.128.32.53]) by revere.sonusnet.com (Switch-2.1.0/Switch-2.1.0) with ESMTP id g1SKqrI06426; Thu, 28 Feb 2002 15:52:53 -0500 (EST) Received: by sonusdc3.sonusnet.com with Internet Mail Service (5.5.2653.19) id <1Z0KBXV9>; Thu, 28 Feb 2002 15:53:00 -0500 Message-ID: <4CF48720B6E9D4118D040060CF2074E904A14400@sonusdc3.sonusnet.com> From: "Farahmand, Fardad" To: "'Dean Willis'" Cc: sipping@ietf.org, "'Henning Schulzrinne'" Subject: RE: [Sipping] sip-911, Preventing dialog termination Date: Thu, 28 Feb 2002 15:52:57 -0500 MIME-Version: 1.0 X-Mailer: Internet Mail Service (5.5.2653.19) Content-Type: text/plain; charset="iso-8859-1" Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org > -----Original Message----- > From: Dean Willis [mailto:dean.willis@softarmor.com] > Sent: Thursday, February 28, 2002 3:12 PM > To: Farahmand, Fardad > Subject: Re: [Sipping] sip-911, Preventing dialog termination > > > Consider the case of "user dials 911, then pulls battery out > of phone"? > > -- > Dean > That's no different from PSTN unplugging the phone. The point is that in E911, even if the phone is unplugged, the call stays up (to the nearest switch). Allowing the operator to keep the call ringing, the police to trace the address, the ambulance to arrive at the door etc. The idea is that in case of SIP maybe the dialog should not be terminated up to the nearest UA, stateful proxy or B2BUA which is E911 aware. This maybe more applicable in some cases such as when the UA is on a SoftSwitch controlling the user's device. This is probably less applicable in other cases such as a softclient UA making a call through the internet. > ----- Original Message ----- > From: "Farahmand, Fardad" > To: "'Henning Schulzrinne'" ; > Sent: Thursday, February 28, 2002 12:43 PM > Subject: [Sipping] sip-911, Preventing dialog termination > > > > > > In draft-schulzrinne-sip-911-01 there is a brief mention of > some existing > > E911 features for disabling call-waiting, three-party-calls and not > allowing > > the caller to hang up the call during an E911 call. But > the document has > no > > recommendation of how to do this with SIP. > > In particular, consider the scenario where someone calls an > emergency > number > > and then hangs up. In PSTN E911 case, the operator is > notified that the > > caller has hung up but the call is not released. The > operator has the > > option of (1) initiate a request by which the caller's device starts > ringing > > or (2) release the call. > > Is there any plan to cover recommendations for this feature > in this (or > > another) draft? > > Thanks, > > Fardad Farahmand > > > > > > > > > > _______________________________________________ > > Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping > > This list is for NEW development of the application of SIP > > Use sip-implementors@cs.columbia.edu for questions on current sip > > Use sip@ietf.org for new developments of core SIP > > > > > _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Thu Feb 28 16:23:04 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id QAA20336 for ; Thu, 28 Feb 2002 16:23:04 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id QAA01499 for sipping-archive@odin.ietf.org; Thu, 28 Feb 2002 16:23:08 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id QAA00662; Thu, 28 Feb 2002 16:09:42 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id QAA00631 for ; Thu, 28 Feb 2002 16:09:40 -0500 (EST) Received: from imr1.ericy.com (imr1.ericy.com [208.237.135.240]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id QAA19784 for ; Thu, 28 Feb 2002 16:09:35 -0500 (EST) Received: from mr7.exu.ericsson.se (mr7u3.ericy.com [208.237.135.122]) by imr1.ericy.com (8.11.3/8.11.3) with ESMTP id g1SL9ah12178; Thu, 28 Feb 2002 15:09:37 -0600 (CST) Received: from newman.exu.ericsson.se (newman.exu.ericsson.se [138.85.75.179]) by mr7.exu.ericsson.se (8.11.3/8.11.3) with ESMTP id g1SL9ae22562; Thu, 28 Feb 2002 15:09:36 -0600 (CST) Received: from lmf.ericsson.se (rmt160160.am.ericsson.se [138.85.160.160]) by newman.exu.ericsson.se (8.7.5/8.7.3) with ESMTP id PAA00403; Thu, 28 Feb 2002 15:09:34 -0600 (CST) Message-ID: <3C7E9CBB.B05E471F@lmf.ericsson.se> Date: Thu, 28 Feb 2002 23:10:19 +0200 From: Gonzalo Camarillo X-Mailer: Mozilla 4.79 [en] (WinNT; U) X-Accept-Language: en MIME-Version: 1.0 To: Jagannathan R CC: sipping@ietf.org Subject: Re: [Sipping] SIP-PSTN Interconnection References: <003401c1bf56$1784db00$7301ff0a@axsppc65> Content-Type: text/plain; charset=us-ascii Content-Transfer-Encoding: 7bit Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit Hi, request routing in SIP works in a different way than in the PSTN. Your proxy do not have to register towards your MGC. Your MGC will decide where to send a particular request based on the host part of the destination SIP URI (that has been obtained from the destination telephone number in ISUP). Regards, Gonzalo Jagannathan R wrote: > > Hi all, > > I have a some basic doubts regarding the SIP-PSTN Interworking, this > is more on SIP-T. Can any one clarify these doubts please. > > 1. Proxy server is connected to MGC (point of SIP-PSTN > Interconnection), > > 1a. Is there a way proxy server registering itself with MGC? > This is required at MGC to know which are the SIP users this > proxy can service, > so that the PSTN originated call can be correctly terminated to > that Proxy. > > 1b. How the MGC knows the status of Proxy (In-Service or > Out-of-Service) in order to terminate the call. > > 1c. How many Proxy servers with different/same set of SIP users can be > linked to one MGC. > > 2. When SIP call arrives at MGC, MGC requires NOA (Nature of Address) > in order to translate the called number and route the call to the > correct PSTN Trunk Group. How the NOA address (like 0+IDDD, IDDD, DDD > calls) can be represented in the SIP messages. > > 3. For SIP-PSTN call, how different digit sequences like ANI, > Authorization code, Account code (these are required for billing at > MGC and to have Dialing plan) can be sent to MGC in SIP Messages. > > > Thanks in advance, > > jagannathan > -- Gonzalo Camarillo Phone : +1 212 939 71 71 Columbia University Mobile: +358 40 702 35 35 472 Computer Science Building Fax : +358 9 299 30 52 1214 Amsterdam Ave., Mail Code 0401 http://www.hut.fi/~gonzalo New York, NY 10027 USA Gonzalo.Camarillo@ericsson.com _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Thu Feb 28 16:47:47 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id QAA21447 for ; Thu, 28 Feb 2002 16:47:46 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id QAA02942 for sipping-archive@odin.ietf.org; Thu, 28 Feb 2002 16:47:49 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id QAA00921; Thu, 28 Feb 2002 16:13:12 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id QAA00879 for ; Thu, 28 Feb 2002 16:13:10 -0500 (EST) Received: from bdsl.66.12.12.130.gte.net (bdsl.66.12.12.130.gte.net [66.12.12.130]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id QAA19953 for ; Thu, 28 Feb 2002 16:13:04 -0500 (EST) Received: from TXDWILLIS2 (bdsl.greycouncil.com [127.0.0.1]) by bdsl.66.12.12.130.gte.net (8.11.6/8.11.6) with SMTP id g1SLCXB14075; Thu, 28 Feb 2002 15:12:33 -0600 Message-ID: <029b01c1c09c$8fe957f0$bc036e3f@TXDWILLIS2> From: "Dean Willis" To: "Farahmand, Fardad" , Subject: Re: [Sipping] sip-911, Preventing dialog termination Date: Thu, 28 Feb 2002 15:11:59 -0600 MIME-Version: 1.0 Content-Type: multipart/alternative; boundary="----=_NextPart_000_0298_01C1C06A.448CD070" X-Priority: 3 X-MSMail-Priority: Normal X-Mailer: Microsoft Outlook Express 6.00.2600.0000 X-MimeOLE: Produced By Microsoft MimeOLE V6.00.2600.0000 Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org This is a multi-part message in MIME format. ------=_NextPart_000_0298_01C1C06A.448CD070 Content-Type: text/plain; charset="iso-8859-1" Content-Transfer-Encoding: quoted-printable Fardad says: > That's no different from PSTN unplugging the phone. > > The point is that in E911, even if the phone is unplugged, the call = stays up > (to the nearest switch). Allowing the operator to keep the call = ringing, > the police to trace the address, the ambulance to arrive at the door = etc. > > The idea is that in case of SIP maybe the dialog should not be = terminated up > to the nearest UA, stateful proxy or B2BUA which is E911 aware. > > This maybe more applicable in some cases such as when the UA is on a > SoftSwitch controlling the user's device. This is probably less applicable > in other cases such as a softclient UA making a call through the = internet. Actually, it is different. In the wireline network, unplugging the phone merely takes the analog instrument off the end of the circuit and does = not affect the actual circuit condition. Recovery consists of plugging the analog device back in -- all state was "in the network" and not affected = by the loss of the terminal. In the mobile case, taking the battery out drops the radio carrier hard, instituting a network-initiated termination process that affects systems = at the radio and application layer. Restoring power to the handset does not provide the restoration of state required to recover the emergency = signaling association. Even if the network state were somehow preserved, the = mobile itself no longer has the context, and some sort of recovery process is indicated. -- Dean ------=_NextPart_000_0298_01C1C06A.448CD070 Content-Type: text/html; charset="iso-8859-1" Content-Transfer-Encoding: quoted-printable
Fardad says:
> That's no=20 different from PSTN unplugging the phone.
>
> The point is = that in=20 E911, even if the phone is unplugged, the call stays
up
> (to = the=20 nearest switch).  Allowing the operator to keep the call = ringing,
>=20 the police to trace the address, the ambulance to arrive at the door=20 etc.
>
> The idea is that in case of SIP maybe the dialog = should not=20 be terminated
up
> to the nearest UA, stateful proxy or B2BUA = which is=20 E911 aware.
>
> This maybe more applicable in some cases = such as=20 when the UA is on a
> SoftSwitch controlling the user's = device.  This=20 is probably less
applicable
> in other cases such as a = softclient UA=20 making a call through the internet.

Actually, it is different. In = the=20 wireline network, unplugging the phone
merely takes the analog = instrument off=20 the end of the circuit and does not
affect the actual circuit = condition.=20 Recovery consists of plugging the
analog device back in -- all state = was "in=20 the network" and not affected by
the loss of the terminal.

In = the=20 mobile case, taking the battery out drops the radio carrier = hard,
instituting=20 a network-initiated termination process that affects systems at
the = radio and=20 application layer. Restoring power to the handset does not
provide = the=20 restoration of state required to recover the emergency = signaling
association.=20 Even if the network state were somehow preserved, the mobile
itself = no longer=20 has the context, and some sort of recovery process=20 is
indicated.

--
Dean

------=_NextPart_000_0298_01C1C06A.448CD070-- _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP From daemon@optimus.ietf.org Thu Feb 28 19:53:56 2002 Received: from optimus.ietf.org (ietf.org [132.151.1.19] (may be forged)) by ietf.org (8.9.1a/8.9.1a) with ESMTP id TAA27873 for ; Thu, 28 Feb 2002 19:53:56 -0500 (EST) Received: (from daemon@localhost) by optimus.ietf.org (8.9.1a/8.9.1) id TAA13454 for sipping-archive@odin.ietf.org; Thu, 28 Feb 2002 19:53:59 -0500 (EST) Received: from optimus.ietf.org (localhost [127.0.0.1]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id TAA12427; Thu, 28 Feb 2002 19:30:49 -0500 (EST) Received: from ietf.org (odin [132.151.1.176]) by optimus.ietf.org (8.9.1a/8.9.1) with ESMTP id TAA12380 for ; Thu, 28 Feb 2002 19:30:46 -0500 (EST) Received: from cs.columbia.edu (cs.columbia.edu [128.59.16.20]) by ietf.org (8.9.1a/8.9.1a) with ESMTP id TAA27562 for ; Thu, 28 Feb 2002 19:30:43 -0500 (EST) Received: from opus.cs.columbia.edu (opus.cs.columbia.edu [128.59.20.100]) by cs.columbia.edu (8.9.3/8.9.3) with ESMTP id TAA03007; Thu, 28 Feb 2002 19:30:43 -0500 (EST) Received: from cs.columbia.edu (cta.cs.columbia.edu [128.59.19.46]) (authenticated bits=0) by opus.cs.columbia.edu (8.12.1/8.12.1) with ESMTP id g210UgNB027413 (version=TLSv1/SSLv3 cipher=RC4-MD5 bits=128 verify=NOT); Thu, 28 Feb 2002 19:30:43 -0500 (EST) Message-ID: <3C7ECBA7.90F71A76@cs.columbia.edu> Date: Thu, 28 Feb 2002 19:30:31 -0500 From: Henning Schulzrinne Organization: Columbia University X-Mailer: Mozilla 4.76 [en] (Windows NT 5.0; U) X-Accept-Language: en MIME-Version: 1.0 To: "Farahmand, Fardad" CC: "'Dean Willis'" , sipping@ietf.org Subject: Re: [Sipping] sip-911, Preventing dialog termination References: <4CF48720B6E9D4118D040060CF2074E904A14400@sonusdc3.sonusnet.com> Content-Type: text/plain; charset=us-ascii Content-Transfer-Encoding: 7bit Content-Transfer-Encoding: 7bit Sender: sipping-admin@ietf.org Errors-To: sipping-admin@ietf.org X-Mailman-Version: 1.0 Precedence: bulk List-Id: SIPPING Working Group (applications of SIP) X-BeenThere: sipping@ietf.org Content-Transfer-Encoding: 7bit I should point out that I had a discussion on this topic with some folks in the (U.S.) 911 community. Opinions diverged - some claimed that switches restricted services, while other disagreed, saying (e.g.,) that this might have been done in the past, but was no longer feasible with modern switches. It appears that this may well be dependent on the LEC and maybe the particulars of the local CO. Apparently, you get some really strange results when three-way calling gets involved, with two 911 operators on the line. As long as the PSAP has a reliable call-back address, even midcall disconnects are not that big a deal. As Dean hints at, that's about the best you can do for mobiles in any event. Here are some quotes: "I can't speak for Call Transfer however Call Waiting and Three Way Calling are still available to the 9-1-1 caller. At least it is here in a DMS environment." "From my experience, most switches (5ESS, DMX-100, and GTD5EAX) disable custom calling features for the entire call when calling to N11 services in a manner similar to when they are establishing a call to another number. In other words, the switch deals with the 911 call like when you are listening to a ringing signal (i.e. no call waiting, no call transfer, no 3-way calling, etc.). ANSI T1S1 is the standards group that is responsible for the PSTN signaling standards for the US and Canada. T1S1.7 is specifcally tasked with the architecture, services and signaling. ANSI T1.628-2000 and ANSI T1.628a-2001 seem to be the governing ANSI documents relating to ECS (Emergency Calling Services). You might look there." I don't have access to the documents. _______________________________________________ Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping This list is for NEW development of the application of SIP Use sip-implementors@cs.columbia.edu for questions on current sip Use sip@ietf.org for new developments of core SIP