[Sip] Draft minutes for SIP WG meeting at IETF 56
"Dean Willis" <dean.willis@softarmor.com> Thu, 10 April 2003 19:52 UTC
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From: Dean Willis <dean.willis@softarmor.com>
To: sip@ietf.org
Cc: Rohan Mahy <rohan@cisco.com>, "Jon Peterson (jon.peterson@NeuStar.com)" <jon.peterson@neustar.biz>
Date: Thu, 10 Apr 2003 14:45:18 -0500
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Subject: [Sip] Draft minutes for SIP WG meeting at IETF 56
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Draft minutes are posted at: http://www.softarmor.com/sipwg/meets/ietf56/notes/minutes.html Please review and send corrections. Thanks. You might notice that the datail is a little light. This could be because I seem not to have received a copy of the notes recorded by the official designated secretary. Oh well, there's less to type this way. So if there are any conclusions that need to be recorded that I failed to capture in my notes, please provide details . . . Text follows: The SIP Working Group met at IETF 56 on Monday, March 17, 1300-1500, in room Continental 6. The posted agenda: 1300 Agenda Bash 1305 Status -- Chairs 1315 Non-Invite Transaction Issues, Robert Sparks draft-sparks-sip-noninvite-00.txt 1330 Referred-By Changes and Open Issues, Robert Sparks draft-ietf-sip-referredby-01.txt 1340 Resource Priority, Henning Schulzrinne draft-polk-sip-resource-02.txt 1355 Caller Preferences, Jonathan Rosenberg draft-ietf-sip-callerprefs-08.txt 1410 Congestion Safety, Dean Willis, Hisham Khartabil draft-ietf-sip-congestsafe-01.txt draft-khartabil-sip-congestionsafe-ci-02.txt 1430 Authenticated ID Bodies, Jon Peterson draft-ietf-sip-authid-body-01.txt 1445 SIPit report on S/MIME and TLS, Rohan Mahy 1455 General Discussion 1500 Break Scribe: Ramu (notes not receivedby chairs) Chart Room Coordinator: Not recorded Topic: Non-invite transactions, Robert Sparks. The slides presented are included in the proceedings. Poll == who as read and understands the draft, app. 20%. Discusssion: Principle problem is the race condition of 64*T1 built into the protocol. Given non-zero latency, the UAS side has an offset on this time, so the client may time out before the server completes. Now we just try and "complete as soon as possible" giving response codes like 408 that make no sense -- by the time it gets back to the UAC, the UAC is assured of having timed out, creating a 408 storm. Other problems exist with intermediary timeouts and with forking convergence. Two solutions offered: A and B. Rohan argues against proposal B, preferring the subscribe/notify style of seperate transactions. Point: What if values of T1 are different? Discussion: If UAS and UAC have different values of T1, it's all broken anyhow. Extended discussion follows. Two problems: the infrequent 1 in 10,000 events, and the problem that the 200 has to be emitted soon enough to have a high probability to make it back to the UAC before the 408s. Most troubling is when the UAS thinks the situation is 200ok, but the UAC has a different view (which can happen with non-invite forking). Point that we should lift the constraint on interoperability with existing proxies. Suggestion we adhoc during this meeting for consensus -- group to email Robert for gathering. Note: a detailed adhoc on this topic was held later and the minutes seperately included in the proceeedings. Topic: Referred-by, Robert Sparks The slides presented are included in the proceedings. Complaint: nobody is giving feedback. Are people mostly happy and just waiting on some of the blocking work to complete? A chairs asks for volunteers to provide feedback: Brian Rosen, Mary Barnes, Alan Johnston, Pekka Pessi all volunteer. Issue: Use of Call-ID seems to be required in auth-ID-body. Is it a leak? Should we put garbage in, change the AID body format, or what? Jon Peterson reports that the newest AIB draft relieves this requirement. Issue: Referred identity binding between REFER and INVITE. Can a referee assert a different identity on the generated INVITE than was referenced in the REFER? Can humans make the distinction? No resolution of this issue in this meeting, but it seems to relate to the URI Leasing work ongoing in SIPPING. Topic: Resource-Priority, Henning Schulzrinne The slides presented ARE NOT currently included in the proceedings but have been requested and may be received in time. Goal: Higher priority of emergency call completion during service disruption of civil emergency, especially in interworking with PSTN. Requirements are establised in RFC3487. List discussion reviewd, best option a new header. Discussion about whether it is "good" to have proxies understand this. Consensus: Work on this problem, starting from this draft as a baseline, but address additional opinions and requirements as raised in the WG process. Topic: Caller Preferences, Jonathan Rosenberg The slides presented are included in the proceedeings. Issue: Needs a thorough review of the current draft, which seems pretty much "done". Volunteers to review -- Robert Sparks, Mary Barnes, Pekka Pessi, Bob Penfield, Cullen Jennings. The plan is to review, feed back, revise, go to 2 week WGLC. Topic: Congestion Safety, Dean Willis and Hisham Khartabil Presentation by Dean Willis. Slides presented included in the proceedings. General poll indicates that after reading the draft, nobody understands how it works. People seem to have the vague feeling that the topic is important but are not cognizant of the implications. Suggestion made that "congestion safe" is a misnomer, as nothing is "safe". No consensus reported. Presentation by Hisham Khartabil. Slides presented included in proceedings. One point made from audience is that implementors do not see immediate value. Proposed by one speaker that we should consider response-CI as a possible alternate mechanism in primary work. No consensus reported. Topic: Auth-ID Issues, Jon Peterson Slides presented are currently NOT in the proceedings but have been requested. The only new "SIP" function is the new 400-response code. Other functions are non-normative. No poll for consensus recorded. _______________________________________________ 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