The IETF 86 meeting was held from March 10-15, 2013 in Orlando, FL, USA
- Magnus Westerlund and Colin Perkins, Multiple RTP Session on a Single Lower-Layer Transport (.txt|.pdf), Internet Engineering Task Force, February 2013, Work in progress (draft-westerlund-avtcore-transport-multiplexing-05.txt).
This version includes minor editorial corrections. There are no technical changes.
- Magnus Westerlund, Bo Burman, Colin Perkins, and Harald Alvestrand, Guidelines for using the Multiplexing Features of RTP (.txt|.pdf), Internet Engineering Task Force, February 2013, Work in progress (draft-westerlund-avtcore-multiplex-architecture-03.txt).
This is a relatively minor update, mostly removing text on new RTP topologies, which is now in the RFC5117bis draft.
- Colin Perkins, On the Use of RTP Control Protocol (RTCP) Feedback for Unicast Multimedia Congestion Control (.txt|.pdf), Internet Engineering Task Force, February 2013, Work in progress (draft-perkins-rmcat-rtp-cc-feedback-00.txt).
- Jonathan Lennox, Magnus Westerlund, Qin Wu, and Colin Perkins, RTP Considerations for Endpoints Sending Multiple Media Streams (.txt|.pdf), Internet Engineering Task Force, February 2013, Work in progress (draft-lennox-avtcore-rtp-multi-stream-02.txt).
This version is merged with draft-wu-avtcore-multisrc-endpoint-adver; changes how Reporting Groups are indicated in RTCP, to make it clear which source(s) is the group's reporting sources; clarifies the rules for when sources can be placed in the same reporting group; and clarifies that mixers and translators need to pass reporting group SDES information if they are forwarding RR and SR traffic from members of a reporting group. I was added as a co-author, based on my input in discussions at IETF 85.
- Colin Perkins, Magnus Westerlund, and Jörg Ott, Web Real-Time Communication (WebRTC): Media Transport and Use of RTP (.txt|.pdf), Internet Engineering Task Force, February 2013, Work in progress (draft-ietf-rtcweb-rtp-usage-06.txt).
The changes in this version of the draft are as follows:
- Expand and clarify discussion of RTP session multiplexing in Section 4.4
- Add Section 7.2 on RTCP extensions for congestion control
- Clarify Section 12.1 on RTP Sessions and PeerConnections
- Expand Section 12.4 on SSRC collision detections
- Rewrite and clarify Section 12.5 on Contributing Sources and the CSRC list
- Rewrite and clarify Section 12.9 on differentiated treatment of flows
- Expand security considerations
- Mark Handley, Van Jacobson, Colin Perkins, and Ali Begen, SDP: Session Description Protocol (.txt|.pdf), Internet Engineering Task Force, February 2013, Work in progress (draft-ietf-mmusic-rfc4566bis-07.txt).
Minor editorial fixes. Note open issues around "a=sdplang:" and "a=lang:" attributes.
- Colin Perkins and Varun Singh, Multimedia Congestion Control: Circuit Breakers for Unicast RTP Sessions (.txt|.pdf), Internet Engineering Task Force, February 2013, Work in progress (draft-ietf-avtcore-rtp-circuit-breakers-02.txt).
The significant changes in this version are as follows:
- In the media timeout circuit breaker, disallow reduction in rate by a factor of 10 as a response when circuit breaker triggered. A media timeout (several reporting intervals when media is being set but not received) signals significant path failure, not a transient problem, and so should stop the RTP media flow, not just reduce it's rate.
- Clarify RTCP Timeout circuit breaker: note that the fixed minimum RTCP reporting intervals SHOULD be used when calculating the RTCP timeout. The rationale is in Section 6.2 of RFC 3550; to avoid premature timeouts if not all participants use reduced minimum interval.
- Clarify congestion circuit breaker: use actual RTCP reporting interval, not the fixed minimum interval, when determining if congestion is occurring. The actual interval, when using the reduced minimum interval, scales with the data rate, and so matches the dynamics of the congestion circuit breaker.
- Break out the description of what it means to cease transmission into a separate section, and expand. When deciding when to restart transmission, clarify that the destination 3-tuple (transport, port, IP addr) rather than the full 5-tuple is used when checking if congestion has eased. The Rationale is that it is not okay to simply change the source port, and try again on the same path; need a different IP-layer path
- Clarify behaviour with reduced-size RTCP: reduced-size RTCP packets containing RTCP SR or RR packets MUST be counted towards the circuit breaker conditions, while reduced size RTCP packets that don't contain SR or RR packets are not counted towards the circuit breaker. The intention is to allow use of low-overhead reduced-size RTP/AVPF NACKs for congestion control without risk of triggering circuit breaker, whilst reacting to significant loss events reported by SR/RR packets.
- Expand discussion of how and when ECN-CE marks are counted towards the circuit breaker. RFC 6679 provides RTCP extensions to feedback ECN-CE marks in RTCP XR, and these are counted towards the circuit breaker. ECN-CE marks reported in a reduced size RTCP packets along with SR or RR blocks are processed; if the SR or RR block is not present, they're ignored. These are conceptually the same rules as for packet loss, but adapted for ECN.
- Update title and abstract
- Clarify that multicast is out of scope
- Clarify why unicast RTP sessions might have more than two SSRCs
- Clarify that RTCP support is required
- Clarify that implementations without a circuit breaker, or congestion control algorithm operating within a circuit breaker envelope, ought not be used on networks subject to congestion
- Note that the RTCP RR jitter estimate is not valid for frames split across multiple RTP packets with the same timestamp
- Expand discussion of competition with TCP flows
- Clarify operation of the congestion circuit breaker if the fraction of packets lost is zero
- Clarify that the circuit breaker at a sender only looks at RTCP SR/RR packets that contain reports for the SSRC values it is using to send
- Magnus Westerlund, Colin Perkins, and Jonathan Lennox, Multiple Media Types in an RTP Session (.txt|.pdf), Internet Engineering Task Force, February 2013, Work in progress (draft-ietf-avtcore-multi-media-rtp-session-02.txt).
This revision adds sections 6.4.1 on Timing out SSRCs and 6.4.2 on Tuning RTCP transmissions, and fixes the use of RFC 2119 normative language in some places, but is otherwise a minor update.
- Ali Begen, Colin Perkins, Dan Wing, and Eric Rescorla, Guidelines for Choosing RTP Control Protocol (RTCP) Canonical Names (CNAMEs) (.txt|.pdf), Internet Engineering Task Force, December 2012, Work in progress (draft-ietf-avtcore-6222bis-00.txt).
Updates RFC 6222 to use a cryptographically secure random number generator, rather than the less secure and more complex alternative suggested.
- Colin Perkins and Magnus Westerlund, Securing the RTP Protocol Framework: Why RTP Does Not Mandate a Single Media Security Solution (.txt|.pdf), Internet Engineering Task Force, February 2013, Work in progress (draft-ietf-avt-srtp-not-mandatory-12.txt).
- Colin Perkins and Magnus Westerlund, Securing the RTP Protocol Framework: Why RTP Does Not Mandate a Single Media Security Solution (.txt|.pdf), Internet Engineering Task Force, November 2012, Work in progress (draft-ietf-avt-srtp-not-mandatory-11.txt).
This version of the draft updates the recommendations in Section 6 based on discussion with the Security Area Directors. There are no changes in the rest of the draft.