Harald pointed out that my first posting of this was officially a reply to an earlier message that some may have missed, so I'm reposting as a top-level item. -- Randell On 10/10/2011 5:22 AM, Magnus Westerlund wrote:
So what are our alternative here?
1) Pick TFRC for now while developing something better? Possibly ensure that the RTP mapping gets published in a reasonable time frame.
2) Try to write clear requirements on the implementation, but no specification and hopes that goes through?
3) Develop something and delay the publication of any part that needs this until it is done?
4) ?
Regarding 3), I don't see how that is going to complete in less than 2 more likely 3 years. Spin up a TSV WG, develop a solution. Simulate and discuss corner cases for a while before getting good enough out.
I vote for 2, and provide a sample implementation and let people innovate. I would also support trying to standardize the sample implementation under 3 in parallel, with the understanding it will take A Long Time. Here's a set of proposed stab at requirements for rtcweb implementations: As part of rtcweb, congestion control must be addressed: 1. All WebRTC media and data streams MUST be congestion-controlled. 2. The congestion algorithms used MUST cause WebRTC streams to act fairly with TCP and other congestion-controlled flows, such as DCCP and TFRC, and other WebRTC flows. Note that WebRTC involves multiple data flows which "normally" would be separately congestion-controlled. 3. In order to support better overall user experiences and to allow applications to have better interaction with congestion control, a new AVPF feedback message [ insert name here] shall be defined to allow reporting of total predicted bandwidth for receiving data, as opposed to TMMBR, which requests a sending rate for a single SSRC flow. [ This is roughly equivalent to b=CT:xxx ] We may want to give the estimation algorithm the option to not include or exclude the data-channel bandwidth, but it SHOULD include that. 4. In order to facilitate better operation of bandwidth-estimation algorithms on the receiving side, the sending side MAY include a transmit-time RTP header extension (TBD) to some or all media streams. Note that this will add about 12 bytes to each RTP packet. An optimization may be to only include these timestamps if they deviate by more than [ some amount TBD from the running average and from the number of bytes preceding it with the same timestamp ]. This is based on the fact that for many devices, the sample->send interval is fairly consistent at the levels of accuracy needed here, and so significant bandwidth savings can be made. 5. The receiver SHOULD attempt to minimize the number of bandwidth reports when there is little or no change, while reporting quickly when there is a significant change. 6. Congestion control MUST work even if there are no media channels, or if the media channels are inactive in one or both directions. 7. The congestion control algorithm SHOULD attempt to keep the total bandwidth controlled so as to minimize the media- stream end-to-end delays between the participants. 8. When receiving a [ insert new AVPF message here ], the sender shall attempt to comply with the overall bandwidth requirements by adjusting parameters it can control, such as codec bitrates and modes, and how much data is sent on the data channels. Not part of our IETF requirements, at the JS level bandwidth changes should be reported to the application along so that it has the option to make changes that we can't make automatically, such as removing or adding a stream, or controlling the parameters of a stream (frame rate, etc). Note that if the application doesn't do anything, the automatic adaptation will still occur. -- Randell Jesup randell-ietf@jesup.org