The content of RTCP

[manphis chen]

Dear all,

Recently I am investigating the feedback mechanism for rate control in webrtc.

Does the receiver use RTCP to inform the sender about the information such as available bandwidth, lossy ratio, .., etc.?

Besides, is the rate control information, such as available bandwidth, appended to RTCP packets in the form of TMMBR (or REMB)?

I check the packet format by wireshark.

I found that RR (Receiver Report) is appended to SR (Sender Report), and there is extra information appended after RR, but wireshark cannot recognize it.

How can I know the exact information in RTCP control packets?

Does anyone has the same experience?

Or is there any document describing the details of the feedback mechanism or the format of the feedback packets in webrtc?

Many thanks

[Dennis]

You're going to want to read the ietf drafts that describe usage of RTP and RTCP in WebRTC https://datatracker.ietf.org/wg/rtcweb/ , specifically the latest 'Web Real-Time Communication (WebRTC): Media Transport and Use of RTP' document.  As far as report blocks after sender reports, that is normal and expected with bi-directional sessions.  The "extra information" after the RTCP reports is related to security (SRTP/crypto/message auth data).  The reports themselves are encrypted so wireshark isn't going to help you out much as far as inspecting the data on the wire in SRs and RRs.  You're also going to want to read the ietf security drafts to learn more about how cryptography works with WebRTC.

[Justin Uberti]

You can read the details on the REMB algorithm used in Chrome and Firefox here: http://tools.ietf.org/html/draft-alvestrand-rmcat-congestion-00

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[ebu...@thrupoint.com]

When I was coding things related to RTCP I found the easiest way to work with wireshark was to compile Chromium with encryption/decryption turned off for RTCP. I also got the bytes to be printed out in logs at other times as this is what you are really working with.

Eric

[Dong Fangshuo]

hello Eric,

I am trying to disable encryption/decryption for RTCP, but could not make it.

Could you give some help?

Thanks,

Wei

[V]

Wei,

I'm not sure if you completed your investigating or even if whats in webrtc now resembles what you did back then.  I'm wondering if you could share some of your findings.

On Thursday, June 13, 2013 9:11:06 PM UTC-4, manphis chen wrote: