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Starred by 13 users
Status: Available
Owner: ----
NextAction: ----
OS: ----
Pri: 3
Type: Bug

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CSRCs audio activity/contribution through getStats or other WebRTC API
Reported by, May 12 2014 Back to list
What steps will reproduce the problem?
In many situations, mixed audio data is necessary in order to achieve efficient transport of audio in multi-party conferences. Audio is usually mixed on a MCU-like server and sent to every subscribing client. Even if audio is mixed, it may be needed to have the possibility to identify which audio source is contributing to the mixed audio at every moment at the subscribing client side.

This can be done in a basic manner indicating CSRCs on RTP packets (e.g. as recommended in 7.1 section of RFC3550), or in a more advanced manner through RFC 6465 spec (enabled by  urn:ietf:params:rtp-hdrext:csrc-audio-level on RFC).

What is the expected result?
Support of CSRC contribution display through getStats or other alternative through the WebRTC API.

What do you see instead?
It seems not supported

What version of the product are you using? On what operating system?
 34.0.1847.131 on Mac OS X

Any plans for supporting this? Timings?

Project Member Comment 1 by, May 14 2014
@justin, any comment?
This will not be supported in WebRTC 1.0. It is currently being debated for future versions of WebRTC.
Project Member Comment 3 by, May 22 2014
Labels: Area-PeerConnection
Status: WontFix
Reopen if/when this lands in the W3C spec. Until then, you can use a datachannel from the mixer to the client. has been merged to 1.0 so this should be reopened.
Project Member Comment 5 by, Oct 12 2015
Status: Available
Project Member Comment 6 by, Oct 19 2015
Labels: EngTriaged
Project Member Comment 7 by, Oct 26 2015
Yes, we'll add support for this as we add RtpSender and RtpReceivers according to the 1.0 spec.
Project Member Comment 8 by, Nov 8 2016
Labels: Pri-3
Project Member Comment 9 by, Dec 14 2016
Owner: ----
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