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Starred by 15 users

Issue metadata

Status: Available
Owner: ----
Cc:
Components:
NextAction: ----
OS: ----
Pri: 3
Type: Bug



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CSRCs audio activity/contribution through getStats or other WebRTC API

Reported by os...@tokbox.com, May 12 2014 Back to list

Issue description

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 braveyao@webrtc.org, May 14 2014

Cc: braveyao@webrtc.org vikasmarwaha@webrtc.org
Owner: juberti@webrtc.org
@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 juberti@webrtc.org, 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.
https://github.com/w3c/webrtc-pc/pull/300 has been merged to 1.0 so this should be reopened.
Project Member

Comment 5 by juberti@webrtc.org, Oct 12 2015

Owner: pthatcher@webrtc.org
Status: Available
Indeed.
Project Member

Comment 6 by honghaiz@google.com, Oct 19 2015

Labels: EngTriaged
Project Member

Comment 7 by pthatcher@webrtc.org, 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 pthatcher@webrtc.org, Nov 8 2016

Labels: Pri-3
Project Member

Comment 9 by anatolid@webrtc.org, Dec 14 2016

Cc: pthatcher@webrtc.org
Owner: ----

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