CSRCs audio activity/contribution through getStats or other WebRTC API
|Reported by os...@tokbox.com, 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?
May 14 2014,
@justin, any comment?
May 14 2014,
This will not be supported in WebRTC 1.0. It is currently being debated for future versions of WebRTC.
May 22 2014,
Reopen if/when this lands in the W3C spec. Until then, you can use a datachannel from the mixer to the client.
Oct 10 2015,
https://github.com/w3c/webrtc-pc/pull/300 has been merged to 1.0 so this should be reopened.
Oct 12 2015,
Oct 19 2015,
Oct 26 2015,
Yes, we'll add support for this as we add RtpSender and RtpReceivers according to the 1.0 spec.
Nov 8 2016,
Dec 14 2016,
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