- From: Iñaki Baz Castillo <ibc@aliax.net>
- Date: Thu, 13 Feb 2014 16:42:36 +0100
- To: Tim Panton new <thp@westhawk.co.uk>
- Cc: "public-webrtc@w3.org" <public-webrtc@w3.org>
2014-02-13 15:59 GMT+01:00 Tim Panton new <thp@westhawk.co.uk>: >> Right, the subject of this thread is "active speaker information in >> mixed streams" so clearly we need the CSRC values inspection in client >> side. That can only be achieved by the WebRTC API (and the WebAudio >> API is totally useless for this subject). >> > > You SIP guys are so funny, you insist that the benefit of SIP is that it decouples > signalling from media, then you go adding more and more "media-meta-data" to the media channel, > till it looks like a signalling channel Well, CSRC field is part of the RTP specification since RFC 3550 (2013) or, even more, since RFC 1889 (1996). It is not a "new addition" at all. WebRTC uses RTP so I don't consider I am requesting something "new". > The 'correct' solution to this is to re-engineer your mixer so it sends active speaker info > over the data channel, not to further delay the standard by adding more VoIP specific legacy features. Indeed that's an option, not so good since it requires perfect synchronization between both the RTP and DataChannel flows, but it is feasible, of course. Anyhow, I'm not requesting "CSRC inspection feature" for 1.0 at all. I am just wondering about how feasible is to add it in some (future) stage of WebRTC. Regards. -- Iñaki Baz Castillo <ibc@aliax.net>
Received on Thursday, 13 February 2014 15:43:23 UTC