[asterisk-dev] Issues with WebRTC going forward - rtcp-mux now required

[Dan Jenkins]

Hi All,

I've been working with a company who utilise WebRTC using Asterisk behind
Kamailio to connect browser users and their SIP infrastructure and just
came across an issue making/receiving calls in Chrome Canary and Chrome Dev.

Long story short; the issue is that rtcp-mux has now been set as required
in Chrome's WebRTC stack -
https://groups.google.com/forum/#!topic/discuss-webrtc/eM57DEy89MY

For now; there is a workaround of being able to pass in a flag to the
RTCPeerConnection call to get the old "negotiate" behaviour and I'm talking
to the Chrome team to find out how long this flag will be around for.

I've been told that Asterisk doesn't support rtcp-mux as of today and so
I'm raising the issue here. It seems - if Asterisk wants to support WebRTC
long term; it will need to support rtcp-mux - I quote Sean Bright from a
conversation we had in IRC where he said it was "non-trivial" to support.

I don't know more than this and I don't mean to say something is difficult
to fix when I honestly don't know the effort levels in order to fix this.
So I'm raising this here for a conversation.

I will be testing the client side flag fix in jssip in a moment and then
writing up a blog post about it if it does indeed fix the issue, at least
temporarily while the flag is available.

Dan
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