Dan, have you open an issue on asterisk jira? did you test the flag in jssip?
thanks best regards On Jan 18, 2017, 10:52 -0600, Dan Jenkins <[email protected]>, wrote: > 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 > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-dev mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-dev
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