[pjsip] PJICE session reseting after negotiating new sip channel

Vytis Valentinavičius xytis2000 at gmail.com
Wed Dec 4 04:02:30 EST 2013


I have an issue with Asterisk transportation mechanism when dealing with
WebRTC clients.
Issue happens when WebRTC client issues Hold request then Resumes the call.
SipMl client running on chrome creates a new channel on 'unhold' event and
correctly negotiates with Asterisk usage of a new port.

What should be done with existing ICE session? Should I scrap and recreate
it, or there is a better way?
Currently Asterisk (svn HEAD) does not alter the ICE session, thus resumes
sending the packets to wrong client port, which in turn get dropped,
resulting in no sound on clients side.

Or I am misunderstanding something?
Basically can I reset ICE session candidates without scraping existing
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