Hello, 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 session? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20131204/d4d09d69/attachment-0001.html>
