I’ve been using webrtc (Jscommunicator) with Asterisk occasionally. Only problem until now which remained was that if dtls_rekey was set to the value other than 0, call hanged up when using chrome after the time where dtls_rekey was set.
I suppose that “bad media description” shown in Chrome’s window which causes call to fail, has appeared with Chromes newer versions (currently
58 beta installed) or with Asterisk 13.15.0. Audio codec I’m using is Opus.
Has somebody else encountered this problem, or more better resolved it?