Webrtc Not Working With Asterisk 11.8 + Jssip/sipml5
users are registering over ws:// but while dialing A -> B , using either jssip/sipml5
I receive an error on B side saying , ice related information is missing , and in INVITE sdp mentioned fields are really missing. Exact error in console is
SetRemoteDescription failed: Called with an SDP without ice-ufrag and ice-pwd
lots of users are using webrtc with jssip/sipml5 + asterisk successfully. Is this some recent chrom related change that is breaking things. Anyone in community using similar setup?