I'm trying to get Asterisk yesterday's trunk and Chrome 36 via WebRTC. The websocket connection is established and the client registers correctly, but when I try to make a call from the browser, I get this message from Asterisk:
chan_sip.c:10535 process_sdp: Rejecting secure audio stream without encryption details: audio 35348 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126
I have followed https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5 and https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial (although admittedly I'm not able to import in Chrome the .pem client certificate, but I'm not sure it's needed. I have imported the server certificate though)
I also tried with the 11.11.0 version but I'm getting the same result.
Help is appreciated!
have you tried using an older version of Chrome? We had everything working till we upgraded to Chrome 36. Peers succeded in registering and composing a call, but as soon as the other party answered, the call died. After some research of which I unfortunately am not able to provide links, we realized that that Chrome update messed things up. We are now compelled to use the version 34.0.1847.137m which works for us. Give it a try.