title: Asterisk trunk, chrome 36, issue with WebRTC
body: <p>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:</p>
  
  <pre><code>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
  </code></pre>
  
  <p>I have followed <a href="https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5" rel="nofollow">https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5</a> and <a href="https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial" rel="nofollow">https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial</a> (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)</p>
  
  <p>I also tried with the 11.11.0 version but I'm getting the same result.</p>
  
  <p>Help is appreciated!</p>