title: No audio in WebRTC and Asterisk
body: <p>I have a strange issue with Asterisk (in this case 13.2 version) and WebRTC.</p>
  
  <p>So, I have latest Asterisk 13.2, latest Crome (with Firefox - same problem) and sip.js (also tried with sipml5) and local network - no nat or firewall.</p>
  
  <p>The problem: if call is answered  immediately - everything works fine. But if there are some delay in answer (say, 10 seconds) - no audio in both directions.</p>
  
  <p>In RTP debug I saw that if there is some delay - destination ip address is incorrect. After removing ice servers from client config - both addresses is correct, but still no audio.</p>
  
  <p>Below is debug for call with audio:<br>
  rtp <a href="http://pastebin.com/EzfByCG5" rel="nofollow">http://pastebin.com/EzfByCG5</a><br>
  sip <a href="http://pastebin.com/1Y08yF5s" rel="nofollow">http://pastebin.com/1Y08yF5s</a>  </p>
  
  <p>and no audio call (answered after 10 seconds delay):<br>
  rtp <a href="http://pastebin.com/TNnFkz6M" rel="nofollow">http://pastebin.com/TNnFkz6M</a><br>
  sip <a href="http://pastebin.com/zG7pjcZD" rel="nofollow">http://pastebin.com/zG7pjcZD</a>  </p>
  
  <p>Also in FreeSwitch everything works fine, no matter when call is answered.</p>