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DoubangoTelecom / webrtc2sip

Smart SIP and Media Gateway to connect WebRTC endpoints to any SIP-legacy network
https://doubango.org
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No Audio if call is accepted after 30 seconds #187

Open vinodpandey23 opened 9 years ago

vinodpandey23 commented 9 years ago

I am using Asterisk 11.6-cert9 with webrtc2sip and sipml5 javascript library for voice over web.

Call is initiated from legacy SIP client and received from sipml5 supported web page. Call is connected and successful audio at both end.

However if call is received after waiting 30 seconds or later of notification; it is connected successful but no audio at both end. I can see below line in webrtc2sip logs: "Audio producer is not yet started"

devkapil007 commented 9 years ago

I am also getting the same issue: I have verified using Chrome and firefox. On firefox, there is no voice if call is picked up after 30 seconds and on chrome there is no voice if call is picked up after 60 seconds(Approx). Please help..

vinodpandey23 commented 9 years ago

One more observation:

  1. Call notification on firefox/chrome browser with permission popup for microphone sharing
  2. If microphone is shared as allow and call is not connected after 30 seconds of that
  3. Then on browser console; we can see log as "ice failed; see about:webrtc for more details" and then no audio.

Since we are removing manual selection of permissions of microphone through about:config in firefox; we might have the situation where we pick the call after 30 seconds and expecting audio successfully. Please let me know if any solution for this problem.