WEBRTC with asterisk 13.1 and freepbx 12

I have installed freepbx with asterisk 13.1.0 along with webrtc phone. All work fine should the video support is not enabled.
WEBRTC phone version is : 12.0.0beta13
User Control Panel is : 12.0.0beta42

The moment video support is enabled webrtc starts experiencing the following:

  1. Sometimes call to webrtc phone lands on the voice mail of that extension
  2. Some calls from webrtc phone to an extension does not go through/ calls from webrtc to a linphone extension does not go through at all, but from a linphone to webrtc works with audio one way
  3. Calls from an extension to the webrtc always have one way audio (audio from webrtc to extension ok, but from extension to webrtc is not)
  4. No video despite the fact that I am using linphone (as advised by some users) with VP8 codec for video and opus for audio
  5. When video support is enabled the (call goes through with one way audio) however (following errors are displayed upon a call in asterisk cli (
    WARNING[13387][C-0000002a]: res_srtp.c:407 ast_srtp_unprotect: SRTP unprotect failed with: authentication failure 10

netsock2.c:303 ast_sockaddr_resolve: getaddrinfo(“vb1v8kpufuva.invalid”, “(null)”, …): Name or service not known
WARNING[30054][C-00000025]: chan_sip.c:16062 __set_address_from
6. At a certain point asterisk crashes should lots of attempts to communicate with webrtc extension.

Important to note that all devices are on the same subnet , no natting enabled or used.

Hope someone can share his experience and knowledge to solve or shed some light on this issue.

I would like to add to the above that, Asterisk 12 current works better.

  1. At Least the asterisk does not crash at all
  2. Video and audio calls work fine with non webrtc client
  3. Webrtc client towards extension (in video enabled @ sip.conf ) works with with 2 ways comminication
  4. Any extension to Webrtc call gives one way communication.
    Error message in sip cli

ERROR[3497][C-00000034]: netsock2.c:303 ast_sockaddr_resolve: getaddrinfo(“73qisucfqtc6.invalid”, “(null)”, …): Name or service not known
WARNING[3497][C-00000034]: chan_sip.c:16053 __set_address_from_contact: Invalid host name in Contact: (can’t resolve in DNS) : ‘73qisucfqtc6.invalid’

The asterisk 12 was compiled ((if that helps)) with --with-crypto --with-ssl --with-srtp

Again the Asterisk box is in internal network along all the extensions (no nat) and no external IP set also consider no internet is involved at this stage.

Problems solved with asterisk 13.2

Webrtc stopped after upgrading forefox from version 36 to version37.
I have been running webrtc with freepbx 12 and firefox version 36 without any issues until firefox was upgraded to version 37.
Unfortunately Chrome works well in one direction (from chrome to any extension) but calling from an extension to a webrtc on chrome has one way voice.

Could someone try to investigate the problem of firefox version37.0.1 with webrtc ? no voice in any direction.
Should we try it with a computer that has not an updated version of firefox things work normally.

Thank you and best regards

This would be a issue between Asterisk and Firefox not anything we could assist or solve.

Thank you , I have posted that to the Asterisk community. Let us see the outcome.