1
votes

I'm trying to get asterisk 11.20.0 running with WebRTC (sip.js 0.72 which I believe is a fork of jssip), but I'm seeing the following (and the called party rings, but when the phone is answered the call gets hung up).

This is my setup:

My setup

What I see:

In the CLI:

[2015-11-24 01:01:53] NOTICE[43619][C-00000002]: res_rtp_asterisk.c:4441 ast_rtp_read: Unknown RTP codec 95 received from '(null)'

In Firefox:

InvalidSessionDescriptionError: Invalid description, no ice-ufrag attribute

Attachments:

What I've tried so far:

  • Changed webRTC implementations (tried chrome and firefox both with SIPML and SIP.JS)
  • Set the STUN server to null on the client side (stunServers: ['stun:null'])
  • Configured properly (I hope) my sip.conf and rtp.conf and httpd.conf
  • Made sure I have libuuid, uuid and their -devel companions and after i've recompiled asterisk.

What I've read:

Please, if you can, give me a hand. I'm about to smash my box with a sledge hammer.

1

1 Answers

2
votes

Faced same issue and followed instruction in http://forums.digium.com/viewtopic.php?f=1&t=90167 realise that:

This issue is caused because you asterisk don't have ICE support, you can solve that by installing the uuid/libuuid and uuid-devel/libuuid-devel packages on your system. Then recompile asterisk(be sure to rerun the configure script before the make command).

I did recompile my Astersik 11.16.0 with patch for ECDH support and fallback to prime256v1 https://issues.asterisk.org/jira/browse/ASTERISK-25265 and looks like lost uuid support at that time. Reverting back to non-patched version (with uuid support, use to be compiled before) resolved my issue with "no ice-ufrag attribute" error in Firefox console and calls are going well now from WebRTC client SIPML5 based to asterisk, but not in opposit direction