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:
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:
- SIP Dialogue (Asterisk CLI)
- Webphone Log
- Config Files (httpd.conf, sip.conf, rtp.conf)
- Asterisk Compiled with Libuuid & Friends
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:
- http://forums.asterisk.org/viewtopic.php?p=201702
- https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support
- https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5
- http://jssip.net/documentation/misc/interoperability/asterisk/
- http://sipjs.com/guides/server-configuration/asterisk/
- https://kunjans.wordpress.com/2015/01/09/web-sip-client-sipml5-with-asterisk-13-on-centos-6-6/
- http://forums.digium.com/viewtopic.php?f=1&t=89798
Please, if you can, give me a hand. I'm about to smash my box with a sledge hammer.