InvalidSessionDescriptionError: Invalid description, no ice-ufrag attribute

[admin]

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).

<strong>This is my setup:</strong>

<a href="

https://imgur.com/lQzDI

" rel="nofollow noreferrer"><img src="

https://imgur.com/lQzDI

" alt="My setup"></a>

What I see:

<strong>In the CLI:</strong>

[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)'

<strong>In Firefox:</strong>

InvalidSessionDescriptionError: Invalid description, no ice-ufrag attribute

Attachments:

<ul>
<li><a href="http://pastebin.com/jsE6UrFk" rel="nofollow noreferrer">SIP Dialogue</a> (Asterisk CLI)</li>
<li><a href="http://pastebin.com/v5Ht7cJw" rel="nofollow noreferrer">Webphone Log</a></li>
<li><a href="http://pastebin.com/PA0jvBgD" rel="nofollow noreferrer">Config Files</a> (httpd.conf, sip.conf, rtp.conf)</li>
<li><a href="http://pastebin.com/C077p6NN" rel="nofollow noreferrer">Asterisk Compiled with Libuuid &amp; Friends</a></li>
</ul>

<strong>What I've tried so far:</strong>

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

<strong>What I've read:</strong>

<ul>
<li><a href="http://forums.asterisk.org/viewtopic.php?p=201702" rel="nofollow noreferrer">http://forums.asterisk.org/viewtopic.php?p=201702</a></li>
<li><a href="https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support" rel="nofollow noreferrer">https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support</a></li>
<li><a href="https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5" rel="nofollow noreferrer">https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5</a></li>
<li><a href="http://jssip.net/documentation/misc/interoperability/asterisk/" rel="nofollow noreferrer">http://jssip.net/documentation/misc/interoperability/asterisk/</a></li>
<li><a href="http://sipjs.com/guides/server-configuration/asterisk/" rel="nofollow noreferrer">http://sipjs.com/guides/server-configuration/asterisk/</a></li>
<li><a href="https://kunjans.wordpress.com/2015/01/09/web-sip-client-sipml5-with-asterisk-13-on-centos-6-6/" rel="nofollow noreferrer">https://kunjans.wordpress.com/2015/01/09/web-sip-client-sipml5-with-asterisk-13-on-centos-6-6/</a></li>
<li><a href="http://forums.digium.com/viewtopic.php?f=1&amp;t=89798" rel="nofollow noreferrer">http://forums.digium.com/viewtopic.php?f=1&amp;t=89798</a></li>
</ul>

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