We can’t do much with part of your debug. You’ll want to post a pastebin link to your full SIP trace, and be sure that it includes at least VERBOSE messages turned up to 5.[1]

Work on WebRTC support is on-going, so you’ll want to test in the very latest Asterisk version in your branch (11 or above). That means you need to be on 11.9.0-rc2[2] at this moment.

[1]: https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
[2]: http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-11.9.0-rc2.tar.gz