[SR-Users] webrtc clients support using rtpengine

[Andrey Utkin]

Hi!
I need to establish calls between WebRTC and usual SIP clients
(exactly, sipml/jssip and linphone-android).
I used configs from https://github.com/caruizdiaz/kamailio-ws and
latest git master HEAD of both kamailio and
rtpengine. I got calls from webrtc to android working correctly (but only with
Firefox browser), even with video. But in other directions i have some
issues because of lack of RTP delivery or RTP timeouts.
I have some logs to show you regarding this:
https://gist.github.com/krieger-od/27c6f3e4924f5e21352e (works),
https://gist.github.com/krieger-od/196bcfbd331d621427ef (doesn't
work).
I would really love to get some quick help from anyone. For direct
manual fixing, I can give a couple of hundreds of bucks.

Looking forward impatiently for reply from anyone having something to say.

-- 
Andrey Utkin