[SR-Users] webrtc clients support using rtpengine

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