[OpenSIPS-Devel] WebRtc and opensips

[Герман Чичикин]

 Hi, All!

Having trouble with testing webrtc with a rtpengin. I use opensips.cfg from opensips.org (Documentation -> Tutorials -> WebSocket Transport using OpenSIPS), opensips v2.1, clients - sipml5, jssip demo webphones or my own script in various combinations. Clients register without problems. When a client calls another, the callee joyfully calls. But when I pick up the phone call is dropped, session ends and no any media. The caller receives a "bye" from opensips. 
Anybody can tell in what direction to dig?

Best regards, Herman.
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