<HTML><BODY>Hi, All!<br><br>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. <br>Anybody can tell in what direction to dig?<br><br>Best regards, Herman.<br></BODY></HTML>