On 05/16/14 02:45, Alexey Rybalko wrote:
Hello!
During a call from classical SIP softphone to WebRTC there's no media from the browser (Mozilla, the same result is for Chrome). In case of a call from the browser to the softphone there's media flow from both sides.
The snippets from kamailio.cfg related to the problem case (SIP-->WebRTC) are below.
There's nothing wrong with the SDP bodies that I can see. I recall that Firefox had or still has a problem with ICE role switching when ice-lite is offered. It never completes ICE negotiation (never sends an STUN packet with "use candidate") and so never starts DTLS handshake.
You can confirm that by doing a packet capture including the RTP ports and inspecting the STUN packets. Chrome shouldn't have that problem though, perhaps do another test run with it? You can send those capture files to me if you'd like me to have a look.
cheers