[SR-Users] [rtpengine] No media from WebRTC UA

Richard Fuchs rfuchs at sipwise.com
Fri May 16 12:53:57 CEST 2014

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.


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