On Fri, Jun 15, 2018 at 08:21:56AM +0000, Pan Christensen wrote:
I based my test on https://github.com/havfo/WEBRTC-to- SIP/blob/master/etc/kamailio/kamailio.cfg but I blamed my failure on an old rtpengine :)
You need to add 'SDES-off' to the rtpengine_manage strings for calls going to WebRTC. Most browsers don't support fallback to SDES (anymore) and will reject the call if both DTLS and SDES are offered.
IIRC the problem I had was that going from RTP to SRTP, there was no key exchange in SDP (a=crypto) being added by rtpengine. I'll look into this again in the near future.