José Lopes writes:
Hello Daniel,
Thanks for your reply.
On the next link, I put the original SDP from webrtc client and the SDP after kamailio that exposes the issue. I am using Kamailio version 5.3.3 with sdpops and rtpengine module.
Perhaps it is this line that the webrtc UAS doesn't like:
m=audio 18184 RTP/AVP 111 103 104 9 0 8 106 105 13 110 112 113 126
If I remember correctly, webrtc mandates UDP/TLS/RTP/SAVPF media transport.
-- Juha