Re: [SR-Users] No audio when calling from SIP to WEBRTC

Hi all, I have configured Kamailio for WebSockets following this guide as an example: https://github.com/havfo/WEBRTC-to-SIP/blob/master/etc/kamailio/kamailio.cfg With sip.js and jssip I'm able to initiate a call from WebRTC to SIP and establish a call successfully. The issue arises when I try to receive a call from a SIP device. In this case the call establishes but there is no audio in either direction. I *think* the issue is with RTP Engine and I've raised a bug there, but I'm not sure why it is misbehaving https://github.com/sipwise/rtpengine/issues/983. There are some logs from RTP engine posted here. The sip device communicates with Kamailio over UDP / RTP, nothing is encrypted. I would appreciate any guidance. Thanks in advance, C _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users