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

Hi Daniel, I set up a brand new install of the latest versions of Kamailio and RTP Engine. They're both up and running but I'm hitting a different error whereby I get the following error from JSSIP: jssip-3.4.3.min.js:9 WebSocket connection to 'wss://mydomain.com:5061/' failed: Connection closed before receiving a handshake response Please note that mydomain.com is set as an alias in Kamailio. I don't see anything in the logs about the websocket connection so I'm at a bit of a loss here. Any advice is most appreciated. Thanks! _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users