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

[Chirag Desai]

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!
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