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

[Sergiu Pojoga]

Are you sure 5061 is your wss port?

At least by default in the webrtc project you used, wss listens on port 4443

On Thu, May 14, 2020 at 7:29 PM Chirag Desai <c.desai at rxhost.co.uk> wrote:

> 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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