[SR-Users] Help with SIP over Websocket audio call

[Brad Johns]

Hi,

New to Kamailio.  I have my Kamailio 4.0 server with websocket support, and
the users can register using the JsSIP Tryit sample WebRTC application.

However, after registration, the users can't place an audio call.  I see no
ringing on the remote browser.  I don't know how to debug this further to
find out what the problem is.  Can anyone help with clues or debug?  In
Debug log I can see the websocket ws_frame.c decode the websocket message
into SIP, and I see normal SIP call flow for an INVITE.  However, nothing
indicating a call.

With this JsSIP, I can do chat through Kamailio SIP over WebSockets.

With this Kamailio server, SIP User Agent Clients work just fine to
register and place SIP call with audio.

It's just that WebRTC audio calls don't work with JsSIP sample application
with Kamailio 4.0 websocket module.

Kamailio websocket configuration borrowed from:

https://gist.github.com/jesusprubio/4066845

Any help debugging this appreciated.
Brad
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