[SR-Users] How to setup Kamailio to dispatch SIP client requests to WebRTC enabled Asterisk

[Daniel-Constantin Mierla]

Hello,

On 18.02.19 10:07, Adesh Pandey wrote:
> Hi Guys,
> I have recently setup a Kamailio server which can accept SIP or
> WebSocket connections but voice is not coming in  the case of SIP client.
>
> I have no idea to change avp in the incoming request, please advise
> how to proceed.

the details you provide here doesn't give any clue of what is the real
problem, where and why, so likely nobody can really help directly at
this stage.

I suggest to start with debug=3 in kamailio cfg and watch the logs when
you do testing calls. Watch also the web browser console/diagnostic
tools to see if you get any hints there.

If you search on the web, you should fine some tutorials about using
kamailio in webrtc -- they might be a good reference to compare with
your config.

Cheers,
Daniel

-- 
Daniel-Constantin Mierla -- www.asipto.com
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