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