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

[Adesh Pandey]

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.
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