[SR-Users] Kamailio WebRTC Call Not Working

[Ankit Jayswal]

Hi,
I have configured Kamailio to register a SIP user over websocket. SIP users
are getting registered over webphone like JSSIP fine but not able to
establish the call scenario below.

>From Twilio Trunk - INVITE - KAMAILIO - ASTERISK - play hello world - Dial
SIP user 101 at Kamailio_IP - Rings 101 and Answer - Hangup.

Here,
101 is registered with Kamailio IP, and I am sending a call from Twilio SIP
trunk to Kamailio server IP.
Asterisk is on port 5080 and Kamailio is on port 5090 both are on the same
host 45.118.163.244.

This above call scenario is NOT working with Webphones registered via
JSSIP. I think the second call leg from Asterisk -> Kamailio is not getting
set up properly and ACK is not getting confirmed here, call connects for
some time and no voice transmits and then disconnects.

The same call scenario is working with users registered at Softphone like
Zoiper perfectly fine. No idea why it is not working with a webphone.

I have attached kamailio.cfg, Kamailio log and Asterisk's pjsip logs here.
So, please guide me on this as soon as possible.

-- 
Thanks & Regards,
*Ankit Jayswal* | Specialist - Software Development
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