12 Dec
2017
12 Dec
'17
5:38 p.m.
Hello All,
I am looking for a Diagram or such that shows the flow of SIP traffic for a WebRTC Client1
=> WebRTC Client2 call using Kamailio in front of Asterisk.
I am unable to get Asterisk to find the correct registered clients, which are registered
in Kamailio and am hoping verifying the flow will help give me a clue as to what is going
on. E.g. Using chrome and tryit-pjsip I have Client1, and Client2 registered in Kamailio.
However when I try to connect Client1 to Client2 (make a call), Asterisk has no clue where
Client1 and Cleint2 are registered to.
Thank you!