[SR-Users] Flow Diagram for WebRTC Client1 => WebRTC Client2 (via Kamailio and Asterisk)

[Joel Serrano]

Hi, can you share with us the asterisk dialplan part where you call the
Dial() application?



On Tue, Dec 12, 2017 at 06:38 Wilkins, Steve <swwilkins at mitre.org> wrote:

> 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!
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