Very interesting development. I was waiting for this to become available.
Excellent job Jitsi team.
I presume that, for now, SIP calling in Jitsi-meet only works with jigasi.
Is that correct?
If we already run our own SIP proxy or B2BUA, like kamailio or Freeswitch,
is it possible to use that instead of jigasi? Any docs out there I can study
on how to implement this functionality?
Jitsi Meet JavaSscript app is using part of Rayo signaling for
starting a call. It sends dial IQ to "config.hosts.call_control" XMPP
component. After Jigasi receives the IQ it joins the conference room
specified in Rayo IQ and waits for session-initiate from the focus.
Note that altough there is code for hanging up a call through Rayo it
is currently not used and handled by Jignle terminate session.
In opposite direction when Jigasi receives SIP invite it looks for
extra header mentioned by Emil and joins the room, then wait for
invite from the focus.
In your own proxy you will need to handle Jingle session with the
focus, MUC presence, do audio mixing and forward it to/from SIP peer.
On Thu, Sep 4, 2014 at 1:04 PM, Peter Villeneuve <firstname.lastname@example.org> wrote: