Hello,
you have to implement a classic-sip-to-webrtc gateway, which can be done
using kamailio and rtpengine. If you search on the web, you can find
some sample configs that can be good starting points to plug into your
existing configuration.
Sometime is better to use a dedicated system for this
classic-sip-to-webrtc gateway function, so the main Kamailio SIP server
configuration stays simpler.
Cheers,
Daniel
On 17.11.20 09:45, Melek Oktay wrote:
Hi,
I am using FreeSwitches behind the Kamailio proxy server and I am
trying to allow multiple registration to my extensions.
So, following configuration is sample of my Kamailio
modparam("registrar", "xavp_cfg", "reg")
......
$xavp(reg=>max_contacts) = 10;
save("location");
....
I saw my phones could register with the same account credentials via
several phones such as Cisco, Zoiper, Yealing etc. When the call is
forward to this extension, all of them are ringing. Very Nice.
But, when I am trying to REGISTER WebRTC supports soft-phones to my
system and with the same account credentials, my extensions are not
ringing like in the previous scenario. WebRTC uses Websocket (WS)
technology and clients register to Kamailio via usrloc module.
When the call is forward to this extension, Kamailio try to replicate
WebRTC'S INVITE packet to other phones (Cisco, Yealing, zoiper etc)
and none of them understand incoming INVITE request because of WebRTC
supported protocols (ICE ,a=candidate) , in a brief, phones could not
recognize/understand incoming WebRTC request.
This is a really tough issue for me, how can I send appropriate
INVITEs for each of them.
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla --
www.asipto.com
www.twitter.com/miconda --
www.linkedin.com/in/miconda
Funding:
https://www.paypal.me/dcmierla