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.

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