[OpenSIPS-Users] Opensips as SIP Proxy and WebRTC Media Gateway

Bogdan-Andrei Iancu bogdan at opensips.org
Thu Jun 29 04:43:48 EDT 2017

Hi Alex,

First, some questions regarding the desired topology:
     1) the WS end-points should register in OpenSIPS or all the way 
into Kamailio ?
     2) also, the calls from the WS end-points should be all the time 
sent to Kamailio ?

More or less, what I'm asking is : is OpenSIPS suppose to act as a 
gateway from WS to UDP , but pass all the resulting traffic to Kamailio ?


Bogdan-Andrei Iancu
   OpenSIPS Founder and Developer

OpenSIPS Bootcamp 2017, Houston, US

On 06/28/2017 12:47 PM, Alex Megalokonomos wrote:
> Hello,
> We have the following scenario: our office call center is an Alcatel 
> OmniPCX Office setup.
> This handles most of our needs and also provides 4 SIP extensions.
> These are provided by what appears to be a Kamailio SIP server v 3.2.2 
> (no webrtc or websockets support)
> What we would like to do is set up an OpenSIPS instance to handle 
> WebRTC and proxy everything to this Kamailio SIP server.
> The idea is to allow a web client (using sip js or something similar) 
> to register / make / receive calls as one of the Kamailio extensions.
> I think half of the configuration is this : 
> http://www.opensips.org/Documentation/Tutorials-WebSocket-2-1
> which I've already completed and indeed, clients can register to 
> opensips and chat/make calls over websockets between them.
> How do I go about proxying registrations/invites/etc to the kamailio 
> server instead?
> best regards
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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