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

[Alex Megalokonomos]

Hello Bogdan,

Yes, a gateway from WS to UDP (as well as DTLS-SRTP to RTP in order for it
to work) is exactly what we're looking for.

Unfortunately our Alcatel OmniPCX call center  is a proprietary system that
only allows for a limited number of SIP extensions (served from what
appears to be an outdated customised  Kamailio 3.2.2 from what I can tell
from the headers.

For our normal internal office use it all works fine.

However we have 3 customer support lines that are currently routed to 3
extensions via OmniPCX.

We want to integrate these to our custom web-based CRM and the best way for
us to do it is to use something like SIP js to handle and log calls,
identify calling parties, bring up customer details etc.

Since the kamailio version inside OmniPCX does not support ws/webrtc we are
looking to set up Opensips in exactly the way you described as a
gateway/proxy for everything in order to convert the UDP-only sip
extensions to ws+ webRTC capable ones.

I have used this tutorial
http://www.opensips.org/Documentation/Tutorials-WebSocket-2-1 to get what I
assume is half the work (for RTP proxying)  but I havent figured out the
rest yet.

Best regards,
Alex

On Thu, Jun 29, 2017 at 11:43 AM, Bogdan-Andrei Iancu <bogdan at opensips.org>
wrote:

> 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 ?
>
> Regards,
>
> Bogdan-Andrei Iancu
>   OpenSIPS Founder and Developer
>   http://www.opensips-solutions.com
>
> OpenSIPS Bootcamp 2017, Houston, US
>   http://opensips.org/training/OpenSIPS_Bootcamp_2017.html
>
> 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>
> 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 listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
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