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

Bogdan-Andrei Iancu bogdan at opensips.org
Fri Jun 30 04:29:26 EDT 2017

Hi Alex,

To make a kind of WS<>UDP gateway you need a complete rework of the 
script presented in the tutorial, as it is a completely different SIP 
scenario. Not sure what are your SIP/OpenSIPS skills.

But, there is a simpler alternative . Instead of a GW, you can make 
OpenSIPS as a sub-server for the WS extensions:

Registration handling:

1) WS extensions register only with OpenSIPS (as right now) - 
authentication is done by OpenSIPS
2) OpenSIPS registers the 3 extensions into OmniPCX using the uac_registrar

By this, we simply add the uac_registration and you achieve kind of 
decoupled 2 steps registration (with a minimum change in the cfg)

Inbound calls:

1) OmniPCX will send all the calls (from other extensions) for the WS 
extension to OpenSIPS (due the registration via uac_registrar) - this is 
default behavior , so nothing to change
2) In OpenSIPS, when receiving calls, you need to authorize (by IP) the 
calls from OmniPCX - and as the current script does, you will handle 
them via the local opensips usrloc -> calls are sent to WS extension

Outbound calls:

1) when you receive a call from a WS extension, you have to check if the 
call is for a local extension (on opensips) or for an extension in OmniPCX
2) if call is local (WS to WS) you will do authentication for the call
3) if the call is to be sent to OmniPCX, simply send the call to OmniPCX 
without auth - the auth will be done by OmniPCX as for any other extension

Hopefully this will work for you :)

Best regards,

Bogdan-Andrei Iancu
   OpenSIPS Founder and Developer

OpenSIPS Bootcamp 2017, Houston, US

On 06/29/2017 11:54 AM, Alex Megalokonomos wrote:
> 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 what appears to be an outdated customised  Kamailio 3.2.2 from what I 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 <mailto: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 <http://www.opensips-solutions.com>
>     OpenSIPS Bootcamp 2017, Houston, US
>        http://opensips.org/training/OpenSIPS_Bootcamp_2017.html
>     <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 list
>>     Users at lists.opensips.org <mailto:Users at lists.opensips.org>
>>     http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>     <http://lists.opensips.org/cgi-bin/mailman/listinfo/users>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.opensips.org/pipermail/users/attachments/20170630/4ca4240e/attachment-0001.html>

More information about the Users mailing list