[OpenSIPS-Users] Opensips as SIP Proxy and WebRTC Media Gateway
Bogdan-Andrei Iancu
bogdan at opensips.org
Fri Jun 30 07:27:46 EDT 2017
I checked the script you mentioned and it does not help you - it has
only UDP (no WS), it is really basic and it does not handle any REGISTER
stuff, which is the trickiest - see
https://blog.opensips.org/2016/12/13/how-to-proxy-sip-registrations/
or
https://blog.opensips.org/2016/12/20/mid-registrar-scalable-registration-and-call-forking/
Maybe you can start with handling REGISTERs - what you need (on top of
the script from the WSS tutorial) is to add this uac_registrant, to have
the WS extensions registered into OmniPCX with a contact URI pointing
back to OpenSIPS IP:
http://www.opensips.org/html/docs/modules/2.3.x/uac_registrant.html
Let me know if you get stuck in this first step.
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 06/30/2017 12:22 PM, Alex Megalokonomos wrote:
> Hello Bogdan,
>
> First of all, thanks for your time.
>
> Unfortunately my SIP/OpensSIPS skills are what I've managed to learn
> in the last couple of days. I am a programmer but I've never had to
> work on SIP stuff before.
>
> Frankly to me, both solutions sound equally difficult since I have no
> idea where to start. (And to be honest, I expected the first to be
> simpler)
>
> I found this
> https://blog.voipxswitch.com/2015/03/27/kamailio-basic-sip-proxy-all-requests-setup/
> and tried to port the config to OpenSIPS since from what I understand
> Kamailio and OpenSIPS share a common codebase to an extent but was
> unsuccesful.
>
> In your second scenario, I am not interested in WS->WS calls so that
> auth part is not an issue.
>
> So I guess I need the uac_registrar, authorize by IP and usrloc parts.
>
> Any relevant documentation to get me started since I'm still not clear
> on what I need to change?
>
> Best regards,
> Alex
>
> On Fri, Jun 30, 2017 at 11:29 AM, Bogdan-Andrei Iancu
> <bogdan at opensips.org <mailto:bogdan at opensips.org>> wrote:
>
> 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
> 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 06/29/2017 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
>> <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>
>>
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