[OpenSIPS-Users] Opensips as SIP Proxy and WebRTC Media Gateway
Bogdan-Andrei Iancu
bogdan at opensips.org
Fri Jun 30 10:20:05 EDT 2017
Good, there is some progress :).
On the incoming calls, if the WS get's the call, we can park the part
with the auth (it seems your opensips script is accepting calls from
unknown sources...we can address this security hole later.
So, if a call drop after 30 secs it usually means there is no ACK. Can
you make a mgrep capture on OpenSIPS to grab the whole call flow ? (grab
5060 and 80 ports)
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 04:52 PM, Alex Megalokonomos wrote:
> I think I set up uac_registrant correctly.
>
> I can dial out from a ws client and the ws extension rings from
> outside calls.
>
> However:
> a) on incoming calls, when ws client accepts, there is no sound and
> the line is dropped after 30 secs or so
> b) on outgoing calls, when the called extension accepts the ws client
> immediately responds with 401 Unauthorised and then BYE
>
> b) I believe is what you mentioned here "In OpenSIPS, when receiving
> calls, you need to authorize (by IP) the calls from OmniPCX "
>
> How do I do this?
>
> and a) seems to be rtp proxy related since I see the following errors
> in the logs¨
>
> ERROR:rtpengine:rtpe_function_call: proxy replied with error: Unknown
> call-id
>
> and
>
> no matching transaction
>
> On Fri, Jun 30, 2017 at 2:27 PM, Bogdan-Andrei Iancu
> <bogdan at opensips.org <mailto:bogdan at opensips.org>> wrote:
>
> 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/
> <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/
> <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
> <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 <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 06/30/2017 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/
>> <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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