[OpenSIPS-Users] Opensips as SIP Proxy and WebRTC Media Gateway
Alex Megalokonomos
alex at clockwork.gr
Tue Jul 4 11:22:04 EDT 2017
As you may have noticed in my last reply, I reached that far as well but
got stuck later on on what appears to be the rtp engine configuration.
Not strictly an Opensips issue but you might be able to help me.
On Tue, Jul 4, 2017 at 6:07 PM, Bogdan-Andrei Iancu <bogdan at opensips.org>
wrote:
> Hi Alex,
>
> Thank you for the offlist provided data. Shortly, the ACK received by
> OpenSIPS from OmniPCX is broken as it is missing all the Route headers.
> According to the pcap, it looks like:
>
> ACK sip:udoioiia at 10.0.1.106:49246;transport=ws SIP/2.0
> Record-Route: <sip:10.0.1.200:5059;ftag=d5de999de446df5165d773dac1f369
> ec;lr=on>
> Contact: "Megalokonomos A." <sip:694 at 10.0.1.200:45698>
> User-Agent: OxO_SPG_103/012.001
> Content-Type: application/sdp
> To: sip:694 at 10.0.1.200;tag=4em4m1ah9r
> From: "Megalokonomos A." <sip:610 at 10.0.1.200>;tag=
> d5de999de446df5165d773dac1f369ec
> Call-ID: af3cc9085db1c8dd86050eb91d747249 at 10.0.1.200
> CSeq: 659214613 ACK
> Via: SIP/2.0/UDP 10.0.1.200:5059;branch=z9hG4bKf3de.
> 2fc1fc65cece765af47f9baf8bf0906e.0;i=c
> Via: SIP/2.0/TCP 10.0.1.200:5080;rport=45698;branch=
> z9hG4bK89fca3417cd4e227b4315145d96657c7
> Max-Forwards: 69
> Content-Length: 2960
>
> v=0
> o=default 14
> .....
>
>
> As OpenSIPS does not find the Route (former Record-Route) it inserted into
> the dialog, the routing logic in the script does not work as expected.
> According to RFC3261, the RR headers MUST be mirrored back in 2xx replies.
>
> Let's try to hack to cope with the broken SIP stack on OmniPCX. In script
> you have something like:
>
> } else {
> # ACK without matching transaction ->
> # ignore and discard
> exit;
> }
>
> Try replacing it with
>
> } else {
> # ACK without matching transaction ->
> # ignore and discard
> t_relay();
> exit;
> }
>
> Let's see if this does the trick. If yes, I can suggest a even better way to fix the broken signaling, using the dialog support in OpenSIPS.
>
> Best 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 07/03/2017 01:08 PM, Bogdan-Andrei Iancu wrote:
>
> Hi Alex,
>
> As suspected, the ACK is not properly routed - see the retransmissions of
> the 200OK + ACK. SImply based on the logs I cannot see what the problem is
> - probably some missing fix_nated_contact() for the replies coming from the
> WS party.
>
> Please make a pcap capture + opensips log (level 4) and send them to me
> *offlist* !
>
> Best 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 05:37 PM, Alex Megalokonomos wrote:
>
> I have attached the debug log so you get a fuller picture. I hope that's
> ok
>
> (Incoming call to WS client 694 is the WS extension...610 is my normal
> desk phone which is connected to OmniPCX) (10.0.1.63-> OpenSIPS
> ,10.0.1.200-> OmniPCX)
>
>
>
> On Fri, Jun 30, 2017 at 5:20 PM, Bogdan-Andrei Iancu <
> <bogdan at opensips.org>bogdan at opensips.org> wrote:
>
>> 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
>> > 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/ 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> 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
>>>>
>>>> OpenSIPS Bootcamp 2017, Houston, US
>>>> 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.o
>>>> rg/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
>>>>> 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
>>>>>
>>>>> _______________________________________________
> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
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