Re: [SR-Users] Kamailio as front proxy for multiple sip servers

2021-05-10 Thread Eliphas Levy Theodoro
Hello,

I now understood that to send the request directly to the socket
instead of via UDP/invalid.ip.address. Succeded by lookup() after
save(), or asterisk's, Dial(PJSIP/local_aor) if registered locally.

I settled with passing registration by domain to the right asterisk
server beneath kamailio, and save()ing on reply 200 OK. On the
asterisks, I set kamailio as outbound proxy; They then dial the
sip:aor@domain and kamailio will lookup($rd) easily.

By the way, why save("$rd") and lookup("$rd") won't substitute for
request uri domain? It is saving a literal $rd on the location table.
In this case I will need to have multiple if()s with each domain name.
Should I use another function/table for saving multidomain contacts?

Now that the signaling is doing alright, I need to figure out about
the media, starting with that config from gitub/havfo (WEBRTC-to-SIP).

Thanks,
Eliphas

Em qui., 6 de mai. de 2021 às 16:03, Yuriy Gorlichenko
 escreveu:
>
> If pjsip path doest work ( which indeed can be a case )
> It is an option for you to mascarade contact on kamailio ( if you need to 
> register phones on asterisk ), but this is not a trivial.
>
> If you do registrar on kamailio then lookup and set up proper $du for webrtc 
> endpoints will workout for you I believe.
>
> On Thu, 6 May 2021, 20:43 Eliphas Levy Theodoro,  wrote:
>>
>> As I have got 4 different answers (thanks!) I will paste them all down there.
>>
>> Em qua., 5 de mai. de 2021 às 18:44, Eliphas Levy Theodoro
>>  escreveu:
>> >
>> > I am trying to config one kamailio as reverse proxy for a bunch of 
>> > internal (no internet address) separate asterisk sip
>> > instances (per domain). The kamailio server would be the only with the 
>> > valid IP address, so would use rtpengine to
>> > force to be in the media path.
>> >
>> > Like this scenario: 
>> > https://opensips.org/pipermail/users/2020-August/043610.html
>> >
>> > I have used as starting point this very basic config:
>> > https://blog.voipxswitch.com/2015/03/27/kamailio-basic-sip-proxy-all-requests-setup/
>> >
>> > Basically just added rtpproxy support, and voilà, inter-SIP is working, 
>> > media always passing into the proxy.
>> >
>> > The problem: I would have WebRTC phones connecting too. I tried setting 
>> > WSS up in kamailio, and asterisk (pjsip)
>> > wouldn't know how to send the message to the proxy: on register it has 
>> > trasnport=wss in the contact: header, looks
>> > like it is confusing the asterisk.
>> >
>> > So, I resort for the wisdom of the list :) What would be the 
>> > good-best-path to take here, hack the header, or put the
>> > webphones registering directly on the asterisks (with a nginx reverse 
>> > proxy maybe)?
>>
>> [..]
>>
>> Daniel-Constantin Mierla mico...@gmail.com por  lists.kamailio.org
>> 06:26 (há 8 horas)
>> >
>> > if both endpoints can do webrtc srtp, then it works with rtpproxy to do 
>> > srtp packet forwarding for nat traversal or networks bridging.
>>
>> Yes, when a pair of softphones (ok) and softphones (not yet) exchange
>> signalling alright in that scenario, I will start on transcoding...
>>
>>
>> Wojtko, Daniel daniel.woj...@rittec.de por  lists.kamailio.org 05:32
>> (há 8 horas)
>> > afaik rtpproxy doesn't support WebRTC but rtpengine does
>>
>> As Daniel said above, I reckon that rtpproxy would work when
>> transcoding/translating sip/webrtc is not needed. But first, need to
>> pass signalling at least :)
>>
>>
>> Yuriy Gorlichenko ovoshl...@gmail.com por  lists.kamailio.org 05:55 (há 8 
>> horas)
>> >
>> > If you looking for examples: you can use this one
>> > https://github.com/havfo/WEBRTC-to-SIP as starting point
>> >
>> > anyway, the Path mentioned by Alex is the best approach.
>>
>> I tried that one but could not figure most of it out... I think I
>> borked it. Tried only changing $du to asterisk instead of doing
>> register locally and got the same results (and lots of rtpengine
>> chattiness) too, so I am using now a very simple config to make
>> finding the signalling problem easier.
>>
>>
>> > Alex Balashov abalas...@evaristesys.com por  lists.kamailio.org 03:26 (há 
>> > 10 horas)
>> > It sounds like you are in need of the Path extension:
>>
>> That was one of the modifications I have made, found out later that
>> the problem is PJSIP not handling Path: anyway:
>> https://community.asterisk.org/t/pjsip-path-module-issues/88046
>> https://issues.asterisk.org/jira/browse/ASTERISK-28211
>> So I have changed back to the older chan_sip interface, problem
>> solved, that one worked with Path: header. Now asterisk sends the
>> invite back to kamailio!
>>
>> Now, the basic signalling of webphone -> kamailio -> asterisk ->
>> kamailio -> otherphone is stopping on kamailio itself, it is sending
>> the packet via UDP like asterisk was, instead of using the socket.
>>
>> This is how the webphone contact looks like:
>> 
>> Kamailio (and asterisk before Path: worked) invites
>> UDP:192.0.2.210:5060, instead of the "local" websocket, and of cour

Re: [SR-Users] Kamailio as front proxy for multiple sip servers

2021-05-06 Thread Yuriy Gorlichenko
If pjsip path doest work ( which indeed can be a case )
It is an option for you to mascarade contact on kamailio ( if you need to
register phones on asterisk ), but this is not a trivial.

If you do registrar on kamailio then lookup and set up proper $du for
webrtc endpoints will workout for you I believe.

On Thu, 6 May 2021, 20:43 Eliphas Levy Theodoro,  wrote:

> As I have got 4 different answers (thanks!) I will paste them all down
> there.
>
> Em qua., 5 de mai. de 2021 às 18:44, Eliphas Levy Theodoro
>  escreveu:
> >
> > I am trying to config one kamailio as reverse proxy for a bunch of
> internal (no internet address) separate asterisk sip
> > instances (per domain). The kamailio server would be the only with the
> valid IP address, so would use rtpengine to
> > force to be in the media path.
> >
> > Like this scenario:
> https://opensips.org/pipermail/users/2020-August/043610.html
> >
> > I have used as starting point this very basic config:
> >
> https://blog.voipxswitch.com/2015/03/27/kamailio-basic-sip-proxy-all-requests-setup/
> >
> > Basically just added rtpproxy support, and voilà, inter-SIP is working,
> media always passing into the proxy.
> >
> > The problem: I would have WebRTC phones connecting too. I tried setting
> WSS up in kamailio, and asterisk (pjsip)
> > wouldn't know how to send the message to the proxy: on register it has
> trasnport=wss in the contact: header, looks
> > like it is confusing the asterisk.
> >
> > So, I resort for the wisdom of the list :) What would be the
> good-best-path to take here, hack the header, or put the
> > webphones registering directly on the asterisks (with a nginx reverse
> proxy maybe)?
>
> [..]
>
> Daniel-Constantin Mierla mico...@gmail.com por  lists.kamailio.org
> 06:26 (há 8 horas)
> >
> > if both endpoints can do webrtc srtp, then it works with rtpproxy to do
> srtp packet forwarding for nat traversal or networks bridging.
>
> Yes, when a pair of softphones (ok) and softphones (not yet) exchange
> signalling alright in that scenario, I will start on transcoding...
>
>
> Wojtko, Daniel daniel.woj...@rittec.de por  lists.kamailio.org 05:32
> (há 8 horas)
> > afaik rtpproxy doesn't support WebRTC but rtpengine does
>
> As Daniel said above, I reckon that rtpproxy would work when
> transcoding/translating sip/webrtc is not needed. But first, need to
> pass signalling at least :)
>
>
> Yuriy Gorlichenko ovoshl...@gmail.com por  lists.kamailio.org 05:55 (há 8
> horas)
> >
> > If you looking for examples: you can use this one
> > https://github.com/havfo/WEBRTC-to-SIP as starting point
> >
> > anyway, the Path mentioned by Alex is the best approach.
>
> I tried that one but could not figure most of it out... I think I
> borked it. Tried only changing $du to asterisk instead of doing
> register locally and got the same results (and lots of rtpengine
> chattiness) too, so I am using now a very simple config to make
> finding the signalling problem easier.
>
>
> > Alex Balashov abalas...@evaristesys.com por  lists.kamailio.org 03:26
> (há 10 horas)
> > It sounds like you are in need of the Path extension:
>
> That was one of the modifications I have made, found out later that
> the problem is PJSIP not handling Path: anyway:
> https://community.asterisk.org/t/pjsip-path-module-issues/88046
> https://issues.asterisk.org/jira/browse/ASTERISK-28211
> So I have changed back to the older chan_sip interface, problem
> solved, that one worked with Path: header. Now asterisk sends the
> invite back to kamailio!
>
> Now, the basic signalling of webphone -> kamailio -> asterisk ->
> kamailio -> otherphone is stopping on kamailio itself, it is sending
> the packet via UDP like asterisk was, instead of using the socket.
>
> This is how the webphone contact looks like:
> 
> Kamailio (and asterisk before Path: worked) invites
> UDP:192.0.2.210:5060, instead of the "local" websocket, and of course
> never succeeding.
>
> I tried save()ing the register locally, but I am sure I am doing it wrong.
>
> if someone wants to look at the actual test config, I pasted it:
> https://pastebin.com/RuXniDTU
>
> Cheers,
> --
> Eliphas
>
> __
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Re: [SR-Users] Kamailio as front proxy for multiple sip servers

2021-05-06 Thread Eliphas Levy Theodoro
As I have got 4 different answers (thanks!) I will paste them all down there.

Em qua., 5 de mai. de 2021 às 18:44, Eliphas Levy Theodoro
 escreveu:
>
> I am trying to config one kamailio as reverse proxy for a bunch of internal 
> (no internet address) separate asterisk sip
> instances (per domain). The kamailio server would be the only with the valid 
> IP address, so would use rtpengine to
> force to be in the media path.
>
> Like this scenario: 
> https://opensips.org/pipermail/users/2020-August/043610.html
>
> I have used as starting point this very basic config:
> https://blog.voipxswitch.com/2015/03/27/kamailio-basic-sip-proxy-all-requests-setup/
>
> Basically just added rtpproxy support, and voilà, inter-SIP is working, media 
> always passing into the proxy.
>
> The problem: I would have WebRTC phones connecting too. I tried setting WSS 
> up in kamailio, and asterisk (pjsip)
> wouldn't know how to send the message to the proxy: on register it has 
> trasnport=wss in the contact: header, looks
> like it is confusing the asterisk.
>
> So, I resort for the wisdom of the list :) What would be the good-best-path 
> to take here, hack the header, or put the
> webphones registering directly on the asterisks (with a nginx reverse proxy 
> maybe)?

[..]

Daniel-Constantin Mierla mico...@gmail.com por  lists.kamailio.org
06:26 (há 8 horas)
>
> if both endpoints can do webrtc srtp, then it works with rtpproxy to do srtp 
> packet forwarding for nat traversal or networks bridging.

Yes, when a pair of softphones (ok) and softphones (not yet) exchange
signalling alright in that scenario, I will start on transcoding...


Wojtko, Daniel daniel.woj...@rittec.de por  lists.kamailio.org 05:32
(há 8 horas)
> afaik rtpproxy doesn't support WebRTC but rtpengine does

As Daniel said above, I reckon that rtpproxy would work when
transcoding/translating sip/webrtc is not needed. But first, need to
pass signalling at least :)


Yuriy Gorlichenko ovoshl...@gmail.com por  lists.kamailio.org 05:55 (há 8 horas)
>
> If you looking for examples: you can use this one
> https://github.com/havfo/WEBRTC-to-SIP as starting point
>
> anyway, the Path mentioned by Alex is the best approach.

I tried that one but could not figure most of it out... I think I
borked it. Tried only changing $du to asterisk instead of doing
register locally and got the same results (and lots of rtpengine
chattiness) too, so I am using now a very simple config to make
finding the signalling problem easier.


> Alex Balashov abalas...@evaristesys.com por  lists.kamailio.org 03:26 (há 10 
> horas)
> It sounds like you are in need of the Path extension:

That was one of the modifications I have made, found out later that
the problem is PJSIP not handling Path: anyway:
https://community.asterisk.org/t/pjsip-path-module-issues/88046
https://issues.asterisk.org/jira/browse/ASTERISK-28211
So I have changed back to the older chan_sip interface, problem
solved, that one worked with Path: header. Now asterisk sends the
invite back to kamailio!

Now, the basic signalling of webphone -> kamailio -> asterisk ->
kamailio -> otherphone is stopping on kamailio itself, it is sending
the packet via UDP like asterisk was, instead of using the socket.

This is how the webphone contact looks like:

Kamailio (and asterisk before Path: worked) invites
UDP:192.0.2.210:5060, instead of the "local" websocket, and of course
never succeeding.

I tried save()ing the register locally, but I am sure I am doing it wrong.

if someone wants to look at the actual test config, I pasted it:
https://pastebin.com/RuXniDTU

Cheers,
-- 
Eliphas

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Re: [SR-Users] Kamailio as front proxy for multiple sip servers

2021-05-06 Thread Daniel-Constantin Mierla
Hello,


if both endpoints can do webrtc srtp, then it works with rtpproxy to do
srtp packet forwarding for nat traversal or networks bridging.


Cheers,
Daniel


On 06.05.21 10:31, Wojtko, Daniel wrote:
>
> ​Hi, 
>
> afaik rtpproxy doesn't support WebRTC but rtpengine does
>
>
> Regards
>
>
> Daniel
>
> 
> *Von:* sr-users  im Auftrag von
> Eliphas Levy Theodoro 
> *Gesendet:* Mittwoch, 5. Mai 2021 23:44
> *An:* sr-users@lists.kamailio.org
> *Betreff:* [SR-Users] Kamailio as front proxy for multiple sip servers
>  
> Hello!
>
> I am trying to config one kamailio as reverse proxy for a bunch of
> internal (no internet address) separate asterisk sip instances (per
> domain). The kamailio server would be the only with the valid IP
> address, so would use rtpengine to force to be in the media path.
>
> Like this scenario:
> https://opensips.org/pipermail/users/2020-August/043610.html
> <https://opensips.org/pipermail/users/2020-August/043610.html>
>
> I have used as starting point this very basic config:
> 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/>
>
> Basically just added rtpproxy support, and voilà, inter-SIP is
> working, media always passing into the proxy.
>
> The problem: I would have WebRTC phones connecting too. I tried
> setting WSS up in kamailio, and asterisk (pjsip) wouldn't know how to
> send the message to the proxy: on register it has trasnport=wss in the
> contact: header, looks like it is confusing the asterisk.
>
> So, I resort for the wisdom of the list :) What would be the
> good-best-path to take here, hack the header, or put the webphones
> registering directly on the asterisks (with a nginx reverse proxy maybe)?
>
> Someone must have already made a blog with such setup, but I could not
> google-unearth it at least until now.
>
> Regards,
> -- 
> Eliphas
>
> __
> Kamailio - Users Mailing List - Non Commercial Discussions
>   * sr-users@lists.kamailio.org
> Important: keep the mailing list in the recipients, do not reply only to the 
> sender!
> Edit mailing list options or unsubscribe:
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-- 
Daniel-Constantin Mierla -- www.asipto.com
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio Advanced Training - Online
May 17-20, 2021 (Europe Timezone) - June 7-10, 2021 (America Timezone)
  * https://www.asipto.com/sw/kamailio-advanced-training-online/

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Re: [SR-Users] Kamailio as front proxy for multiple sip servers

2021-05-06 Thread Yuriy Gorlichenko
If you looking for examples: you can use this one
https://github.com/havfo/WEBRTC-to-SIP as starting point

anyway, the Path mentioned by Alex is the best approach.

чт, 6 мая 2021 г. в 10:34, Wojtko, Daniel :

> ​Hi,
>
> afaik rtpproxy doesn't support WebRTC but rtpengine does
>
>
> Regards
>
>
> Daniel
> --
> *Von:* sr-users  im Auftrag von
> Eliphas Levy Theodoro 
> *Gesendet:* Mittwoch, 5. Mai 2021 23:44
> *An:* sr-users@lists.kamailio.org
> *Betreff:* [SR-Users] Kamailio as front proxy for multiple sip servers
>
> Hello!
>
> I am trying to config one kamailio as reverse proxy for a bunch of
> internal (no internet address) separate asterisk sip instances (per
> domain). The kamailio server would be the only with the valid IP address,
> so would use rtpengine to force to be in the media path.
>
> Like this scenario:
> https://opensips.org/pipermail/users/2020-August/043610.html
>
> I have used as starting point this very basic config:
>
> https://blog.voipxswitch.com/2015/03/27/kamailio-basic-sip-proxy-all-requests-setup/
>
> Basically just added rtpproxy support, and voilà, inter-SIP is
> working, media always passing into the proxy.
>
> The problem: I would have WebRTC phones connecting too. I tried setting
> WSS up in kamailio, and asterisk (pjsip) wouldn't know how to send the
> message to the proxy: on register it has trasnport=wss in the contact:
> header, looks like it is confusing the asterisk.
>
> So, I resort for the wisdom of the list :) What would be the
> good-best-path to take here, hack the header, or put the webphones
> registering directly on the asterisks (with a nginx reverse proxy maybe)?
>
> Someone must have already made a blog with such setup, but I could not
> google-unearth it at least until now.
>
> Regards,
> --
> Eliphas
> __
> Kamailio - Users Mailing List - Non Commercial Discussions
>   * sr-users@lists.kamailio.org
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> Edit mailing list options or unsubscribe:
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Re: [SR-Users] Kamailio as front proxy for multiple sip servers

2021-05-06 Thread Wojtko, Daniel
?Hi,

afaik rtpproxy doesn't support WebRTC but rtpengine does


Regards


Daniel


Von: sr-users  im Auftrag von Eliphas Levy 
Theodoro 
Gesendet: Mittwoch, 5. Mai 2021 23:44
An: sr-users@lists.kamailio.org
Betreff: [SR-Users] Kamailio as front proxy for multiple sip servers

Hello!

I am trying to config one kamailio as reverse proxy for a bunch of internal (no 
internet address) separate asterisk sip instances (per domain). The kamailio 
server would be the only with the valid IP address, so would use rtpengine to 
force to be in the media path.

Like this scenario: https://opensips.org/pipermail/users/2020-August/043610.html

I have used as starting point this very basic config:
https://blog.voipxswitch.com/2015/03/27/kamailio-basic-sip-proxy-all-requests-setup/

Basically just added rtpproxy support, and voilà, inter-SIP is working, media 
always passing into the proxy.

The problem: I would have WebRTC phones connecting too. I tried setting WSS up 
in kamailio, and asterisk (pjsip) wouldn't know how to send the message to the 
proxy: on register it has trasnport=wss in the contact: header, looks like it 
is confusing the asterisk.

So, I resort for the wisdom of the list :) What would be the good-best-path to 
take here, hack the header, or put the webphones registering directly on the 
asterisks (with a nginx reverse proxy maybe)?

Someone must have already made a blog with such setup, but I could not 
google-unearth it at least until now.

Regards,
--
Eliphas
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Re: [SR-Users] Kamailio as front proxy for multiple sip servers

2021-05-05 Thread Alex Balashov

It sounds like you are in need of the Path extension:

https://tools.ietf.org/html/rfc3327

https://kamailio.org/docs/modules/stable/modules/path.html

-- Alex

On 5/5/21 5:44 PM, Eliphas Levy Theodoro wrote:


Hello!

I am trying to config one kamailio as reverse proxy for a bunch of 
internal (no internet address) separate asterisk sip instances (per 
domain). The kamailio server would be the only with the valid IP 
address, so would use rtpengine to force to be in the media path.


Like this scenario: 
https://opensips.org/pipermail/users/2020-August/043610.html 



I have used as starting point this very basic config:
https://blog.voipxswitch.com/2015/03/27/kamailio-basic-sip-proxy-all-requests-setup/ 



Basically just added rtpproxy support, and voilà, inter-SIP is 
working, media always passing into the proxy.


The problem: I would have WebRTC phones connecting too. I tried setting 
WSS up in kamailio, and asterisk (pjsip) wouldn't know how to send the 
message to the proxy: on register it has trasnport=wss in the contact: 
header, looks like it is confusing the asterisk.


So, I resort for the wisdom of the list :) What would be the 
good-best-path to take here, hack the header, or put the webphones 
registering directly on the asterisks (with a nginx reverse proxy maybe)?


Someone must have already made a blog with such setup, but I could not 
google-unearth it at least until now.


Regards,
--
Eliphas

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--
Alex Balashov | Principal | Evariste Systems LLC

Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

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[SR-Users] Kamailio as front proxy for multiple sip servers

2021-05-05 Thread Eliphas Levy Theodoro
Hello!

I am trying to config one kamailio as reverse proxy for a bunch of internal
(no internet address) separate asterisk sip instances (per domain). The
kamailio server would be the only with the valid IP address, so would use
rtpengine to force to be in the media path.

Like this scenario:
https://opensips.org/pipermail/users/2020-August/043610.html

I have used as starting point this very basic config:
https://blog.voipxswitch.com/2015/03/27/kamailio-basic-sip-proxy-all-requests-setup/

Basically just added rtpproxy support, and voilà, inter-SIP is
working, media always passing into the proxy.

The problem: I would have WebRTC phones connecting too. I tried setting WSS
up in kamailio, and asterisk (pjsip) wouldn't know how to send the message
to the proxy: on register it has trasnport=wss in the contact: header,
looks like it is confusing the asterisk.

So, I resort for the wisdom of the list :) What would be the good-best-path
to take here, hack the header, or put the webphones registering directly on
the asterisks (with a nginx reverse proxy maybe)?

Someone must have already made a blog with such setup, but I could not
google-unearth it at least until now.

Regards,
-- 
Eliphas
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