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

Yuriy Gorlichenko ovoshlook at gmail.com
Thu May 6 21:02:50 CEST 2021


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, <eliphas at gmail.com> 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
> <eliphas at gmail.com> 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 miconda at 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.wojtko at 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 ovoshlook at 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 abalashov at 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:
> <sip:cakrtk0i at 192.0.2.210;transport=wss>
> 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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