rtpengine just a proxy. You can use kamailio just a webrtc proxy to freeswitch if want to use FS as backend server that will handle voice and convert it from SRTP to RTP. websocket just a transport like TCP,UDP and TLS, so you also can send SIP over websocket from kamailio using for example $fs valriable for it. You will need configure needed proto:ip:port to freeswitch for using websocket in dispatcher. Среда, 7 июня 2017, 21:18 +03:00 от Dmitri Savolainen savolainen@erinaco.ru:

webrtc kamailio  for example here  https://github.com/havfo/WEBRTC-to-SIP

By the way rtpengine is not mandatory with FreeSwitch. It is possible to use a set of FS(1.6) and balancing by dispatcher module

2017-06-07 14:47 GMT+03:00 Karsten Horsmann < khorsmann@gmail.com > :

...

Hello List,

is there any howto about webrtc loadbalance in combination with kamailio and FreeSWITCH?

I want to share one WSS address/endpoint to multiple FreeSWITCH backends. Or is there any other best practice?

My callflow is mostly that my internal SIP Servers called my registered webrtc clients.

Would be nice to get some input.

-- Kind Regards *Karsten Horsmann*

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-- Savolainen Dmitri _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

Hello,

On 18.06.17 10:25, Karsten Horsmann wrote:

Hi,

RTPengine is very Debian depending and I want to avoid running it.

if you refer to the kernel module, you can run without it, having rtpengine in the user space like tpproxy. Mayb be slower, but it works. Not sure about dependencies to compile it, but I expect to be everywhere in a main linux distro.

Cheers, Daniel

I tried to run RTP engine on my CentOS 6 SBCs. But then I realized that is an too old starting point. My Freeswitch webrtc systems run CentOS 7 and in my setup I could handel websocket and srtp stuff.

But in that setup I am limited to one box and here comes Kamailio for the rescue.

Thanks for the links I will read them and try to figure out what kind of setup I can run.

First step is to setup two Freeswitch webrtc boxes (with public ips) and loadblance webrtc signaling to them via Kamailio (hmm to plain SIP then or plain WS).

In this first step it should be only for signaling.

Am 08.06.2017 7:58 vorm. schrieb "Юрий Горличенко" <gorlichenko_uv@inbox.ru mailto:gorlichenko_uv@inbox.ru>:

rtpengine just a proxy. You can use kamailio just a webrtc proxy
to freeswitch if want to use FS as backend server that will handle
voice and convert it from SRTP to RTP.

websocket just a transport like TCP,UDP and TLS, so you also can
send SIP over websocket from kamailio using for example $fs
valriable for it. You will need configure needed proto:ip:port to
freeswitch for using websocket in dispatcher.

    Среда, 7 июня 2017, 21:18 +03:00 от Dmitri Savolainen
    <savolainen@erinaco.ru <mailto:savolainen@erinaco.ru>>:


    webrtc kamailio  for example
    here https://github.com/havfo/WEBRTC-to-SIP

    By the way rtpengine is not mandatory with FreeSwitch. It is
    possible to use a set of FS(1.6) and balancing by dispatcher
    module

    2017-06-07 14:47 GMT+03:00 Karsten Horsmann <khorsmann@gmail.com>:

        Hello List,


        is there any howto about webrtc loadbalance in combination
        with kamailio and FreeSWITCH?

        I want to share one WSS address/endpoint to multiple
        FreeSWITCH backends.
        Or is there any other best practice?

        My callflow is mostly that my internal SIP Servers called
        my registered webrtc clients.

        Would be nice to get some input.

        -- 
        Kind Regards
        *Karsten Horsmann*

        _______________________________________________
        Kamailio (SER) - Users Mailing List
        sr-users@lists.kamailio.org
        https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users




    -- 
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    _______________________________________________
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    sr-users@lists.kamailio.org
    https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users



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Hello,

that combination of dispatcher+websocket+xhttp modules works just fine...

So to load balance the SIP signaling with Kamailio towards FreeSwitch, just use the dispatcher module as usual. A sample config is available at:

- https://www.kamailio.org/docs/modules/stable/modules/dispatcher.html#dispatc...

You need to add the support for websocket traffic via websocket module:

- https://www.kamailio.org/docs/modules/stable/modules/websocket.html#idp42826...

or extract from the tutorial linked in a previous email on this thread.

Cheers, Daniel

On 13.06.17 16:38, Joel Serrano wrote:

Hi Karsten,

Have you tried with regular Kamailio (w/ dispatcher+websocket+xhttp modules)? I don't see why it wouldn't work...

Joel.

On Wed, Jun 7, 2017 at 4:47 AM, Karsten Horsmann <khorsmann@gmail.com mailto:khorsmann@gmail.com> wrote:

Hello List,


is there any howto about webrtc loadbalance in combination with
kamailio and FreeSWITCH?

I want to share one WSS address/endpoint to multiple FreeSWITCH
backends.
Or is there any other best practice?

My callflow is mostly that my internal SIP Servers called my
registered webrtc clients.

Would be nice to get some input.

-- 
Kind Regards
*Karsten Horsmann*

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