[SR-Users] Multiple FreeSWITCH servers behind kamailio-websocket

[Karsten Horsmann]

Hi,

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

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 at 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 at 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 at 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
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