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>:


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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