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

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