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 module2017-06-07 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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