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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