<HTML><BODY><p><br>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.</p><p>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.</p><blockquote style="border-left:1px solid #0857A6; margin:10px; padding:0 0 0 10px;">
Среда, 7 июня 2017, 21:18 +03:00 от Dmitri Savolainen <savolainen@erinaco.ru>:<br>
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<div id="style_14968594970000000016_BODY"><div dir="ltr">webrtc kamailio for example here <a href="https://github.com/havfo/WEBRTC-to-SIP" target="_blank">https://github.com/havfo/WEBRTC-to-SIP</a><div><br></div><div>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</div></div><div><br><div>2017-06-07 14:47 GMT+03:00 Karsten Horsmann <span dir="ltr"><<a href="mailto:khorsmann@gmail.com">khorsmann@gmail.com</a>></span>:<br><blockquote style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Hello List,<div><br></div><div><br></div><div>is there any howto about webrtc loadbalance in combination with kamailio and FreeSWITCH?</div><div><br></div><div>I want to share one WSS address/endpoint to multiple FreeSWITCH backends.<br></div><div>Or is there any other best practice?</div><div><br></div><div>My callflow is mostly that my internal SIP Servers called my registered webrtc clients.</div><div><br></div><div>Would be nice to get some input.</div><span><font color="#888888"><div><div><br></div>-- <br><div data-smartmail="gmail_signature">Kind Regards<br>*Karsten Horsmann*<br></div>
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<br></blockquote></div><br><br clear="all"><div><br></div>-- <br><div data-smartmail="gmail_signature"><div dir="ltr"><div><div dir="ltr"><div>Savolainen Dmitri</div></div></div></div></div>
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