<div dir="auto"><div>Hi, </div><div dir="auto"><br></div><div dir="auto">RTPengine is very Debian depending and I want to avoid running it.</div><div dir="auto"><br></div><div dir="auto">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. </div><div dir="auto"><br></div><div dir="auto">But in that setup I am limited to one box and here comes Kamailio for the rescue. </div><div dir="auto"><br></div><div dir="auto">Thanks for the links I will read them and try to figure out what kind of setup I can run. </div><div dir="auto"><br></div><div dir="auto">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). </div><div dir="auto"><br></div><div dir="auto">In this first step it should be only for signaling. <br><div class="gmail_extra" dir="auto"><br><div class="gmail_quote">Am 08.06.2017 7:58 vorm. schrieb "Юрий Горличенко" <<a href="mailto:gorlichenko_uv@inbox.ru">gorlichenko_uv@inbox.ru</a>>:<br type="attribution"><blockquote class="quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div><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 <<a href="mailto:savolainen@erinaco.ru">savolainen@erinaco.ru</a>>:<div class="elided-text"><br>
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<div id="m_-1192002882757016333style_14968594970000000016_BODY"><div dir="ltr">webrtc kamailio for example here <a>https://github.com/havfo/<wbr>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>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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