<div dir="ltr">webrtc kamailio for example here <a href="https://github.com/havfo/WEBRTC-to-SIP">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 class="gmail_extra"><br><div class="gmail_quote">2017-06-07 14:47 GMT+03:00 Karsten Horsmann <span dir="ltr"><<a href="mailto:khorsmann@gmail.com" target="_blank">khorsmann@gmail.com</a>></span>:<br><blockquote class="gmail_quote" 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 class="HOEnZb"><font color="#888888"><div><div><br></div>-- <br><div class="m_2473207808299382131gmail_signature" 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 class="gmail_signature" data-smartmail="gmail_signature"><div dir="ltr"><div><div dir="ltr"><div>Savolainen Dmitri</div></div></div></div></div>
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