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<p>Hi, <br>
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<p>afaik rtpproxy doesn't support WebRTC but rtpengine does<br>
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<p>Regards<br>
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<p>Daniel<br>
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<div id="divRplyFwdMsg" dir="ltr"><font face="Calibri, sans-serif" color="#000000" style="font-size:11pt"><b>Von:</b> sr-users <sr-users-bounces@lists.kamailio.org> im Auftrag von Eliphas Levy Theodoro <eliphas@gmail.com><br>
<b>Gesendet:</b> Mittwoch, 5. Mai 2021 23:44<br>
<b>An:</b> sr-users@lists.kamailio.org<br>
<b>Betreff:</b> [SR-Users] Kamailio as front proxy for multiple sip servers</font>
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<div dir="ltr">Hello!
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<div>I am trying to config one kamailio as reverse proxy for a bunch of internal (no internet address) separate asterisk sip instances (per domain). The kamailio server would be the only with the valid IP address, so would use rtpengine to force to be in the
media path.</div>
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<div>Like this scenario: <a href="https://opensips.org/pipermail/users/2020-August/043610.html">
https://opensips.org/pipermail/users/2020-August/043610.html</a><br>
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<div>I have used as starting point this very basic config:</div>
<div><a href="https://blog.voipxswitch.com/2015/03/27/kamailio-basic-sip-proxy-all-requests-setup/">https://blog.voipxswitch.com/2015/03/27/kamailio-basic-sip-proxy-all-requests-setup/</a><br>
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<div>Basically just added rtpproxy support, and voilą, inter-SIP is working, media always passing into the proxy.</div>
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<div>The problem: I would have WebRTC phones connecting too. I tried setting WSS up in kamailio, and asterisk (pjsip) wouldn't know how to send the message to the proxy: on register it has trasnport=wss in the contact: header, looks like it is confusing the
asterisk.</div>
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<div>So, I resort for the wisdom of the list :) What would be the good-best-path to take here, hack the header, or put the webphones registering directly on the asterisks (with a nginx reverse proxy maybe)?</div>
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<div>Someone must have already made a blog with such setup, but I could not google-unearth it at least until now.</div>
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<div>Regards,</div>
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<div dir="ltr">Eliphas</div>
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