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<p>Hello,</p>
<p><br>
</p>
<p>if both endpoints can do webrtc srtp, then it works with rtpproxy
to do srtp packet forwarding for nat traversal or networks
bridging.</p>
<p><br>
</p>
<p>Cheers,<br>
Daniel</p>
<p><br>
</p>
<div class="moz-cite-prefix">On 06.05.21 10:31, Wojtko, Daniel
wrote:<br>
</div>
<blockquote type="cite" cite="mid:1620289860202.89627@rittec.de">
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<p>Hi, <br>
</p>
<p>afaik rtpproxy doesn't support WebRTC but rtpengine does<br>
</p>
<p><br>
</p>
<p>Regards<br>
</p>
<p><br>
</p>
<p>Daniel<br>
</p>
<div style="color: rgb(33, 33, 33);">
<hr tabindex="-1" style="display:inline-block; width:98%">
<div id="divRplyFwdMsg" dir="ltr"><font style="font-size:11pt"
face="Calibri, sans-serif" color="#000000"><b>Von:</b>
sr-users <a class="moz-txt-link-rfc2396E" href="mailto:sr-users-bounces@lists.kamailio.org"><sr-users-bounces@lists.kamailio.org></a> im
Auftrag von Eliphas Levy Theodoro <a class="moz-txt-link-rfc2396E" href="mailto:eliphas@gmail.com"><eliphas@gmail.com></a><br>
<b>Gesendet:</b> Mittwoch, 5. Mai 2021 23:44<br>
<b>An:</b> <a class="moz-txt-link-abbreviated" href="mailto:sr-users@lists.kamailio.org">sr-users@lists.kamailio.org</a><br>
<b>Betreff:</b> [SR-Users] Kamailio as front proxy for
multiple sip servers</font>
<div> </div>
</div>
<div>
<div dir="ltr">Hello!
<div><br>
<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>
<div><br>
</div>
<div>Like this scenario: <a
href="https://opensips.org/pipermail/users/2020-August/043610.html"
moz-do-not-send="true">
https://opensips.org/pipermail/users/2020-August/043610.html</a><br>
</div>
<div><br>
</div>
<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/"
moz-do-not-send="true">https://blog.voipxswitch.com/2015/03/27/kamailio-basic-sip-proxy-all-requests-setup/</a><br>
</div>
<div><br>
</div>
<div>Basically just added rtpproxy support, and voilà,
inter-SIP is working, media always passing into the
proxy.</div>
<div><br>
</div>
<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>
<div><br>
</div>
<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>
<div><br>
</div>
<div>Someone must have already made a blog with such
setup, but I could not google-unearth it at least until
now.</div>
<div><br>
</div>
<div>Regards,</div>
<div>-- </div>
<div>
<div>
<div dir="ltr">Eliphas</div>
</div>
</div>
</div>
</div>
</div>
</div>
<br>
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