<div dir="ltr"><a href="https://github.com/2600hz/kazoo-configs-kamailio/blob/master/kamailio/websockets-role.cfg">https://github.com/2600hz/kazoo-configs-kamailio/blob/master/kamailio/websockets-role.cfg</a></div><div class="gmail_extra"><br><div class="gmail_quote">2018-06-21 18:35 GMT+03:00 Emanuel Gianico <span dir="ltr"><<a href="mailto:emanuelgianico@gmail.com" target="_blank">emanuelgianico@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"><div><div>I don't think so... The idea is to use FS as a media server. Why I would use RTPEngine? FS offers the same and more:</div><div><br></div><div>- WebRTC support</div><div>- VoiceMail Server</div><div>- Queues</div><div>- IVR</div><div>- Announcements</div><div><br></div><div>And so on...</div><div><br></div><div>The idea is to use Kamailio in front and throw all media related stuff to FS.</div><div><br></div><div>There is a way to accomplish this? I searched a lot but I couldn't find nothing about it, only about Kamailio and RTPProxy (or RTPEngine)</div><div><br></div><div>Best regards,</div><div>Emanuel.</div></div><div><div class="h5"><div dir="auto"></div><br><div class="gmail_quote"><div dir="ltr">El vie., 15 de jun. de 2018 09:32, Pan Christensen <<a href="mailto:pan.christensen@phonect.no" target="_blank">pan.christensen@phonect.no</a>> escribió:<br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
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<p class="MsoNormal">Or maybe FreeSwitch is redundant if you use rtpengine…<u></u><u></u></p>
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<p class="MsoNormal"><span style="font-size:9pt;font-family:Helvetica,sans-serif;color:black">With kind regards<br>
<b>Pan B. Christensen</b><br>
Developer<br>
Phonect AS </span><span><u></u><u></u></span></p>
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<p class="MsoNormal"><b>From:</b> sr-users <<a href="mailto:sr-users-bounces@lists.kamailio.org" rel="noreferrer" target="_blank">sr-users-bounces@lists.kamail<wbr>io.org</a>> <b>
On Behalf Of </b>Emanuel Gianico<br>
<b>Sent:</b> fredag 15. juni 2018 13:29<br>
<b>To:</b> Kamailio (SER) - Users Mailing List <<a href="mailto:sr-users@lists.kamailio.org" rel="noreferrer" target="_blank">sr-users@lists.kamailio.org</a>><br>
<b>Subject:</b> Re: [SR-Users] Kamailio + FreeSwitch + WebRTC<u></u><u></u></p>
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<p class="MsoNormal">I'm going to investigate Kazoo samples as Gorlichenko suggested because I think using RTPEngine or rtp proxy seems redundant/unnecesary to me since FreeSwitch fully supports WebRTC<u></u><u></u></p>
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<p class="MsoNormal">El jue., 14 de jun. de 2018 17:42, Yuriy Gorlichenko <<a href="mailto:ovoshlook@gmail.com" rel="noreferrer" target="_blank">ovoshlook@gmail.com</a>> escribió:<u></u><u></u></p>
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<p class="MsoNormal">You can watch at the kazoo project examples if you want to avoid rtp proxy<u></u><u></u></p>
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<p class="MsoNormal">On Thu, Jun 14, 2018, 23:26 Daniel Tryba <<a href="mailto:d.tryba@pocos.nl" rel="noreferrer" target="_blank">d.tryba@pocos.nl</a>> wrote:<u></u><u></u></p>
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<p class="MsoNormal">On Thu, Jun 14, 2018 at 04:48:40PM -0300, Emanuel Gianico wrote:<br>
> From the logs I see the jssip throw this error:<br>
> <br>
> "Failed to set remote offer sdp: Called with SDP without DTLS fingerprint."<br>
> <br>
> I would like to avoid RTPEngine, because from what I understand, FreeSwitch<br>
> can handle the media part.<br>
<br>
IIRC I got the same error in my tries to transcode/bridge SIP over TLS<br>
with SRTP to just plain old SIP with RTP. I haven't put any effort in it<br>
to get it working. You'll need to play around with rtpengine<br>
offer/answer, I based my test on<br>
<a href="https://github.com/havfo/WEBRTC-to-SIP/blob/master/etc/kamailio/kamailio.cfg" rel="noreferrer" target="_blank">https://github.com/havfo/WEBRT<wbr>C-to-SIP/blob/master/etc/kamai<wbr>lio/kamailio.cfg</a><br>
but I blamed my failure on an old rtpengine :)<br>
<br>
<br>
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