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<p class="MsoNormal">Or maybe FreeSwitch is redundant if you use rtpengine…<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal"><span style="font-size:9.0pt;font-family:"Helvetica",sans-serif;color:black;mso-fareast-language:NO-BOK">With kind regards<br>
<b>Pan B. Christensen</b><br>
Developer<br>
Phonect AS </span><span style="mso-fareast-language:NO-BOK"><o:p></o:p></span></p>
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<p class="MsoNormal"><b>From:</b> sr-users <sr-users-bounces@lists.kamailio.org> <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 <sr-users@lists.kamailio.org><br>
<b>Subject:</b> Re: [SR-Users] Kamailio + FreeSwitch + WebRTC<o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></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<o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></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" target="_blank">ovoshlook@gmail.com</a>> escribió:<o:p></o:p></p>
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<p class="MsoNormal">You can watch at the kazoo project examples if you want to avoid rtp proxy<o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
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<p class="MsoNormal">On Thu, Jun 14, 2018, 23:26 Daniel Tryba <<a href="mailto:d.tryba@pocos.nl" target="_blank">d.tryba@pocos.nl</a>> wrote:<o:p></o:p></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" target="_blank">https://github.com/havfo/WEBRTC-to-SIP/blob/master/etc/kamailio/kamailio.cfg</a><br>
but I blamed my failure on an old rtpengine :)<br>
<br>
<br>
_______________________________________________<br>
Kamailio (SER) - Users Mailing List<br>
<a href="mailto:sr-users@lists.kamailio.org" target="_blank">sr-users@lists.kamailio.org</a><br>
<a href="https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users" target="_blank">https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users</a><o:p></o:p></p>
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<p class="MsoNormal">_______________________________________________<br>
Kamailio (SER) - Users Mailing List<br>
<a href="mailto:sr-users@lists.kamailio.org" target="_blank">sr-users@lists.kamailio.org</a><br>
<a href="https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users" target="_blank">https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users</a><o:p></o:p></p>
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