https://github.com/2600hz/kazoo-configs-kamailio/blob/master/kamailio/websockets-role.cfg
2018-06-21 18:35 GMT+03:00 Emanuel Gianico <[email protected]>: > 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: > > - WebRTC support > - VoiceMail Server > - Queues > - IVR > - Announcements > > And so on... > > The idea is to use Kamailio in front and throw all media related stuff to > FS. > > 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) > > Best regards, > Emanuel. > > El vie., 15 de jun. de 2018 09:32, Pan Christensen < > [email protected]> escribió: > >> Or maybe FreeSwitch is redundant if you use rtpengine… >> >> >> >> With kind regards >> *Pan B. Christensen* >> Developer >> Phonect AS >> >> >> >> *From:* sr-users <[email protected]> * On Behalf Of >> *Emanuel >> Gianico >> *Sent:* fredag 15. juni 2018 13:29 >> *To:* Kamailio (SER) - Users Mailing List <[email protected]> >> *Subject:* Re: [SR-Users] Kamailio + FreeSwitch + WebRTC >> >> >> >> 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 >> >> >> >> El jue., 14 de jun. de 2018 17:42, Yuriy Gorlichenko <[email protected]> >> escribió: >> >> You can watch at the kazoo project examples if you want to avoid rtp proxy >> >> >> >> On Thu, Jun 14, 2018, 23:26 Daniel Tryba <[email protected]> wrote: >> >> On Thu, Jun 14, 2018 at 04:48:40PM -0300, Emanuel Gianico wrote: >> > From the logs I see the jssip throw this error: >> > >> > "Failed to set remote offer sdp: Called with SDP without DTLS >> fingerprint." >> > >> > I would like to avoid RTPEngine, because from what I understand, >> FreeSwitch >> > can handle the media part. >> >> IIRC I got the same error in my tries to transcode/bridge SIP over TLS >> with SRTP to just plain old SIP with RTP. I haven't put any effort in it >> to get it working. You'll need to play around with rtpengine >> offer/answer, I based my test on >> https://github.com/havfo/WEBRTC-to-SIP/blob/master/etc/kamai >> lio/kamailio.cfg >> but I blamed my failure on an old rtpengine :) >> >> >> _______________________________________________ >> Kamailio (SER) - Users Mailing List >> [email protected] >> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >> >> _______________________________________________ >> Kamailio (SER) - Users Mailing List >> [email protected] >> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >> >> _______________________________________________ >> Kamailio (SER) - Users Mailing List >> [email protected] >> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >> > > _______________________________________________ > Kamailio (SER) - Users Mailing List > [email protected] > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > >
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