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]<mailto:[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]<mailto:[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/kamailio/kamailio.cfg but I blamed my failure on an old rtpengine :) _______________________________________________ Kamailio (SER) - Users Mailing List [email protected]<mailto:[email protected]> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ Kamailio (SER) - Users Mailing List [email protected]<mailto:[email protected]> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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