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/kamailio/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 >
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