Or maybe FreeSwitch is redundant if you use rtpengine…
With kind regards Pan B. Christensen Developer Phonect AS
From: sr-users sr-users-bounces@lists.kamailio.org On Behalf Of Emanuel Gianico Sent: fredag 15. juni 2018 13:29 To: Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org 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 <ovoshlook@gmail.commailto:ovoshlook@gmail.com> 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 <d.tryba@pocos.nlmailto:d.tryba@pocos.nl> 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 :)
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