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.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.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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