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