14 Jun
2018
14 Jun
'18
10:37 p.m.
Yeah, you need to set the correct offer, i did that a while ago, but i
can't remember how i did it.
Check out https://github.com/havfo/WEBRTC-to-SIP
Hope it help.
David
On Thu, Jun 14, 2018, 22:26 Daniel Tryba <d.tryba(a)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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