14 Jun
2018
14 Jun
'18
8:25 p.m.
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 :)