Hello
You can start with this config, it's for WebRTC, but will hold same concepts
https://github.com/igorolhovskiy/webrtc-full-proxy/blob/master/kamailio/kamailio.cfg
Point, you might would need an rtpengine as well if your media would be
decoded in SRTP <-> RTP way.
Regards,
Igor
Le 25.03.2022 à 21:44, jaflong a écrit :
Hi Kamailio community
Any advise on how to archive this
sip.mydomain.com sip.mypbx.com
sip_phone =======sip/tls=======> KAMAILIO =========sip/udp=======>PBX
sip_phone should register to Kamilio with tls
Kamailio should proxy register to PBX with udp
sip_phone invites should be proxied to PBX
in sip_client setting he will point to sip.mydomain.com as his server
Regards
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