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/kama...

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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