My scenario is the following:

External clients register to a Kamailio server (doesn't have to be Kamailio, just what I'm using in my test environment) via TLS and calls between external clients are via SRTP.

There is another internal Kamailio server where users register via standard SIP and media over RTP.

Can Freeswitch act as a gateway for calls between internal and external clients and vice versa, where the SRTP stream is initiated/terminated (depending on the call direction) on the Freeswitch server?

Thanks.