[Freeswitch-users] Achieving TLS + SRTP for inbound calls

[David P]

We use conferences to allow a verto user to call and connect with an
Asterisk channel. We would like to secure both signalling and media via TLS
+ SRTP, and I've read
https://freeswitch.org/confluence/display/FREESWITCH/SIP+TLS a few times to
understand how to do this. Note that that page has a broken link:
https://wiki.freeswitch.org/wiki/Secure_RTP

First, is it still true that FS doesn't offer prebuilt installs (for
Ubuntu) to support this kind of security?

Assuming that it must be compiled, I began to follow
https://freeswitch.org/confluence/display/FREESWITCH/Debian+8+Jessie with
the additional first step of:  apt-get install libssl-dev

I soon ran into "Unable to locate package freeswitch-video-deps-most".

What should I try next?

Cheers,
David
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