The TLS wiki page talks about [1] Freeswitch being able to act as a past though proxy for a SIP phone. Turning an unencrypted SIP+RTP session into a SIPS+SRTP session.
Is there howto guide for the above?
I'm also wondering if Freeswitch could do a drop-in encryption. ie. Phone <-> Asterisk replaced with Phone <-> FS <==> FS <-> Asterisk, where <==> is encrypted.
Nicholas