Or you could do Phone <-> FS <=====> FS <-> Phone...

;)  Less complex.

/b

On Jan 27, 2010, at 11:54 PM, Nicholas Lee wrote:

> 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

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