[Freeswitch-users] Conceptual Question about Freeswitch and SIP

[Kurt Marasco]

It's my understanding that when I call from one SIP address to another 
that Freeswitch manages the invites and then directly connects the two 
sip devices, such that freeswitch is no longer involved in the 
conversation. Is this how it is supposed to work?

The reason for my question is that the rtp traffic is going through my 
freeswitch ip address at all times. It seems that both ends are speaking 
g711u, so I don't believe that any transcoding is going on.

I must be missing something because having all data flow through 
Freeswitch would not be scalable and I know that one of the big 
differences that Freeswitch offers is scalability.

What am I missing....and I know that I am:)
Kurt