[Asterisk-Users] Get asterisk out of the RTP stream?

[Matthew Boehm]

Here is the setup:

Phone A (in NYC) on own bandwidth.
Phone B (in LA) on own bandwidth.
Asterisk box in Houston,TX on own bandwidth.

Both phones contact asterisk to register. Not much bandwidth used for this
as it is a few packets every hour or so.

Phone A calls Phone B. Phone A sends a call request to asterisk and asterisk
calls phone B. Both phones are connected and both people are talking.

Is all of the data/voice comming from phone A going into asterisk box and
then from asterisk box to phone B? If so, then using g711, phone A would
send/recieve 64Kbps to/from asterisk and phone B would also send/recieve
64Kbps to/from asterisk. Asterisk would then be sending/recieving 128Kbps
for this one call right? So with 1 T1 you could only get 12 calls going
right?

If I use canreinvite=yes on both phones, will phone A connect to phone B
directly therefore lowering the bandwidth usage in/out of the asterisk box
right?

If so, what is the "signalling" bandwidth usage in/out of asterisk in this
case? Even if the phones are connected directly to eachother, they still
have to pass some data to asterisk so asterisk still knows that the call is
up and has to know when the call goes away. We need to know this bandwidth
usage on a T1 because lets say it was 10Kbps, you could actually do a bunch
of calls on 1 T1 provided that all phones use canreinvite right?

Thanks,
Matthew