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

[C F]

On Thu, 16 Dec 2004 14:51:53 -0600, Matthew Boehm <mboehm at cytelcom.com> wrote:
> 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
> 
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The answer is yes. If the a reinvite is issued then * is out of it but
stays in there for the signaling.
look at the following:
http://www.voip-info.org/wiki-Asterisk+SIP+media+path
http://www.voip-info.org/wiki-Asterisk+sip+canreinvite
http://www.voip-info.org/tiki-index.php?page=Asterisk:%20Letting%20SIP%20clients%20connect%20directly