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

[Kevin P. Fleming]

Matthew Boehm wrote:

> 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?

Nope, you've misunderstood. If these phones are connecting via SIP or 
IAX, and Asterisk is allowed to reinvite them to talk to each other, 
then Asterisk will be _completely_ out of the conversation. The only way 
that Asterisk would become involved again is if one of the phone users 
decided to transfer their end of the call.

Given this, if you allow reinvites, you _cannot_ have accurate and 
complete CDR information. Many of us would like to see Asterisk support 
this mode of operation (reinvite only the media stream, not the control 
stream), and some of those many think it's actually possible... but 
there are others who feel that since Asterisk is not a SIP proxy it 
cannot be done. I am not in that group, I just don't have the time to 
try to implement it myself :-(

In any case, you have two choices: avoid the bandwidth consumption on 
the Asterisk server's link, or have accurate CDR.