asterisk users - Dec 2004 - Get asterisk out of the RTP stream?

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

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.

On Thu, 16 Dec 2004 14:51:53 -0600, Matthew Boehm <mboehm@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

RTP re-invite is possible.  The mess that is Cisco CallManager
supports SCCP, SIP, H.323 and MGCP.  Calls from anyone of those
technologies to any other technology works with the signalling
passed through the CCM server and RTP re-invites occur between
the endpoints.  So CDR works and scalability does not suffer.

I've been meaning to test to see if Asterisk supported this mode
of operation, and you've saved me the trouble.

I'd hate to predict the odds that such support would be added
since IAX doesn't use RTP, but I'd love to see it anyway.  And
if I understood more than the basics of programming in C, I'd
try it.

Dan


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.
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