[Asterisk-Users] Calculating required bandwidth

[Damon Estep]

A TDM channel (POTS/PRI) consists of 64k of real-time data, VoIP with no
compression (g.711) consists of 64k plus IP protocol overhead for a
total bandwidth or 80 to 90k required per uncompressed channel. So a IP
T1 carrying VoIP without compression has lower capacity that a Voice T1.
A t1 for voice typically carries 23 b channels and 1 d channel, so 23
conversations not 24.

If you use compression on the VoIP traffic you gain capacity, but loose
CPU performance as the RTP data stream has to be transcoded by *.

If compression is used, and the box has the CPU power, significantly
more than 23 is the answer, probably limited more by then number that
your * can setup, transcode, and tear down. The exact answer depends on
your use and can only be determined through testing.

Uncompressed the answer is probably closer to 15 to 18 RTP streams
across a dedicate T1 IP link.

> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com 
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of 
> Ed Greenberg
> Sent: Thursday, December 16, 2004 10:45 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] Calculating required bandwidth
> 
> I was posed this question:
> 
> A T1 set up for voice carries 24 conversations on a circuit 
> that is 1.544 megabits/second. Right?
> 
> Well, if you set that T1 up to carry data and run a link 
> between two IP networks over it, how many SIP conversations 
> could it be expected to carry? 
> How about IAX?
> 
> How would one extend this calculation to varying bandwidth 
> circuits and various VOIP protocols (MGCP, SCCP and H323 come 
> to mind)?
> 
> Rather than asking for a full education here, can somebody 
> point me at a suitable practical reference? Of course, if 
> somebody wants to actually post the answer that'd be fine too :)
> 
> THanks,
> </edg>
> 
> 
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