[Asterisk-Users] Calculating required bandwidth

[Jim Van Meggelen]

asterisk-users-bounces at lists.digium.com wrote:
> I was posed this question:
> 
> A T1 set up for voice carries 24 conversations on a circuit that is
> 1.544 megabits/second. Right?

Yes and no. If the T1 is channelized, then yes. If it's a PRI circuit,
then it has only 23 channels to carry voice, as the 24th channel is used
for the D-channel (signalling channel).

PRI is superior, because it offers far more flexible use of the circuit,
and provides far more information (like CallerID).


> 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?

Interesting question. I'll tell you this, it won't have so much to do
with the 24 channels as it will with how efficiently the circuit is
used. When you run data on a T1, all of the pipe is treated as one big
channel by the upper layers. The 24 timeslots are all still there, but
the network doesn't have any knowledge of them.

> How would one extend this calculation to varying bandwidth circuits
> and various VOIP protocols (MGCP, SCCP and H323 come to mind)?

Each network layer (think of the OSI model) will add overhead, so the
calculation has to take into account how the data (in this case, the
voice packets) is encapsulated at each layer. 

Of the protocols, IAX would probably utilize the circuit most
efficiently, due to it's trunking. Naturally, the codec you use will be
another key factor.

> 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 :)

I've always found Newton's Telecom Dictionary to be a great reference.
It's not too technical, packed with humour, and very comprehensive.