[Asterisk-Users] Calculating required bandwidth

[Race Vanderdecken]

Thank you Peasants,

In general the original question was answered. I am software guy, if the
network slobs can't fit all the data in the pipe that is not my problem.

The basic idea in the answer was that you can get more calls by using
compression; much like the automobile manufacture's gas mileage may
vary.

Also remember that a telephone conversation is 2/3's silence. ( I speak,
silence, then you speak. See the book at bought on Amazon 4 years ago
but can't remember the name of the book.)IP only sends the data when
there is noise versus the T1 which is a constant TDM stream. So I
predict in testing with good VoIP equipment you can get more then 24
G.711 calls per T1. So take that and comment. You should be able to get
more VoIP calls, my prediction is 40 G.711 well behaved calls with
silence suppression per T1. Why else would the Baby Bells move to VoIP?

But it is nice to know there are some intelligent folks monitoring the
list, thank you.

Race

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Andrew
Kohlsmith
Sent: 16 December 2004 14:18
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] Calculating required bandwidth

On December 16, 2004 01:52 pm, Race Vanderdecken wrote:
> The quick tyrannical answer,

And wrong -- I am taking the time to correct it not so much to slam you
but 
more for list posterity -- just because the codec rate is 64kbps doesn't
mean 
that's what's actually on the wire, even if you ignore signalling.

> Each T1 channels is 64Kbps, 1,544,000 / 24 = 64,333

each T1 has 24 channels of 8 bit data plus one frame bit.
24*8+1 = 193 bits per T1 frame.  Frames are sent 8000 per second.
8000*193 = 
1544000 bits per second.  There's your T1 raw rate.

You can't use that frame bit for yourself so 24*8*8000 = 1536000 bits
per 
second.  That's your T1 data rate; that's what you can actually use.

Now.  Running IP on a T1 you have certain overheads.  UDP frame overhead
is 4 
bytes, plus your TCP overhead of 12 bytes, for a total of 16 bytes (128 
bits).  G.711 is 64kbps data rate, but Asterisk sends only 20ms per
packet in 
an attempt to balance data throughput and effect of lost packets.

so 64kbps / 50 is 1280 bits of audio per packet, plus 128 bits of
overhead for 
1408 bits per packet.  50 of these per second of audio gives you
70400bps for 
one second of G.711 VOIP audio.

so now take your T1 data rate of 1536000bps and divide your audio rate
into it 
for an answer of 21 channels of G.711 VOIP audio.

Now that was straight UDP audio -- there was no signalling overhead and
it 
wasn't SIP RTP.

RTP has 12 octets all its own, and still need 12 bytes of IP overhead,
so it 
is actually costlier: I'll spare you all the calculations but it's 20 
channels of SIP G.711 audio per T1, likely with enough room for 
signalling.  :-)

Regards,
Andrew "the tyrant's tyrant" Kohlsmith
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