[afnog] Bandwidth Utilization On VoIP

[Mark Tinka]

On Tuesday 06 March 2007 10:42, byaruhj at mtn.co.ug wrote:

> Can any one on the forum tell me of some wise ways of reducing on
> bandwidth utilization over an internet circuit? I am currently
> running a circuit with G.729 codecs and i believe i am using abt
> 18K per call, However i believe there must be a way i can utilize
> less bandwidth, Why i try compression on G.723, i think i can
> utilize 16k per call but that's still high.

You would use slightly higher bandwidth than is advertised for the 
codec (8Kbps for G.729, 5.3Kbps to 6.3Kbps for G.723.1) because the 
amount of encapsulated data in the UDP and RTP headers can often be 
larger than the audio data payload.

<someone double-check with me on this>

In order to minimize latency, let's say your sample rate for voice 
frames encoded with G.723.1 is 30ms, carrying audio packets each 
containing a 24-byte encoded audio frame. You audio bandwidth 
becomes:

(24 * 8) / (30 * 0.001) = 6,400bps.

But remember that each frame would have an IP header of 20 bytes, a 
UDP header of 8 bytes and an RTP header of 12 bytes - so you have 
an overhead of 40 bytes. To recalculate how much bandwidth you'll 
end up using:

( (40 + 24) * 8) / (30 * 0.001) = 17,066bps

This is more than double the amount of bandwidth the codec is 
advertised with. This excludes Layer 2 overheads such as Ethernet 
headers, PPP headers, e.t.c., which increases the (link) overhead.

One documented way to reduce the bandwidth utilized is by increasing 
the latency (trade-off), although I have not tried to do this. For 
instance, assume your frame rate is now 120ms, your recalculated 
bandwidth becomes:

( (40 + (4 * 24) * 8) / (4 * (30 * 0.001) ) = 9,067bps

</someone double-check with me on this>

Mark.
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