[afnog] Bandwidth Utilization On VoIP
mtinka at africaonline.co.zw
Tue Mar 6 09:41:04 UTC 2007
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
(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
( (40 + (4 * 24) * 8) / (4 * (30 * 0.001) ) = 9,067bps
</someone double-check with me on this>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Size: 827 bytes
Desc: not available
Url : http://afnog.org/mailman/private/afnog/attachments/20070306/d1c11a55/attachment.bin
More information about the afnog