[asterisk-biz] Low Bandwidth VoIP

[Stelios Koroneos]

On Tue, 2012-07-17 at 05:09 +0100, David Knell wrote:
> On Mon, 2012-07-16 at 11:11 +0200, Martin Vit wrote: 
> >         The short answer is no you can't get 32 *concurrent* channels
> >         on a
> >         250kbps uplink
> >         
> >         With G729 as a codec you need around 32kbits per channel
> >         including the
> >         overheads from the voip protocol and tcp etc, so that would
> >         give you
> >         around 7 *concurrent* channels. 
> 
> This is correct if you're using vanilla RTP and no silence suppression.
> The G.729 payload runs at 8kbits/sec during talk and 1.6kbits/sec otherwise;
> assuming 50% silence gives an average bitrate of 4.8kbits/sec.
> Alternatively,
> just dropping silent frames brings that down to 4kbits/sec.
> 
> Multiplexing 32 of these over a 250kbits/sec connection should not be a 
> problem, provided that they're not individually wrapped with RTP, UDP
> and IP headers.
> 
> Incidentally, at 50% talk/silence (which is an over-estimate - people listen
> more than they talk), the chance of everyone at one end of 32 calls talking
> at the same time is 2*10^-10.  So there's not going to be many frames 
> dropped as a result.
> 


You are also correct in the calculations.
The main issue is that the asterisk g729 codec is annex A so there is no
silence suppression and thus you get a constant bit rate.
http://www1.digium.com/en/products/software/g729-codec

If you notice Digium says you can get 140 g729 calls in a T1 (1.5Mbit)
using iax trunking, which comes around 10kbit per channel.
And this very close to what i have found.

After going back and reading my replies, what i should have clarified is
that i was talking about *asterisk* getting the 32 channels in 250kbit.

g723, or another low bitrate codec might be an option but the voice
quality is usually not what the clients expect.