[asterisk-biz] Low Bandwidth VoIP

[David Knell]

G. Jacobsen wrote:

> is it possible simply to drop silent packets in the straight g729 (not a,
b) protocol?
> How do I recognize silent packets ?

Good question - you'd need to recognise them before they reach the encoder,
I 
guess.  A moving power average compared to a threshold is a simple and not
entirely
useless speech/silence differentiator when combined with a holdover timer.

It's fine to start and stop the RTP stream - i.e. just to drop packets which

would represent silence - provided that the timestamps are correct and,
ideally, 
the bit which marks a new burst gets correctly set.

> Thanks for this informative posting

Glad you found it of use!

--Dave

On Jul 17, 2012, at 7:09 AM, 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.
> 
> --Dave
> 
> 
> 
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