asterisk users - Apr 2005 - Is There Media Accelerator For Better Asterisk Calls

Dear Asterisk Users:

Please give me your opinion or advise of how I can
optimize my internet bandwidth to get better calls.

During morning hours, calls are fine. But during high
traffic time, I have calls problems. So I experimented
with changing the codecs in iax file and monitoring
the internet bandwidth using Mandrake linux control
Ctr.
When I used ulaw, the sending speed rate was at 9.3
Kbs and people can hear me fine. The receiving speed
rate was oscillating around 5kbs and 9.3 kbs where at
9.3 kbs the quality of the call is good. If I changed
the codec to a less bandwidth codec like gsm, the
sending speed is 3.1 kbs and people can hear me fine
and the receiving call alternates around 1 kbs and 3.1
kbs where at 3.1 kbs i can hear people fine. In both
codecs I experienced breaking up sounds when the rate
goes beyond 9.3 and 3.1.

The bandwidth for gsm is there, as it showed in ulaw.
Why gsm can't use the bandwidth consistently. Can I
download something like a media accelerator? Please
tell me what you think. Any suggestions.

Thanks;
 

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chawki hammoud wrote:
> When I used ulaw, the sending speed rate was at 9.3
> Kbs and people can hear me fine. The receiving speed
> rate was oscillating around 5kbs and 9.3 kbs where at
> 9.3 kbs the quality of the call is good. If I changed
> the codec to a less bandwidth codec like gsm, the
> sending speed is 3.1 kbs and people can hear me fine
> and the receiving call alternates around 1 kbs and 3.1
> kbs where at 3.1 kbs i can hear people fine. In both
> codecs I experienced breaking up sounds when the rate
> goes beyond 9.3 and 3.1.
It looks to me like packet loss rather than bandwidth only.  I.E. if you 
get to max bandwidth you will see lost packets.  However routers further 
down the line can also cause packet loss (at times of high load).

5/9.3 * 100 = 53.763440860215053763440860215054
1/3.1 * 100 = 32.258064516129032258064516129032

So, you are probably getting a max of 53% packet loss with Ulaw and 32% 
with GSM.

I would guess that you have something in the region of 25% packet loss 
at a remote router and 10-25% loss on the local connection depending on 
load.

In order to try and confirm this, see if small packets get loss.

I.E. if you sent something that was trying to use 1 kps do you still get 
25% packet loss?

The other option (if you have access to the egress point - router etc) 
is to enable QOS on outgoing data so that you shape the traffic. 
Basically the idea of this is as follows:

You have multiple bins:

[ 1 ][ 2 ][ 3 ][ 4 ]  (for arguments sake)

The idea is that the traffic shaper will check bin 1 and if there is 
something in it, send it.  If not check bin 2 and send etc etc until you 
reach the last bin.

So you can see that if you were to put VoIP traffic into bin 1 and say 
downloads into bin 4, and SSH into bin 2 etc then whenever there is VoIP 
traffic it will get sent first.

So, how do you decide what goes into each bin?

Well, deciding on what goes into each bin can be done in multiple ways. 
  For example you can do it based on IP (src/dest), port (src/dest), 
tcp/udp, TOS flag etc.

A TOS flag is a small number that is included in every packet.  TOS 
stands for Type Of Service.  Basically you can mark packets as having 
different types of requirements.  I.E. lowdelay, maxreliability etc.

You can set the TOS flags for traffic in Asterisk.

So, the bins actually come from tc or similar, but the easiest way is to 
use a script like wondershaper ( http://lartc.org/wondershaper/ ).

Basically if you use wondershaper you can just edit the file to decide 
how your traffic gets managed.

Let us know how you go.

-- 
Cheers,

Matt Riddell
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--- Matt Riddell <matt.riddell@sineapps.com> wrote:

 
> In order to try and confirm this, see if small
> packets get loss.
 
There was a packet loss in all the codecs i used, but
it's hard to tell whether the packet percentage loss
is gretaer or less at different codecs and how that
help in solving the issue. 
> The other option (if you have access to the egress
> point - router etc) 
My * box is behind the nat from a small isp and i
don't have access to anything besides my box.

I appreciate your feedback about IP compression. If
all my voice calls' destinations are known, what can I
do to reduce the ip header bandwidth. That might not
help me here, but it will in other type of internet
providers.

The other option I like to try is to to install some
accelerator to make more bandwidth available to my
asterisk box. 

I appreciate any more suggestions.


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