[asterisk-users] how to find out one way latency

[Hans Witvliet]

On Wed, 2011-11-30 at 20:03 -0500, Adam Moffett wrote:

> 
> You can make a pretty good prediction with ping.
> "sudo ping -f -i .02 -s 180 -Q 0xb8 [ip]" gives a tolerable simulation 
> of voip traffic.  let it run for awhile, then press ctrl+c and see how 
> many packets were dropped and also check the mdev number.  If mdev is 
> low and packet loss is almost nothing then you can expect decent voice 
> quality.  It may not be a 100% perfect test, but I'll bet you a vast 
> majority of the time I can do that test and tell you whether it's going 
> to suck.
> 
> latency by itself with low jitter and no packet loss just means delay.  
> It's a matter of opinion and circumstance how tolerable delay is, but I 
> think your 230ms ping is at the upper edge of what most people can live 
> with.  Much more than that and you'll be tempted to say 'over' at the 
> end of sentence.
> 
> --
Fully agree,

Actually, you can do better than just a ping, but it takes some time,
equipment and experience:

What you can do, is adding an extra box inbetween your voip-client and
voip-server, and introduce all kinds of "real-life" circumstances.
I mean artificial delay, loss, resequencing, duplicating packages,
reduced bandwith. We've done it some time ago as an "satelite simulator"
You can build it aroud any *bsd/linux box with multiple nics.

The basic idea's you can find at http://lartc.org/
If you combine it with the echo function from asterisk, you can decide
for yourself what it acceptable and what not.

For one of my projects i push the echo destination as the "default" sip
connection to their soft phone, as i noticed that people at the other
side of town regularly have a worse connection then people using umts or
satelite. Main culprit (in my case) is ill-configured WIFI-setup.
Latencies of over 10,000 ms and loss of 80% are daily events.
And people complaining....


hw