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

[NaJIm]

WOW.. That is the most complicated Ping I have ever seen.. :)

This is the result I got.

# ping -f -i .02 -s 180 -Q 0xb8 xx.xx.xx.xx
*PING xx.xx.xx.xx (xx.xx.xx.xx) 180(208) bytes of data.
.............
--- xx.xx.xx.xx ping statistics ---
15338 packets transmitted, 15325 received, 0% packet loss, time 352748ms
rtt min/avg/max/mdev = 276.499/286.185/310.118/7.248 ms, pipe 15, ipg/ewma
22.999/284.882 ms
*

The same test with my Present SIP Provider gave me the result below.

*10926 packets transmitted, 10913 received, 0% packet loss, time 244048ms
rtt min/avg/max/mdev = 289.514/292.668/316.350/2.336 ms, pipe 15, ipg/ewma
22.338/292.941 ms
*

I suppose the value of mdev is much higher in the first case but 0% packet
loss in both the cases.
Does this mean that the voice quality is going to be real bad??

Thanks,
Najim

On Thu, Dec 1, 2011 at 6:33 AM, Adam Moffett <adamlists at plexicomm.net>wrote:

>
>  a ping is the time a packet needs for travelling to a destination and
>>> back to you. So the one way latency you are refering to, should be half
>>> the time your ping took.
>>>
>>> In your case this will be 130ms, I would say this is still reasonable.
>>>
>> I am probably splitting hairs, but that's not always true because there's
> no guarantee that the reply traveled the same path as the echo request.  If
> you dig into BGP issues you'll see sometimes that traffic one direction
> takes a different route than traffic the other direction.  I don't know of
> any simple and accurate way to learn the "one way" latency so I'm surprised
> they specified anything other than round trip time.
>
>
>  'Ping time' is not an accurate predictor of SIP quality.
>>
>> A 'ping' is an ICMP Echo/reply packet and some routers consider them less
>> important than 'data' packets and service them on an 'as resources permit'
>> basis.
>>
> That's possibly maybe true if someone's router or connection is overloaded
> and they are trying to make up for it with CoS policies while they save up
> for an upgrade.  Otherwise it's an apology for a crappy network.  That's
> the brutally honest truth.
>
> 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.
>
>
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