Re: [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


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

[Adam Moffett]

I would bet you get about the same result with the two providers.all 
else being equal.
mdev (mean deviation) is a simple way to measure jitter, and you have to 
put in context with the min/avg/max numbers.  If I had 7ms of deviation 
and average times of 4ms, that would be an issue because you would be 
likely to get packets out of order.  But 7ms compared to 286ms probably 
means nothing.


Your biggest problem with both providers is delay, but if you can 
tolerate the delay you have now, then you can probably tolerate the 
delay with the other provider.


Also note that although packet loss is 0%, some packets are still 
dropped in both cases.  One dropped packet means a small amount of audio 
is lost (depends on codec, but often 20ms).  If those handful of dropped 
packets are scattered evenly then you wouldn't notice it, but it's 
common for them to occur in a cluster.  If the 13 packets dropped in the 
first example all happened at once you would have lost 260ms of 
audioand you would certainly hear that.  You may be able to tell by 
watching the periods appear on the screen when you run the ping 
command.  Each period is a dropped packetif they accumulate in a 
burst then something is happening that you would hear on the phone.



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

[Adam Moffett]

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

[NaJIm]

My ping requests show 0% packet loss. How do we find out packet
re-ordering.??

Najim.

On Thu, Dec 1, 2011 at 5:18 AM, Hans Witvliet  wrote:

> On Thu, 2011-12-01 at 04:52 +0530, NaJIm wrote:
> > Is there anything else that I should be concerned about, when looking
> > to signup for a SIP provider. ??
> Latency is important, but packet loss also, likewise packet re-ordering.
>
> hw
>
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Re: [asterisk-users] how to find out one way latency

[Hans Witvliet]

On Thu, 2011-12-01 at 04:52 +0530, NaJIm wrote:
> Is there anything else that I should be concerned about, when looking
> to signup for a SIP provider. ?? 
Latency is important, but packet loss also, likewise packet re-ordering.

hw

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

[NaJIm]

Is there anything else that I should be concerned about, when looking to
signup for a SIP provider. ??

Regards,
Najim

On Thu, Dec 1, 2011 at 4:49 AM, NaJIm  wrote:

> Does that mean I can expect lesser delays with my Voice packets ?? That
> would be even better.
>
> Regards,
> Najim
>
> On Thu, Dec 1, 2011 at 3:42 AM, Steve Edwards 
> wrote:
>
>> Am 30.11.2011 21:47, schrieb NaJIm:
>>>
>>
>>  Ping request to their IP Address gives me a response in approx. 260ms.

>>>
>>  Will that be good enough for a SIP Trunk.

>>>
>> On Wed, 30 Nov 2011, Ruben Rögels 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.
>>>
>>
>> '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.
>>
>> --
>> Thanks in advance,
>> --**--**
>> -
>> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
>> Newline  Fax: +1-760-731-3000
>> --
>> _
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>
>
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Re: [asterisk-users] how to find out one way latency

[NaJIm]

Does that mean I can expect lesser delays with my Voice packets ?? That
would be even better.

Regards,
Najim

On Thu, Dec 1, 2011 at 3:42 AM, Steve Edwards wrote:

> Am 30.11.2011 21:47, schrieb NaJIm:
>>
>
>  Ping request to their IP Address gives me a response in approx. 260ms.
>>>
>>
>  Will that be good enough for a SIP Trunk.
>>>
>>
> On Wed, 30 Nov 2011, Ruben Rögels 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.
>>
>
> '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.
>
> --
> Thanks in advance,
> --**--**
> -
> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
> Newline  Fax: +1-760-731-3000
> --
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Re: [asterisk-users] how to find out one way latency

[Steve Edwards]

Am 30.11.2011 21:47, schrieb NaJIm:



Ping request to their IP Address gives me a response in approx. 260ms.



Will that be good enough for a SIP Trunk.


On Wed, 30 Nov 2011, Ruben Rögels 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.


'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.


--
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-
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Newline  Fax: +1-760-731-3000--
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Re: [asterisk-users] how to find out one way latency

[NaJIm]

Thank you Ruben.

 Is there anything else that I should be concerned about when looking for a
SIP provider. ??

Regards,
Najim.

On Thu, Dec 1, 2011 at 2:34 AM, Ruben Rögels  wrote:

> Am 30.11.2011 21:47, schrieb NaJIm:
> > Hi All,
> >
> > How can I find out One way latency from my PBX to my SIP Trunk Provider.
> > My SIP provider recommends a One way latency of 100ms for good Voice
> > quality. Ping request to their IP Address gives me a response in approx.
> > 260ms.
> > Will that be good enough for a SIP Trunk.
> >
> > Please help. We are trying to sign up with a new SIP Provider.
> >
> > Thanks,
> > Najim
> >
> >
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> >http://lists.digium.com/mailman/listinfo/asterisk-users
>
> Hi Najim,
>
> 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.
>
>
> regards,
> Ruben
>
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Re: [asterisk-users] how to find out one way latency

[Ruben Rögels]

Am 30.11.2011 21:47, schrieb NaJIm:
> Hi All,
> 
> How can I find out One way latency from my PBX to my SIP Trunk Provider.
> My SIP provider recommends a One way latency of 100ms for good Voice
> quality. Ping request to their IP Address gives me a response in approx.
> 260ms.
> Will that be good enough for a SIP Trunk.
> 
> Please help. We are trying to sign up with a new SIP Provider.
> 
> Thanks,
> Najim
> 
> 
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Hi Najim,

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.


regards,
Ruben

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

[NaJIm]

Hi All,

How can I find out One way latency from my PBX to my SIP Trunk Provider.
My SIP provider recommends a One way latency of 100ms for good Voice
quality. Ping request to their IP Address gives me a response in approx.
260ms.
Will that be good enough for a SIP Trunk.

Please help. We are trying to sign up with a new SIP Provider.

Thanks,
Najim
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