how to calculate the roundtrip time, delay and jitter

Hi there:

Ok I presume this is a very stupid question, but I cant figure it
out,how can I calculate the roundtrip time time, delay and jitter in a
wlan?
I have ethereal, but i dont know where to get this info.


Regards

flo

(E-Mail Removed) hath wroth:

>Ok I presume this is a very stupid question, but I cant figure it
>out,how can I calculate the roundtrip time time, delay and jitter in a
>wlan?
>I have ethereal, but i dont know where to get this info.

Since you didn't bother to mention the operating system, I'll assume
you're a Windoze user. Mac and Linux users always seem to remember to
mention their operating system.

Time delay can be done with just ping.
Start -> run -> cmd <enter>
ping ip_address

For better resolution, I suggest FPING:
http://www.kwakkelflap.com/fping.html

You can use ping to obtain a series of latency values and feed them to
a spreadsheet or statistics calculator to obtain variations, which is
jitter. A simple histogram of ping results is usually sufficiently
interesting.

IPerf will measure jitter but only with UDP packets. See:
http://dast.nlanr.net/Projects/Iperf/
It will not do it as a "round trip" type of measurement because jitter
is normally measured one way. You'll need two computers, one running
the server half to get jitter results.

Ethereal is a packet sniffer and decoder. It offers nothing in the
way of performance measuring tools.

What are you trying to accomplish?


--
Jeff Liebermann (E-Mail Removed)
150 Felker St #D http://www.LearnByDestroying.com
Santa Cruz CA 95060 http://802.11junk.com
Skype: JeffLiebermann AE6KS 831-336-2558

hi Jeff:
thanks, yes i m a windows user, 2000 to be more accurate,
i need this to measure the performance of voip calls, i don't think
ping is what I could use, i need something that can give me the round
trip time ,delay and jitter at the time when i m sending the packets,
any ideas? if it was free much better.
Thanks

flo

Jeff Liebermann wrote:
> (E-Mail Removed) hath wroth:
>
> >Ok I presume this is a very stupid question, but I cant figure it
> >out,how can I calculate the roundtrip time time, delay and jitter in a
> >wlan?
> >I have ethereal, but i dont know where to get this info.
>
> Since you didn't bother to mention the operating system, I'll assume
> you're a Windoze user. Mac and Linux users always seem to remember to
> mention their operating system.
>
> Time delay can be done with just ping.
> Start -> run -> cmd <enter>
> ping ip_address
>
> For better resolution, I suggest FPING:
> http://www.kwakkelflap.com/fping.html
>
> You can use ping to obtain a series of latency values and feed them to
> a spreadsheet or statistics calculator to obtain variations, which is
> jitter. A simple histogram of ping results is usually sufficiently
> interesting.
>
> IPerf will measure jitter but only with UDP packets. See:
> http://dast.nlanr.net/Projects/Iperf/
> It will not do it as a "round trip" type of measurement because jitter
> is normally measured one way. You'll need two computers, one running
> the server half to get jitter results.
>
> Ethereal is a packet sniffer and decoder. It offers nothing in the
> way of performance measuring tools.
>
> What are you trying to accomplish?
>
>
> --
> Jeff Liebermann (E-Mail Removed)
> 150 Felker St #D http://www.LearnByDestroying.com
> Santa Cruz CA 95060 http://802.11junk.com
> Skype: JeffLiebermann AE6KS 831-336-2558

On 3 Jan 2007 13:27:15 -0800, "(E-Mail Removed)" <(E-Mail Removed)>
wrote:

>thanks, yes i m a windows user, 2000 to be more accurate,
>i need this to measure the performance of voip calls, i don't think
>ping is what I could use, i need something that can give me the round
>trip time ,delay and jitter at the time when i m sending the packets,
>any ideas? if it was free much better.

Welcome to the VoIP benchmarking can of worms. The problem is that
VoIP quality measurements are subjective and at best guesswork.
Attempts have been made to provide a statistical basis for these
measurements. They work, but the results are generally useless
because they either measure a short time interval or averaged over too
long a period to see momentary dropouts.

You also have the problem of having your test software interact with
the VoIP traffic. The only way around this is to use the VoIP packets
themselves as the test packets. That requires a specialized VoIP
tool. If you use something like IPerf *WHILE* you're sending VoIP
packets, you'll get the conglomerated result of both programs traffic,
which is useless.

I couldn't find much in the way of free VoIP simulation tools.
<http://www.packetizer.com/voip/diagnostics/testtools/>
Seach Google for "voip simulator".

Also note that the common Softphone products (Skype and Gizmoproject)
have built in diagnostics which display some of the performance
parameters.

There are some online measuring tools for VoIP. For example:
<http://myspeed.visualware.com/voip/>
The server is free for the first 2 weeks. Then, it's $400 to $1,200
per year. Ouch.
<http://www.myvoipspeed.com>

I don't think I really answered your question, but my comments might
give you a start in the right direction. For VoIP questions, you
might get better results in a VoIP specific blog or newsgroup. I
found plenty with Google and a few with Google Groups, but I don't
which ones are the best for VoIP questions.

Also, you might want to get the shift key on your keyboard repaired.
It seems to be non-functional.

--
# Jeff Liebermann 150 Felker St #D Santa Cruz CA 95060
# 831-336-2558 (E-Mail Removed)
# http://802.11junk.com (E-Mail Removed)
# http://www.LearnByDestroying.com AE6KS

Hi Jeff:

Thanks for your answers, I will have a look at the links you showed me.
I ll come back later to say what I used in the end, so the topic is
finished.
Regards

flo

for jitter and traffic stats im using Ethereal. There is an option in
stats -> RTP -> analize stream
this jitter value is calculated according to the RFC 3550 for RTP

I dont know how to get end-to-end delay yet

On Jan 4, 10:27 am, "vop...@gmail.com" <vop...@gmail.com> wrote:
> Hi Jeff:
>
> Thanks for your answers, I will have a look at the links you showed me.
> I ll come back later to say what I used in the end, so the topic is
> finished.
> Regards
>
> flo

On Jan 8, 5:29 am, "vop...@gmail.com" <vop...@gmail.com> wrote:
> for jitter and traffic stats im using Ethereal. There is an option in
> stats -> RTP -> analize stream
> this jitter value is calculated according to the RFC 3550 for RTP

Thanks. I didn't know it would do that. I just tried it with both a
GizmoProject and a Skype call using WireShark 0.99.4. I couldn't
capture any RTP packets. I'm doing something wrong. These might be of
interest:
<http://wiki.wireshark.org/VoIP_calls>
<http://wiki.wireshark.org/RTP_statistics>
<http://wiki.wireshark.org/RTP>
I'll play with it some more when the phone stops ringing. (I hate
mondays).

that is because skype use propietary coding scheemes, which means
ethereal cant see whats the payload, If you make the test with a sip
based client you can.
I have my own server with asterisk, which uses UDP/RTP

Cheers

flo
On Jan 8, 6:12 pm, "Jeff Liebermann" <j...@comix.santa-cruz.ca.us>
wrote:
> On Jan 8, 5:29 am, "vop...@gmail.com" <vop...@gmail.com> wrote:
>
> > for jitter and traffic stats im using Ethereal. There is an option in
> > stats -> RTP -> analize stream
> > this jitter value is calculated according to the RFC 3550 for RTPThanks. I didn't know it would do that. I just tried it with both a
> GizmoProject and a Skype call using WireShark 0.99.4. I couldn't
> capture any RTP packets. I'm doing something wrong. These might be of
> interest:
> <http://wiki.wireshark.org/VoIP_calls>
> <http://wiki.wireshark.org/RTP_statistics>
> <http://wiki.wireshark.org/RTP>
> I'll play with it some more when the phone stops ringing. (I hate
> mondays).