Jitter is variation in delay. All networks, encoding and decoding
processes add delay. If you only had delay (and no packet loss) you
would not need buffering as you would know always when the next voice
frame was going to arrive you could jjust play the voice sample. But
because of congestion, multiple network paths, and other variations
there will be always times when a voice frame arrives late. Because you
always want to have something play (not just silence) the designers of a
VoIP system always will have buffer at least as long maximum tolerable
delay variation or jitter, plus some breathing space. Unfortunately I
expect that jitter is subject to bell curves, etc where while average
jitter might by say 20ms, occasionaly you might get jitter of 50 ms and
very occasionally 100ms. The VoIP designer needs to make a compromise
in choosing the tolerable jitter - between buffer size which increases
the minimum delay and voice sample loss.
----Original Message-----
From: ethereal-users-bounces@xxxxxxxxxxxx
[mailto:ethereal-users-bounces@xxxxxxxxxxxx] On Behalf Of John Graves
Sent: Tuesday, 12 April 2005 7:19 AM
To: Ethereal user support
Subject: Re: [Ethereal-users] Jitter Measurement
Scott,
If you highlight a RTP stream, and go Statistics > RTP > Stream Analysis
It shows you both sides of an RTP conversation, and one of the
statistics is Jitter. I was looking at that and going back to the
inter-packet times and trying to figure out how Jitter is measured.
What I see in a lot of sessions is a growth and decline of Jitter across
the session, and usually one side is worse than the other. I am trying
to understand if jitter is an important variable in diagnosing voice
that goes into the toilet or just another interesting set of numbers.
Is this an estimated amount of jitter or is this absolute? Can I use
this as a sign that the voip traffic starts to degrade with only 10-15
ms of jitter. Specs on the equipment say that it can tolerate up to 100
ms of jitter before the buffer blows.
John G.
Scott Lowrey wrote:
> I don't know how Ethereal measures jitter. (Does it? :)
>
> RFC 3550 explains the concept and the required formulas for RTP/RTCP.
> In a nutshell, jitter is a statistical measurement of the packet
> inter-arrival time variation in a stream, and is expressed in units of
> time.
>
> There are other definitions of jitter, most dealing with electronic
> communications and circuitry.
>
> John Graves wrote:
>
>> Looking at the numbers for packets included in the RTP stream
>> analysis, it is not clear to me how jitter is measured. Can someone
>> point me to an explanation of this or explain how it is determined?
>>
>
> --
> *Scott Lowrey*
> Test Engineering Manager
> NexTone Communications <http://nextone.com> Gaithersburg, Maryland USA
>
> /1.240.912.1369/
>
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--
John Graves
Dynamic Devices Inc.
781-245-9100
Your source for
Security, Wireless LANS & Wide Area Networking for converged Data, Voice
and Video
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