Re: [Bloat] DSLReports Speed Test has latency measurement built-in

[jb <justin@dslr.net>]

[-- Attachment #1: Type: text/plain, Size: 2744 bytes --]

I am working on a multi-location jitter test (sorry PDV!) and it is showing
a lot of promise.
For the purposes of reporting jitter, what kind of time measurement horizon
is acceptable
and what is the +/- output actually based on, statistically ?

For example - is one minute or more of jitter measurements, with the +/-
being
the 2rd std deviation, reasonable or is there some generally accepted
definition ?

ping reports an "mdev" which is
SQRT(SUM(RTT*RTT) / N – (SUM(RTT)/N)^2)
but I've seen jitter defined as maximum and minimum RTT around the average
however that seems very influenced by one outlier measurement.

thanks

On Thu, May 7, 2015 at 2:29 PM, Mikael Abrahamsson <swmike@swm.pp.se> wrote:

> On Wed, 6 May 2015, Jonathan Morton wrote:
>
>  So, as a proposed methodology, how does this sound:
>>
>> Determine a reasonable ballpark figure for typical codec and jitter-buffer
>> delay (one way). Fix this as a constant value for the benchmark.
>>
>
> Commercial grade VoIP systems running in a controlled environment
> typically (in my experience) come with 40ms PDV (Packet Delay Variation,
> let's not call it jitter, the timing people get upset if you call it
> jitter) buffer. These systems typically do not work well over the Internet
> as we here all know, 40ms is quite low PDV on a FIFO based Internet access.
> Applications actually designed to work on the Internet have PDV buffers
> that adapt according to what PDV is seen, and so they can both increase and
> decrease in size over the time of a call.
>
> I'd say ballpark reasonable figure for VoIP and video conferencing of
> reasonable PDV is in the 50-100ms range or so, where lower of course is
> better. It's basically impossible to have really low PDV on a 1 megabit/s
> link because a full size 1500 byte packet will take close to 10ms to
> transmit, but it's perfectly feasable to keep it under 10-20ms when the
> link speed increases. If we say that 1 megabit/s (typical ADSL up speed)is
> the lower bound of speed where one can expect VoIP to work together with
> other Internet traffic, then 50-100ms should be technically attainable if
> the vendor/operator actually tries to reduce bufferbloat/PDV.
>
>  Measure the maximum induced delays in each direction.
>>
>
> Depending on the length of the test, it might make sense to aim for 95th
> or 99th percentile, ie throw away the one or few worst values as these
> might be outliers. But generally I agree with your proposed terminology.
>
> --
> Mikael Abrahamsson    email: swmike@swm.pp.se
> _______________________________________________
> Bloat mailing list
> Bloat@lists.bufferbloat.net
> https://lists.bufferbloat.net/listinfo/bloat
>

[-- Attachment #2: Type: text/html, Size: 3666 bytes --]