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

[Simon Barber]

The key figure for VoIP is maximum latency, or perhaps somewhere around 
99th percentile. Voice packets cannot be played out if they are late, so 
how late they are is the only thing that matters. If many packets are early 
but more than a very small number are late, then the jitter buffer has to 
adjust to handle the late packets. Adjusting the jitter buffer disrupts the 
conversation, so ideally adjustments are infrequent. When maximum latency 
suddenly increases it becomes necessary to increase the buffer fairly 
quickly to avoid a dropout in the conversation. Buffer reductions can be 
hidden by waiting for gaps in conversation. People get used to the acoustic 
round trip latency and learn how quickly to expect a reply from the other 
person (unless latency is really too high), but adjustments interfere with 
this learned expectation, so make it hard to interpret why the other person 
has paused. Thus adjustments to the buffering should be as infrequent as 
possible.

Codel measures and tracks minimum latency in its inner 'interval' loop. For 
VoIP the maximum is what counts. You can call it minimum + jitter, but the 
maximum is the important thing (not the absolute maximum, since a very 
small number of late packets are tolerable, but perhaps the 99th percentile).

During a conversation it will take some time at the start to learn the 
characteristics of the link, but ideally the jitter buffer algorithm will 
quickly get to a place where few adjustments are made. The more 
conservative the buffer (higher delay above minimum) the less likely a 
future adjustment will be needed, hence a tendency towards larger buffers 
(and more delay).

Priority queueing is perfect for VoIP, since it can keep the jitter at a 
single hop down to the transmission time for a single maximum size packet. 
Fair Queueing will often achieve the same thing, since VoIP streams are 
often the lowest bandwidth ongoing stream on the link.

Simon

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On May 7, 2015 6:16:00 AM jb <justin at dslr.net> wrote:

> I thought would be more sane too. I see mentioned online that PDV is a
> gaussian distribution (around mean) but it looks more like half a bell
> curve, with most numbers near the the lowest latency seen, and getting
> progressively worse with
> less frequency.
> At least for DSL connections on good ISPs that scenario seems more frequent.
> You "usually" get the best latency and "sometimes" get spikes or fuzz on
> top of it.
>
> by the way after I posted I discovered Firefox has an issue with this test
> so I had
> to block it with a message, my apologies if anyone wasted time trying it
> with FF.
> Hopefully i can figure out why.
>
>
> On Thu, May 7, 2015 at 9:44 PM, Mikael Abrahamsson <swmike at swm.pp.se> wrote:
>
> > On Thu, 7 May 2015, jb wrote:
> >
> >  There is a web socket based jitter tester now. It is very early stage but
> >> works ok.
> >>
> >> http://www.dslreports.com/speedtest?radar=1
> >>
> >> So the latency displayed is the mean latency from a rolling 60 sample
> >> buffer, Minimum latency is also displayed. and the +/- PDV value is the
> >> mean difference between sequential pings in that same rolling buffer. It is
> >> quite similar to the std.dev actually (not shown).
> >>
> >
> > So I think there are two schools here, either you take average and display
> > + / - from that, but I think I prefer to take the lowest of the last 100
> > samples (or something), and then display PDV from that "floor" value, ie
> > PDV can't ever be negative, it can only be positive.
> >
> > Apart from that, the above multi-place RTT test is really really nice,
> > thanks for doing this!
> >
> >
> > --
> > Mikael Abrahamsson    email: swmike at swm.pp.se
> >
>
>
>
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