[Bloat] DSLReports Speed Test has latency measurement built-in
Jim Gettys
jg at freedesktop.org
Wed May 6 11:30:22 EDT 2015
On Wed, May 6, 2015 at 4:50 AM, Sebastian Moeller <moeller0 at gmx.de> wrote:
> Hi Simon,
>
> On May 6, 2015, at 07:08 , Simon Barber <simon at superduper.net> wrote:
>
> > Hi Sebastian,
> >
> > My numbers are what I've personally come up with after working for many
> years with VoIP - they have no other basis.
>
> I did not intend to be-little such numbers at all, I just wanted
> to propose that we either use generally accepted scientifically measured
> numbers or make such measurements our self.
>
> > One thing is that you have to compare apples to apples - the ITU numbers
> are for acoustic one way delay.
>
> True, and this is why we easily can estimate the delay cost of different
> stages of the whole voip one-way pipeline to deuce how much latent budget
> we have for the network (aka buffer bloat on the way), but still bases our
> numbers on some reference for mouth-to-ear-delay. I think we can
> conservatively estimate the latency cost of the sampling, sender processing
> and receiver processing (outside of the de-jitter buffering) seem harder to
> estimate reliably, to my untrained eye.
>
> > The poor state of jitter buffer implementations that almost every VoIP
> app or device has means that to hit these acoustic delay numbers you need
> significantly lower network delays.
>
> I fully agree, and if we can estimate this I think we can justify
> deductions from the mouth-to-ear budget. I would as first approximation
> assume that what we call latency under load increase to be tightly
> correlated with jitter, so we could take our “bloat-measurement” in ms an
> directly deduct it from the budget (or if we want to accept occasional
> voice degradation we can pick a sufficiently high percentile, but that is
> implementation detail).
>
> > Also note that these numbers are worst case, which must include trip
> halfway around the globe - if you can hit the numbers with half globe
> propagation then you will hit much better numbers for 'local calls’.
>
> We could turn this around by estimating to what distance voip
> quality will be good/decent/acceptable/lughable…
>
>
>
Mean RTT is almost useless for VOIP and teleconferencing. What matters is
the RTT + jitter; a VOIP or teleconferencing application cannot function at
a given latency unless the "drop outs" caused by late packets is low enough
to not be obnoxious to human perception; there are a number of techniques
to hide such late packet dropouts but all of them (short of FEC) damage the
audio stream.
So ideally, not only do you measure the delay, you also measure the jitter
to be able to figure out a realistic operating point for such applications.
- Jim
>
>
>
> >
> > Simon
> >
> >
> > On 4/24/2015 11:03 PM, Sebastian Moeller wrote:
> >> Hi Simon, hi List
> >>
> >> On Apr 25, 2015, at 06:26 , Simon Barber <simon at superduper.net> wrote:
> >>
> >>> Certainly the VoIP numbers are for peak total latency, and while
> Justin is measuring total latency because he is only taking a few samples
> the peak values will be a little higher.
> >> If your voip number are for peak total latency they need
> literature citations to back them up, as they are way shorter than what the
> ITU recommends for one-way-latency (see ITU-T G.114, Fig. 1). I am not
> "married” to the ITU numbers but I think we should use generally accepted
> numbers here and not bake our own thresholds (and for all I know your
> numbers are fine, I just don’t know where they are coming from ;) )
> >>
> >> Best Regards
> >> Sebastian
> >>
> >>
> >>> Simon
> >>>
> >>> Sent with AquaMail for Android
> >>> http://www.aqua-mail.com
> >>>
> >>>
> >>> On April 24, 2015 9:04:45 PM Dave Taht <dave.taht at gmail.com> wrote:
> >>>
> >>>> simon all your numbers are too large by at least a factor of 2. I
> >>>> think also you are thinking about total latency, rather than induced
> >>>> latency and jitter.
> >>>>
> >>>> Please see my earlier email laying out the bands. And gettys'
> manifesto.
> >>>>
> >>>> If you are thinking in terms of voip, less than 30ms *jitter* is what
> >>>> you want, and a latency increase of 30ms is a proxy for also holding
> >>>> jitter that low.
> >>>>
> >>>>
> >>>> On Fri, Apr 24, 2015 at 8:15 PM, Simon Barber <simon at superduper.net>
> wrote:
> >>>>> I think it might be useful to have a 'latency guide' for users. It
> would say
> >>>>> things like
> >>>>>
> >>>>> 100ms - VoIP applications work well
> >>>>> 250ms - VoIP applications - conversation is not as natural as it
> could be,
> >>>>> although users may not notice this.
> >> The only way to detect whether a conversation is natural is if
> users notice, I would say...
> >>
> >>>>> 500ms - VoIP applications begin to have awkward pauses in
> conversation.
> >>>>> 1000ms - VoIP applications have significant annoying pauses in
> conversation.
> >>>>> 2000ms - VoIP unusable for most interactive conversations.
> >>>>>
> >>>>> 0-50ms - web pages load snappily
> >>>>> 250ms - web pages can often take an extra second to appear, even on
> the
> >>>>> highest bandwidth links
> >>>>> 1000ms - web pages load significantly slower than they should, taking
> >>>>> several extra seconds to appear, even on the highest bandwidth links
> >>>>> 2000ms+ - web browsing is heavily slowed, with many seconds or even
> 10s of
> >>>>> seconds of delays for pages to load, even on the highest bandwidth
> links.
> >>>>>
> >>>>> Gaming.... some kind of guide here....
> >>>>>
> >>>>> Simon
> >>>>>
> >>>>>
> >>>>>
> >>>>>
> >>>>> On 4/24/2015 1:55 AM, Sebastian Moeller wrote:
> >>>>>> Hi Toke,
> >>>>>>
> >>>>>> On Apr 24, 2015, at 10:29 , Toke Høiland-Jørgensen <toke at toke.dk>
> wrote:
> >>>>>>
> >>>>>>> Sebastian Moeller <moeller0 at gmx.de> writes:
> >>>>>>>
> >>>>>>>> I know this is not perfect and the numbers will probably require
> >>>>>>>> severe "bike-shedding”
> >>>>>>> Since you're literally asking for it... ;)
> >>>>>>>
> >>>>>>>
> >>>>>>> In this case we're talking about *added* latency. So the ambition
> should
> >>>>>>> be zero, or so close to it as to be indiscernible. Furthermore, we
> know
> >>>>>>> that proper application of a good queue management algorithm can
> keep it
> >>>>>>> pretty close to this. Certainly under 20-30 ms of added latency.
> So from
> >>>>>>> this, IMO the 'green' or 'excellent' score should be from zero to
> 30 ms.
> >>>>>> Oh, I can get behind that easily, I just thought basing the
> limits
> >>>>>> on externally relevant total latency thresholds would directly tell
> the user
> >>>>>> which applications might run well on his link. Sure this means that
> people
> >>>>>> on a satellite link most likely will miss out the acceptable voip
> threshold
> >>>>>> by their base-latency alone, but guess what telephony via satellite
> leaves
> >>>>>> something to be desired. That said if the alternative is no
> telephony I
> >>>>>> would take 1 second one-way delay any day ;).
> >>>>>> What I liked about fixed thresholds is that the test would
> give a
> >>>>>> good indication what kind of uses are going to work well on the
> link under
> >>>>>> load, given that during load both base and induced latency come
> into play. I
> >>>>>> agree that 300ms as first threshold is rather unambiguous though
> (and I am
> >>>>>> certain that remote X11 will require a massively lower RTT unless
> one likes
> >>>>>> to think of remote desktop as an oil tanker simulator ;) )
> >>>>>>
> >>>>>>> The other increments I have less opinions about, but 100 ms does
> seem to
> >>>>>>> be a nice round number, so do yellow from 30-100 ms, then start
> with the
> >>>>>>> reds somewhere above that, and range up into the deep red / purple
> /
> >>>>>>> black with skulls and fiery death as we go nearer and above one
> second?
> >>>>>>>
> >>>>>>>
> >>>>>>> I very much think that raising peoples expectations and being quite
> >>>>>>> ambitious about what to expect is an important part of this. Of
> course
> >>>>>>> the base latency is going to vary, but the added latency
> shouldn't. And
> >>>>>>> sine we have the technology to make sure it doesn't, calling out
> bad
> >>>>>>> results when we see them is reasonable!
> >>>>>> Okay so this would turn into:
> >>>>>>
> >>>>>> base latency to base latency + 30 ms:
> green
> >>>>>> base latency + 31 ms to base latency + 100 ms: yellow
> >>>>>> base latency + 101 ms to base latency + 200 ms: orange?
> >>>>>> base latency + 201 ms to base latency + 500 ms: red
> >>>>>> base latency + 501 ms to base latency + 1000 ms: fire
> >>>>>> base latency + 1001 ms to infinity:
> >>>>>> fire & brimstone
> >>>>>>
> >>>>>> correct?
> >>>>>>
> >>>>>>
> >>>>>>> -Toke
> >>>>>> _______________________________________________
> >>>>>> Bloat mailing list
> >>>>>> Bloat at lists.bufferbloat.net
> >>>>>> https://lists.bufferbloat.net/listinfo/bloat
> >>>>>
> >>>>> _______________________________________________
> >>>>> Bloat mailing list
> >>>>> Bloat at lists.bufferbloat.net
> >>>>> https://lists.bufferbloat.net/listinfo/bloat
> >>>>
> >>>>
> >>>> --
> >>>> Dave Täht
> >>>> Open Networking needs **Open Source Hardware**
> >>>>
> >>>> https://plus.google.com/u/0/+EricRaymond/posts/JqxCe2pFr67
> >>>
> >>> _______________________________________________
> >>> Bloat mailing list
> >>> Bloat at lists.bufferbloat.net
> >>> https://lists.bufferbloat.net/listinfo/bloat
> >
>
> _______________________________________________
> Bloat mailing list
> Bloat at lists.bufferbloat.net
> https://lists.bufferbloat.net/listinfo/bloat
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://lists.bufferbloat.net/pipermail/bloat/attachments/20150506/0e23dae6/attachment-0002.html>
More information about the Bloat
mailing list