Hi Dave, On Apr 27, 2015, at 18:39 , Dave Taht <[email protected]> wrote:
> On Fri, Apr 24, 2015 at 11:03 PM, Sebastian Moeller <[email protected]> wrote: >> Hi Simon, hi List >> >> On Apr 25, 2015, at 06:26 , Simon Barber <[email protected]> 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 ;) ) > > At one level I am utterly prepared to set new (And lower) standards > for latency, and not necessarily pay attention to compromise driven > standards processes established in the 70s and 80s, but to the actual > user experience numbers that jim cited in the fq+aqm manefesto on his > blog. I am not sure I git the right one, could you please post a link to the document you are referring to? My personal issue with new standards is that it is going to be harder to convince others that these are real and not simply selected to push our agenda., hence using other peoples numbers, preferably numbers backed up by research ;) I also note that in the ITU numbers I dragged into the discussion the measurement pretends to be mouth to ear (one way) delay, so for intermediate buffering the thresholds need to be lower to allow for sampling interval (I think typically 10ms for the usual codecs G.711 and G.722), further sender processing and receiver processing, so I guess for the ITU thresholds we should subtract say 30ms for processing and then doube it to go from one-way delay to RTT. Now I am amazed how large the resulting RTTs actually are, so I assume I need to scrutinize the psycophysics experiments that hopefully underlay those numbers... > > I consider induced latencies of 30ms as a "green" band because that is > the outer limit of the range modern aqm technologies can achieve (fq > can get closer to 0). There was a lot of debate about 20ms being the > right figure for induced latency and/or jitter, a year or two back, > and we settled on 30ms for both, so that number is already a > compromise figure. Ah, I think someone brought this up already, do we need to make allowances for slow links? If a full packet traversal is already 16ms can we really expect 30ms? And should we even care, I mean, a slow link is a slow link and will have some drawbacks maybe we should just expose those instead of rationalizing them away? On the other hand I tend to think that in the end it is all about the cumulative performance of the link for most users, i.e. if the link allows glitch-free voip while heavy up- and downloads go on, normal users should not care one iota what the induced latency actually is (aqm or no aqm as long as the link behaves well nothing needs changing) > > It is highly likely that folk here are not aware of the extra-ordinary > amount of debate that went into deciding the ultimate ATM cell size > back in the day. The eu wanted 32 bytes, the US 48, both because that > was basically a good size for the local continental distance and echo > cancellation stuff, at the time. > > In the case of voip, jitter is actually more important than latency. > Modern codecs and coding techniques can tolerate 30ms of jitter, just > barely, without sound artifacts. >60ms, boom, crackle, hiss. Ah, and here is were I understand why my simplistic model from above fails; induced latency will contribute significantly to jitter and hence is a good proxy for link-suitability for real-time applications. So I agree using the induced latency as measure to base the color bands from sounds like a good approach. > > >> Best Regards >> Sebastian >> >> >>> >>> Simon >>> >>> Sent with AquaMail for Android >>> http://www.aqua-mail.com >>> >>> >>> On April 24, 2015 9:04:45 PM Dave Taht <[email protected]> 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 <[email protected]> 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 <[email protected]> wrote: >>>>>> >>>>>>> Sebastian Moeller <[email protected]> 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 >>>>>> [email protected] >>>>>> https://lists.bufferbloat.net/listinfo/bloat >>>>> >>>>> >>>>> _______________________________________________ >>>>> Bloat mailing list >>>>> [email protected] >>>>> 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 >>> [email protected] >>> 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 [email protected] https://lists.bufferbloat.net/listinfo/bloat
