[Bloat] Updated Bufferbloat Test

[Sina Khanifar]

> So perhaps this can feed into the rating system, total latency < 50mS is an A, < 150mS is a B, 600mS is a C or something like that.

The "grade" we give is purely a measure of bufferbloat. If you start
with a latency of 500 ms on your connection, it wouldn't be fair for
us to give you an F grade even if there is no increase in latency due
to bufferbloat.

This is why we added the "Real-World Impact" table below the grade -
in many cases people may start with a connection that is already
problematic for video conferencing, VoIP, and gaming.

I think we're going to change the conditions on that table to have
high 95%ile latency trigger the degraded performance shield warnings.
In the future it might be neat for us to move to grades on the table
as well.


On Thu, Feb 25, 2021 at 5:53 AM Simon Barber <simon at superduper.net> wrote:
>
> So perhaps this can feed into the rating system, total latency < 50mS is an A, < 150mS is a B, 600mS is a C or something like that.
>
> Simon
>
> On February 25, 2021 5:49:26 AM Mikael Abrahamsson <swmike at swm.pp.se> wrote:
>
>> On Thu, 25 Feb 2021, Simon Barber wrote:
>>
>>> The ITU say voice should be <150mS, however in the real world people are
>>> a lot more tolerant. A GSM -> GSM phone call is ~350mS, and very few
>>> people complain about that. That said the quality of the conversation is
>>> affected, and staying under 150mS is better for a fast free flowing
>>> conversation. Most people won't have a problem at 600mS and will have a
>>> problem at 1000mS. That is for a 2 party voice call. A large group
>>> presentation over video can tolerate more, but may have issues with
>>> talking over when switching from presenter to questioner for example.
>>
>>
>> I worked at a phone company 10+ years ago. We had some equipment that
>> internally was ATM based and each "hop" added 7ms. This in combination
>> with IP based telephony at the end points that added 40ms one-way per
>> end-point (PDV buffer) caused people to complain when RTT started creeping
>> up to 300-400ms. This was for PSTN calls.
>>
>> Yes, people might have more tolerance with mobile phone calls because they
>> have lower expectations when out and about, but my experience is that
>> people will definitely notice 300-400ms RTT but they might not get upset
>> enough to open a support ticket until 600ms or more.
>>
>> --
>> Mikael Abrahamsson    email: swmike at swm.pp.se
>
>