[Bloat] Updated Bufferbloat Test

[Simon Barber]

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

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