this is what bufferbloat has been fighting. The default was that
'data is important, don't throw it away, hang on to it and send it
later'
In practice, this has proven to be suboptimal as the buffers grew
large enough that the data being buffered was retransmitted anyway
(among other problems)
And because the data was buffered, new data arriving was delayed
behind the buffered data.
This is measurable as 'latency under load' for light connections. So
while latency isn't everything, it turns out to be a good proxy to
detect when the standard queuing mechansisms are failing to give you
good performance.
It turns out that not all data is equally important. Active Queue
Management is the art of deciding priorities, both in deciding what
data to throw away, but also in allowing some later arriving data to
be transmitted ahead of data in another connection that arrived
before it.
With fq_codel and cake, this involves tracking the different
connections and their behavior. connections that send relatively
little data (DNS lookups, video chat) have priority over connections
that send a lot of data (ISO downloads), not based on classifying
the data, but by watching the behavior.
connections with a lot of data can buffer a bit, but aren't allowed
to use all the available buffer space, after they have used 'their
share', packets get marked/dropped to signal the sender to slow
down.
While it is possible for an implementation to 'cheat' by detecting
the latency probes and prioritizing them, measuring the latency on
real data works as well and they can't cheat on that without
actually addressing the problem
That's why you see such a significant focus on latency in this
group, it's not latency for the sake of latency, it's latency as a
sign that new and sparse flows can get a reasonable share of
bandwith even in the face of heavy/hostile users on the same links.
David Lang
On Fri, 8 Mar 2024, Jack Haverty via Nnagain wrote:
Date: Fri, 8 Mar 2024 15:44:05 -0800
From: Jack Haverty via Nnagain <nnag...@lists.bufferbloat.net>
To: nnag...@lists.bufferbloat.net, Starlink@lists.bufferbloat.net
Cc: Jack Haverty <j...@3kitty.org>
Subject: [NNagain] When Flows Collide?
It's great to see that latency is getting attention as well as
action to control it. But it's only part of the bigger picture of
Internet performance.
While performance across a particular network is interesting, most
uses of the Internet involve data flowing through several separate
networks. That's pretty much the definition of "Internet". The
endpoints might be some kind of LAN in a home or corporate IT
facility or public venue. In between there might be fiber, radio,
satellite, or other (even whimsically avian!?) networks carrying a
users data. This kind of system configuration has existed since
the genesis of The Internet and seems likely to continue. Technology
has advanced a lot, with bigger and bigger "pipes" invented to carry
more data, but fundamental issues remain.
System configurations we used in the early research days were real
experiments to be measured and tested, or often just "thought
experiments" to imagine how the system would behave, what algorithms
would be appropriate, and what protocols had to exist to coordinate
the activities of all the components.
One such configuration was very simple. Imagine there are three
very fast computers, each attached to a very fast LAN. The
computers and LAN can send and receive data as fast as you can
imagine, so that they are not a limiting factor. The LANs are
attached to some "ISP" which isn't as fast (in bandwidth or latency)
as a LAN. ISPs are interconnected at various points, forming a
somewhat rich mesh of topology with several, or many, possible
routes from any source to any destination.
Now imagine a user configuration in which two of the computers send
a constant stream of data to the third computer at a predefined
rate. Perhaps it is a UDP datagram every N milliseconds, each
datagram containing a frame of video. If N=20 it corresponds to a
50Hz frame rate, which is common for video.
Somewhere along the way to that common destination, those two data
streams collide, and there is a bottleneck. All the data coming in
cannot fit in the pipe going out. Something has to give.
Thought experiment -- What should happen? Does the bottleneck
discard datagrams it can't handle? How does it decide which ones to
discard? Does the bottleneck buffer the excess datagrams, hoping
that the situation is just temporary? Does the bottleneck somehow
signal back to the sources to reduce their data rate? Does th
ebottleneck discard datagrams that it knows won't reach the
destination in time to be useful? Does the bottleneck trigger some
kind of network reconfiguration, perhaps to route "low priority"
data along some alternate path to free up capacity for the video
streams that requires low latency?
Real experiment -- set up such a configuration and observe what
happens, especially from the end-users' perspectives. What kind of
video does the end-user see?
Second thought experiment -- Using the same configuration, send data
using TCP instead of UDP. This adds more mechanisms, but now in the
end-users' computers. How should the ISPs and TCPs involved behave?
How should they cooperate? What should happen? What mechanisms
(algorithms, protocols, etc.) are needed to make the system behave
that way?
Second real Experiment -- How do the specific TCP implementations
actually behave? What kind of video quality do the end users
experience? What kind of data flows actually travel through the
network components?
Of course we all observe such real experiments every day, whenever
we see or participate in various kinds of videoconferences. Perhaps
someone has instrumented and gathered performance data...?
These questions were discussed and debated at great length more than
40 years ago as TCP V4 was designed. We couldn't figure out the
appropriate algorithms and protocols, and didn't have computer
equipment or communications capabilities to implement anything more
than the simplest mechanisms anyway. So the topic became an item
on the "future study" list.
But we did put various "placeholder" mechanisms in place in TCP/IP
V4, as a reminder that a "real" solution was needed for some future
next generation release. Time-to-live (TTL) would likely need to be
based on actual time instead of hops - which were silly but the best
we could do with available equipment at the time. Source Quench
(SQ) needed to be replaced by a more effective mechanism, and
include details of how all the components should act when sending or
receiving an SQ. Routing needed to be expanded to add the ability
to send different data flows over different routes, so that bulk and
interactive data could more readily coexist. Lots of such issues
to be resolved.
In the meanwhile, the general consensus was that everything would
work OK as long as the traffic flows only rarely created
"bottleneck" situations, and such events would be short and
transitory. There wasn't a lot of data flow yet; the Internet was
still an Experiment. We figured we'd be OK for a while as the
research continued and found solutions.
Meanwhile, the Web happened. Videoconferencing, vlogs, and other
generators of high traffic exploded. Clouds have formed, with users
now interacting with very remote computers instead of the ones on
their desks or down the hall.
As Dorothy would say, "We're not in Kansas anymore".
Jack Haverty
On 3/8/24 12:31, Dave Taht via Nnagain wrote:
I am deeply appreciative of everyones efforts here over the past 3
years, and within starlink burning the midnight oil on their 20ms
goal, (especially nathan!!!!) to make all the progress made on their
systems in these past few months. I was so happy to burn about 12
minutes, publicly, taking apart Oleg's results here, last week:
https://www.youtube.com/watch?v=N0Tmvv5jJKs&t=1760s
But couldn't then and still can't talk better to the whys and the
problems remaining. (It's not a kernel problem, actually)
As for starlink/space support of us, bufferbloat.net, and/or
lowering
latency across the internet in general, I don't know. I keep hoping
a
used tesla motor for my boat will arrive in the mail one day, that's
all. :)
It is my larger hope that with this news, all the others doing FWA,
and for that matter, cable, and fiber, will also get on the stick,
finally. Maybe someone in the press will explain bufferbloat. Who
knows what the coming days hold!?
13 herbs and spices....
On Fri, Mar 8, 2024 at 3:10 PM the keyboard of geoff goodfellow
via
Starlink<starlink@lists.bufferbloat.net> wrote:
it would be a super good and appreciative gesture if they would
disclose what/if any of the stuff they are making use of and then
also to make a donation :)
On Fri, Mar 8, 2024 at 12:50 PM J Pan<p...@uvic.ca> wrote:
they benefited a lot from this mailing list and the research and
even
user community at large
--
J Pan, UVic CSc, ECS566, 250-472-5796 (NO VM),p...@uvic.ca,
Web.UVic.CA/~pan
On Fri, Mar 8, 2024 at 11:40 AM the keyboard of geoff goodfellow
via
Starlink<starlink@lists.bufferbloat.net> wrote:
Super excited to be able to share some of what we have been working
on over the last few months!
EXCERPT:
Starlink engineering teams have been focused on improving the
performance of our network with the goal of delivering a service
with stable 20 millisecond (ms) median latency and minimal packet
loss.
Over the past month, we have meaningfully reduced median and
worst-case latency for users around the world. In the United States
alone, we reduced median latency by more than 30%, from 48.5ms to
33ms during hours of peak usage. Worst-case peak hour latency (p99)
has dropped by over 60%, from over 150ms to less than 65ms. Outside
of the United States, we have also reduced median latency by up to
25% and worst-case latencies by up to 35%...
[...]
https://api.starlink.com/public-files/StarlinkLatency.pdf
via
https://twitter.com/Starlink/status/1766179308887028005
&
https://twitter.com/VirtuallyNathan/status/1766179789927522460
--
geoff.goodfel...@iconia.com
living as The Truth is True
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