Re: [aqm] Is bufferbloat a real problem?

Daniel Havey <dhavey@gmail.com> Fri, 27 February 2015 19:29 UTC

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Date: Fri, 27 Feb 2015 11:28:58 -0800
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From: Daniel Havey <dhavey@gmail.com>
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Subject: Re: [aqm] Is bufferbloat a real problem?
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Switching to gmail because Yahoo is too horrible to use.

Like I said, I agree that bufferbloat is only 1 part of a two sided
problem.  The queue with respect to each flow can be either two large
or too small (and sometimes just right?).  Anyways, I'm just wondering
if there are any measurement studies out there that show the effects
of this problem in a formal way.

It might be an impossible problem.  What provider is going to want to
help publish data that shows they have poor queue sizing?  Anyways,
I'm looking for a way to demonstrate that this is a real problem in a
dissertation defense.  Perhaps when asked the question I should just
launch into a long explanation about how the queue can be too small or
too large or even both if there are multiple flows in the queue and
that solving the problem requires more than just shortening the queues
or using AQM to tune them to a particular RTT?  I dunno.

It would be nice if we had some measurement studies, but, that might
not be possible given the realities that we face on the Internet
today.

...Daniel





On Fri, Feb 27, 2015 at 10:52 AM, Curtis Villamizar
<curtis@ipv6.occnc.com> wrote:
> In message <2134947047.1078309.1424979858723.JavaMail.yahoo@mail.yahoo.com>
> Daniel Havey writes:
>
>>
>> I know that this question is a bit ridiculous in this community.  Of
>> course bufferbloat is a real problem.  However, it would be nice to
>> formally address the question and I think this community is the right
>> place to do so.
>>
>> Does anybody have a measurement study?  I have some stuff from the FCC
>> Measuring Broadband in America studies, but, that doesn't address
>> bufferbloat directly.
>>
>> Let me repeat for clarity.  I don't need to be convinced.  I need
>> evidence that I can use to convince my committee.
>>
>> ...Daniel
>>
>> _______________________________________________
>> aqm mailing list
>> aqm@ietf.org
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>
>
> Daniel,
>
> You are convinced.  So am I but we need to temper the message to avoid
> going too far in the opposite direction - too little buffer.
>
> If you are not interested in the details, just skip to the last
> paragraph.
>
> Bufferbloat should in principle only be a problem for interactive
> realtime traffic.  Interactive means two way or multiway.  This is
> SIP, Skype, audio and video conferencing, etc.  In practice it is also
> bad for TCP flows with short RTT and max window set small.
>
> One way realtime (such as streaming audio and video) should be
> unnafected by all but huge bufferbloat.  That is should be.  For
> example, youtube video is carried over TCP and is typically either way
> ahead of the playback or choppy (inadequate aggregate bandwidth or
> marigninal and/or big drop with TCP stall).  It would be nice if
> marginal aggregate bandwidth was dealt with by switching to a lower
> bandwidth encoding, but too often this is not the case.  This doesn't
> mean that some streaming formats don't manage to get this wrong and
> end up delay sensitive.
>
> TCP needs buffering to function correctly.  Huge bufferbloat is bad
> for TCP, particularly for small transfers that never get out of TCP
> slow start and for short RTT flows.  For long RTT flows too little
> buffer causes problems.
>
> [ aside: For example, if TCP starts with 4 segments at 1K segment
> size, it will take 4 RTT to hit 64KB window, the typical max window
> without TCP large window (TCPLW) option.  During that time, 60KB will
> be sent.  After that 64KB will be sent each RTT.  With geographic RTT
> is 70 msec (approximate US continental RTT due to finite speed of
> light in fiber and fiber distance), 60 KB is sent in the first 280
> msec and 64KB gets sent every 70 msec yielding 7 mb/s.  OTOH if there
> is a server 2 msec RTT away (1 msec one way is 125mi = 200km), then
> 60KB in first 8 msec and 256 Mb/s after that.  If there is 100 msec
> buffer at a bottleneck, then this low RTT TCP flow will be slowed by a
> factor of 50.  OTOH, if bottlenecks have a lot less than 1 RTT of
> buffer, then the long TCP flows will get even further slowed. ]
>
> One of the effects of some buffer, but not excessive, is short RTT
> flows which given the no-TCPLW max window get slowed down while longer
> RTT are less affected.  This becomes more fair wrt to transfer rates
> among TCP flows.  The same holds true if TCPLW gets turned on in
> commodity gadgets and the commonly deploed max window increases, but
> the number change.
>
> If the buffer grows a little and the deployed window sizes become the
> limiting factor, then this is very light congestion with delay but
> absolutely zero loss due to queue drops (not considering AQM for the
> moment).
>
> Some uses of TCP increase the window to work better over long RTT.  It
> takes a bit longer to hit the max window but the rate once it has been
> hit is greater.  Setting TCP window large on short RTT flows is
> counterproductive since one or a small number of flows can cause a
> bottleneck on a slow provider link (ie: 10-100 Mb/s range typical of
> home use).  On a LAN RTT can be well under 1 msec on Ethernet and
> highly variable on WiFi.  On WiFi larger window can contribute to some
> real trouble.  So best that the default window be changed.
>
> [ Note that the work on automatic sizing of tcp_sndbuf and scp_recvbuf
> may create a tendency to saturate links as the window can go up to 2MB
> with default parameters.  Has this hit consumer devices yet?  This
> could be bad it this rolls out before widespread use of AQM. ]
>
> When a small amount of loss occurs, such as one or much less than the
> current window size, TCP cuts the current window size in half and
> retransmits the packet for the window in flight (ignoring selective
> acknowledgment extension aka SACK for the moment).
>
> If the buffer is way too small, then a large amount of premature drop
> occurs when the buffer limit is hit.  Lots of TCP flows slow down.
> The long RTT flows slow down the most.  Some retransmission occurs
> (which doesn't help congestion).  If there is a long period of drop
> relative to a short RTT, then a entire window can be dropped and this
> is terrible for TCP (slow start is initiated after delay based on an
> estimate of RTT and RTT stdev, or 3 sec if RTT estimate is stale -
> this is a TCP stall).  So with too little buffer some TCP flows get
> hammered and stall.  TCP flows with long RTT tend to stall less but
> are more sensitve to the frequency of drop events and can get
> extremely slow due to successively cutting window in half and then
> growing the window linearly rather than exponentially.
>
> With tiny buffers really bad things tend to happen.  The rate of
> retransmission can drive goodput (the amount of non-retransmit traffic
> per time) can drop substantially.  Long RTT flows can become
> hopelessly slow.  Stalls become more common.  In the worst case (which
> has been observed in a ISP network during a tiny buffer experiment
> about a decade ago, details in private email) TCP synchronization can
> occur, and utilization and goodput drop dramatically.
>
> A moderate amount of buffer is good for all TCP.  A large buffer is
> good for long RTT TCP flows, particularly those that have increased
> max window.  As mentioned before, any but a very small buffer is bad
> for interactive real time applications.
>
> Enter AQM.  A large buffer can be used but with a lower target delay
> and some form of AQM to introduce a low rate of isolated drops as
> needed to slow the senders.  Avoiding queue tail drop events where a
> lot of drops occur over an interval lowers the amount of
> retransmission and avoids stalls.  Long RTT flows tend to get
> penalized the most.
>
> Fairness is not great with a single queue and AQM but this is much
> better than a single queue with either small or large buffer and tail
> drop.  Fairness is greatly improved with some form of FQ or SFQ.
>
> Ideally with FQ each flow would get its own queue.  In practice this
> is not the case but the situation is greatly improved.  A real time
> flow, which is inherently rate limited, would see minimal delay and no
> loss.  A short RTT flow would see a moderate increase in delay and a
> low level of loss (ie: typically much less than 1%) enough to slow it
> down enough to avoid congestion.  a long RTT flow would see a moderate
> increase in delay and no loss if still running slower than the small
> RTT flows.  This does wonders for fairness and provides the best
> possible service for each service type.
>
> In practice, some FQ or SFQ queues have a mix of real time, low RTT
> TCP, and high RTT TCP.  If any such queue is taking a smaller share
> than other queues, delay is low and loss is low or zero.  If such a
> queue is taking more than its share, then the situation is similar to
> the single queue case.  Less flows end up in such a queue.  Cascaded
> queues have been proposed and in some cases (no longer existing) have
> been implemented.  In a cascaded SFQ scheme, the queues taking more
> than their share are further subdivied.  Repeat the subdivision a few
> times and you can end up with the large bandwidth contributors in
> their own queue and getting a fair share of capacity.
>
> So excuse the length of this but solving bufferbloat is *not* a silver
> bullet.  Not understanding that point and just making buffers really
> small could result in an even worse situation than we have now.
>
> Curtis
>
> ps - Some aspects of this may not reflect WG direction.  IMHO- the
> down sides of just making buffers smaller and/or setting low delay
> targets may not be getting enough (or any) attention in the WG.  Maybe
> discussion wouldn't hurt.
>
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