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Bufferbloat — the Submarine That's Sinking the Net
gottabeme writes "Jim Gettys, one of the original X Window System developers and editor of the HTTP/1.1 spec, has posted a series of articles on his blog detailing his research on the relatively unknown problem of bufferbloat. Bufferbloat is affecting the entire Internet, slowly worsening as RAM prices drop and buffers enlarge, and is causing latency and jitter to spike, especially for home broadband users. Unchecked, this problem may continue to deteriorate the usability of interactive applications like VOIP and gaming, and being so widespread, will take years of engineering and education efforts to resolve. Being like 'frogs in heating water,' few people are even aware of the problem. Can bufferbloat be fixed before the Internet and 3G networks become nearly unusable for interactive apps?"
http://en.wikipedia.org/wiki/X_Window_System
I'm so glad the term has been defined so that I know what the hell we're talking about here. Oh wait, no it hasn't.
Okay, then I'll RTFA. Oh wait, two screens worth of text later and it still hasn't.
I'd like to change the topic now to the submarine that's sinking the English language: jargonbloat.
#naabhaprzrag, #sverubfr-000, #agi-fcbafberq, negvpyr[pynff*=' negvpyr-ary-'] { qvfcynl: abar !vzcbegnag; }
Jim Getty, one of the original X Window System developers and editor of the HTTP/1.1 spec
I'd murder four people just to have TTY in my name. Five if I could capitalize them, and postfix with a number. I'd name my son Dev.
You'd get a business card with something like Dev GeTTY1, Armadillo Avenue 64, Seattle, Washington
8 of 13 people found this answer helpful. Did you?
Just start RTFAing: "In my last post I outlined the general bufferbloat problem."
Follow the link:
"Each of these initial experiments were been designed to clearly demonstrate a now very common problem: excessive buffering in a network path. I call this bufferbloat
I read TFA and I'm not seeing the problem. He can't duplicate this issue unless he maxes out his connection and then his latency goes to hell. No shit Sherlock, that's what happens when your pipe is full and the packets have to wait in the queue to be transmitted. Am I stupid or could he avoid this issue entirely by using QoS and/or rate-limiting his connection to some amount <100% of it's maximum throughout? I have QoS at the office that keeps our connection from pegging (it's limited to around 75% on the download and 90% on upload) and have never once encountered an issue with latency or jitter. At home I only throttle the upload (to 90% of maximum) and have successfully ran VPNs, bittorrent uploads and VoIP calls all at the same time without any headaches.
Really, what's the problem here?
I want peace on earth and goodwill toward man.
We are the United States Government! We don't do that sort of thing.
http://www.cringely.com/2011/01/2011-predictions-one-word-bufferbloat-or-is-that-two-words/
Red Leader Standing By!
RAM is cheap.
High speed uplink is not cheap.
Peering agreements are manipulative, expensive, and sometimes extortionate.
So...
The poorly designed, poorly peered, under allocated back haul links can't handle the traffic that routers want to push through them -- but since RAM is cheap, operators just add RAM to the buffers so that when those back-haul lines slow down for a second the packets can get pushed through.
And we're blaming the buffer for the problem?
The problem with quotes on the internet, is that nobody bothers to check their veracity. -- Abraham Lincoln
Several issues:
1. People who aren't networking engineers don't know about QoS, or don't know/want to know how to configure it.
2. QoS used that way is a hack to work around an issue that doesn't have to be there in the first place
3. How do you determine the maximum throughput? It's not necessarily the official line's speed. The nice thing about TCP is that it's supposed to figure out on its own how much bandwidth there is. You're proposing a regression to having to tell the system by hand.
4. QoS is most effective on stuff you're sending, but in the current consumer-oriented internet most people download a lot more than they upload.
What Jim is saying is that TCP flows try to train themselves to the dynamically available bandwidth, such that there is a minimum of dropped packets, retransmits, etc.
But in order for TCP to do this, packets must be dropped _fast_.
When TCP was designed, the assumptions about the price of ram (and thus, the amount of onboard memory in all the devices in the virtual circuit) were different -- namely, buffers were going to be smaller, fill up faster, and send "i'm full" messages backwards much sooner.
What the experimentation has determined is that many network devices will buffer 1 megabyte or MORE of traffic before finally dropping something and telling the tcp originator to slow down. And yet with a 1 meg buffer and a rate of 1 megabyte per second.. it will take 1 second simply to drain the buffer.
The pervasive presence of large buffers all along the tcp vc, and the non-speified or tail-drop drop behavior of these large queues means that tcp's ability to rate limit is effectively nullified, and in situations where the link is highly utilized, many degenerate behaviors occur, such that the overall link has extremely high latency, and that bulk traffic will cause interesting traffic to be randomly dropped.
Personally, I used pf/altQ on openBSD to try and manage this somewhat.. but its a dicey business.
My opinions are my own, and do not necessarily represent those of my employer.
Really, what's the problem here?
You really don't see the problem? How can you be so naive. Maybe you're new to this. All signs show to the fact there is a problem.
Of course the problem is not obvious. The article itself says it'll completely surprise us. They know we won't believe it at first. But that's why we must believe it, or else it's Armageddon.
Would you risk an Armageddon, because of your inability to understand and see?
And that's, in short, why we must attack Iraq.
Wait, what were we talking about :P?
As an extreme example, say you request a 1GB file from a download site. That site has a monster internet connection, and manages to transmit the entire file in 1 second. The file makes it to the ISP at that speed, who then buffers the packets for slow transmission over your ADSL link, which will take 1 hour. During that time you try to browse the web, and your PC tries to do a dns lookup. The request goes out ok, but the response gets added to the buffer on the ISP side of your internet connection, so you won't get it until your original transfer completes. How's 1 hour for latency?
The situation is only not that bad because:
A: Most download sites serve so many people at once and/or rate limit so they won't saturate most peoples' connections
B: Most buffers in network hardware are still quite small
He's not arguing against application-level caching. He's saying that too much caching at the IP layer is confusing TCP's algorithm for deciding how fast the link between two points is. This in turn causes massive variability in how fast the data can be downloaded; or in your terms, how fast the video can be buffered (and, in fact, how much buffer the video player needs).
Rgasuya aata! : I have been coding Perl and cannot tell where my fingers are now!
After reading TFSeries, the problem is excessive buffering (as in 1-10 or more seconds worth of data) screwing up TCP/IP's automatic bandwidth detection. QoS helps a little bit by getting the important packets (especially ACKs) through, but high-bandwidth TCP connections are still going nuts when they hit a slower link with excessive buffering.
And one of the major offenders is Linux commonly defaulting to a txqueuelen of 1000.
#naabhaprzrag, #sverubfr-000, #agi-fcbafberq, negvpyr[pynff*=' negvpyr-ary-'] { qvfcynl: abar !vzcbegnag; }
but in the current consumer-oriented internet most people download a lot more than they upload.
Because the current consumer infrastructure forces it onto you. I would happily seed my torrents all year long, except I only have 1/12th the uploading bandwidth as I have for downloading. Since I need some of it for other things, uploading becomes impractical.
It's easy to blame the consumer, but there's a certain model imposed on him from the start.
Seven puppies were harmed during the making of this post.
If you put a frog in a pot of water and slowly raise the temperature it will try to jump out before the water reaches a temperature that is fatal to the frog.
Best Slashdot Co
Mod parent up. Half way down the comments, and this is the first post that actually explains why 'bufferbloat' is something I should care about.
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You haven't read the article (or the many others around on LWN.net on the same topic). Basically, large buffers in networking gear, from DSL routers on your home network through to ISP's, mean that interactivity is *shite*. You might download Gb's but in terms of interactive applications it's useless and we're facing ever-increasing latency and problems through wanting to cope too much with errors and delays (e.g. huge buffers to keep resending instead of just letting packets drop and having TCP sort it out by retransmission). TCP windows never shrink because errors and buffered and retried so much from intermediate devices that any sort of window scaling is worthless because it doesn't *see* any packet-loss.
Same devices, smaller buffers, everything works fine and "faster" / "more responsive" all around. It actually would *save* money on new devices because you don't need some huge artificial buffer, you can just drop the occasional packet. But the problem is so deeply embedded into run-of-the-mill hardware that it's almost impossible to escape at the moment and thus EVERYONE from large businesses to home users are running on a completely sub-optimal setup because of it. Almost every networking device made in the last few years has buffers so large that they cause problems with interactivity, bandwidth control, QoS, etc. It's NOT just that a "faster connection" solves the problem - we are getting a percentage of optimal service that's steadily decreasing as buffers increase even though we're improving all the time. That's the point. And it *is* caused by memory prices because memory is so cheap that a huge thoughtless buffer costs no more than a tiny, thought-out buffer.
So naturally, I instantly get modded down.
Given that most traffic on a domestic connection is incoming, that doesn't help much.
It's not that hard to shape downstream traffic. Take a Linux router with two ethernet cards. eth0 is the LAN and eth1 is the internet. You shape eth0 with a maximum throughput of 75%-80% of your line speed. All of the downstream traffic has to go out on that interface so that's your opportunity to shape it. I do this at work and successfully share a 3.0mbit/s connection with 60+ employees. We use latency sensitive services like VoIP and RDP alongside streaming video and other large downloads without any major hassles. It stinks to lose some of your bandwidth because of this (you have to shape it to a number less than 100% of your line speed, otherwise buffering occurs at your ISP and your QoS scheme is defeated) but I'll take responsiveness over throughout any day of the week.
I want peace on earth and goodwill toward man.
We are the United States Government! We don't do that sort of thing.
This is an excellent explanation of what issues are happening here. I can clearly see that this is an issue, and the problem is something that over time will impact everybody.
The problem is really focused on trying to deal with differences in bandwidth between computers... always a problem but in this case trying to match up slow connections with fast connections is particularly difficult. Since memory is cheap, a 1 GB buffer certainly can be found in some devices now and perhaps much more. I don't see this example as being really too far off the mark in the near future.... which is the point being raised and why buffer bloat is such a big deal.
More to the point, some of the complaints that triggered the "quality of service" debate are rooted in this problem. As mentioned in the original article triggering this whole slashdot thread, setting up "quality of service" priorities only creates multiple buffer queues.... it doesn't solve the problem of the monster queue to begin with. That is why the author of the blog post suggests that the debate over network neutrality is not based upon the real problem that is facing network engineering and why it is a political solution in search of a problem.
It takes awhile to "grok" this problem, but once you do it becomes obvious why this is such a huge deal.
I've no idea if this post explains it correctly or not (one of the replies implies that it doesn't), but if it does, it should be nearer the top of the page, hence my posting it here. :-)
OK, not on the (intentionally ridiculous) scale used in the example, but people are doing something very similar to what you describe, even though they "can't do that". http://slashdot.org/article.pl?sid=10/11/26/1729218
# cat
Damn, my RAM is full of llamas.
2. QoS used that way is a hack to work around an issue that doesn't have to be there in the first place
3. How do you determine the maximum throughput? It's not necessarily the official line's speed. The nice thing about TCP is that it's supposed to figure out on its own how much bandwidth there is. You're proposing a regression to having to tell the system by hand.
4. QoS is most effective on stuff you're sending, but in the current consumer-oriented internet most people download a lot more than they upload.
While the Internet in-theory is beautiful, our modern implementation really is a series of layered hacks. And the solution to Bufferbloat is going to be another hack. You're crazy if you think that the solution to the Bufferbloat 'problem' is going to be some fundamental redesign of the TCP protocol (how would you force 10 people to use it?), or the total re-architecture of millions of consumer devices to remove buffering. You're also crazy if you think the ISPs and backbone providers are going to stand by while this thing kills the Internet.
So the question is: which hack will it be? The GP poster already identified one that works well enough --- using QoS to control flows. Your final objection about content providers stressing connections is the real one. But there's some probably a good hack to deal with it --- or more likely a series of hacks, some at the content providers themselves (e.g., Netflix), some in the backbone, and some at your ISP. It won't be elegant, but it will keep this problem from ever becoming anything more than a few cranky blog posts.
let me summarize the problem that is being observed: On a given interface, if you have more buffer memory than is needed as packet buffer on the transmit side, it can induce latency. As an example, consider a 1Mb/s link. If you want to have a peak of .1s latency added by buffering at high load, then you want 1Mb*.1=12,500 bytes of buffer. If you have 1MB of buffer, then you have 8 seconds of buffer, therefore triggering the "buffer bloat" issue. Part of the problem is that buffer size would be set based on the top speed a piece of hardware could drive, i.e. if you want a 1Gb/s interface to be able to buffer .1s, then you use it at 100Mb/s, then it has 1s worth of buffer. In most home deployments where you have a router that may have a 1Gbps upstream, maybe 4 100Mb/s physical connections, and a 54Mbps wireless router, you probably have a shared buffer for all the interfaces. The result of this is that when using the 54Mb/s wireless, you can easily have the buffer over saturated, while the buffer size may be just right for the 100Mb/s interfaces.
What is the solution to this? Realistically, the alternative is to drop packets that have resided in the buffer longer than a configured amount of time, which causes it's own performance issues. Net result: TCP would slowdown for a period of time, but would speed up again resulting in a sawtooth behavior. This would result in periodic issues with other protocols as well, i.e. VOIP would have dropped packets every time TCP ramps up again, etc.
Solution: Don't download porn when you are trying to do VOIP calls.
Latency is bad? Bigger buffers = more latency?
Buffers increasing latency is not exactly a new phenomena. Its been observed and taken into design considerations for quite some time. For example back-in-the-day serial chips essentially had a buffer of one byte. The CPU fed data one byte at a time as the buffer became available and latency was pretty low since data was immediately transmitted. As more capable serial chips became available larger buffers were introduced. A newer chip may have a larger buffer but it may also not transmit data as soon as it has a single byte. It was common to have two programmable thresholds to begin a data transmission, (1) when a certain amount of data has accumulated in the buffer or (2) when a certain amount of time has elapsed. So if a "packet" to transmit was small enough it may sit in the buffer until (2), hence more latency with larger buffers. Software that cared generally began to issue flush commands to cause anything in the buffer to be sent immediately.
Network cards and/or the operating system may try to similarly accumulate data before transmitting a packet.
I discovered this series of blog posts about 2 months ago, when he accidentally published one of his blog posts prematurely. I started reading it and followed the links and saw that this was a like a sleuth tale-if I had started reading this with his very first blog on the topic I would have had no idea where he was going with this. Now as to why this contribution by Jim Gettys does the world a great service:
Hats of to Jim Gettys. Thanks for your service.
Within a router it would be the actual IP data packets that are being buffered. A standard router has a number of network interfaces (token ring, ethernet, wireless, ISDN, whatever....) . Each network interface is piece of hardware that is memory mapped to allow the CPU to send and receive packets. Each hardware device also has a small online memory buffer to store the most recently received or transmitted addressed data packets (every protocol layer down to the MAC source and destination address, IP address, sequence number as well as the data). Depending on system and packet size, that could be anything between 1 and 16.
The usual implementation was to have each hardware device generate an interrupt whenever some data had been received and to transfer the data from internal memory to a common pool in system RAM. The latter was divided up into pre-allocated blocks with a few large blocks (>1000 bytes) and many smaller blocks (512 bytes). Some one might have done a statistical analysis onto the theoretical distribution of the size of packet data being sent through the network. Most of the time this worked out, but there were problems that happened some times. If all the smaller blocks were in use, then the larger blocks were used instead. For efficiency, these wouldn't be transferred through the system, until all the entire block has been filled up with data, so if you have a stream of 128 byte packets, it would take eight of them before the larger block was filled. For some systems, packet sizes were enhanced to 4K or even 8K. A constant high-speed stream of small packets was most likely to do this.
Also, many of the hardware devices would simply overwrite the contents of one unprocessed data packet with the contents of the latest arrival if it wasn't collected fast enough. So that could really mess up sequence numbers.
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There is no 'bufferbloat because RAM is getting cheaper'. What he is seeing is what happens when you want to saturate your link. ... ...you get either a buffered or a dropped packet.
Yes, and if a link is saturated, there should be packet drops, which TCP senses, then automatically throttles back to reduce the required bandwidth and avoid saturation. But what is happening, is that these huge buffers are holding packets that would otherwise be dropped, and so TCP doesn't get the feedback it needs to detect saturation. So it continues transmitting at full speed, believing it has uncongested pipes, which in turn continues to fill the buffers, and so on.
Because of the buffers, most of these packets are eventually getting through, but maybe in seconds instead of tens or low hundreds of milliseconds. Thus you're getting huge latency.
Jitter is caused by the buffers eventually filling or TCP timing out (registering packet loss), dropping the rate for a little bit, the buffers draining, then TCP upping the rate again as the buffers refill, hiding the saturation, until they're full again. Rinse and repeat.
It's related to the "bloat" of buffering (due to the increasing affordability of RAM and the "more of a good thing must be better than a little of a good thing - QED" mindset) because, if the size of the buffer is kept below a certain point related to the pipe bandwidth and number of traffic streams, it tends to act just as a temporary "buffer" against spikes in the traffic (the intention of buffering), and can't cause the scenario above, having insufficient capacity to overload the bandwidth just from buffer contents alone. Above this threshold, the latency issues and back-and-forth thrashing noted above occurs. The bigger the buffers, the worse the effect.
And it's not just a "well, keep your traffic below x mbit if you're on ADSL2" issue, because it happens anywhere a high capacity pipe interfaces with a low capacity or otherwise congested (of any capacity) pipe. This might be your ISP's backbone which is getting hit by several thousand people downloading the latest WOW patch simultaneously, causing your 300kbps Skype call to go to hell through latency and jitter. If the ISP's equipment had smaller buffers, the servers would be throttling back as packet loss occurred. You'd probably still be losing packets, but they'd be detected and re-transmitted pretty quickly and you possibly wouldn't notice the latency or have jitter.
What he is seeing is what happens when you want to saturate your link.
So, no, what you get with appropriate buffers is your TCP connection moderating itself to the appropriate link capacity and availability, and latency remaining approximately the same (relative to what you're seeing in bufferbloat, but worse than an uncongested link, obviously).
With bufferbloat, your bandwidth appears to remain about the same, but your latency balloons massively and you get jitter effects as above.
How much bandwidth can I have, though? Take the link between my desktop and a Slashdot server; is the correct answer "1GBit/s, no more" (speed of my network card)? Is is "20MBit/s, no more" (speed of my current Internet connection)? Is it "0.5MBit/s, no more" (my fair share of this office's Internet connection)? In practice, you need the answer to change rapidly, depending on network conditions - maybe I can have the full 20MBit/s if no-one else is using the Internet, maybe I should slow down briefly while someone else handles their e-mail.
TCP doesn't slam the network; it starts off slowly (TCP slow start currently sends just two packets initially), and gradually ramps up as it finds that packets aren't dropped. When packet drop happens, it realises that it's pushing too hard, and drops back. If there's been no packet drop for a while, it goes back to trying to ramp up. RFC 5681 talks about the gory details. It's possible (bar idiots with firewalls that block it) to use ECN (explicit congestion notification) instead of packet drop to indicate congestion, but the presence of people who think that ECN-enabled packets should be dropped (regardless of whether congestion has happened) means that you can't implement ECN on the wider Internet.
This works well in practice, given sane buffers; it dynamically shares the link bandwidth, without overflowing it. Bufferbloat destroys this, because TCP no longer gets the feedback it expects until the latency is immense. As a result, instead of sending typically 20MBit/s (assuming I'm the only user of the connection), and occasionally trying 20.01MBit/s, my TCP stack tries 20.01MBit/s, finds it works (thanks to the queue), speeds up to 20.10MBit/s, and still no failure, until it's trying to send at (say) 25MBit/s over a 20MBit/s bottleneck. Then packet loss kicks in, and brings it back down to 20MBit/s, but now the link latency is 5 seconds, not 5 milliseconds.
I appear to have a blog. Odd.