Got (Buffer) Bloat? 121
mtaht writes, "After a very intense month of development, the Bufferbloat project has announced the debloat-testing git kernel tree, featuring (via suggestion of Van Jacobson) a wireless network latency smashing algorithm, (called eBDP), the SFB and CHOKe packet schedulers, and a slew of driver level fixes that reduce network latency across the Linux kernel by over 2 orders of magnitude. Got Bloat?"
what it is (Score:1, Insightful)
Re:what it is (Score:5, Informative)
It's about the downside of memory becoming cheap causing latency problems with congestion control mechinisms that rely on the endpoints being able to inform the sender when it's sending too fast.
Jim Getty's research blog entry [wordpress.com] explains the problem in detail.
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oblig. car analogy, by Eric Raymond no less:
https://lists.bufferbloat.net/pipermail/bloat/2011-February/000050.html [bufferbloat.net]
== Packets on the Highway ==
To fix bufferbloat, you first have to understand it. Start by
imagining cars traveling down an imaginary road. They're trying to get
from one end to the other as fast as possible, so they travel nearly
bumper to bumper at the road's highest safe speed.
[snipped]
What bufferbloat is (Score:5, Informative)
My understanding may not be correct but:
Bufferbloat (I first came across the term bufferbloat in this blog post by Jim Gettys [wordpress.com]) is the nickname that has been given to the high latency that can occur in modern network connections due to large buffers in the network. An example could be the way that a network game on one computer starts to stutter if another computer starts to use a protocol like bittorrent to transfer files on the same network connection.
The large buffers seem to have arisen from a desire to maximise download throughput regardless of network condition. This can give rise to the situation where small urgent packets are delayed because big packets (which perhaps should not have been sent) are queued up in front of them. The system sending the big packets is not told to stop sending them so quickly because its packets are being delivered...
The linked article sounds like people have modified the Linux kernel source to allow people who know how to compile their own kernels to test ideas people have had for reducing the bufferbloat effect on their hardware and to report back their results.
Does this help explain things a bit?
Re:What bufferbloat is (Score:5, Interesting)
high latency that can occur in modern network connections due to large buffers in the network
Nobody ever explained this to me but I was using ping to measure latency on a network where I was actually most interested in ssh. Ping times went something like 10ms, 50ms, 90ms, 130ms... up to about 500ms, then started again at 10ms, 50ms and so on. Maybe some of my pings shared a buffer with a large, periodic data transfer and when that transfer filled a buffer somewhere my latency dropped.
I am pretty sure the people actually operating the WAN in question had no idea what was going on either.
Re:What bufferbloat is (Score:5, Informative)
I should have also linked to a definition of bufferbloat by Jim Gettys [wordpress.com]. For the curious here's a page of links to bufferbloat resources [bufferbloat.net] and a 5 minute animation that shows the impact of large buffers on network communication (.avi) [bufferbloat.net].
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Better link to the animation (Score:1)
Sorry Dave. A Coral cache version of the animation [nyud.net].
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Re:What bufferbloat is (Score:5, Informative)
The code changes to the Linux kernel also reduce the size and ill effects of buffers inside the kernel and drivers.
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My home internet connections have previously suffered from enormous buffers in the DSLAM - setting off a big download could cause ping times to increase to about 2 seconds, rendering other interactive use of the connection impossible.
Still, either this has been fixed now or more modern versions of TCP are more sophisticated, because it doesn't seem to happen any more - at least not to the same degree.
Re:What bufferbloat is (Score:4, Informative)
I think what you were seeing was more due to ATM overhead than the DSLAM trying to be cute with throttling. Because ADSL encapsulates everything in ATM even small IP / Ethernet frames get broken up into lots of ATM cells which can add upwards of 20% overhead. So an ADSL line trained at 8Mb/s will never provide 8Mb/s of usable throughput to the end user. Some ISPs actually advertise targeted throughput instead of train rate and set the train rate a certain percentage above the target throughput to compensate. Others just advertise train rates and have disclaimers in the fine print.
I've had my hands inside most Gen 1, 1.5 and 2nd Gen DSLAMs and never seen any with automatic throttling like you described.
(Gen 1 being units that just function at the ATM layer requiring an external system to bridge to Ethernet or IP. Gen 1.5's being upgraded Gen 1s with crude bridging, and Gen 2 being units that were designed to terminate connections directly from the ground up.)
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To be fair, it was probably the pppoe terminator that throttled your connection. It's the only device in the network that has the correct position to be able to throttle efficiently on a per-customer basis, for both the upstream and the downstream direction, as well as the information required to do this "dynamic" throttling you talk about.
The issue technical support has which leads to bufferbloat is equally simple "but your add said 10 Mbit download speed and I'm getting 9.8 according to speedtest.net". Th
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Re:what it is (Score:5, Informative)
Re:what it is (Score:5, Insightful)
What routers should do is keep track of how long packets have been in the router (in milliseconds or even microseconds) and use that with QoS stuff (and maybe some heuristics) to figure out which packets to send first, or to drop.
For example, "bulk/throughput" packets might be kept around for hundreds of milliseconds, but while latency sensitive packets get priority they are dropped if they cannot be sent within tens of milliseconds (then the sender will faster realize that it should slow down).
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That's a much more complex solution than "don't buffer so much damn stuff for no good reason."
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But you do need big buffers to be able to do fast single tcp transfers! You need at least rtt * bandwidth in buffer in any place that has a faster uplink than downlink, like distribution switches for instance. And that's several megabytes, per port, in the today's gigabit ethernet world. Otherwise you're going to get bad to horrible throughput for high latency transfers.
Now, big buffers also need a decent buffer management (just trivial RED is orders of magnitudes better than "lets just fill the buffer up a
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The problem with switches is that most switches have not merely small buffers, which would be ok, but microscopic ones. E.g. Cisco 3560G loses traffic on a gigabit port when faced with 50Mbps of bursty traffic in total coming from two ports. 10ms of buffer at 1Gbps is ~1MB, and most switches have nothing near that per port.
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This is an important point - and one most people are confused about. I'd like to add a nuance: my understanding is TX buffers are OK IF the router is very smarting about its queuing algorithm within the buffer, so that it drops packets early for any given sender so that the senders don't mistake a large buffer for a large pipe and overspeed the transmission. I believe Gettys &c just released (this article) is such a router/buffer queuing algorithm - smart enough to be effectively utilized with large buf
Who pointed it out? (Score:2)
Have you done the research to see just who you're disagreeing with about this?
And why they engineered TCP the way they did?
I won't pretend that I've walked through the experiments to try to verify their conclusions. I'm not even sure I know enough to interpret to interpret But...the people shouting the warnings aren't your average Chicken Littles.
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Have you considered they still could be wrong?
Personally I've never seen a buffer in software designed they way they described. I've never heard of hardware acting that way, but as you said they certainly know more than I.
I stopped reading when they said 'it waits for the buffer to fill up until sending', which is true on a per packet level for a lot of things at a end point, in transit everything I've ever dealt with will forward packets without waiting UNTIL the output link becomes too congested to do so
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Of course they could be wrong!
My only point here was: are you looking at their credentials?
As I understand it, you just re-stated the problem. Which is why they engineered TCP to work the way it does in the first place
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Your alternative isn't as simple as you'd like when you have many self interested clients playing zero sum competition over the router's bandwidth.
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Keeping big buffers but managing them better (Score:3)
The first problem is that a ton of transit systems on the Internet (like indeed a ton of systems everywhere) are effectively running the default behaviors in this respect, with no special tuning. That means FIFO with whatever queue size is available.
The second is that even if all the ISP operators decided to fix this, "QoS stuff" has the potential to run afoul of Network Neutrality. The current thi
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I'm proposing they use an AQM algorithm that isn't that stupid/random but rather based on the QoS AND _age_of_packet_. The latter I believe is important.
One can determine the QoS by fields in the packet header and/or guessing.
Guessing isn't necessarily that difficult or error prone - latency sensitive stuff uses mostly small packets (because bigger packets = higher latency). And high throughput stuff uses mostly big "max size" packets.
With my proposal if say a 1Mbps ADSL user gets a quick burs
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Please feel free to write some code. Writing a new qdisc for Linux and BSD is not very hard.
Writing a good qdisc, insanely so.
That said, I tend to feel that time-stamping more packets and doing more guessing may make sense, as does concepts in TCP vegas.
But: Before starting, read these:
http://pollere.net/Pdfdocs/bcit_6.2001.pdf [pollere.net] Kathleen Nichols - who proved to Van Jacobson that RED was wrong -
And Van Jacobson: http://pollere.net/Pdfdocs/QrantJul06.pdf [pollere.net]
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You may be right about some of your points, but (if I understand your point) you're wrong about the QoS and net neutral stuff. FCC has never indicated that net neutrality regulations will impinge on "reasonable network management" practices. If folks need to route certain kinds of packets or manage certain kinds of buffers in specific ways to get performant networks, that's just great as far they're concerned (or anyone else with a legislative/regulatory angle I've ever read about or talked with).
I correspo
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Indeed, when I referenced "net neutrality", I wasn't referring to the specific implementation by the FCC, but rather the concept itself. The actual language does include an exception for "reasonable network management", but many network neutrality proponents were (I think somewhat rightfully) concerned at the size and flexibility of such a loophole.
Several providers saw their networks being loaded up with bit-torrent, and believed that limiting that specific protocol would constitute "reasonable network
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I understand that QoS policies can be written neutral, my concern is that they won't be. Dismissing the actions of ISPs and carriers as "a market problem" doesn't really constitute useful policy. Similarly, ignoring all network hosts who aren't FLOSS apps written after 2011 is also not helpful.
The issues being examined in the bufferbloat discussions are not about whether there is a technically possible solution somewhere in the world (indeed, both ECN and AQM have been around for a while). The issues
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2) where you see a problem others may see a genuine business opportunity (I claim my proposal will give a better user experience than smaller buffers and/or stuff like RED).
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Particularly, with carriers throwing bandwidth at the core, this should be an interesting project for DD-WRT, since gateway
Re:what it is (Score:4, Informative)
re:"Interesting problem for dd-wrt"
Agreed.
We are throwing efforts at both the mainline kernel and openwrt.
Openwrt is foundational for dd-wrt and several other (commercial) distributions of Linux on the router. I have a large set of debloated routers already, I'm just awaiting further work on the eBDP algorithm to make better....
http://www.bufferbloat.net/projects/bloat/wiki/Experiment_-_Bloated_LAGN_vs_debloated_WNDR5700 [bufferbloat.net]
re: "using pings"
httpping is a much saner approach than ping, in many cases. Get it from:
http://www.vanheusden.com/httping/ [vanheusden.com]
re: RED & AQM
SFB and CHOKEe are in the debloat-testing kernel, as is eBDP.
RED 93 isn't going to work. nRED may. Experimentation and scripts highly desired. See the bloat and bloat-devel mailing lists for discussions.
Also:
http://www.bufferbloat.net/projects/bloat/wiki/Dogfood_Principle [bufferbloat.net]
Also:
I've seen some VERY interesting behavior with tcp vegas over bloated connections.
http://www.bufferbloat.net/projects/bloat/wiki/Experiment_-_TCP_cubic_vs_TCP_vegas [bufferbloat.net]
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1. Does said router properly respect QoS when deciding what data gets "rushed"?
2. Does said client have to pay a premium when sending out packets with elevated QoS?
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It's a myth that routers have the ability to tell a client to slow down, at least in the majority of environments (particularly ones with Ethernet segments, but other network types too).
Ethernet Flow Control has very limited utility here. You'll see it kick in in a rare few congestion cases - like if a switch backplane becomes overloaded - but it is used in a very limited number of situations and definitely will not be used end-to-end on a network for a router or host to tell another host (client) to do sl
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And you know, in my experience that's where one of the real problems - and one of the most commonly undiagnosed problems - exists. In nearly 99% of networks I've looked at where buffer overflows were occurring and drops were happening, network admins were not only unaware of the severity of packet drops and didn't understand the impact this was having on their *critical* workloads, they had no idea how to even look for it.
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Re:what it is (Score:4, Informative)
Oh come on, you only have to follow four links to get to the definition [wordpress.com]. What are you, lazy?
Seriously, this is a true failure of web design. You click from the summary, then you go to the wiki, then you go to the faq, and the faq doesn't even tell you, it references a blog post.
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Solution: Use a proper protocol (aka ISO) (Score:1)
Instead of using TCP/IP (bastardized version of ISO), people should start using real OSI implementations such as the ISO protocol, with 20 byte addresses and QoS level settings for each of the 7 OSI layers.
Once upon a time it was an issue of cost of h/w logic, IP was the cheaper alternative, today the difference is nil and the benefits of ISO are orders of magnitude better than IP.
bufferbloat, IP address exhaustion, etc are just a few of the reason why we should drop IP altogether.
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The difference is that you can write an smtp server by reading in strings line by line and treating them as commands, then watch the logs and kludge it until it seems to interoperate well enough. With the OSI way of doing things you have to wear a blue tie for a start then you have to print out all the interface definition documents and spread them out on your desk and write the software to the interface.
You correctly point out that IP is cheaper, but that means all the people who work with it will be cheap
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The difference is that you can write an smtp server by reading in strings line by line and treating them as commands, then watch the logs and kludge it until it seems to interoperate well enough. With the OSI way of doing things you have to wear a blue tie for a start then you have to print out all the interface definition documents and spread them out on your desk and write the software to the interface.
man.. I want your desk if you can spread out all the iso interface definition documents on it and be able to read them
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I'm not sure if this has anything to do with it (or we were just victims of slick salespeople):
Bach in the early days of networking, when I was at Boeing, we (engineering) were starting to write some client-server stuff. Every time our IT folks approached us with ISO/OSI networking products as recommendations, there always seemed to be licensing fees attached. Per seat, per process, per user, per CPU, per whatever. While the software gurus were negotiating licenses and contracts, we just said "Screw it. Gi
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Erm, that doesn't seem to be the problem here IMHO & as per RTFA the problem is a maladaptive response to packet loss by throwing cheap memory into dumb buffers that effectively break the whole packet loss concept. Packet loss is not the enemy of throughput it is 'big idea' behind maintaing it. But sitting in a bloated buffer is the enemy of throughput, seeing the Internet as as series of sealed pipes is the enemy of throughput, missing the point completely and connecting huge dumb buffers into you OS y
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Why is this only rated a 1?
This may be the best summary of the problem that I've seen yet.
That is a battle which was lost 20 years ago (Score:4, Interesting)
A lot of our problems today would not be here if.
OSI stack instead of TCP/IP.
DCE & DFS instead of passwd/whatever + the bastard abomination which is NFS.
Meh. People are lazy and cheap. Free with the network effect always wins. The Lowest Common Denominator. It's going to take another 15 years before we are near where we were 15 years ago. But this time it will be in Java!
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OSI stack instead of TCP/IP
Can you please elaborate?
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Call me an idiot, but I thought TCP/IP was part of the OSI stack.
I'd also like to hear an explanation.
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Well, no. In my (limited) experience, you'd use CLNP [wikipedia.org] instead of IP if you were using OSI. And instead of IP addresses you would have NSAP [wikipedia.org] addresses. It's a whole different world actually.
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Did you ever use those things? I've never used the OSI stack (though I have had the misfortune of looking at some of the specs), but
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Having never worked with the originals (Kerberos and Andrew File System), I don't know if this was a problem added in the "standardization" or if it came with the territory.
I can't speak about historical implementations, but the current (and I assume most modern implementations elsewhere) implementations of Kerberos used by Microsoft and the FreeBSD project can be configured for a system with a 5 line config file that could be generated from the output of a hostname -f call if the client is otherwise configured properly (Has its domain name set properly). It does require a proper DNS setup which can be obnoxious if you try to configure it by hand, but there again, its an impl
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Sure it will. You simply cap the customer at the bandwidth he/she is paying for and do QoS within the customer's allocation. Then you don't oversell the bandwidth. :-) Yeah, right. That'll happen.
Alternatively (and more realistically), you design things such that high priority packets must be sent isochronously (that is, every nth time slot) plus or minus a little jitter. A video codec will have no problem delivering frames at a constant data rate. Random bulk communications will not be able to do so,
Re:Solution: Use a proper protocol (aka ISO) (Score:4, Interesting)
You further penalize the endpoint by cutting it down to 56k modem speeds for a month if you see it making too many short-lived (<5s) connections to multiple IPs and declaring them isoch. By definition, any such connections are an abuse of the standard.
But isn't that exactly how channel surfing would work on IP-based television?
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No. That should generally look like a single, continuous stream from the head end at your cable company. Your machine sends back-channel messages to the head end to indicate that a channel change should occur, and it sends down a different stream. There's really no good reason to disconnect and reconnect when changing channels; doing so would constitute a rather serious abuse of the network due to all the overhead of setting up and tearing down an isochronous transport stream.
Now if by "changing channels
Which streams does this head end carry? (Score:2)
[Changing channels] should generally look like a single, continuous stream from the head end at your cable company. [...] Now if by "changing channels" you mean rapidly switching between separate streams from content sources on the Internet, then yes, but on the other hand, most streaming videos are almost invariably not being transmitted live
In other words, streams MUST NOT be live unless the viewer subscribes to a cable television service and the streams have been approved by the operator of such service.
In general, isoch connections should be used for live streams from special events, video chat, VoIP, and a single isoch connection to the head end for anything that resembles IPTV.
This raises the question: should each IPTV user have access to multiple head ends? News and sporting events appear to qualify as "special events" for this purpose, but someone might want to watch news and sports offered by a provider without a specific carriage agreement with the head end owned by your particular ISP. For example, a small or r
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No, it was an issue of the ISO specs being bloated and incomprehensible. The human cost had much more to do with their failure than the hardware cost.
Re:Two orders of magnitude! ? (Score:4, Interesting)
It is about reducing latency. So it helps prevent problems with VoIP failing when there is a lot of other data flowing over the same connection(s).
Some buffers are really large and it turns out some introduce latency of several seconds (!).
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Re:Two orders of magnitude! ? (Score:5, Interesting)
Most network throughput is at least 80-90% efficient already, so it won't get much faster. It will make it more responsive though, which is good if you're browsing the web, playing an online game or something else interactive.
I assume this is under load though, because on ping there's not much to be saved. On local sites I have 8-12 ms ping, on slashdot I have 140-150 ms. Since the theoretical round trip in a straight line at light speed is some 110 ms, there's not even room for a 50 ms drop. A lot of weirdness can happen under load though if stuff gets buffered up various places.
Re:Two orders of magnitude! ? (Score:5, Informative)
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Most network throughput is at least 80-90% efficient already, so it won't get much faster. It will make it more responsive though, which is good if you're browsing the web, playing an online game or something else interactive.
I assume this is under load though, because on ping there's not much to be saved. On local sites I have 8-12 ms ping, on slashdot I have 140-150 ms. Since the theoretical round trip in a straight line at light speed is some 110 ms, there's not even room for a 50 ms drop. A lot of weirdness can happen under load though if stuff gets buffered up various places.
Doesn't this also mean that under high bittorrent pressure, pings will not time out and and ssh will remain responsive.
Latency again (Score:4, Insightful)
I've seen it time and time again, people just generally don't care about latency, or even deny it exists in many cases (buffer bloat is certainly one cause of latency).
Everything from changing channel on your TV remote, to a mobile phone number entry, to the frame delays you get from LCD monitors, to the soundcard delay, to the GUI widgets you click on;......... it's all over the place, and it can wreck the experience, or reduce it somewhat according to how big the delay is. Just because latency is harder to measure, that doesn't mean it isn't very important, especially when it builds up with lots of other 'tiny' delays to make one big delay.
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That said, I wish we could settle on an Internet-wide QOS implementation and get both. Some packets have a legitimate need to cut in line. It would be workable if ISPs advertized both 'total' bandwidth and a smaller amount of 'turbo' bandwidth, or whatever stupid name they want to use for it, which is the fraction of your bandwidth that is not over-subscribed. By setti
Re:Latency again (Score:4, Insightful)
Re:Latency again (Score:4, Informative)
"Why does the web site load so slowly?" is the classic question - caused in many cases by the "eagleholic" having 4 live eagle nest video streams running in one window while trying to post observations and screencaps to the web site in another.
Believe me - there is ample reason to deal with the problem as most of today's home networks are used for more than just one thing at a time. Mom is watching video, sis is uploading pictures of her party to Facebook, son is playing online games and dad is trying to listen to streaming audio - and NOTHING is working correctly despite the fact that this is a trivial load for even a T1 (1.45Mbps) let alone today's high-speed cable (30Mbps down and 5Mbps up). We used to run 30+ modems and web sites and email and all manner of stuff over bonded 56K ISDN lines for pity sake - and we got better latency than the links today.
What's the problem? The latency for the "twitch" game packets has gone from 10ms to 4000ms or more - and the isochronos audio stream is jerky because it's bandwidth starved and the upload takes forever because the ACKs from FB can't get through the incoming video dump from YouTube (with its fast start window pushed from default 3 to 11 or 12) and by the time the video is half over, the link to YouTube has dropped because it took 30 seconds or more for the buffer to drain after the first push and the link had timed out.
That's the problem - you need low latency for some things at the same time you need high throughput for others - and it is possible and can be done - and IS done if things are tuned correctly. But correctly tuning the use of buffers is an art today, not a science - and the ever-changing (by 3-4 orders of magnitude) needs of today's end-point routers has pushed the limits of what AQM (automated queue management) algorithms are currently available, even if they're turned on (which in most cases they're not it seems)
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"I want everything louder than everything else" (Meat Loaf) epitomizes the net today - we have Google screwing with the fast start window and Microsoft pretty much ignoring it and setting it as large as possible in some cases (they do other things right though it seems)
The buffer bloat problem is one born of history and ignorance:
History - it used to be that we could not put enough buffer RAM i
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A poorly designed router can sabotage the performance TCP, causing overall slowness to your connections. Particularly, those 10Mbps you're paying for and want them to properly work.
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(Reasonable) packet loss or ECN - pick one - and then tell your up and downstream neighbors why you picked it (hopefully ECN will find its way into near 100% deployment ASAP) and why they should respec
Every packet is sacred (Score:5, Funny)
Every packet is great.
If a packet is wasted,
TCP gets quite irate.
Let the heathen drop theirs
When their RAM is spent.
TCP shall make them pay for
Each packet that can't be sent.
Every packet is wanted.
To this we are sworn.
From real-time data from CERN
To the filthiest of porn.
Every packet is sacred.
Every packet is great.
If a packet is wasted,
TCP gets quite irate.
.
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Best. Comment. Ever. That's going on my door at work - thanks.
Most delays are due to the ethernet packet buffers (Score:1)
Most delays are due to users connecting to their ADSL modem via Ethernet and not traffic managing properly.
On a congested link this can cause large delays as Ethernet normally has a 1000 packet buffer in the Linux kernel and the ADSL modem has a similar buffer. You only need a couple of heavy connections which want to go faster than the ADSL will support and those buffers start to fill up real fast. You can easily end up with latencies measured in seconds if you have a lot of connections running (say bitto
Buffer bloat is (not) an illusion... (Score:2)
After reading this guys buffer bloat rant I largly agree with him with some exceptions:
1. What does multiple TCP sessions have to do with circumvention of congestion avoidance? TCP congestion avoidance needs to work with lots and lots of TCP sessions at once
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Sort of in answer to both of your questions the bufferbloat.net servers are configured as follows:
http://www.bufferbloat.net/projects/bloat/wiki/Dogfood_Principle [bufferbloat.net]
trying at every point to make sure http 1.1 actually got used.
We survived today's slashdotting. Handily.
That said, your points are well made. SPDY is part of the chromium browser and looks to have some potential.
In my case, I like the idea of smarter - and eventually sctp-enabled - proxies, especially on wireless hops. See thread at:
https://lists.b [bufferbloat.net]
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Re: 1. I've always thought that the congestion window to the same end-point should be shared: but that's not the way TCP implementations work, and wishing they worked that way won't make the problem go away. And, as I've shown, bufferbloat is not a TCP phenomena in any case.
Re: 2. HTTP is a lousy protocol in and of itself, and having to do it on top of TCP makes it yet harder. It is the fact that HTTP is so ugly that makes so much else difficult. And I disagree with your claim that high latency links won't
Corrected Git URL (Score:4, Informative)
The link in the /. story to debloat-testing should go here: git://git.infradead.org/debloat-testing.git [git].
git:gitinfradeadorgdebloat-testinggit is not a valid URL.
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