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Boosting Socket Performance on Linux
Cop writes "The Sockets API lets you develop client and server applications that can communicate across a local network or across the world via the Internet. Like any API, you can use the Sockets API in ways that promote high performance -- or inhibit it. This article explores four ways to use the Sockets API to squeeze the greatest performance out your application and to tune the GNU/Linux® environment to achieve the best results."
Agreed. In fact, as someone who learned socket coding around 1999/2000 (and as a result do not have a good grasp on how to actively define register variables, compilers do that stuff for you these days) I did all of these things out of habit, and didnt fully understand them until this article.
In the same line - where is the discussion of different FD table polling mechanisms? select() versus poll(), and wheres the writeup about Linux's epoll(). I would have been interested in an epoll() article, especially how it compares to FreeBSD's kqueue().
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...on developerWorks, not the least of which, if I may say so, is the GLib tutorial I wrote for them this past summer. If you wanted how to use various GLib collections and utilities - lists, tables, trees, quarks, relations, and all that - check it out. You can even download a nice PDF file for offline perusing.
Folks who are thinking about writing something technical - give dW a shot. The editors are savvy folks and there's lots of good stuff up there already.
Oh, and book plug!
The Army reading list
Fasterfox also trips a lot of traps intended to catch content stealing bots.
To get around the above problems, I came up with the following scheme: Leave Nagle's algorithm enabled, but create a FlushSocket() function that merely disables Nagle on the socket, then calls send() on the socket with a 0-byte buffer, then enables Nagle again. This apparently forces the TCP stack to immediately send any data that it may have accumulated in its Nagle-buffer. Therefore the only thing the calling code has to remember to do is to call FlushSocket() whenever it has called send() one or more times and doesn't think it will be sending any more data any time soon.
The above technique seems to work pretty well under Linux, Windows, and OS/X (and is more portable than Linux-specific flags like TCP_CORK, etc), but I haven't seen it documented anywhere. Is that simply an oversight, or is there some nasty downside to this technique that I'm overlooking?
I don't care if it's 90,000 hectares. That lake was not my doing.
imbsc but I vaguely recall in the early days of web browsers, they would pull down the base page, and then one image at a time. Netscape opening multiple requests in parallel seemed like a massive abuse of webserver resources at the time, to me at least.
455fe10422ca29c4933f95052b792ab2
Documentation like this is great and extremely valuable. It would be much more valuable, however, if it remained current. For example, can the ABISS project (which improves block I/O) be used at all? What do the numbers look like, when using profiling tools like Web100 (which profiles TCP communications)?
Has anyone run the Linux or one of the *BSD kernels through DAKOTA, KOJAK or PAPI to determine where, precisely, bottlenecks are within the kernels? It's easy to theorise, but isn't it cleaner to measure?
Now, I'm not saying these things aren't being done. They probably are, somewhere, by someone, but if the results aren't getting published we don't really know what impact what changes are going to have. The current method of evolving Operating System code in general is often a mix of personal theory and subjective experience based on non-random samples of activity. That can't really be a good way to do things, can it?
If I'm wrong, feel free to say. If I'm right, then maybe it would be a good thing if someone (possibly me) put together some kind of testing kit for measuring Linux kernel performance and actually measured the stats for Linux kernels on some kind of regular basis.
It's a small world and it smells funny; I'd buy another if it wasn't for the money; Take back what I paid (SoM)
I don't know if that really makes sense for networking though... the reason you'd want fflush() to block until the data makes it to disk is so that once your call to fflush() returns you know that your written data is safe in the event of a crash or power failure. (Although with too-clever hard drive firmware I'm not so sure even that's true anymore!). With networking on the other hand, even once the data has left your Ethernet port there is no guarantee that it will get to its destination... so what would be the purpose is waiting?
I don't care if it's 90,000 hectares. That lake was not my doing.
Here's the real problem, and its solution.
The concept behind delayed ACKs is to bet, when receiving some data from the net, that the local application will send a reply very soon. So there's no need to send an ACK immediately; the ACK can be piggybacked on the next data going the other way. If that doesn't happen, after a 500ms delay, an ACK is sent anyway.
The concept behind the Nagle algorithm is that if the sender is doing very tiny writes (like single bytes, from Telnet), there's no reason to have more than one packet outstanding on the connection. This prevents slow links from choking with huge numbers of outstanding tinygrams.
Both are reasonable. But they interact badly in the case where an application does two or more small writes to a socket, then waits for a reply. (X-Windows is notorious for this.) When an application does that, the first write results in an immediate packet send. The second write is held up until the first is acknowledged. But because of the delayed ACK strategy, that acknowledgement is held up for 500ms. This adds 500ms of latency to the transaction, even on a LAN.
The real problem is that 500ms unconditional delay. (Why 500ms? That was a reasonable response time for a time-sharing system of the 1980s.) As mentioned above, delaying an ACK is a bet that the local application will reply to the data just received. Some apps, like character echo in Telnet servers, do respond every time. Others, like X-Windows "clients" (really servers, but X is backwards about this), only reply some of the time.
TCP has no strategy to decide whether it's winning or losing those bets. That's the real problem.
The right answer is that TCP should keep track of whether delayed ACKs are "winning" or "losing". A "win" is when, before the 500ms timer runs out, the application replies. Any needed ACK is then coalesced with the next outgoing data packet. A "lose" is when the 500ms timer runs out and the delayed ACK has to be sent anyway. There should be a counter in TCP, incremented on "wins", and reset to 0 on "loses". Only when the counter exceeds some number (5 or so), should ACKs be delayed. That would eliminate the problem automatically, and the need to turn the "Nagle algorithm" on and off.
So that's the proper fix, at the TCP internals level. But I haven't done TCP internals in years, and really don't want to get back into that. If anyone is working on TCP internals for Linux today, I can be reached at the e-mail address above. This really should be fixed, since it's been annoying people for 20 years and it's not a tough thing to fix.
The user-level solution is to avoid write-write-read sequences on sockets. write-read-write-read is fine. write-write-write is fine. But write-write-read is a killer. So, if you can, buffer up your little writes to TCP and send them all at once. Using the standard UNIX I/O package and flushing write before each read usually works.
John Nagle
best for the internet as a whole
are you sure?
From a paper written by Phil Dykstra, back in 1999.
"A recent example comes from the Pacific Northwest Gigapop in Seattle which is based on a collection of Foundry gigabit ethernet switches. At Supercomputing '99, Microsoft and NCSA demonstrated HDTV over TCP at over 1.2 Gbps from Redmond to Portland. In order to achieve that performance they used 9000 byte packets and thus had to bypass the switches at the NAP! Let's hope that in the future NAPs don't place 1500 byte packet limitations on applications."
Ok, forget it mentions the M word, this article is about using jumbo frames (9000 byte packets) instead of the 1500 byte ones that were originally specced in 1980 (back when ethernet was.. not quite as fast as it is today).
Seeing as how the internet as a whole is based on this packet size, and the article (http://sd.wareonearth.com/~phil/jumbo.html) describes the stunning performance gains that can be had with jumbo frames, the internet as a whole is actually being held back significantly by it (ie. increase the frame buffer by 6, you get about a 40 times throughput)(bigger frames than 9000 bytes are not practical due to other TCP design limitations).
His recommendations are - if you're on a LAN, enable jumbo frames today.
IPv6 will not have this restriction and so will be faster, maybe things like HDTV on demand will drive its adoption on the internet.