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Yandrek Vegueria Vega. Beneath the java. TCP handels all package synchronization, retransmitting the lost packets etc whereas if you use UDP protocol, it is all application developers responsibility.
So I would suggest you stick with TCP. Stan James. A connection consumes resources on the server. A simple socket might not need much depending on your server, but some systems like databases use huge amounts of memory per connection. Connecting and disconnecting saves that memory but takes a little time. It might be many thousands of times longer than using an open connection, but still the time on the clock is not that much. Many web pages do it dozens of times for images and frames and such.
Holding a connection for a long time can fail. If there is a network burp or the server shuts down and restarts while you're not looking the client's connection is bad. You have to detect such things and try to refresh the connection. If a server is holding resources for a connection and the client never calls again, how do we reclaim the resources? Maybe you need a keep-alive signal or a time out. What exactly does the Nagle algorithm do? What is the difference between read and recv? Is there any advantage to handling the signal, rather than just ignoring it and checking for the EPIPE error?
Are there any useful parameters passed to the signal catching function? After the chroot , calls to socket are failing. What are socket exceptions? What is out-of-band data? How do I convert a string into an internet address?
Why does connect succeed even before my server did an accept? Why do I sometimes lose a server's address when using more than one server? How can I set the timeout for the connect system call? Should I bind a port number in my client program, or let the Why do I get "connection refused" when the server isn't running?
RFC specifies that this timeout if it exists must be configurable. On the majority of Unix variants, this configuration may only be done globally, affecting all TCP connections which have keepalive enabled. If you're sending to the peer, though, you have some better guarantees; since sending data implies receiving ACKs from the peer, then you will know after the retransmit timeout whether the peer is still alive.
But the retransmit timeout is designed to allow for various contingencies, with the intention that TCP connections are not dropped simply as a result of minor network upsets. So you should still expect a delay of several minutes before getting notification of the failure. The approach taken by most application protocols currently in use on the Internet e.
Protocols where the connection is maintained even if idle for long periods have two choices:. Polling should usually be avoided since it uses more CPU time than other techniques. Using SIGIO allows your application to do what it does and have the operating system tell it with a signal that there is data waiting for it on a socket. The only drawback to this soltion is that it can be confusing, and if you are dealing with multiple sockets you will have to do a select anyway to find out which one s is ready to be read.
Using select is great if your application has to accept data from more than one socket at a time since it will block until any one of a number of sockets is ready with data.
One other advantage to select is that you can set a time-out value after which control will be returned to you whether any of the sockets have data for you or not.
And an addition note from Andrew andrew erlenstar. On some other implementations, accept seemed to be capable of blocking if this occured. This is important, since if select said the listening socket was readable, then you would normally expect not to block in the accept call. The fix is, of course, to set nonblocking mode on the listening socket if you are going to use select on it. You can't force it.
TCP makes up its own mind as to when it can send data. Now, normally when you call write on a TCP socket, TCP will indeed send a segment, but there's no guarantee and no way to force this. There are lots of reasons why TCP will not send a segment: a closed window and the Nagle algorithm are two things to come immediately to mind. Setting this only disables one of the many tests, the Nagle algorithm. But if the original poster's problem is this, then setting this socket option will help.
Now from Dr. Charles E. Campbell Jr. As you've surmised, I've never had any problem with disabling Nagle's algorithm. Its basically a buffering method; there's a fixed overhead for all packets, no matter how small. Hence, Nagle's algorithm collects small packets together no more than. This approach works well for rcp, for example: the. Helps in university settings where most folks using the network are using standard tools such as rcp and ftp, and programs such as telnet may use it, too.
However, Nagle's algorithm is pure havoc for real-time control and not much better for keystroke interactive applications control-C, anyone? It has seemed to me that the types of new programs using sockets that people write usually do have problems with small packet delays. One way to bypass Nagle's algorithm selectively is to use "out-of-band" messaging, but that is limited in its content and has other effects such as a loss of sequentiality by the way, out-of-band is often used for that ctrl-C, too.
If it doesn't, you will have to use out-of-band messaging, but according to Andrew, "out-of-band data has its own problems, and I don't think it works well as a solution to buffering delays haven't tried it though.
It is not 'expedited data' in the sense that exists in some other protocols; it is transmitted in-stream, but with a pointer to indicate where it is. I asked Andrew something to the effect of " What promises does TCP make about when it will get around to writing data to the network? The first of the interesting cases is "window closed" ie.
OK, it makes sense that if the client isn't reading, the data isn't going to make it across the connection. I take it this causes the sender to block after the recieve queue is filled? While the window is closed, the sending TCP sends window probe packets. This ensures that when the window finally does open again, the sending TCP detects the fact. The second interesting case is "Nagle algorithm" small segments, e. Since the amount of data is likely to be small relative to the socket send buffer, then it is likely since the connection is idle at that point that the entire request will require only one call to write , and that the TCP layer will immediately dispatch the request as a single segment with the PSH flag, see point 2.
The Nagle algorithm only has an effect when a second write call is made while data is still unacknowledged. In the normal case, this data will be left buffered until either: a there is no unacknowledged data; or b enough data is available to dispatch a full-sized segment.
The delay cannot be indefinite, since condition a must become true within the retransmit timeout or the connection dies. Since this delay has negative consequences for certain applications, generally those where a stream of small requests are being sent without response, e. The other possible sources of delay in the TCP are not really controllable by the program, but they can only delay the data temporarily.
By temporarily, you mean that the data will go as soon as it can, and I won't get stuck in a position where one side is waiting on a response, and the other side hasn't recieved the request? Or at least I won't get stuck forever.
You can only deadlock if you somehow manage to fill up all the buffers in both directions If it is possible to do this, can't think of a good example though , the solution is to use nonblocking mode, especially for writes. Then you can buffer excess data in the program as necessary. Campbell, Jr. The file is called ssl. Thanks to Bill McKinnon for tracking it down for me!
My link to their web site has gone stale, and I don't know where their new site is. Please send me an email if you find it. Philippe Jounin has developed a cross platform library which includes high level support for http and ftp protocols, with more to come. The data that causes select to return is the EOF because the other side has closed the connection.
This causes read to return zero. For more information see 2. With poll , however, the user must allocate an array of pollfd structures, and pass the number of entries in this array, so there's no fundamental limit.
As Casper notes, fewer systems have poll than select , so the latter is more portable. Also, with original implementations SVR3 you could not set the descriptor to -1 to tell the kernel to ignore an entry in the pollfd structure, which made it hard to remove entries from the array; SVR4 gets around this.
Someone could write an implementation of poll that uses select , for these environments, but I've never seen one. Anything other than single bytes of data will probably get mangled unless you take care. For integer values you can use htons and friends, and strings are really just a bunch of single bytes, so those should be OK. Be careful not to send a pointer to a string though, since the pointer will be meaningless on another machine.
If you need to send a struct, you should write sendthisstruct and readthisstruct functions for it that do all the work of taking the structure apart on one side, and putting it back together on the other. If you need to send floats, you may have a lot of work ahead of you. You should read RFC which is about portable ways of getting data from one machine to another thanks to Andrew Gabriel for pointing this out.
First off, be sure you really want to use it in the first place. It will disable the Nagle algorithm see 2. It should only be set for applications that send frequent small bursts of information without getting an immediate response, where timely delivery of data is required the canonical example is mouse movements.
It groups together as much data as it can between ACK's from the other end of the connection. I found this really confusing until Andrew Gierth andrew erlenstar. Case 1 : client writes 1 byte per write call. The program on host B is tcpserver. Total segments: 5. Note that in case 2, you don't want the implementation to gratuitously delay sending the data, since that would add straight onto the response time.
Other values for the flags parameter change the behaviour of recv. Is there any advantage to handling the signal, rather than just ignoring it and checking for the EPIPE error?
In general, the only parameter passed to a signal handler is the signal number that caused it to be invoked. Some systems have optional additional parameters, but they are no use to you in this case.
There is one situation where you should not ignore SIGPIPE ; if you are going to exec another program with stdout redirected to a socket. On systems where sockets are implemented on top of Streams e. Your system documentation may or may not specify exactly which device nodes are required; I can't help you there sorry.
Editors note: Adrian Hall adrian hottub. A less-obvious issue with chroot is if you call syslog , as many daemons do; syslog opens depending on the system either a UDP socket, a FIFO or a Unix-domain socket. The partition must not be mounted nosuid because in Solaris nosuid implies nodev. Ouch, or use BSD.
You would have to actually syslog something. This will obviously work only if you have a single chroot'ed daemon. This isn't really so much an error as an exit condition. It means that the call was interrupted by a signal. Sample Source Code. A common case is when the peer closes the connection sending you a FIN but you ignore it because you're writing and not reading. You should be using select. Socket exceptions usually refer to the notification that out-of-band data has arrived.
Out-of-band data called "urgent data" in TCP looks to the application like a separate stream of data from the main data stream. This can be useful for separating two different kinds of data. Note that just because it is called "urgent data" does not mean that it will be delivered any faster, or with higher priorety than data in the in-band data stream. Also beware that unlike the main data stream, the out-of-bound data may be lost if your application can't keep up with it.
Some systems set the hostname to the FQDN and others set it to just the unqualified host name. Regardless, the way around this is to first get the host's name perhaps an FQDN, perhaps unaualified. Most systems support the Posix way to do this using uname , but older BSD systems only provide gethostname.
Call gethostbyname to find your IP address. Then take the IP address and call gethostbyaddr. If the program is supposed to be portable then GNU autoconf would be the way to go. The results of not doing this can be quite annoying: e. If you are reading a host's address from the command line, you may not know if you have an aaa.
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