Партнёры:
Хостинг:
Archive-name: unix-faq/socket
Posting-Frequency: monthly
Last-modified: 1998/01/22
URL: http://www.ibrado.com/sock-faq/
Programming UNIX Sockets in C - Frequently Asked Questions
Created by Vic Metcalfe, Andrew Gierth and other con-
tributers
January 22, 1998
This is a list of frequently asked questions, with answers about pro-
gramming TCP/IP applications in unix with the sockets interface.
______________________________________________________________________
Table of Contents:
1. General Information and Concepts
1.1. About this FAQ
1.2. Who is this FAQ for?
1.3. What are Sockets?
1.4. How do Sockets Work?
1.5. Where can I get source code for the book [book title]?
1.6. Where can I get more information?
2. Questions regarding both Clients and Servers (TCP/SOCK_STREAM)
2.1. How can I tell when a socket is closed on the other end?
2.2. What's with the second parameter in bind()?
2.3. How do I get the port number for a given service?
2.4. If bind() fails, what should I do with the socket descriptor?
2.5. How do I properly close a socket?
2.6. When should I use shutdown()?
2.7. Please explain the TIME_WAIT state.
2.8. Why does it take so long to detect that the peer died?
2.9. What are the pros/cons of select(), non-blocking I/O and
SIGIO?
2.10. Why do I get EPROTO from read()?
2.11. How can I force a socket to send the data in its buffer?
2.12. Where can a get a library for programming sockets?
2.13. How come select says there is data, but read returns zero?
2.14. Whats the difference between select() and poll()?
2.15. How do I send [this] over a socket?
2.16. How do I use TCP_NODELAY?
2.17. What exactly does the Nagle algorithm do?
2.18. What is the difference between read() and recv()?
2.19. I see that send()/write() can generate SIGPIPE. 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?
2.20. After the chroot(), calls to socket() are failing. Why?
2.21. Why do I keep getting EINTR from the socket calls?
2.22. When will my application receive SIGPIPE?
2.23. What are socket exceptions? What is out-of-band data?
2.24. How can I find the full hostname (FQDN) of the system I'm
running on?
3. Writing Client Applications (TCP/SOCK_STREAM)
3.1. How do I convert a string into an internet address?
3.2. How can my client work through a firewall/proxy server?
3.3. Why does connect() succeed even before my server did an
accept()?
3.4. Why do I sometimes lose a server's address when using more
than one server?
3.5. How can I set the timeout for the connect() system call?
3.6. Should I bind() a port number in my client program, or let the
system choose one for me on the connect() call?
3.7. Why do I get "connection refused" when the server isn't
running?
3.8. What does one do when one does not know how much information
is comming over the socket ? Is there a way to have a dynamic buffer ?
4. Writing Server Applications (TCP/SOCK_STREAM)
4.1. How come I get "address already in use" from bind()?
4.2. Why don't my sockets close?
4.3. How can I make my server a daemon?
4.4. How can I listen on more than one port at a time?
4.5. What exactly does SO_REUSEADDR do?
4.6. What exactly does SO_LINGER do?
4.7. What exactly does SO_KEEPALIVE do?
4.8. How can I bind() to a port number < 1024?
4.9. How do I get my server to find out the client's address /
hostname?
4.10. How should I choose a port number for my server?
4.11. What is the difference between SO_REUSEADDR and SO_REUSEPORT?
4.12. How can I write a multi-homed server?
4.13. How can I read only one character at a time?
4.14. I'm trying to exec() a program from my server, and attach my
socket's IO to it, but I'm not getting all the data across. Why?
5. Writing UDP/SOCK_DGRAM applications
5.1. When should I use UDP instead of TCP?
5.2. What is the difference between "connected" and "unconnected"
sockets?
5.3. Does doing a connect() call affect the receive behaviour of
the socket?
5.4. How can I read ICMP errors from "connected" UDP sockets?
5.5. How can I be sure that a UDP message is received?
5.6. How can I be sure that UDP messages are received in order?
5.7. How often should I re-transmit un-acknowleged messages?
5.8. How come only the first part of my datagram is getting
through?
5.9. Why does the socket's buffer fill up sooner than expected?
6. Advanced Socket Programming
6.1. How would I put my socket in non-blocking mode?
6.2. How can I put a timeout on connect()?
7. Sample Source Code
______________________________________________________________________
1. General Information and Concepts
1.1. About this FAQ
This FAQ is maintained by Vic Metcalfe ([email protected]), with lots of
assistance from Andrew Gierth ([email protected]). I am
depending on the true wizards to fill in the details, and correct my
(no doubt) plentiful mistakes. The code examples in this FAQ are
written to be easy to follow and understand. It is up to the reader
to make them as efficient as required. I started this faq because
after reading comp.unix.programmer for a short time, it became evident
that a FAQ for sockets was needed.
The FAQ is available at the following locations:
Usenet: (Posted on the 21st of each month)
news.answers, comp.answers, comp.unix.answers,
comp.unix.programmer
FTP:
ftp://rtfm.mit.edu/pub/usenet/news.answers/unix-faq/socket
WWW:
http://www.ibrado.com/sock-faq
http://kipper.york.ac.uk/~vic/sock-faq http://www.ntua.gr/sock-
faq
Please email me if you would like to correct or clarify an answer. I
would also like to hear from you if you would like me to add a
question to the list. I may not be able to answer it, but I can add
it in the hopes that someone else will submit an answer. Every hour I
seem to be getting even busier, so if I am slow to respond to your
email, please be patient. If more than a week passes you may want to
send me another one as I often put messages aside for later and then
forget about them. I'll have to work on dealing with my mail better,
but until then feel free to pester me a little bit.
1.2. Who is this FAQ for?
This FAQ is for C programmers in the Unix environment. It is not
intended for WinSock programmers, or for Perl, Java, etc. I have
nothing against Windows or Perl, but I had to limit the scope of the
FAQ for the first draft. In the future, I would really like to
provide examples for Perl, Java, and maybe others. For now though I
will concentrate on correctness and completeness for C.
This version of the FAQ will only cover sockets of the AF_INET family,
since this is their most common use. Coverage of other types of
sockets may be added later.
1.3. What are Sockets?
Sockets are just like "worm holes" in science fiction. When things go
into one end, they (should) come out of the other. Different kinds of
sockets have different properties. Sockets are either connection-
oriented or connectionless. Connection-oriented sockets allow for
data to flow back and forth as needed, while connectionless sockets
(also known as datagram sockets) allow only one message at a time to
be transmitted, without an open connection. There are also different
socket families. The two most common are AF_INET for internet
connections, and AF_UNIX for unix IPC (interprocess communication).
As stated earlier, this FAQ deals only with AF_INET sockets.
1.4. How do Sockets Work?
The implementation is left up to the vendor of your particular unix,
but from the point of view of the programmer, connection-oriented
sockets work a lot like files, or pipes. The most noticeable
difference, once you have your file descriptor is that read() or
write() calls may actually read or write fewer bytes than requested.
If this happens, then you will have to make a second call for the rest
of the data. There are examples of this in the source code that
accompanies the faq.
1.5. Where can I get source code for the book [book title]?
Here is a list of the places I know to get source code for network
programming books. It is very short, so please mail me with any
others you know of.
Title: Unix Network Programming
Author: W. Richard Stevens ([email protected])
Publisher: Prentice Hall, Inc.
ISBN: 0-13-949876-1
URL: http://www.noao.edu/~rstevens
Title: Power Programming with RPC
Author: John Bloomer
Publisher: O'Reilly & Associates, Inc.
ISBN: 0-937175-77-3
URL: ftp://ftp.uu.net/published/oreilly/nutshell/rpc/rpc.tar.Z
Recommended by: Lokmanm Merican (lokmanm#[email protected])
Title: UNIX PROGRAM DEVELOPMENT for IBM PC'S Including OSF/Motif
Author: Thomas Yager
Publisher: Addison Wesley, 1991
ISBN: 0-201-57727-5
1.6. Where can I get more information?
I keep a copy of the resources I know of on my socks page on the web.
I don't remember where I got most of these items, but some day I'll
check out their sources, and provide ftp information here. For now,
you can get them at http://www.ibrado.com/sock-faq.
There is a good TCP/IP FAQ maintained by George Neville-Neil
([email protected]) which can be found at
http://www.visi.com/~khayes/tcpipfaq.html
2. Questions regarding both Clients and Servers (TCP/SOCK_STREAM)
2.1. How can I tell when a socket is closed on the other end?
From Andrew Gierth ([email protected]):
AFAIK:
If the peer calls close() or exits, without having messed with
SO_LINGER, then our calls to read() should return 0. It is less clear
what happens to write() calls in this case; I would expect EPIPE, not
on the next call, but the one after.
If the peer reboots, or sets l_onoff = 1, l_linger = 0 and then
closes, then we should get ECONNRESET (eventually) from read(), or
EPIPE from write().
I should also point out that when write() returns EPIPE, it also
raises the SIGPIPE signal - you never see the EPIPE error unless you
handle or ignore the signal.
If the peer remains unreachable, we should get some other error.
I don't think that write() can legitimately return 0. read() should
return 0 on receipt of a FIN from the peer, and on all following
calls.
So yes, you must expect read() to return 0.
As an example, suppose you are receiving a file down a TCP link; you
might handle the return from read() like this:
rc = read(sock,buf,sizeof(buf));
if (rc > 0)
{
write(file,buf,rc);
/* error checking on file omitted */
}
else if (rc == 0)
{
close(file);
close(sock);
/* file received successfully */
}
else /* rc < 0 */
{
/* close file and delete it, since data is not complete
report error, or whatever */
}
2.2. What's with the second parameter in bind()?
The man page shows it as "struct sockaddr *my_addr". The sockaddr
struct though is just a place holder for the structure it really
wants. You have to pass different structures depending on what kind
of socket you have. For an AF_INET socket, you need the sockaddr_in
structure. It has three fields of interest:
sin_family
Set this to AF_INET.
sin_port
The network byte-ordered 16 bit port number
sin_addr
The host's ip number. This is a struct in_addr, which contains
only one field, s_addr which is a u_long.
2.3. How do I get the port number for a given service?
Use the getservbyname() routine. This will return a pointer to a
servent structure. You are interested in the s_port field, which
contains the port number, with correct byte ordering (so you don't
need to call htons() on it). Here is a sample routine:
/* Take a service name, and a service type, and return a port number. If the
service name is not found, it tries it as a decimal number. The number
returned is byte ordered for the network. */
int atoport(char *service, char *proto) {
int port;
long int lport;
struct servent *serv;
char *errpos;
/* First try to read it from /etc/services */
serv = getservbyname(service, proto);
if (serv != NULL)
port = serv->s_port;
else { /* Not in services, maybe a number? */
lport = strtol(service,&errpos,0);
if ( (errpos[0] != 0) || (lport < 1) || (lport > 5000) )
return -1; /* Invalid port address */
port = htons(lport);
}
return port;
}
2.4. If bind() fails, what should I do with the socket descriptor?
If you are exiting, I have been assured by Andrew that all unixes will
close open file descriptors on exit. If you are not exiting though,
you can just close it with a regular close() call.
2.5. How do I properly close a socket?
This question is usually asked by people who try close(), because they
have seen that that is what they are supposed to do, and then run
netstat and see that their socket is still active. Yes, close() is
the correct method. To read about the TIME_WAIT state, and why it is
important, refer to ``2.7 Please explain the TIME_WAIT state.''.
2.6. When should I use shutdown()?
From Michael Hunter ([email protected]):
shutdown() is useful for deliniating when you are done providing a
request to a server using TCP. A typical use is to send a request to
a server followed by a shutdown(). The server will read your request
followed by an EOF (read of 0 on most unix implementations). This
tells the server that it has your full request. You then go read
blocked on the socket. The server will process your request and send
the necessary data back to you followed by a close. When you have
finished reading all of the response to your request you will read an
EOF thus signifying that you have the whole response. It should be
noted the TTCP (TCP for Transactions -- see R. Steven's home page)
provides for a better method of tcp transaction management.
S.Degtyarev ([email protected]) wrote a nice in-depth message to me
about this. He shows a practical example of using shutdown() to aid
in synchronization of client processes when one is the "reader"
process, and the other is the "writer" process. A portion of his
message follows:
Sockets are very similar to pipes in the way they are used for data
transfer and client/server transactions, but not like pipes they are
bidirectional. Programs that use sockets often fork() and each
process inherits the socket descriptor. In pipe based programs it is
strictly recommended to close all the pipe ends that are not used to
convert the pipe line to one-directional data stream to avoid data
losses and deadlocks. With the socket there is no way to allow one
process only to send data and the other only to receive so you should
always keep in mind the consequences.
Generally the difference between close() and shutdown() is: close()
closes the socket id for the process but the connection is still
opened if another process shares this socket id. The connection stays
opened both for read and write, and sometimes this is very important.
shutdown() breaks the connection for all processes sharing the socket
id. Those who try to read will detect EOF, and those who try to write
will reseive SIGPIPE, possibly delayed while the kernel socket buffer
will be filled. Additionally, shutdown() has a second argument which
denotes how to close the connection: 0 means to disable further
reading, 1 to disable writing and 2 disables both.
The quick example below is a fragment of a very simple client process.
After establishing the connection with the server it forks. Then
child sends the keyboard input to the server until EOF is received and
the parent receives answers from the server.
/*
* Sample client fragment,
* variables declarations and error handling are omitted
*/
s=connect(...);
if( fork() ){ /* The child, it copies its stdin to
the socket */
while( gets(buffer) >0)
write(s,buf,strlen(buffer));
close(s);
exit(0);
}
else { /* The parent, it receives answers */
while( (l=read(s,buffer,sizeof(buffer)){
do_something(l,buffer);
/* Connection break from the server is assumed */
/* ATTENTION: deadlock here */
wait(0); /* Wait for the child to exit */
exit(0);
}
What do we expect? The child detects an EOF from its stdin, it closes
the socket (assuming connection break) and exits. The server in its
turn detects EOF, closes connection and exits. The parent detects
EOF, makes the wait() system call and exits. What do we see instead?
The socket instance in the parent process is still opened for writing
and reading, though the parent never writes. The server never detects
EOF and waits for more data from the client forever. The parent never
sees the connection is closed and hangs forever and the server hangs
too. Unexpected deadlock! ( any deadlock is unexpected though :-)
You should change the client fragment as follows:
if( fork() ) { /* The child */
while( gets(buffer) }
write(s,buffer,strlen(buffer));
shutdown(s,1); /* Break the connection
for writing, The server will detect EOF now. Note: reading from
the socket is still allowed. The server may send some more data
after receiving EOF, why not? */
exit(0);
}
I hope this rough example explains the troubles you can have with
client/server syncronization. Generally you should always remember
all the instances of the particular socket in all the processes that
share the socket and close them all at once if you whish to use
close() or use shutdown() in one process to break the connection.
2.7. Please explain the TIME_WAIT state.
Remember that TCP guarantees all data transmitted will be delivered,
if at all possible. When you close a socket, the server goes into a
TIME_WAIT state, just to be really really sure that all the data has
gone through. When a socket is closed, both sides agree by sending
messages to each other that they will send no more data. This, it
seemed to me was good enough, and after the handshaking is done, the
socket should be closed. The problem is two-fold. First, there is no
way to be sure that the last ack was communicated successfully.
Second, there may be "wandering duplicates" left on the net that must
be dealt with if they are delivered.
Andrew Gierth ([email protected]) helped to explain the
closing sequence in the following usenet posting:
Assume that a connection is in ESTABLISHED state, and the client is
about to do an orderly release. The client's sequence no. is Sc, and
the server's is Ss. The pipe is empty in both directions.
Client Server
====== ======
ESTABLISHED ESTABLISHED
(client closes)
ESTABLISHED ESTABLISHED
<CTL=FIN+ACK><SEQ=Sc><ACK=Ss> ------->>
FIN_WAIT_1
<<-------- <CTL=ACK><SEQ=Ss><ACK=Sc+1>
FIN_WAIT_2 CLOSE_WAIT
<<-------- <CTL=FIN+ACK><SEQ=Ss><ACK=Sc+1> (server closes)
LAST_ACK
<CTL=ACK>,<SEQ=Sc+1><ACK=Ss+1> ------->>
TIME_WAIT CLOSED
(2*msl elapses...)
CLOSED
Note: the +1 on the sequence numbers is because the FIN counts as one
byte of data. (The above diagram is equivalent to fig. 13 from RFC
793).
Now consider what happens if the last of those packets is dropped in
the network. The client has done with the connection; it has no more
data or control info to send, and never will have. But the server does
not know whether the client received all the data correctly; that's
what the last ACK segment is for. Now the server may or may not care
whether the client got the data, but that is not an issue for TCP; TCP
is a reliable rotocol, and must distinguish between an orderly
connection close where all data is transferred, and a connection abort
where data may or may not have been lost.
So, if that last packet is dropped, the server will retransmit it (it
is, after all, an unacknowledged segment) and will expect to see a
suitable ACK segment in reply. If the client went straight to CLOSED,
the only possible response to that retransmit would be a RST, which
would indicate to the server that data had been lost, when in fact it
had not been.
(Bear in mind that the server's FIN segment may, additionally, contain
data.)
DISCLAIMER: This is my interpretation of the RFCs (I have read all the
TCP-related ones I could find), but I have not attempted to examine
implementation source code or trace actual connections in order to
verify it. I am satisfied that the logic is correct, though.
More commentarty from Vic:
The second issue was addressed by Richard Stevens ([email protected],
author of "Unix Network Programming", see ``1.5 Where can I get source
code for the book [book title]?''). I have put together quotes from
some of his postings and email which explain this. I have brought
together paragraphs from different postings, and have made as few
changes as possible.
From Richard Stevens ([email protected]):
If the duration of the TIME_WAIT state were just to handle TCP's full-
duplex close, then the time would be much smaller, and it would be
some function of the current RTO (retransmission timeout), not the MSL
(the packet lifetime).
A couple of points about the TIME_WAIT state.
o The end that sends the first FIN goes into the TIME_WAIT state,
because that is the end that sends the final ACK. If the other
end's FIN is lost, or if the final ACK is lost, having the end that
sends the first FIN maintain state about the connection guarantees
that it has enough information to retransmit the final ACK.
o Realize that TCP sequence numbers wrap around after 2**32 bytes
have been transferred. Assume a connection between A.1500 (host A,
port 1500) and B.2000. During the connection one segment is lost
and retransmitted. But the segment is not really lost, it is held
by some intermediate router and then re-injected into the network.
(This is called a "wandering duplicate".) But in the time between
the packet being lost & retransmitted, and then reappearing, the
connection is closed (without any problems) and then another
connection is established between the same host, same port (that
is, A.1500 and B.2000; this is called another "incarnation" of the
connection). But the sequence numbers chosen for the new
incarnation just happen to overlap with the sequence number of the
wandering duplicate that is about to reappear. (This is indeed
possible, given the way sequence numbers are chosen for TCP
connections.) Bingo, you are about to deliver the data from the
wandering duplicate (the previous incarnation of the connection) to
the new incarnation of the connection. To avoid this, you do not
allow the same incarnation of the connection to be reestablished
until the TIME_WAIT state terminates.
Even the TIME_WAIT state doesn't complete solve the second problem,
given what is called TIME_WAIT assassination. RFC 1337 has more
details.
o The reason that the duration of the TIME_WAIT state is 2*MSL is
that the maximum amount of time a packet can wander around a
network is assumed to be MSL seconds. The factor of 2 is for the
round-trip. The recommended value for MSL is 120 seconds, but
Berkeley-derived implementations normally use 30 seconds instead.
This means a TIME_WAIT delay between 1 and 4 minutes. Solaris 2.x
does indeed use the recommended MSL of 120 seconds.
A wandering duplicate is a packet that appeared to be lost and was
retransmitted. But it wasn't really lost ... some router had
problems, held on to the packet for a while (order of seconds, could
be a minute if the TTL is large enough) and then re-injects the packet
back into the network. But by the time it reappears, the application
that sent it originally has already retransmitted the data contained
in that packet.
Because of these potential problems with TIME_WAIT assassinations, one
should not avoid the TIME_WAIT state by setting the SO_LINGER option
to send an RST instead of the normal TCP connection termination
(FIN/ACK/FIN/ACK). The TIME_WAIT state is there for a reason; it's
your friend and it's there to help you :-)
I have a long discussion of just this topic in my just-released
"TCP/IP Illustrated, Volume 3". The TIME_WAIT state is indeed, one of
the most misunderstood features of TCP.
I'm currently rewriting "Unix Network Programming" (see ``1.5 Where
can I get source code for the book [book title]?''). and will include
lots more on this topic, as it is often confusing and misunderstood.
An additional note from Andrew:
Closing a socket: if SO_LINGER has not been called on a socket, then
close() is not supposed to discard data. This is true on SVR4.2 (and,
apparently, on all non-SVR4 systems) but apparently not on SVR4; the
use of either shutdown() or SO_LINGER seems to be required to
guarantee delivery of all data.
2.8. Why does it take so long to detect that the peer died?
From Andrew Gierth ([email protected]):
Because by default, no packets are sent on the TCP connection unless
there is data to send or acknowledge.
So, if you are simply waiting for data from the peer, there is no way
to tell if the peer has silently gone away, or just isn't ready to
send any more data yet. This can be a problem (especially if the peer
is a PC, and the user just hits the Big Switch...).
One solution is to use the SO_KEEPALIVE option. This option enables
periodic probing of the connection to ensure that the peer is still
present. BE WARNED: the default timeout for this option is AT LEAST 2
HOURS. This timeout can often be altered (in a system-dependent
fashion) but not normally on a per-connection basis (AFAIK).
RFC1122 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. The method of changing the value, moreover, is
often difficult and/or poorly documented, and in any case is different
for just about every version in existence.
If you must change the value, look for something resembling
tcp_keepidle in your kernel configuration or network options
configuration.
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.g. FTP, SMTP etc.) is to implement read timeouts on
the server end; the server simply gives up on the client if no
requests are received in a given time period (often of the order of 15
minutes). Protocols where the connection is maintained even if idle
for long periods have two choices:
1. use SO_KEEPALIVE
2. use a higher-level keepalive mechanism (such as sending a null
request to the server every so often).
2.9. What are the pros/cons of select(), non-blocking I/O and SIGIO?
Using non-blocking I/O means that you have to poll sockets to see if
there is data to be read from them. 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.
2.10. Why do I get EPROTO from read()?
From Steve Rago ([email protected]):
EPROTO means that the protocol encountered an unrecoverable error for
that endpoint. EPROTO is one of those catch-all error codes used by
STREAMS-based drivers when a better code isn't available.
And an addition note from Andrew ([email protected]):
Not quite to do with EPROTO from read(), but I found out once that on
some STREAMS-based implementations, EPROTO could be returned by
accept() if the incoming connection was reset before the accept
completes.
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.
2.11. How can I force a socket to send the data in its buffer?
From Richard Stevens ([email protected]):
You can't force it. Period. 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.
(Snipped suggestion from Andrew Gierth to use TCP_NODELAY)
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.
A quick glance at tcp_output() shows around 11 tests TCP has to make
as to whether to send a segment or not.
Now from Dr. Charles E. Campbell Jr. ([email protected]):
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 .2sec delay) and thereby
reduces the amount of overhead bytes being transferred. This approach
works well for rcp, for example: the .2 second delay isn't humanly
noticeable, and multiple users have their small packets more
efficiently transferred. 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).
More from Vic:
So to sum it all up, if you are having trouble and need to flush the
socket, setting the TCP_NODELAY option will usually solve the problem.
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?" I
thought his reply should be put under this question:
Not many promises, but some.
I'll try and quote chapter and verse on this:
References:
RFC 1122, "Requirements for Internet Hosts" (also STD 3)
RFC 793, "Transmission Control Protocol" (also STD 7)
1. The socket interface does not provide access to the TCP PUSH flag.
2. RFC1122 says (4.2.2.2):
A TCP MAY implement PUSH flags on SEND calls. If PUSH flags are
not implemented, then the sending TCP: (1) must not buffer data
indefinitely, and (2) MUST set the PSH bit in the last buffered
segment (i.e., when there is no more queued data to be sent).
3. RFC793 says (2.8):
When a receiving TCP sees the PUSH flag, it must not wait for more
data from the sending TCP before passing the data to the receiving
process.
[RFC1122 supports this statement.]
4. Therefore, data passed to a write() call must be delivered to the
peer within a finite time, unless prevented by protocol
considerations.
5. There are (according to a post from Stevens quoted in the FAQ
[earlier in this answer - Vic]) about 11 tests made which could
delay sending the data. But as I see it, there are only 2 that are
significant, since things like retransmit backoff are a) not under
the programmers control and b) must either resolve within a finite
time or drop the connection.
The first of the interesting cases is "window closed" (ie. there is
no buffer space at the receiver; this can delay data indefinitely, but
only if the receiving process is not actually reading the data that is
available)
Vic asks:
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?
The sender blocks when the socket send buffer is full, so buffers will
be full at both ends.
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. [RFC1122, ss 4.2.2.17]
The second interesting case is "Nagle algorithm" (small segments, e.g.
keystrokes, are delayed to form larger segments if ACKs are expected
from the peer; this is what is disabled with TCP_NODELAY)
Vic Asks:
Does this mean that my tcpclient sample should set TCP_NODELAY to
ensure that the end-of-line code is indeed put out onto the network
when sent?
No. tcpclient.c is doing the right thing as it stands; trying to write
as much data as possible in as few calls to write() as is feasible.
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 2.2 above).
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.g. mouse movements, the standards specify that an
option must exist to disable it. [RFC1122, ss 4.2.3.4]
Additional note: RFC1122 also says:
[DISCUSSION]:
When the PUSH flag is not implemented on SEND calls, i.e., when
the application/TCP interface uses a pure streaming model,
responsibility for aggregating any tiny data fragments to form
reasonable sized segments is partially borne by the application
layer.
So programs should avoid calls to write() with small data lengths
(small relative to the MSS, that is); it's better to build up a
request in a buffer and then do one call to sock_write() or
equivalent.
The other possible sources of delay in the TCP are not really
controllable by the program, but they can only delay the data
temporarily.
Vic asks:
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... not easy.
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.
2.12. Where can a get a library for programming sockets?
There is the Simple Sockets Library by Charles E. Campbell, Jr. PhD.
and Terry McRoberts. The file is called ssl.tar.gz, and you can
download it from this faq's home page. For c++ there is the Socket++
library which is on ftp://ftp.virginia.edu/pub/socket++-1.10.tar.gz.
There is also C++ Wrappers. The file is called
ftp://ftp.huji.ac.il/pub/languages/C++/C++_wrappers.tar.gz. Thanks to
Bill McKinnon for tracking it down for me! From
http://www.cs.wustl.edu/~schmidt you should be able to find the ACE
toolkit. PING Software Group has some libraries that include a
sockets interface among other things. You can find them at
http://love.geology.yale.edu/~markl/ping.
I don't have any experience with any of these libraries, so I can't
recomend one over the other.
2.13. How come select says there is data, but read returns zero?
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 ``2.1 How can I tell when a socket is closed on
the other end?''
2.14. Whats the difference between select() and poll()?
From Richard Stevens ([email protected]):
The basic difference is that select()'s fd_set is a bit mask and
therefore has some fixed size. It would be possible for the kernel to
not limit this size when the kernel is compiled, allowing the
application to define FD_SETSIZE to whatever it wants (as the comments
in the system header imply today) but it takes more work. 4.4BSD's
kernel and the Solaris library function both have this limit. But I
see that BSD/OS 2.1 has now been coded to avoid this limit, so it's
doable, just a small matter of programming. :-) Someone should file a
Solaris bug report on this, and see if it ever gets fixed.
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.
Personally, I always use select() and rarely poll(), because I port my
code to BSD environments too. Someone could write an implementation
of poll() that uses select(), for these environments, but I've never
seen one. Both select() and poll() are being standardized by POSIX
1003.1g.
2.15. How do I send [this] over a socket?
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 1014 which is about portable ways of
getting data from one machine to another (thanks to Andrew Gabriel for
pointing this out).
2.16. How do I use TCP_NODELAY?
First off, be sure you really want to use it in the first place. It
will disable the Nagle algorithm (see ``2.11 How can I force a socket
to send the data in its buffer?''), which will cause network traffic
to increase, with smaller than needed packets wasting bandwidth.
Also, from what I have been able to tell, the speed increase is very
small, so you should probably do it without TCP_NODELAY first, and
only turn it on if there is a problem.
Here is a code example, with a warning about using it from Andrew
Gierth:
int flag = 1;
int result = setsockopt(sock, /* socket affected */
IPPROTO_TCP, /* set option at TCP level */
TCP_NODELAY, /* name of option */
(char *) &flag, /* the cast is historical
cruft */
sizeof(int)); /* length of option value */
if (result < 0)
... handle the error ...
TCP_NODELAY is for a specific purpose; to disable the Nagle buffering
algorithm. 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).
2.17. What exactly does the Nagle algorithm do?
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 ([email protected]) drew the following diagram, and
explained:
This diagram is not intended to be complete, just to illustrate the
point better...
Case 1: client writes 1 byte per write() call. The program on host B
is tcpserver.c from the FAQ examples.
CLIENT SERVER
APP TCP TCP APP
[connection setup omitted]
"h" ---------> [1 byte]
------------------>
-----------> "h"
[ack delayed]
"e" ---------> [Nagle alg. .
now in effect] .
"l" ---------> [ditto] .
"l" ---------> [ditto] .
"o" ---------> [ditto] .
"\n"---------> [ditto] .
.
.
[ack 1 byte]
<------------------
[send queued
data]
[5 bytes]
------------------>
------------> "ello\n"
<------------ "HELLO\n"
[6 bytes, ack 5 bytes]
<------------------
"HELLO\n" <----
[ack delayed]
.
.
. [ack 6 bytes]
------------------>
Total segments: 5. (If TCP_NODELAY was set, could have been up to 10.)
Time for response: 2*RTT, plus ack delay.
Case 2: client writes all data with one write() call.
CLIENT SERVER
APP TCP TCP APP
[connection setup omitted]
"hello\n" ---> [6 bytes]
------------------>
------------> "hello\n"
<------------ "HELLO\n"
[6 bytes, ack 6 bytes]
<------------------
"HELLO\n" <----
[ack delayed]
.
.
. [ack 6 bytes]
------------------>
Total segments: 3.
Time for response = RTT (therefore minimum possible).
Hope this makes things a bit clearer...
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.
2.18. What is the difference between read() and recv()?
From Andrew Gierth ([email protected]):
read() is equivalent to recv() with a flags parameter of 0. Other
values for the flags parameter change the behaviour of recv().
Similarly, write() is equivalent to send() with flags == 0.
It is unlikely that send()/recv() would be dropped; perhaps someone
with a copy of the POSIX drafts for socket calls can check...
Portability note: non-unix systems may not allow read()/write() on
sockets, but recv()/send() are usually ok. This is true on Windows and
OS/2, for example.
2.19. I see that send()/write() can generate SIGPIPE. 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?
From Andrew Gierth ([email protected]):
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.
My advice is to just ignore SIGPIPE as you suggest. That's what I do
in just about all of my socket code; errno values are easier to handle
than signals (in fact, the first revision of the FAQ failed to mention
SIGPIPE in that context; I'd got so used to ignoring it...)
There is one situation where you should not ignore SIGPIPE; if you are
going to exec() another program with stdout redirected to a socket. In
this case it is probably wise to set SIGPIPE to SIG_DFL before doing
the exec().
2.20. After the chroot(), calls to socket() are failing. Why?
From Andrew Gierth ([email protected]):
On systems where sockets are implemented on top of Streams (e.g. all
SysV-based systems, presumably including Solaris), the socket()
function will actually be opening certain special files in /dev. You
will need to create a /dev directory under your fake root and populate
it with the required device nodes (only).
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 ([email protected]) suggested checking the man
page for ftpd, which should list the files you need to copy and
devices you need to create in the chroot'd environment.)
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. So if you use it after a
chroot() call, make sure that you call openlog() *before* the chroot.
2.21. Why do I keep getting EINTR from the socket calls?
This isn't really so much an error as an exit condition. It means
that the call was interrupted by a signal. Any call that might block
should be wrapped in a loop that checkes for EINTR, as is done in the
example code (See ``6. Sample Source Code'').
2.22. When will my application receive SIGPIPE?
From Richard Stevens ([email protected]):
Very simple: with TCP you get SIGPIPE if your end of the connection
has received an RST from the other end. What this also means is that
if you were using select instead of write, the select would have
indicated the socket as being readable, since the RST is there for you
to read (read will return an error with errno set to ECONNRESET).
Basically an RST is TCP's response to some packet that it doesn't
expect and has no other way of dealing with. 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.)
So you write to a connection that has been closed by the other end and
the oether end's TCP responds with an RST.
2.23. What are socket exceptions? What is out-of-band data?
Unlike exceptions in C++, socket exceptions do not indicate that an
error has occured. 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.
2.24. running on? How can I find the full hostname (FQDN) of the
system I'm
From Richard Stevens ([email protected]):
Some systems set the hostname to the FQDN and others set it to just
the unqualified host name. I know the current BIND FAQ recommends the
FQDN, but most Solaris systems, for example, tend to use only 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(). The h_name member of
the hostent{} should then be your FQDN.
3. Writing Client Applications (TCP/SOCK_STREAM)
3.1. How do I convert a string into an internet address?
If you are reading a host's address from the command line, you may not
know if you have an aaa.bbb.ccc.ddd style address, or a
host.domain.com style address. What I do with these, is first try to
use it as a aaa.bbb.ccc.ddd type address, and if that fails, then do a
name lookup on it. Here is an example:
/* Converts ascii text to in_addr struct. NULL is returned if the
address can not be found. */
struct in_addr *atoaddr(char *address) {
struct hostent *host;
static struct in_addr saddr;
/* First try it as aaa.bbb.ccc.ddd. */
saddr.s_addr = inet_addr(address);
if (saddr.s_addr != -1) {
return &saddr;
}
host = gethostbyname(address);
if (host != NULL) {
return (struct in_addr *) *host->h_addr_list;
}
return NULL;
}
3.2. How can my client work through a firewall/proxy server?
If you are running through separate proxies for each service, you
shouldn't need to do anything. If you are working through sockd, you
will need to "socksify" your application. Details for doing this can
be found in the package itself, which is available at:
ftp://ftp.net.com/socks.cstc/socks.cstc.4.2.tar.gz
you can get the socks faq at:
ftp://coast.cs.purdue.edu/pub/tools/unix/socks/FAQ
3.3. Why does connect() succeed even before my server did an
accept()?
From Andrew Gierth ([email protected]):
Once you have done a listen() call on your socket, the kernel is
primed to accept connections on it. The usual UNIX implementation of
this works by immediately completing the SYN handshake for any
incoming valid SYN segments (connection attempts), creating the socket
for the new connection, and keeping this new socket on an internal
queue ready for the accept() call. So the socket is fully open before
the accept is done.
The other factor in this is the 'backlog' parameter for listen(); that
defines how many of these completed connections can be queued at one
time. If the specified number is exceeded, then new incoming connects
are simply ignored (which causes them to be retried).
3.4. Why do I sometimes lose a server's address when using more than
one server?
From Andrew Gierth ([email protected]):
Take a careful look at struct hostent. Notice that almost everything
in it is a pointer? All these pointers will refer to statically
allocated data.
For example, if you do:
struct hostent *host = gethostbyname(hostname);
then (as you should know) a subsequent call to gethostbyname() will
overwrite the structure pointed to by 'host'.
But if you do:
struct hostent myhost;
struct hostent *hostptr = gethostbyname(hostname);
if (hostptr) myhost = *host;
to make a copy of the hostent before it gets overwritten, then it
still gets clobbered by a subsequent call to gethostbyname(), since
although myhost won't get overwritten, all the data it is pointing to
will be.
You can get round this by doing a proper 'deep copy' of the hostent
structure, but this is tedious. My recommendation would be to extract
the needed fields of the hostent and store them in your own way.
Robin Paterson ([email protected]) has added:
It might be nice if you mention MT safe libraries provide
complimentary functions for multithreaded programming. On the solaris
machine I'm typing at, we have gethostbyname and gethostbyname_r (_r
for reentrant). The main difference is, you provide the storage for
the hostent struct so you always have a local copy and not just a
pointer to the static copy.
3.5. How can I set the timeout for the connect() system call?
From Richard Stevens ([email protected]):
Normally you cannot change this. Solaris does let you do this, on a
per-kernel basis with the ndd tcp_ip_abort_cinterval parameter.
The easiest way to shorten the connect time is with an alarm() around
the call to connect(). A harder way is to use select(), after setting
the socket nonblocking. Also notice that you can only shorten the
connect time, there's normally no way to lengthen it.
From Andrew Gierth ([email protected]):
First, create the socket and put it into non-blocking mode, then call
connect(). There are three possibilities:
o connect succeeds: the connection has been successfully made (this
usually only happens when connecting to the same machine)
o connect fails: obvious
o connect returns -1/EINPROGRESS. The connection attempt has begun,
but not yet completed.
If the connection succeeds:
o the socket will select() as writable (and will also select as
readable if data arrives)
If the connection fails:
o the socket will select as readable *and* writable, but either a
read or write will return the error code from the connection
attempt. Also, you can use getsockopt(SO_ERROR) to get the error
status - but be careful; some systems return the error code in the
result parameter of getsockopt, but others (incorrectly) cause the
getsockopt call *itself* to fail with the stored value as the
error.
3.6. system choose one for me on the connect() call? Should I bind()
a port number in my client program, or let the
From Andrew Gierth ([email protected]):
** Let the system choose your client's port number **
The exception to this, is if the server has been written to be picky
about what client ports it will allow connections from. Rlogind and
rshd are the classic examples. This is usually part of a Unix-specific
(and rather weak) authentication scheme; the intent is that the server
allows connections only from processes with root privilege. (The
weakness in the scheme is that many O/Ss (e.g. MS-DOS) allow anyone to
bind any port.)
The rresvport() routine exists to help out clients that are using this
scheme. It basically does the equivalent of socket() + bind(),
choosing a port number in the range 512..1023.
If the server is not fussy about the client's port number, then don't
try and assign it yourself in the client, just let connect() pick it
for you.
If, in a client, you use the naive scheme of starting at a fixed port
number and calling bind() on consecutive values until it works, then
you buy yourself a whole lot of trouble:
The problem is if the server end of your connection does an active
close. (E.G. client sends 'QUIT' command to server, server responds
by closing the connection). That leaves the client end of the
connection in CLOSED state, and the server end in TIME_WAIT state. So
after the client exits, there is no trace of the connection on the
client end.
Now run the client again. It will pick the same port number, since as
far as it can see, it's free. But as soon as it calls connect(), the
server finds that you are trying to duplicate an existing connection
(although one in TIME_WAIT). It is perfectly entitled to refuse to do
this, so you get, I suspect, ECONNREFUSED from connect(). (Some
systems may sometimes allow the connection anyway, but you can't rely
on it.)
This problem is especially dangerous because it doesn't show up unless
the client and server are on different machines. (If they are the same
machine, then the client won't pick the same port number as before).
So you can get bitten well into the development cycle (if you do what
I suspect most people do, and test client & server on the same box
initially).
Even if your protocol has the client closing first, there are still
ways to produce this problem (e.g. kill the server).
3.7. Why do I get "connection refused" when the server isn't running?
The connect() call will only block while it is waiting to establish a
connection. When there is no server waiting at the other end, it gets
notified that the connection can not be established, and gives up with
the error message you see. This is a good thing, since if it were not
the case clients might wait for ever for a service which just doesn't
exist. Users would think that they were only waiting for the
connection to be established, and then after a while give up,
muttering something about crummy software under their breath.
3.8. over the socket ? Is there a way to have a dynamic buffer ?
What does one do when one does not know how much information is com-
ming
This question asked by Niranjan Perera ([email protected]).
When the size of the incoming data is unknown, you can either make the
size of the buffer as big as the largest possible (or likely) buffer,
or you can re-size the buffer on the fly during your read. When you
malloc() a large buffer, most (if not all) varients of unix will only
allocate address space, but not physical pages of ram. As more and
more of the buffer is used, the kernel allocates physical memory.
This means that malloc'ing a large buffer will not waste resources
unless that memory is used, and so it is perfectly acceptable to ask
for a meg of ram when you expect only a few K.
On the other hand, a more elegant solution that does not depend on the
inner workings of the kernel is to use realloc() to expand the buffer
as required in say 4K chunks (since 4K is the size of a page of ram on
most systems). I may add something like this to sockhelp.c in the
example code one day.
4. Writing Server Applications (TCP/SOCK_STREAM)
4.1. How come I get "address already in use" from bind()?
You get this when the address is already in use. (Oh, you figured
that much out?) The most common reason for this is that you have
stopped your server, and then re-started it right away. The sockets
that were used by the first incarnation of the server are still
active. This is further explained in ``2.7 Please explain the
TIME_WAIT state.'', and ``2.5 How do I properly close a socket?''.
4.2. Why don't my sockets close?
When you issue the close() system call, you are closing your interface
to the socket, not the socket itself. It is up to the kernel to close
the socket. Sometimes, for really technical reasons, the socket is
kept alive for a few minutes after you close it. It is normal, for
example for the socket to go into a TIME_WAIT state, on the server
side, for a few minutes. People have reported ranges from 20 seconds
to 4 minutes to me. The official standard says that it should be 4
minutes. On my Linux system it is about 2 minutes. This is explained
in great detail in ``2.7 Please explain the TIME_WAIT state.''.
4.3. How can I make my server a daemon?
There are two approaches you can take here. The first is to use inetd
to do all the hard work for you. The second is to do all the hard
work yourself.
If you use inetd, you simply use stdin, stdout, or stderr for your
socket. (These three are all created with dup() from the real socket)
You can use these as you would a socket in your code. The inetd
process will even close the socket for you when you are done.
If you wish to write your own server, there is a detailed explanation
in "Unix Network Programming" by Richard Stevens (see ``1.5 Where can
I get source code for the book [book title]?''). I also picked up
this posting from comp.unix.programmer, by Nikhil Nair
([email protected]). You may want to add code to ignore SIGPIPE,
because if this signal is not dealt with, it will cause your
application to exit. (Thanks to [email protected] for pointing
this out).
I worked all this lot out from the GNU C Library Manual (on-line
documentation). Here's some code I wrote - you can adapt it as necessary:
#include <stdio.h>
#include <stdlib.h>
#include <ctype.h>
#include <unistd.h>
#include <fcntl.h>
#include <signal.h>
#include <sys/wait.h>
/* Global variables */
volatile sig_atomic_t keep_going = 1; /* controls program termination */
/* Function prototypes: */
void termination_handler (int signum); /* clean up before termination */
int
main (void)
{
...
if (chdir (HOME_DIR)) /* change to directory containing data
files */
{
fprintf (stderr, "`%s': ", HOME_DIR);
perror (NULL);
exit (1);
}
/* Become a daemon: */
switch (fork ())
{
case -1: /* can't fork */
perror ("fork()");
exit (3);
case 0: /* child, process becomes a daemon: */
close (STDIN_FILENO);
close (STDOUT_FILENO);
close (STDERR_FILENO);
if (setsid () == -1) /* request a new session (job control) */
{
exit (4);
}
break;
default: /* parent returns to calling process: */
return 0;
}
/* Establish signal handler to clean up before termination: */
if (signal (SIGTERM, termination_handler) == SIG_IGN)
signal (SIGTERM, SIG_IGN);
signal (SIGINT, SIG_IGN);
signal (SIGHUP, SIG_IGN);
/* Main program loop */
while (keep_going)
{
...
}
return 0;
}
void
termination_handler (int signum)
{
keep_going = 0;
signal (signum, termination_handler);
}
4.4. How can I listen on more than one port at a time?
The best way to do this is with the select() call. This tells the
kernel to let you know when a socket is available for use. You can
have one process do i/o with multiple sockets with this call. If you
want to wait for a connect on sockets 4, 6 and 10 you might execute
the following code snippet:
fd_set socklist;
FD_ZERO(&socklist); /* Always clear the structure first. */
FD_SET(4, &socklist);
FD_SET(6, &socklist);
FD_SET(10, &socklist);
if (select(11, NULL, &socklist, NULL, NULL) < 0)
perror("select");
The kernel will notify us as soon as a file descriptor which is less
than 11 (the first parameter to select()), and is a member of our
socklist becomes available for writing. See the man page on select()
for more details.
4.5. What exactly does SO_REUSEADDR do?
This socket option tells the kernel that even if this port is busy (in
the TIME_WAIT state), go ahead and reuse it anyway. If it is busy,
but with another state, you will still get an address already in use
error. It is useful if your server has been shut down, and then
restarted right away while sockets are still active on its port. You
should be aware that if any unexpected data comes in, it may confuse
your server, but while this is possible, it is not likely.
It has been pointed out that "A socket is a 5 tuple (proto, local
addr, local port, remote addr, remote port). SO_REUSEADDR just says
that you can reuse local addresses. The 5 tuple still must be
unique!" by Michael Hunter ([email protected]). This is true, and this
is why it is very unlikely that unexpected data will ever be seen by
your server. The danger is that such a 5 tuple is still floating
around on the net, and while it is bouncing around, a new connection
from the same client, on the same system, happens to get the same
remote port. This is explained by Richard Stevens in ``2.7 Please
explain the TIME_WAIT state.''.
4.6. What exactly does SO_LINGER do?
On some unixes this does nothing. On others, it instructs the kernel
to abort tcp connections instead of closing them properly. This can
be dangerous. If you are not clear on this, see ``2.7 Please explain
the TIME_WAIT state.''.
4.7. What exactly does SO_KEEPALIVE do?
From Andrew Gierth ([email protected]):
The SO_KEEPALIVE option causes a packet (called a 'keepalive probe')
to be sent to the remote system if a long time (by default, more than
2 hours) passes with no other data being sent or received. This packet
is designed to provoke an ACK response from the peer. This enables
detection of a peer which has become unreachable (e.g. powered off or
disconnected from the net). See ``2.8 Why does it take so long to
detect that the peer died?'' for further discussion.
Note that the figure of 2 hours comes from RFC1122, "Requirements for
Internet Hosts". The precise value should be configurable, but I've
often found this to be difficult. The only implementation I know of
that allows the keepalive interval to be set per-connection is SVR4.2.
4.8. How can I bind() to a port number < 1024?
From Andrew Gierth ([email protected]):
The restriction on access to ports < 1024 is part of a (fairly weak)
security scheme particular to UNIX. The intention is that servers (for
example rlogind, rshd) can check the port number of the client, and if
it is < 1024, assume the request has been properly authorised at the
client end.
The practical upshot of this, is that binding a port number < 1024 is
reserved to processes having an effective UID == root.
This can, occasionally, itself present a security problem, e.g. when a
server process needs to bind a well-known port, but does not itself
need root access (news servers, for example). This is often solved by
creating a small program which simply binds the socket, then restores
the real userid and exec()s the real server. This program can then be
made setuid root.
4.9. How do I get my server to find out the client's address / host-
name?
From Andrew Gierth ([email protected]):
After accept()ing a connection, use getpeername() to get the address
of the client. The client's address is of course, also returned on
the accept(), but it is essential to initialise the address-length
parameter before the accept call for this will work.
Jari Kokko ([email protected]) has offered the following code to
determine the client address:
int t;
int len;
struct sockaddr_in sin;
struct hostent *host;
len = sizeof sin;
if (getpeername(t, (struct sockaddr *) &sin, &len) < 0)
perror("getpeername");
else {
if ((host = gethostbyaddr((char *) &sin.sin_addr,
sizeof sin.sin_addr,
AF_INET)) == NULL)
perror("gethostbyaddr");
else printf("remote host is '%s'\n", host->h_name);
}
4.10. How should I choose a port number for my server?
The list of registered port assignments can be found in STD 2 or RFC
1700. Choose one that isn't already registered, and isn't in
/etc/services on your system. It is also a good idea to let users
customize the port number in case of conflicts with other un-
registered port numbers in other servers. The best way of doing this
is hardcoding a service name, and using getservbyname() to lookup the
actual port number. This method allows users to change the port your
server binds to by simply editing the /etc/services file.
4.11. What is the difference between SO_REUSEADDR and SO_REUSEPORT?
SO_REUSEADDR allows your server to bind to an address which is in a
TIME_WAIT state. It does not allow more than one server to bind to
the same address. It was mentioned that use of this flag can create a
security risk because another server can bind to a the same port, by
binding to a specific address as opposed to INADDR_ANY. The
SO_REUSEPORT flag allows multiple processes to bind to the same
address provided all of them use the SO_REUSEPORT option.
From Richard Stevens ([email protected]):
This is a newer flag that appeared in the 4.4BSD multicasting code
(although that code was from elsewhere, so I am not sure just who
invented the new SO_REUSEPORT flag).
What this flag lets you do is rebind a port that is already in use,
but only if all users of the port specify the flag. I believe the
intent is for multicasting apps, since if you're running the same app
on a host, all need to bind the same port. But the flag may have
other uses. For example the following is from a post in February:
From Stu Friedberg ([email protected]):
SO_REUSEPORT is also useful for eliminating the
try-10-times-to-bind hack in ftpd's data connection setup
routine. Without SO_REUSEPORT, only one ftpd thread can
bind to TCP (lhost, lport, INADDR_ANY, 0) in preparation for
connecting back to the client. Under conditions of heavy
load, there are more threads colliding here than the
try-10-times hack can accomodate. With SO_REUSEPORT, things
work nicely and the hack becomes unnecessary.
I have also heard that DEC OSF supports the flag. Also note that
under 4.4BSD, if you are binding a multicast address, then
SO_REUSEADDR is condisered the same as SO_REUSEPORT (p. 731 of "TCP/IP
Illustrated, Volume 2"). I think under Solaris you just replace
SO_REUSEPORT with SO_REUSEADDR.
From a later Stevens posting, with minor editing:
Basically SO_REUSEPORT is a BSD'ism that arose when multicasting was
added, even thought it was not used in the original Steve Deering
code. I believe some BSD-derived systems may also include it (OSF,
now Digital Unix, perhaps?). SO_REUSEPORT lets you bind the same
address *and* port, but only if all the binders have specified it.
But when binding a multicast address (its main use), SO_REUSEADDR is
considered identical to SO_REUSEPORT (p. 731, "TCP/IP Illustrated,
Volume 2"). So for portability of multicasting applications I always
use SO_REUSEADDR.
4.12. How can I write a multi-homed server?
The original question was actually from Shankar Ramamoorthy
([email protected]):
I want to run a server on a multi-homed host. The host is
part of two networks and has two ethernet cards. I want to
run a server on this machine, binding to a pre-determined
port number. I want clients on either subnet to be able to
send broadcast packates to the port and have the server
receive them.
And answered by Andrew Gierth ([email protected]):
Your first question in this scenario is, do you need to know which
subnet the packet came from? I'm not at all sure that this can be
reliably determined in all cases.
If you don't really care, then all you need is one socket bound to
INADDR_ANY. That simplifies things greatly.
If you do care, then you have to bind multiple sockets. You are
obviously attempting to do this in your code as posted, so I'll assume
you do.
I was hoping that something like the following would work.
Will it? This is on Sparcs running Solaris 2.4/2.5.
I don't have access to Solaris, but I'll comment based on my
experience with other Unixes.
[Shankar's original code omitted]
What you are doing is attempting to bind all the current hosts unicast
addresses as listed in hosts/NIS/DNS. This may or may not reflect
reality, but much more importantly, neglects the broadcast addresses.
It seems to be the case in the majority of implementations that a
socket bound to a unicast address will not see incoming packets with
broadcast addresses as their destinations.
The approach I've taken is to use SIOCGIFCONF to retrieve the list of
active network interfaces, and SIOCGIFFLAGS and SIOCGIFBRDADDR to
identify broadcastable interfaces and get the broadcast addresses.
Then I bind to each unicast address, each broadcast address, and to
INADDR_ANY as well. That last is necessary to catch packets that are
on the wire with INADDR_BROADCAST in the destination. (SO_REUSEADDR
is necessary to bind INADDR_ANY as well as the specific addresses.)
This gives me very nearly what I want. The wrinkles are:
o I don't assume that getting a packet through a particular socket
necessarily means that it actually arrived on that interface.
o I can't tell anything about which subnet a packet originated on if
its destination was INADDR_BROADCAST.
o On some stacks, apparently only those with multicast support, I get
duplicate incoming messages on the INADDR_ANY socket.
4.13. How can I read only one character at a time?
This question is usually asked by people who are testing their server
with telnet, and want it to process their keystrokes one character at
a time. The correct technique is to use a psuedo terminal (pty).
More on that in a minute.
According to Roger Espel Llima ([email protected]), you can have
your server send a sequence of control characters: 0xff 0xfb 0x01 0xff
0xfb 0x03 0xff 0xfd 0x0f3, which translates to IAC WILL ECHO IAC WILL
SUPPRESS-GO-AHEAD IAC DO SUPPRESS-GO-AHEAD. For more information on
what this means, check out std8, std28 and std29. Roger also gave the
following tips:
o This code will suppress echo, so you'll have to send the characters
the user types back to the client if you want the user to see them.
o Carriage returns will be followed by a null character, so you'll
have to expect them.
o If you get a 0xff, it will be followed by two more characters.
These are telnet escapes.
Use of a pty would also be the correct way to execute a child process
and pass the i/o to a socket.
I'll add pty stuff to the list of example source I'd like to add to
the faq. If someone has some source they'd like to contribute
(without copyright) to the faq which demonstrates use of pty's, please
email me!
4.14. I'm trying to exec() a program from my server, and attach my
socket's IO to it, but I'm not getting all the data across. Why?
If the program you are running uses printf(), etc (streams from
stdio.h) you have to deal with two buffers. The kernel buffers all
socket IO, and this is explained in ``section 2.11''. The second
buffer is the one that is causing you grief. This is the stdio
buffer, and the problem was well explained by Andrew:
(The short answer to this question is that you want to use a pty
rather than a socket; the remainder of this article is an attempt to
explain why.)
Firstly, the socket buffer controlled by setsockopt() has absolutly
nothing to do with stdio buffering. Setting it to 1 is guaranteed to
be the Wrong Thing(tm).
Perhaps the following diagram might make things a little clearer:
Process A Process B
+---------------------+ +---------------------+
| | | |
| mainline code | | mainline code |
| | | | ^ |
| v | | | |
| fputc() | | fgetc() |
| | | | ^ |
| v | | | |
| +-----------+ | | +-----------+ |
| | stdio | | | | stdio | |
| | buffer | | | | buffer | |
| +-----------+ | | +-----------+ |
| | | | ^ |
| | | | | |
| write() | | read() |
| | | | | |
+-------- | ----------+ +-------- | ----------+
| | User space
------------|-------------------------- | ---------------------------
| | Kernel space
v |
+-----------+ +-----------+
| socket | | socket |
| buffer | | buffer |
+-----------+ +-----------+
| ^
v |
(AF- and protocol- (AF- and protocol-
dependent code) dependent code)
Assuming these two processes are communicating with each other (I've
deliberately omitted the actual comms mechanisms, which aren't really
relevent), you can see that data written by process A to its stdio
buffer is completely inaccessible to process B. Only once the decision
is made to flush that buffer to the kernel (via write()) can the data
actually be delivered to the other process.
The only guaranteed way to affect the buffering within process A is to
change the code. However, the default buffering for stdout is
controlled by whether the underlying FD refers to a terminal or not;
generally, output to terminals is line-buffered, and output to non-
terminals (including but not limited to files, pipes, sockets, non-tty
devices, etc.) is fully buffered. So the desired effect can usually be
achieved by using a pty device; this, for example, is what the
'expect' program does.
Since the stdio buffer (and the FILE structure, and everything else
related to stdio) is user-level data, it is not preserved across an
exec() call, hence trying to use setvbuf() before the exec is
ineffective.
A couple of alternate solutions were proposed by Roger Espel Llima
([email protected]):
If it's an option, you can use some standalone program that will just
run something inside a pty and buffer its input/output. I've seen a
package by the name pty.tar.gz that did that; you could search around
for it with archie or AltaVista.
Another option (**warning, evil hack**) , if you're on a system that
supports this (SunOS, Solaris, Linux ELF do; I don't know about
others) is to, on your main program, putenv() the name of a shared
executable (*.so) in LD_PRELOAD, and then in that .so redefine some
commonly used libc function that the program you're exec'ing is known
to use early. There you can 'get control' on the running program, and
the first time you get it, do a setbuf(stdout, NULL) on the program's
behalf, and then call the original libc function with a dlopen() +
dlsym(). And you keep the dlsym() value on a static var, so you can
just call that the following times.
(Editors note: I still haven't done an expample for how to do pty's,
but I hope I will be able to do one after I finish the non-blocking
example code.)
5. Writing UDP/SOCK_DGRAM applications
5.1. When should I use UDP instead of TCP?
UDP is good for sending messages from one system to another when the
order isn't important and you don't need all of the messages to get to
the other machine. This is why I've only used UDP once to write the
example code for the faq. Usually TCP is a better solution. It saves
you having to write code to ensure that messages make it to the
desired destination, or to ensure the message ordering. Keep in mind
that every additional line of code you add to your project in another
line that could contain a potentially expensive bug.
If you find that TCP is too slow for your needs you may be able to get
better performance with UDP so long as you are willing to sacrifice
message order and/or reliability.
UDP must be used to multicast messages to more than one other machine
at the same time. With TCP an application would have to open separate
connections to each of the destination machines and send the message
once to each target machine. This limits your application to only
communicate with machines that it already knows about.
5.2. What is the difference between "connected" and "unconnected"
sockets?
From Andrew Gierth ([email protected]):
If a UDP socket is unconnected, which is the normal state after a
bind() call, then send() or write() are not allowed, since no
destination address is available; only sendto() can be used to send
data.
Calling connect() on the socket simply records the specified address
and port number as being the desired communications partner. That
means that send() or write() are now allowed; they use the destination
address and port given on the connect call as the destination of the
packet.
5.3. of the socket? Does doing a connect() call affect the receive
behaviour
From Richard Stevens ([email protected]):
Yes, in two ways. First, only datagrams from your "connected peer"
are returned. All others arriving at your port are not delivered to
you.
But most importantly, a UDP socket must be connected to receive ICMP
errors. Pp. 748-749 of "TCP/IP Illustrated, Volume 2" give all the
gory details on why this is so.
5.4. How can I read ICMP errors from "connected" UDP sockets?
If the target machine discards the message because there is no process
reading on the requested port number, it sends an ICMP message to your
machine which will cause the next system call on the socket to return
ECONNREFUSED. Since delivery of ICMP messages is not guarenteed you
may not recieve this notification on the first transaction.
Remember that your socket must be "connected" in order to receive the
ICMP errors. I've been told, and Alan Cox has verified that Linux
will return them on "unconnected" sockets. This may cause porting
problems if your application isn't ready for it, so Alan tells me
they've added a SO_BSDCOMPAT flag which can be set for Linux kernels
after 2.0.0.
5.5. How can I be sure that a UDP message is received?
You have to design your protocol to expect a confirmation back from
the destination when a message is received. Of course is the
confirmation is sent by UDP, then it too is unreliable and may not
make it back to the sender. If the sender does not get confirmation
back by a certain time, it will have to re-transmit the message, maybe
more than once. Now the receiver has a problem because it may have
already received the message, so some way of dropping duplicates is
required. Most protocols use a message numbering scheme so that the
receiver can tell that it has already processed this message and
return another confirmation. Confirmations will also have to
reference the message number so that the sender can tell which message
is being confirmed. Confused? That's why I stick with TCP.
5.6. How can I be sure that UDP messages are received in order?
You can't. What you can do is make sure that messages are processed
in order by using a numbering system as mentioned in ``5.5 How can I
be sure that a UDP message is received?''. If you need your messages
to be received and be received in order you should really consider
switching to TCP. It is unlikely that you will be able to do a better
job implementing this sort of protocol than the TCP people already
have, without a significant investment of time.
5.7. How often should I re-transmit un-acknowleged messages?
The simplest thing to do is simply pick a fairly small delay such as
one second and stick with it. The problem is that this can congest
your network with useless traffic if there is a problem on the lan or
on the other machine, and this added traffic may only serve to make
the problem worse.
A better technique, described with source code in "UNIX Network
Programming" by Richard Stevens (see ``1.5 Where can I get source code
for the book [book title]?''), is to use an adaptive timeout with an
exponential backoff. This technique keeps statistical information on
the time it is taking messages to reach a host and adjusts timeout
values accordingly. It also doubles the timeout each time it is
reached as to not flood the network with useless datagrams. Richard
has been kind enough to post the source code for the book on the web.
Check out his home page at http://www.noao.edu/~rstevens.
5.8. How come only the first part of my datagram is getting through?
This has to do with the maximum size of a datagram on the two machines
involved. This depends on the sytems involved, and the MTU (Maximum
Transmission Unit). According to "UNIX Network Programming", all
TCP/IP implementations must support a minimum IP datagram size of 576
bytes, regardless of the MTU. Assuming a 20 byte IP header and 8 byte
UDP header, this leaves 548 bytes as a safe maximum size for UDP
messages. The maximum size is 65516 bytes. Some platforms support IP
fragmentation which will allow datagrams to be broken up (because of
MTU values) and then re-assembled on the other end, but not all
implementations support this.
This information is taken from my reading of "UNIX Netowrk
Programming" (see ``1.5 Where can I get source code for the book [book
title]?'').
Andrew has pointed out the following regarding large UDP messages:
Another issue is fragmentation. If a datagram is sent which is too
large for the network interface it is sent through, then the sending
host will fragment it into smaller packets which are reassembled by
the receiving host. Also, if there are intervening routers, then they
may also need to fragment the packet(s), which greatly increases the
chances of losing one or more fragments (which causes the entire
datagram to be dropped). Thus, large UDP datagrams should be avoided
for applications that are likely to operate over routed nets or the
Internet proper.
5.9. Why does the socket's buffer fill up sooner than expected?
From Paul W. Nelson ([email protected]):
In the traditional BSD socket implementation, sockets that are atomic
such as UDP keep received data in lists of mbufs. An mbuf is a fixed
size buffer that is shared by various protocol stacks. When you set
your receive buffer size, the protocol stack keeps track of how many
bytes of mbuf space are on the receive buffer, not the number of
actual bytes. This approach is used because the resource you are
controlling is really how many mbufs are used, not how many bytes are
being held in the socket buffer. (A socket buffer isn't really a
buffer in the traditional sense, but a list of mbufs).
For example: Lets assume your UNIX has a small mbuf size of 256
bytes. If your receive socket buffer is set to 4096, you can fit 16
mbufs on the socket buffer. If you receive 16 UDP packets that are 10
bytes each, your socket buffer is full, and you have 160 bytes of
data. If you receive 16 UDP packets that are 200 bytes each, your
socket buffer is also full, but contains 3200 bytes of data. FIONREAD
returns the total number of bytes, not the number of messages or bytes
of mbufs. Because of this, it is not a good indicator of how full
your receive buffer is.
Additionaly, if you receive UDP messages that are 260 bytes, you use
up two mbufs, and can only recieve 8 packets before your socket buffer
is full. In this case, only 2080 bytes of the 4096 are held in the
socket buffer.
This example is greatly simplified, and the real socket buffer
algorithm also takes into account some other parameters. Note that
some older socket implementations use a 128 byte mbuf.
6. Advanced Socket Programming
6.1. How would I put my socket in non-blocking mode?
From Andrew Gierth ([email protected]):
Technically, fcntl(soc, F_SETFL, O_NONBLOCK) is incorrect since it
clobbers all other file flags. Generally one gets away with it since
the other flags (O_APPEND for example) don't really apply much to
sockets. In a similarly rough vein, you would use fcntl(soc, F_SETFL,
0) to go back to blocking mode.
To do it right, use F_GETFL to get the current flags, set or clear the
O_NONBLOCK flag, then use F_SETFL to set the flags.
And yes, the flag can be changed either way at will.
6.2. How can I put a timeout on connect()?
Andrew Gierth ([email protected]) has outlined the
following procedure for using select() with connect(), which will
allow you to put a timeout on the connect() call:
First, create the socket and put it into non-blocking mode, then call
connect(). There are three possibilities:
o connect succeeds: the connection has been successfully made (this
usually only happens when connecting to the same machine)
o connect fails: obvious
o connect returns -1/EINPROGRESS. The connection attempt has begun,
but not yet completed.
If the connection succeeds:
o the socket will select() as writable (and will also select as
readable if data arrives)
If the connection fails:
o the socket will select as readable *and* writable, but either a
read or write will return the error code from the connection
attempt. Also, you can use getsockopt(SO_ERROR) to get the error
status - but be careful; some systems return the error code in the
result parameter of getsockopt(), but others (incorrectly) cause
the getsockopt call itself to fail with the stored value as the
error.
Sample code that illustrates this can be found in the socket-faq
examples, in the file connect.c.
7. Sample Source Code
The sample source code is no longer included in the faq. To get it,
please download it from one of the unix-socket-faq www pages:
http://www.ibrado.com/sock-faq
http://kipper.york.ac.uk/~vic/sock-faq
http://www.ntua.gr/sock-faq
If you don't have web access, you can ftp it with ftpmail by following
the following instructions. Please do not use the ftp server if you
have access to the web, since computain.com is connected only by a
28.8 modem, and you'd be amazed how much traffic this faq generates.
To get the sample source by mail, send mail to [email protected],
with no subject line and a body like this:
reply <put your email address here>
connect ftp.computain.com
binary
uuencode
get pub/sockets/examples.tar.gz
quit
Save the reply as examples.uu, and type:
% uudecode examples.uu
% gunzip examples.tar.gz
% tar xf examples.tar
This will create a directory called socket-faq-examples which contains
the sample code from this faq, plus a sample client and server for
both tcp and udp.
Note that this package requires the gnu unzip program to be installed
on your system. It is very common, but if you don't have it you can
get the source for it from:
ftp://prep.ai.mit.edu/pub/gnu/gzip-1.2.4.tar
If you don't have ftp access, you can obtain it in a way similar to
obtaining the sample source. I'll leave the exact changes to the body
of the message as an excersise for the reader.
|
Закладки на сайте Проследить за страницей |
Created 1996-2024 by Maxim Chirkov Добавить, Поддержать, Вебмастеру |