As per my understanding a simple TCP server will be coded as follows.
socket() - bind() - listen() - accept() - read() - write()
The clients will be written as follows.
socket() - bind()(Optional) - connect() - write() - read()
Please note the order difference in read() and write() calls between client and server program.
Is it a requirement to always read() before write() in a server program and if, then why?
Thanks,
Naga
That isn't mandatory, but it makes sense for the server to read the request before writing a response. Note that it is necessary to read on both sides often enough to prevent a distributed deadlock: for example, if the both sides are trying to write and not reading, then the buffers in-between will get full and neither one's write will be able to proceed. One solution for this is to have a separate thread which keeps reading, if there is something to read (this applies to both the client and the server).
The simple answer is no. You are free to do whatever you like.
However, I'll caveat that quickly with the fact that most protocols are designed to wait for the client to send something. After all, the server, by nature, serves requests and needs to wait to know what that request is, be it "GET /" or "HELO" or whatever. So, it is fairly natural for a sever to read before writing any response back to the client.
That said, you could if you felt like it dump version information down to the client before you do any reading. To see the effect, connect to your server using telnet.
You can perform them in either order. However, a server will normally generate a response from the read() operation, then write it with the write() operation, so this order makes sense.
If you're handling multiple clients, you should use a multiplexer like select to notify you when clients have data ready to read, so your server won't lock up the every time you try to read() from a client who hasn't sent anything.
It isn't a requirement, server program can write to socket without reading first. But in many cases server program must know what client wants - so it calls read() first.
Related
Let's suppose, I have a custom server that listens to connections on some port and once it has received a connection, it starts sending data (sort of a logger). Here's the first question:
Can it be just binary data? Actually, I need just two non-zero 8-bit values, and I was thinking of 0-value byte to separate each new portion of data.
These three bytes will be sent once or may be twice a second.
So, now I am looking for some code snippet in Swift 2 to properly read this data. Normally, I would expect calling
connectSocket(IP,port)
which would connect to the socket, and once it receives the first chunk of data,
socketCallBack()
is called, or something like that.
Intuitively, I don't like the idea of checking data in a while (true) loop. Or is this the proper way?
I've seen an example, when it first sends 'get' request to the server and immediately starts waiting for response. Probably, I can call it using a timer, once a second? Will it be correct?
What I am concerned about is trafic. Right now I have impemented it through a web-server, but I don't like that it spends way too much trafic for that overhead http data.
Probably, with that tcp connections on timer that would be much less, and it would save even more trafic if I establish just one connection in the beginning and transmit the data within this connection. Am I right?
I'm trying to understand the idea of non-blocking web server and it seems like there is something I miss.
I can understand there are several reasons for "block" web request(psuedocode):
CPU bound
string on_request(arg)
{
DO_SOME_HEAVY_CPU_CALC
return "done";
}
IO bound
string on_request(arg)
{
DO_A_CALL_TO_EXTERNAL_RESOURCE_SUCH_AS_WEB_IO
return "done";
}
sleep
string on_request(arg)
{
sleep(VERY_VERY_LONG_TIME);
return "done";
}
are all the three can benefit from non-blocking server?
how the situation that do benefit from the non-blocking web server really do that?
I mean, when looking at the Tornado server documentation, it seems
like it "free" the thread. I know that a thread can be put to sleep
and wait for a signal from the operation system (at least in Linux),
is this the meaning of "freeing" the thread? is this some higher
level implementation? something that actually create a new thread
that is waiting for new request instead of the "sleeping" one?
Am I missing something here?
Thanks
Basically the way the non-blocking sockets I/O work is by using polling and the state machine. So your scheme for many connections would be something like that:
Create many sockets and make them nonblocking
Switch the state of them to "connect"
Initiate the connect operation on each of them
Poll all of them until some events fire up
Process the fired up events (connection established or connection failed)
Switch the state those established to "sending"
Prepare the Web request in a buffer
Poll "sending" sockets for WRITE operation
send the data for those who got the WRITE event set
For those which have all the data sent, switch the state to "receiving"
Poll "receiving" sockets for READ operation
For those which have the READ event set, perform read and process the read data according to the protocol
Repeat if the protocol is bidirectional, or close the socket if it is not
Of course, at each stage you need to handle errors, and that the state of each socket is different (one may be connecting while another may be already reading).
Regarding polling I have posted an article about how different polling methods work here: http://www.ulduzsoft.com/2014/01/select-poll-epoll-practical-difference-for-system-architects/ - I suggest you check it.
To benefit from a non-blocking server, your code must also be non-blocking - you can't just run blocking code on a non-blocking server and expect better performance. For example, you must remove all calls to sleep() and replace them with non-blocking equivalents like IOLoop.add_timeout (which in turn involves restructuring your code to use callbacks or coroutines).
How To Use Linux epoll with Python http://scotdoyle.com/python-epoll-howto.html may give you some points about this topic.
I am writing a Client/Server application in C++ with the help of Boost Asio. I have a working server, and the server workflow is something I understand well.
My client application handles the connect gracefully as shown in Asio examples, after which, it exchanges a handshake with the server. After that however, the users should be able to send requests to the server when and how they want, which is where I have a problem understanding the paradigm.
The initial workflow goes like a little like this:
OnConnected() { SendHandshake() }
SendHandshake() { async.write_some(handshake...), async_read_some(&OnRead) }
OnRead() { ReadServerHandshake() *** }
And users would send messages by using Write(msg):
Write (msg) { async_write_some(msg,&OnWrite), async_Read_some(&OnRead) }
OnWrite() {}
EDIT: Rephrasing the question to be clearer, here is the scenario:
After the initial handshaking is complete, the Client is only used to send requests to the server, on which it will get a reply. So, for instance, a user sends a write. Client waits for the read operation to complete, reads the reply and does something with it. The next user write will only come after, say, 5 minutes. Will the io_service stop working in the meanwhile because there are no outstanding asynchronous operations in between the last reply read and the next write?
On an informative note, you can provide it with io_service::work to stop an io_service from running out of work. This will ensure that the io_service::run never returns until the work object is destroyed.
To control the lifetime of the work object, you can use a shared_ptr pointer and reset it once the work is done, or you can use boost::optional as outlined here.
Of course you still need to handle the case where either the server closes the TCP connection, or the connection dies for whatever reason. To handle this case, one solution would be to have an outstanding async_read on the socket to the server. The read handler should be called with an error_code when/if something goes wrong with the connection. If you have the outstanding read on the connection, you do not need to use the work object.
If you want the IO service to complete a read, you must start a read. If you want to read data any time the client sends it, you must have an asynchronous read operation pending at all times. Otherwise, how would the library know what to do with the data?
Is there a way to only close "one end" of a TCP socket to cleanly indicate one side of a connection is done writing to the connection? (Just like you do with a pipe in every Unix pipe tutorial ever.) Or should I use some in-band solution like a sentinel value or some such?
You can shutdown a socket for read or write using the second parameter to the method:
shutdown(sock, SHUT_RD)
shutdown(sock, SHUT_WR)
If the server is doing the writing, and does a shutdown() for write, the client should get an end of file when it tries to read (rather than blocking and waiting for data to arrive). It will however still be able to write to the socket.
I was wondering how tcp/ip communication is implemented in unix. When you do a send over the socket, does the tcp/level work (assembling packets, crc, etc) get executed in the same execution context as the calling code?
Or, what seems more likely, a message is sent to some other daemon process responsible for tcp communication? This process then takes the message and performs the requested work of copying memory buffers and assembling packets etc.? So, the calling code resumes execution right away and tcp work is done in parallel? Is this correct?
Details would be appreciated. Thanks!
The TCP/IP stack is part of your kernel. What happens is that you call a helper method which prepares a "kernel trap". This is a special kind of exception which puts the CPU into a mode with more privileges ("kernel mode"). Inside of the trap, the kernel examines the parameters of the exception. One of them is the number of the function to call.
When the function is called, it copies the data into a kernel buffer and prepares everything for the data to be processed. Then it returns from the trap, the CPU restores registers and its original mode and execution of your code resumes.
Some kernel thread will pick up the copy of the data and use the network driver to send it out, do all the error handling, etc.
So, yes, after copying the necessary data, your code resumes and the actual data transfer happens in parallel.
Note that this is for TCP packets. The TCP protocol does all the error handling and handshaking for you, so you can give it all the data and it will know what to do. If there is a problem with the connection, you'll notice only after a while since the TCP protocol can handle short network outages by itself. That means you'll have "sent" some data already before you'll get an error. That means you will get the error code for the first packet only after the Nth call to send() or when you try to close the connection (the close() will hang until the receiver has acknowledged all packets).
The UDP protocol doesn't buffer. When the call returns, the packet is on it's way. But it's "fire and forget", so you only know that the driver has put it on the wire. If you want to know whether it has arrived somewhere, you must figure out a way to achieve that yourself. The usual approach is have the receiver send an ack UDP packet back (which also might get lost).
No - there is no parallel execution. It is true that the execution context when you're making a system call is not the same as your usual execution context. When you make a system call, such as for sending a packet over the network, you must switch into the kernel's context - the kernel's own memory map and stack, instead of the virtual memory you get inside your process.
But there are no daemon processes magically dispatching your call. The rest of the execution of your program has to wait for the system call to finish and return whatever values it will return. This is why you can count on return values being available right away when you return from the system call - values like the number of bytes actually read from the socket or written to a file.
I tried to find a nice explanation for how the context switch to kernel space works. Here's a nice in-depth one that even focuses on architecture-specific implementation:
http://www.ibm.com/developerworks/linux/library/l-system-calls/