Non-blocking UDP recv in Haskell - sockets

I'm still learning the basics of Haskell and currently working through porting some Java code to Haskell. My current problem is in UDP recvFrom using Network.Socket.ByteString.
The problem is with this method:
public abstract SocketAddress receive(ByteBuffer dst) throws IOException
Receives a datagram via this channel.
If a datagram is immediately available, or if this channel
is in blocking mode and one eventually becomes available,
then the datagram is copied into the given byte buffer and
its source address is returned. If this channel is in
non-blocking mode and a datagram is not immediately available
then this method immediately returns null.
The thing is that when I use Network.Socket.ByteString.recvFrom my code blocks at this point waiting for the packet to come. It doesn't return something like Maybe to indicate if something was received or not (the same way in Java there is a null returned when no data was available)
I found this thread: https://mail.haskell.org/pipermail/haskell-cafe/2010-August/082725.html
At the end of it there are a couple of ways suggested: 1) FFI 2) run recvFrom in its own thread
I'm not sure I'm capable of using any of those approaches at this moment (not enough knowledge). What I want is to get something similar to Java way of non-blocking receive: get needed info if it is available or just nothing if there is no single UDP packet. Anyone can point out what would be a better approach, any code snippets, someone already handled this problem?

You could use socketToHandle together with hGetNonBlocking:
recvNonBlocking :: Socket -> Int -> IO ByteString
recvNonBlocking s n = do
hnd <- socketToHandle s ReadMode
hGetNonBlocking hnd n
However, keep in mind that you cannot use the Socket after a call to socketToHandle, so this is only feasible if you would close the Socket either way afterwards.

Related

Non-blocking TCP socket fails to connect using socket2

I am currently in the process of converting some of my code from blocking to non-blocking using the sockets2 crate, however I am running into issues with connecting the socket. The socket always fails to connect before the timeout is exceeded. Despite my attempts to search for examples, I have yet to find any Rust code showing how a non-blocking TCP stream is created.
To give you an idea what I am attempting to do, the code I am currently converting looks looks roughly like this. This gives me no issues and works fine, but it is getting too costly to create a new thread for every socket.
let address = SocketAddr::from(([x, y, z, v], port));
let mut socket = TcpStream::connect_timeout(&address, timeout)?;
At the moment, my code to connect the socket looks like this. Since connect_timeout can only be executed in blocking mode, I use connect instead and regularly poll the socket to check if it is connected. At the moment, I keep getting WouldBlock errors when calling connect, but I do not know what this means. At first I assumed that the connect was proceeding, but returning the result immediately would require blocking so a WouldBlock error was given instead. However, due to the issues getting the socket to connect, I am second guessing those assumptions.
let address = SocketAddr::from(([x, y, z, v], port));
// Create socket equivalent to TcpStream
let socket = Socket::new(Domain::IPV4, Type::STREAM, Some(Protocol::TCP))?;
// Enable non-blocking mode on the socket
socket.set_nonblocking(true)?;
// What response should I expect? Do I need to bind an address first?
match socket.connect(&address.into()) {
Ok(_) => {}
Err(e) if e.kind() == ErrorKind::WouldBlock => {
// I keep getting this error, but I don't know what this means.
// Is non-blocking connect unavailable?
// Do I need to keep trying to connect until it succeeds?
},
// Are there any other types of errors I should be looking for before failing the connection?
Err(e) => return Err(e),
}
I am also unsure what the correct approach is to determine if a socket is connected. At the moment, I attempt to read to a zero length buffer and check if I get a NotConnected error. However, I am unsure what WouldBlock means in this context and I have never gotten a positive response from this approach.
let mut buffer = [0u8; 0];
// I also tried self.socket.peer_addr(), but ran into issues where it returned a positive
// response despite not being connected.
match self.socket.read(&mut buffer) {
Ok(_) => Ok(true),
// What does WouldBlock mean in this context?
Err(e) if e.kind() == ErrorKind::WouldBlock => Ok(false),
Err(e) if e.kind() == ErrorKind::NotConnected => Ok(false),
Err(e) => Err(e),
}
Each socket is periodically checked until an arbitrary timeout is reached to determine if it has connected. So far, no socket has passed the connected before reaching its timeout (20 sec) when connecting to a known-good server. These tests are all performed in a single threaded application on Windows using a known-good server that has been checked to work with the blocking version of my program.
Edit: Here is a minimum reproducible example for this issue. However, it likely won't work if you run it on Rust playground due to network restrictions. https://play.rust-lang.org/?version=stable&mode=debug&edition=2021&gist=a08c22574a971c0032fd9dd37e10fd94
WouldBlock is the expected error when a non-blocking connect() (or other operation) is successfully started in the background. You can then wait up to your desired timeout interval for the operation to finish (use select() or epoll() or other platform-specific notification to detect this). If the timeout elapses, close the socket and handle the timeout accordingly. Otherwise, check the socket's SO_ERROR option to see if the operation was successful or failed, and act accordingly.
To give you an idea what I am attempting to do, the code I am currently converting looks looks roughly like this. This gives me no issues and works fine, but it is getting too costly to create a new thread for every socket.
This sounds to me strongly like an XY-Problem.
I think you misunderstand what 'nonblocking' means. What it does not mean is that you can simply and without worrying run multiple sockets in parallel. What it does mean is that every operation that would block returns an error instead and you have to retry it at a later time.
Actual non-blocking sockets usually don't get used at enduser level. They are meant for libraries that depend on them and provide some higher level interface for asynchronism. Non-blocking sockets are hard to get right. They need to be paired with events, because otherwise you can only implement them with 100% cpu hungry busy loops, which is most likely not what you want.
There's good news, though! Remember the high-level libraries I talked about that use nonblocking sockets internally? The most famous one right now is called tokio and does exactly what you want. It will require you to learn a programming mechanism called asynchronism, but you will grasp it, I'm sure :)
I recommend this read: https://tokio.rs/tokio/tutorial

How to deal with ZMQ sockets lack of thread safety?

I've been using ZMQ in some Python applications for a while, but only very recently I decided to reimplement one of them in Go and I realized that ZMQ sockets are not thread-safe.
The original Python implementation uses an event loop that looks like this:
while running:
socks = dict(poller.poll(TIMEOUT))
if socks.get(router) == zmq.POLLIN:
client_id = router.recv()
_ = router.recv()
data = router.recv()
requests.append((client_id, data))
for req in requests:
rep = handle_request(req)
if rep:
replies.append(rep)
requests.remove(req)
for client_id, data in replies:
router.send(client_id, zmq.SNDMORE)
router.send(b'', zmq.SNDMORE)
router.send(data)
del replies[:]
The problem is that the reply might not be ready on the first pass, so whenever I have pending requests, I have to poll with a very short timeout or the clients will wait for more than they should, and the application ends up using a lot of CPU for polling.
When I decided to reimplement it in Go, I thought it would be as simple as this, avoiding the problem by using infinite timeout on polling:
for {
sockets, _ := poller.Poll(-1)
for _, socket := range sockets {
switch s := socket.Socket; s {
case router:
msg, _ := s.RecvMessage(0)
client_id := msg[0]
data := msg[2]
go handleRequest(router, client_id, data)
}
}
}
But that ideal implementation only works when I have a single client connected, or a light load. Under heavy load I get random assertion errors inside libzmq. I tried the following:
Following the zmq4 docs I tried adding a sync.Mutex and lock/unlock on all socket operations. It fails. I assume it's because ZMQ uses its own threads for flushing.
Creating one goroutine for polling/receiving and one for sending, and use channels in the same way I used the req/rep queues in the Python version. It fails, as I'm still sharing the socket.
Same as 2, but setting GOMAXPROCS=1. It fails, and throughput was very limited because replies were being held back until the Poll() call returned.
Use the req/rep channels as in 2, but use runtime.LockOSThread to keep all socket operations in the same thread as the socket. Has the same problem as above. It doesn't fail, but throughput was very limited.
Same as 4, but using the poll timeout strategy from the Python version. It works, but has the same problem the Python version does.
Share the context instead of the socket and create one socket for sending and one for receiving in separate goroutines, communicating with channels. It works, but I'll have to rewrite the client libs to use two sockets instead of one.
Get rid of zmq and use raw TCP sockets, which are thread-safe. It works perfectly, but I'll also have to rewrite the client libs.
So, it looks like 6 is how ZMQ was really intended to be used, as that's the only way I got it to work seamlessly with goroutines, but I wonder if there's any other way I haven't tried. Any ideas?
Update
With the answers here I realized I can just add an inproc PULL socket to the poller and have a goroutine connect and push a byte to break out of the infinite wait. It's not as versatile as the solutions suggested here, but it works and I can even backport it to the Python version.
I opened an issue a 1.5 years ago to introduce a port of https://github.com/vaughan0/go-zmq/blob/master/channels.go to pebbe/zmq4. Ultimately the author decided against it, but we have used this in production (under VERY heavy workloads) for a long time now.
This is a gist of the file that had to be added to the pebbe/zmq4 package (since it adds methods to the Socket). This could be re-written in such a way that the methods on the Socket receiver instead took a Socket as an argument, but since we vendor our code anyway, this was an easy way forward.
The basic usage is to create your Socket like normal (call it s for example) then you can:
channels := s.Channels()
outBound := channels.Out()
inBound := channels.In()
Now you have two channels of type [][]byte that you can use between goroutines, but a single goroutine - managed within the channels abstraction, is responsible for managing the Poller and communicating with the socket.
The blessed way to do this with pebbe/zmq4 is with a Reactor. Reactors have the ability to listen on Go channels, but you don't want to do that because they do so by polling the channel periodically using a poll timeout, which reintroduces the same exact problem you have in your Python version. Instead you can use zmq inproc sockets, with one end held by the reactor and the other end held by a goroutine that passes data in from a channel. It's complicated, verbose, and unpleasant, but I have used it successfully.

how do sockets not missing arriving data?

a typical socket program example would be like this:
while(1){
data = socket.recv()
//do some work
}
since you don't know when package arrive,it must block to wait until get some data from the listening port,suppose if the program start a heavy work after received the command from another side,during the work , another package arrived,but because at that moment you are doing the work,you are not listening the port, you might missed the package ,no matter how fast you handle the work.
so how does the socket work to handle all the data without any lost?
The operating system has a receive buffer which holds packets that have been received from the network but not yet recv()ed by the application. If that buffer fills up packets will be lost. You don't have to be in a recv() call when packets arrive, though you should make sure you call it often enough to keep the buffer from overflowing.

Send a zero-data TCP/IP packet using Java

My goal is to send a TCP packet with empty data field, in order to test the socket with the remote machine.
I am using the OutputStream class's method of write(byte[] b).
my attempt:
outClient = ClientSocket.getOutputStream();
outClient.write(("").getBytes());
Doing so, the packet never show up on the wire. It works fine if "" is replaced by " " or any non-zero string.
I tried jpcap, which worked with me, but didn't serve my goal.
I thought of extending the OutputStream class, and implementing my own OutputStream.write method. But this is beyond my knowledge. Please give me advice if someone have already done such a thing.
If you just want to quickly detect silently dropped connections, you can use Socket.setKeepAlive( true ). This method ask TCP/IP to handle heartbeat probing without any data packets or application programming. If you want more control on the frequency of the heartbeat, however, you should implement heartbeat with application level packets.
See here for more details: http://mindprod.com/jgloss/socket.html#DISCONNECT
There is no such thing as an empty TCP packet, because there is no such thing as a TCP packet. TCP is a byte-stream protocol. If you send zero bytes, nothing happens.
To detect broken connections, short of keepalive which only helps you after two hours, you must rely on read timeouts and write exceptions.

How to set a timeout in connect/send ? ( as400 iseries v5r4, rpg )

From this rpg socket tutorial we created a socket client in rpg that calls a java server socket
The problem is that connect()/send() operations blocks and we have a requirement that if the connect/send couldn't be done in a matter of a second per say, we have to just log it and finish.
If I set the socket to non-blocking mode (I think with fnctl), we are not fully understanding how to proceed, and can't find any useful documentation with examples for it.
I think if I do connect to a non-blocking socket I have to do select(..., timeout) which tells us if the connect succeed and/ we are able to send(bytes). But, if we send(bytes) afterwards, as it is now a non-blocking socket (which will immediately return after the call), how do I know that send() did the actual sending of the bytes to the server before closing the socket ?
I can fall back to have the client socket in AS400 as a Java or C procedure, but I really want to just keep it in a simple RPG program.
Would somebody help me understand how to do that please ?
Thanks !
In my opinion, that RPG tutorial you mention has a slight defect. What I believe is causing your confusion is the following section's code:
...
Consequently, we typically call the
send() API like this:
D miscdata S 25A
D rc S 10I 0
C eval miscdata = 'The data to send goes here'
C eval rc = send(s: %addr(miscdata): 25: 0)
c if rc < 25
C* for some reason we weren't able to send all 25 bytes!
C endif
...
If you read the documentation of send() you will see that the return value does not indicate an error if it is greater than -1 yet in the code above it seems as if an error has occurred. In fact, the sum of the return values must equal the size of the buffer assuming that you keep moving the pointer into the buffer to reflect what has been sent. Look here in Beej's Guide to Network Programming. You might also like to look at Richard Stevens' book UNIX Network Programming, Volume 1 for really detailed explanations.
As to the problem of determining if the last send before close() did the actual send ... well the paragraph above explains how to determine what portion of the data was sent. However, calling close() will attempt to send all unsent data unless SO_LINGER is set.
fnctl() is used to control blocking while setsockopt() is used to set SO_LINGER.
The abstraction of network communications being used is BSD sockets. There are some slight differences in implementations across OS's but it is generally quite homogeneous. This means that one can generally use documentation written for other OS's for the broad overview. Most of the time.