Will NetworkStream.Write block only until it places the data to be sent into the TCP send buffer, or will it block until the data is actually ACK'd by the receiving host?
Note: The socket is configured for blocking I/O.
Edit: Whoops, there's no such thing as TcpClient.Write of course! We all understood we were talking about TcpClient.GetStream().Write, which is actually NetworkStream.Write!
Unless .net is using something other than winsock, then according to the winsock reference:
The successful completion of a send function does not indicate that the data was successfully delivered and received to the recipient. This function only indicates the data was successfully sent.
If no buffer space is available within the transport system to hold the data to be transmitted, send will block unless the socket has been placed in nonblocking mode. On nonblocking stream oriented sockets, the number of bytes written can be between 1 and the requested length, depending on buffer availability on both the client and server computers.
Assuming that write is calling send underneath, then a strict interpretation of the winsock documentation would indicate that there is no gurantee that the data made it to the other end of the pipe when it returns.
Here is the link to the winsock docs I am quoting from:
http://msdn.microsoft.com/en-us/library/windows/desktop/ms741416(v=VS.85).aspx
I disagree with both answers [that state it blocks]. Writing to TCP/IP socket does not block unless the underlying buffer is already full of unacknowledge data. Generally, it doesn't block but just gets handed off to the TCP implementation. But of course now I have to go track down some references to back this up :)
From SO
TcpClient.Write will block until the packet buffer has been flushed to the network and the appropriate ACK(s) have been received. You'll notice that a dropped connection will usually end up throwing an exception on the Write operation, since it waits for the ACK but doesn't get one within the defined timeout period.
Related
I have several points regarding .NET socket implementation so I will state them sequentially:
My understanding is that an instance of a Socket has a buffer of a changeable size in its internal class implementation, and is actually a queue of bytes, and also is different than the application buffer that you declare and define in your application.
In synchronous mode using socket type:stream and protocol type:tcp , when using the method Receive (which is blocking the process), with a parameter application byte buffer is actually dequeuing the socket buffer in chunks that have the same size of the application byte buffer you declared and defined in your application then assigns this chunk to your application byte buffer you sent to Receive function.
If the above is true, then what happens when the byte buffer is larger in length than the byte elements in the socket queue?
Also, if 2 is correct then Send method of a socket sends the the data to the end point connected host Socket buffer and not application buffer.
Finally, since the Socket method Accept is non-blocking, a thread is created for it in the underlying implementation, and it has a queue of its own that is dequeued when Accept method is called.
I ask all of this to check if my understanding so far is right, or if it's mostly wrong and need correcting.
First of all .net's implementation is mostly just a managed wrapper around winsock.
My understanding is that an instance of a Socket has a buffer of a changeable size in its internal class implementation, and is actually a queue of bytes, and also is different than the application buffer that you declare and define in your application.
Ok so far.
In synchronous mode using socket type: ... when using the method Receive
When you call Receive, data will be copied into the supplied buffer and the number of bytes written will be returned. This may well be less than the size of the buffer. If your buffer is not large enough to accomodate all of the data queued by the TCP stack only as many bytes as can be copied into your buffer will be copied, the remaining bytes will be returned on your next call to Receive.
Sockets treat all data sent (or received) as being a continuous stream without breaks. However, data sent across the network is subject to networks or hosts splitting the data to meet constraints like a maximum packet size. Your code should assumes data may arrive in arbitrarily sized chunks. Incidentally, this kind of message is more likely to appear in a production environment than in a development/testing one.
socket sends the the data to the end point connected host Socket buffer and not application buffer
Send will return when the data is queued by the TCP stack. If the TCP window is full and the remote endpoint is not reading off the socket (say because it is waiting for its own send to complete) this could potentially be a long time.
Finally, since the Socket method Accept is non-blocking
Per the documentation, Accept will either block until a connection is received or (in non-blocking mode) either synchronously accept the first available connection or throw if no connection is available.
this is still relevant and would still be recommended reading for anyone about to start writing network code.
I use on a connected socket on my server something like this to send data to the client:
IAsyncResult asyncRes = connectionSocket.BeginSend(data, 0, length, SocketFlags.None,
out error, new AsyncCallback(SendDataDone), signalWhenDataSent);
As it seems, when there is a slow internet connection between the server and the client I receive an exception description like this: NoBufferSpaceAvailable
What exactly does this error mean ? The internal OS buffer for the socket connectionSocket is full ? What are the means to make it work. As a context where this appears is in a http proxy server. This might indicate, I suppose, that the rate at which data is coming from the origin server is higher than the rate my server can handle with the proxy client. How would you deal with it ?
I am using tcp.
The way to fix this problem is to correlate the way one reads from one socket to the speed one writes to the other socket because if you do no buffering you cannot write to a socket at a higher speed than the client connected at that end can read.
You one uses synchronous sockets the problem does not appear because they block as long as the operation is still pending but this is not the case with async calls.
Exactly, most likely the kernel socket buffer holding outgoing data is full. You're sending too "fast" for the client. You can try to increase the send buffer size, but that does not guarantee you won't bump into this problem again.
The simple answer is that you should be prepared that a send operation might fail, and retry it later. It's not ideal to maintain an ever growing buffer inside you application either, but the origin server should also slow down if you stop receiving (depending on the TCP window size and your receive buffer size).
I'm writing a server application for an iPhone application im designing. iPhone app is written in C# (MonoTouch) and the server is written in C# too (.NET 4.0)
I'm using asynchronous sockets for the network layer. The server allows two or more iPhones ("devices") to connect to each other and be able to send data bi-directionally.
Depending on the incoming message, the server either processes the message itself , or relays the data through to the other device(s) in the same group as the sending device. It can make this decision by decoding the header of the packet first, and deciding what type of packet it is.
This is done by framing the stream in a way that the first 8 bytes are two integers, the length of the header and the length of the payload (which can be much larger than the header).
The server reads (asynchronously) from the socket the first 8 bytes so it has the lengths of the two sections. It then reads again, up to the total length of the header section.
It then deserializes the header, and based on the information within, can see if the remaining data (payload) should be forwarded onto another device, or is something that the server itself needs to work with.
If it needs to be forwarded onto another device, then the next step is to read data coming into the socket in chunks of say, 1024 bytes, and write these directly using an async send via another socket that is connected to the recipient device.
This reduces the memory requirements of the server, as i'm not loading in the entire packet into a buffer, then re-sending it down the wire to the recipient.
However, because of the nature of async sockets, I am not guaranteed to receive the entire payload in one read, so have to keep reading until I receive all the bytes. In the case of relaying onto its final destination, this means that i'm calling BeginSend() for each chunk of bytes I receive from the sender, and forwarding that chunk onto the recipient, one chunk at a time.
The issue with this is that because I am using async sockets, this leaves the possibility of another thread doing a similar operation with the same recipient (and therefore same final destination socket), and so it is likely that the chunks coming from both threads will get mixed up and corrupt all the data going to that recipient.
For example: If the first thread sends a chunk, and is waiting for the next chunk from the sender (so it can relay it onwards), the second thread could send one of its chunks of data, and corrupt the first thread's (and the second thread's for that matter) data.
As I write this, i'm just wondering is it as simple as just locking the socket object?! Would this be the correct option, or could this cause other issues (e.g.: issues with receiving data through the locked socket that's being sent BACK from the remote device?)
Thanks in advance!
I was facing a similar scenario a while back, I don't have the complete solution anymore, but here's pretty much what I did :
I didn't use sync sockets, decided to explore the async sockets in C# - fun ride
I don't allow multiple threads to share a single resource unless I really have to
My "packets" were containing information about size, index and total packet count for a message
My packet's 1st byte was unique to signify that it's a start of a message, I used 0xAA
My packets's last 2 bytes were a result of a CRC-CCITT checksum (ushort)
The objects that did the receiving bit contained a buffer with all received bytes. From that buffer I was extracting "complete" messages once the size was ok, and the checksum matched
The only "locking" I needed to do was in the temp buffer so I could safely analyze it's contents between write/read operations
Hope that helps a bit
Not sure where the problem is. Since you mentioned servers, I assume TCP, yes?
A phone needs to communicate some of your PDU to another phone. It connects as a client to the server on the other phone. A socket-pair is established. It sends the data off to the server socket. The socket-pair is unique - no other streams that might be happening between the two phones should interrupt this, (will slow it up, of course).
I don't see how async/sync sockets, assuming implemented correctly, should affect this, either should work OK.
Is there something I cannot see here?
BTW, Maciek's plan to bolster up the protocol by adding an 'AA' start byte is an excellent idea - protocols depending on sending just a length as the first element always seem to screw up eventually and result in a node trying to dequeue more bytes that there are atoms in the universe.
Rgds,
Martin
OK, now I understand the problem, (I completely misunderstood the topology of the OP network - I thought each phone was running a TCP server as well as client/s, but there is just one server on PC/whatever a-la-chatrooms). I don't see why you could not lock the socket class with a mutex, so serializing the messages. You could queue the messages to the socket, but this has the memory implications that you are trying to avoid.
You could dedicate a connection to supplying only instructions to the phone, eg 'open another socket connection to me and return this GUID - a message will then be streamed on the socket'. This uses up a socket-pair just for control and halves the capacity of your server :(
Are you stuck with the protocol you have described, or can you break your messages up into chunks with some ID in each chunk? You could then multiplex the messages onto one socket pair.
Another alternative, that again would require chunking the messages, is introduce a 'control message', (maybee a chunk with 55 at start instead of AA), that contains a message ID, (GUID?), that the phone uses to establish a second socket connection to the server, passes up the ID and is then sent the second message on the new socket connection.
Another, (getting bored yet?), way of persuading the phone to recognise that a new message might be waiting would be to close the server socket that the phone is receiving a message over. The phone could then connect up again, tell the server that it only got xxxx bytes of message ID yyyy. The server could then reply with an instruction to open another socket for new message zzzz and then resume sending message yyyy. This might require some buffering on the server to ensure no data gets lost during the 'break'. You might want to implement this kind of 'restart streaming after break' functionality anyway since phones tend to go under bridges/tunnels just as the last KB of a 360MB video file is being streamed :( I know that TCP should take care of dropped packets, but if the phone wireless layer decides to close the socket for whatever reason...
None of these solutions is particularly satisfying. Interested to see whay other ideas crop up..
Rgds,
Martin
Thanks for the help everyone, i've realised the simpliest approach is to use synchronous send commands on the client, or at least a send command that must complete before the next item is sent. Im handling this with my own send queue on the client, rather than various parts of the app just calling send() when they need to send something.
My asking is quite simple and is about asynchronous sockets, working with TCP protocol.
When I send some data with the "BeginSend" method, when will the callback be called?
Will it be called when the data is just sent out to the network, or when we are ensured that the data as reached its destination (like it should be regarding to TCP specification) ?
Thanks for your answers.
KiTe.
ps : I'm sorry if my english is a bit bad ^^.
From MSDN:
"When your application calls BeginSend, the system will use a separate thread to execute the specified callback method, and will block on EndSend until the Socket sends the number of bytes requested or throws an exception."
"The successful completion of a send does not indicate that the data was successfully delivered. If no buffer space is available within the transport system to hold the data to be transmitted, send will block unless the socket has been placed in nonblocking mode."
http://msdn.microsoft.com/en-us/library/38dxf7kt.aspx
When the callback is called you can be sure that the data has been cleared from the output buffer (the asynchronous operation uses a separate thread to ensure that your calling thread is not blocked in case there is no room in the transmit buffer and it has to wait to send the date) and that it will reach it's destination - but not that it has reached it yet.
Because of the TCP protocol's nature however, you can be sure (well, I guess almost sure) that it will get to the destination, eventually.
However, for timing purposes you should not consider the time of the callback as being the same as the time the data reaches the other party.
Despite the documentation, NetworkStream.Write does not appear to wait until the data has been sent. Instead, it waits until the data has been copied to a buffer and then returns. That buffer is transmitted in the background.
This is the code I have at the moment. Whether I use ns.Write or ns.BeginWrite doesn't matter - both return immediately. The EndWrite also returns immediately (which makes sense since it is writing to the send buffer, not writing to the network).
bool done;
void SendData(TcpClient tcp, byte[] data)
{
NetworkStream ns = tcp.GetStream();
done = false;
ns.BeginWrite(bytWriteBuffer, 0, data.Length, myWriteCallBack, ns);
while (done == false) Thread.Sleep(10);
}
public void myWriteCallBack(IAsyncResult ar)
{
NetworkStream ns = (NetworkStream)ar.AsyncState;
ns.EndWrite(ar);
done = true;
}
How can I tell when the data has actually been sent to the client?
I want to wait for 10 seconds(for example) for a response from the server after sending my data otherwise I'll assume something was wrong. If it takes 15 seconds to send my data, then it will always timeout since I can only start counting from when NetworkStream.Write returns - which is before the data has been sent. I want to start counting 10 seconds from when the data has left my network card.
The amount of data and the time to send it could vary - it could take 1 second to send it, it could take 10 seconds to send it, it could take a minute to send it. The server does send an response when it has received the data (it's a smtp server), but I don't want to wait forever if my data was malformed and the response will never come, which is why I need to know if I'm waiting for the data to be sent, or if I'm waiting for the server to respond.
I might want to show the status to the user - I'd like to show "sending data to server", and "waiting for response from server" - how could I do that?
I'm not a C# programmer, but the way you've asked this question is slightly misleading. The only way to know when your data has been "received", for any useful definition of "received", is to have a specific acknowledgment message in your protocol which indicates the data has been fully processed.
The data does not "leave" your network card, exactly. The best way to think of your program's relationship to the network is:
your program -> lots of confusing stuff -> the peer program
A list of things that might be in the "lots of confusing stuff":
the CLR
the operating system kernel
a virtualized network interface
a switch
a software firewall
a hardware firewall
a router performing network address translation
a router on the peer's end performing network address translation
So, if you are on a virtual machine, which is hosted under a different operating system, that has a software firewall which is controlling the virtual machine's network behavior - when has the data "really" left your network card? Even in the best case scenario, many of these components may drop a packet, which your network card will need to re-transmit. Has it "left" your network card when the first (unsuccessful) attempt has been made? Most networking APIs would say no, it hasn't been "sent" until the other end has sent a TCP acknowledgement.
That said, the documentation for NetworkStream.Write seems to indicate that it will not return until it has at least initiated the 'send' operation:
The Write method blocks until the requested number of bytes is sent or a SocketException is thrown.
Of course, "is sent" is somewhat vague for the reasons I gave above. There's also the possibility that the data will be "really" sent by your program and received by the peer program, but the peer will crash or otherwise not actually process the data. So you should do a Write followed by a Read of a message that will only be emitted by your peer when it has actually processed the message.
TCP is a "reliable" protocol, which means the data will be received at the other end if there are no socket errors. I have seen numerous efforts at second-guessing TCP with a higher level application confirmation, but IMHO this is usually a waste of time and bandwidth.
Typically the problem you describe is handled through normal client/server design, which in its simplest form goes like this...
The client sends a request to the server and does a blocking read on the socket waiting for some kind of response. If there is a problem with the TCP connection then that read will abort. The client should also use a timeout to detect any non-network related issue with the server. If the request fails or times out then the client can retry, report an error, etc.
Once the server has processed the request and sent the response it usually no longer cares what happens - even if the socket goes away during the transaction - because it is up to the client to initiate any further interaction. Personally, I find it very comforting to be the server. :-)
In general, I would recommend sending an acknowledgment from the client anyway. That way you can be 100% sure the data was received, and received correctly.
If I had to guess, the NetworkStream considers the data to have been sent once it hands the buffer off to the Windows Socket. So, I'm not sure there's a way to accomplish what you want via TcpClient.
I can not think of a scenario where NetworkStream.Write wouldn't send the data to the server as soon as possible. Barring massive network congestion or disconnection, it should end up on the other end within a reasonable time. Is it possible that you have a protocol issue? For instance, with HTTP the request headers must end with a blank line, and the server will not send any response until one occurs -- does the protocol in use have a similar end-of-message characteristic?
Here's some cleaner code than your original version, removing the delegate, field, and Thread.Sleep. It preforms the exact same way functionally.
void SendData(TcpClient tcp, byte[] data) {
NetworkStream ns = tcp.GetStream();
// BUG?: should bytWriteBuffer == data?
IAsyncResult r = ns.BeginWrite(bytWriteBuffer, 0, data.Length, null, null);
r.AsyncWaitHandle.WaitOne();
ns.EndWrite(r);
}
Looks like the question was modified while I wrote the above. The .WaitOne() may help your timeout issue. It can be passed a timeout parameter. This is a lazy wait -- the thread will not be scheduled again until the result is finished, or the timeout expires.
I try to understand the intent of .NET NetworkStream designers, and they must design it this way. After Write, the data to send are no longer handled by .NET. Therefore, it is reasonable that Write returns immediately (and the data will be sent out from NIC some time soon).
So in your application design, you should follow this pattern other than trying to make it working your way. For example, use a longer time out before received any data from the NetworkStream can compensate the time consumed before your command leaving the NIC.
In all, it is bad practice to hard code a timeout value inside source files. If the timeout value is configurable at runtime, everything should work fine.
How about using the Flush() method.
ns.Flush()
That should ensure the data is written before continuing.
Bellow .net is windows sockets which use TCP.
TCP uses ACK packets to notify the sender the data has been transferred successfully.
So the sender machine knows when data has been transferred but there is no way (that I am aware of) to get that information in .net.
edit:
Just an idea, never tried:
Write() blocks only if sockets buffer is full. So if we lower that buffers size (SendBufferSize) to a very low value (8? 1? 0?) we may get what we want :)
Perhaps try setting
tcp.NoDelay = true