I have a bitmap, I converted it to bytes using BinaryFormatter and MemoryStream then sent the bytes to the TCP server, when I try to convert it back to BitMapI get this error System.Runtime.Serialization.SerializationException: End of Stream encountered before parsing was completed. I tried converting the bitmap to bytes then converting the bytes to bitmap on the client side just to check if the error is due to the conversion but everything worked just fine. So I think the problem is that the server is receiving the byte of array in chunks not in 1 big array, so my question is how check if byte array is fully sent?
As you say the data can absolutely be sent in multiple chunks and you need to have a way of knowing when all the data is received. For HTTP you use Content-Length in the headers to let the client know when all data is received. Since you control both sides you can transform your image into a byte array, check the size and lets say its 5000 bytes.
Then you create an int (or long if necessary, not in this case probably) and set it to 5000 and send that first (as bytes) and then the rest of the data. You will then have created your own header. On the other side you start by reading the exact amount of bytes for an int (or other if you chose long etc). Then transform the bytes into an int and you know it will now come 5000 bytes. Then start reading until you have 5000 bytes. This can always be optimized but this is a simple way you can do it.
Related
If i send 1000 bytes in TCP, does it guarantee that the receiver will get the entire 1000 bytes "togther"? or perhaps he will first only get 500 bytes, and later he'll receive the other bytes?
EDIT: the question comes from the application's point of view. If the 1000 bytes are reassembles into a single buffer before they reach the application .. then i don't care if it was fragmented in the way..
See Transmission Control Protocol:
TCP provides reliable, ordered delivery of a stream of bytes from a program on one computer to another program on another computer.
A "stream" means that there is no message boundary from the receiver's point of view. You could get one 1000 byte message or one thousand 1 byte messages depending on what's underneath and how often you call read/select.
Edit: Let me clarify from the application's point of view. No, TCP will not guarantee that the single read would give you all of the 1000 bytes (or 1MB or 1GB) packet the sender may have sent. Thus, a protocol above the TCP usually contains fixed length header with the total content length in it. For example you could always send 1 byte that indicates the total length of the content in bytes, which would support up to 255 bytes.
As other answers indicated, TCP is a stream protocol -- every byte sent will be received (once and in the same order), but there are no intrinsic "message boundaries" -- whether all bytes are sent in a single .send call, or multiple ones, they might still be received in one or multiple .receive calls.
So, if you need "message boundaries", you need to impose them on top of the TCP stream, IOW, essentially, at application level. For example, if you know the bytes you're sending will never contain a \0, null-terminated strings work fine; various methods of "escaping" let you send strings of bytes which obey no such limitations. (There are existing protocols for this but none is really widespread or widely accepted).
Basically as far as TCP goes it only guarantees that the data sent from one end to the other end will be sent in the same order.
Now usually what you'll have to do is have an internal buffer that keeps looping until it has received your 1000 byte "packet".
Because the recv command as mentioned returns how much has actually been received.
So usually you'll have to then implement a protocol on top of TCP to make sure you send data at an appropriate speed. Because if you send() all the data in one run through it will overload the under lying networking stack, and which will cause complications.
So usually in the protocol there is a tiny acknowledgement packet sent back to confirm that the packet of 1000 bytes are sent.
You decide, in your message that how many bytes your message shall contain. For instance in your case its 1000. Following is up and running C# code to achieve the same. The method returns with 1000 bytes. The abort code is 0 bytes; you can tailor that according to your needs.
Usage:
strMsg = ReadData(thisTcpClient.Client, 1000, out bDisconnected);
Following is the method:
string ReadData(Socket sckClient, int nBytesToRead, out bool bShouldDisconnect)
{
bShouldDisconnect = false;
byte[] byteBuffer = new byte[nBytesToRead];
Array.Clear(byteBuffer, 0, byteBuffer.Length);
int nDataRead = 0;
int nStartIndex = 0;
while (nDataRead < nBytesToRead)
{
int nBytesRead = sckClient.Receive(byteBuffer, nStartIndex, nBytesToRead - nStartIndex, SocketFlags.None);
if (0 == nBytesRead)
{
bShouldDisconnect = true;
//0 bytes received; assuming disconnect signal
break;
}
nDataRead += nBytesRead;
nStartIndex += nBytesRead;
}
return Encoding.Default.GetString(byteBuffer, 0, nDataRead);
}
Let us know this didn't help you (0: Good luck.
Yes, there is a chance for receiving packets part by part. Hope this msdn article and following example (taken from the article in msdn for quick review) would be helpful to you if you are using windows sockets.
void CChatSocket::OnReceive(int nErrorCode)
{
CSocket::OnReceive(nErrorCode);
DWORD dwReceived;
if (IOCtl(FIONREAD, &dwReceived))
{
if (dwReceived >= dwExpected) // Process only if you have enough data
m_pDoc->ProcessPendingRead();
}
else
{
// Error handling here
}
}
TCP guarantees that they will recieve all 1000 bytes, but not necessarily in order (though, it will appear so to the recieving application) and not necessarily all at once (unless you craft the packet yourself and make it so.).
That said, for a packet as small as 1000 bytes, there is a good chance it'll send in one packet as long as you do it in one call to send, though for larger transmissions it may not.
The only thing that the TCP layer guarantees is that the receiver will receive:
all the bytes transmitted by the sender
in the same order
There are no guarantees at all about how the bytes might be split up into "packets". All the stuff you might read about MTU, packet fragmentation, maximum segment size, or whatever else is all below the layer of TCP sockets, and is irrelevant. TCP provides a stream service only.
With reference to your question, this means that the receiver may receive the first 500 bytes, then the next 500 bytes later. Or, the receiver might receive the data one byte at a time, if that's what it asks for. This is the reason that the recv() function takes a parameter that tells it how much data to return, instead of it telling you how big a packet is.
The transmission control protocol guarantees successful delivery of all packets by requiring acknowledgment of the successful delivery of each packet to the sender by the receiver. By this definition the receiver will always receive the payload in chunks when the size of the payload exceeds the MTU (maximum transmission unit).
For more information please read Transmission Control Protocol.
The IP packets may get fragmented during retransmission.
So the destination machine may receive multiple packets - which will be reassembled back by TCP/IP stack. Depending on the network API you are using - the data will be given to you either reassembled or in RAW packets.
It depends of the stablished MTU (Maximum transfer unit). If your stablished connection (once handshaked) refers to a MTU of 512 bytes you will need two or more TCP packets to send 1000 bytes.
I have a one-on-one connection between a server and a client. The server is streaming real-time audio/video data.
My question may sound weird, but should I use multiple ports/socket or only one? Is it faster to use multiple ports or a single one offer better performance? Should I have a port only for messages, one for video and one for audio or is it more simple to package the whole thing in a single port?
One of my current problem is that I need to first send the size of the current frame as the size - in bytes - may change from one frame to the next. I'm fairly new to Networking, but I haven't found any mechanism that would automatically detect the correct range for a specific object being transmitted. For example, if I send a 2934 bytes long packet, do I really need to tell the receiver the size of that packet?
I first tried to package the frame as fast as they were coming in, but I found out the receiving end would sometime not get the appropriated number of bytes. Most of the time, it would read faster than I send them, getting only a partial frame. What's the best way to get only the appropriated number of bytes as quickly as possible?
Or am I looking too low and there's a higher-level class/framework used to handle object transmission?
I think it is better to use an object mechanism and send data in an interleaved fashion. This mechanism may work faster than multiple port mechanism.
eg:
class Data {
DataType, - (Adio/Video)
Size, - (Size of the Data buffer)
Data Buffer - (Data depends on the type)
}
'DataType' and 'Size' always of constant size. At the client side take the 'DataType' and 'Size' and then read the specifed size of corresponding sent data(Adio/Video).
Just making something up off the top of my head. Shove "packets" like this down the wire:
1 byte - packet type (audio or video)
2 bytes - data length
(whatever else you need)
|
| (raw data)
|
So whenever you get one of these packets on the other end, you know exactly how much data to read, and where the beginning of the next packet should start.
[430 byte audio L packet]
[430 byte audio R packet]
[1000 byte video packet]
[20 byte control packet]
[2000 byte video packet]
...
But why re-invent the wheel? There are protocols to do these things already.
Beginner Question again: Kind of a follow up to a question I asked not long ago.
I am trying to understand this synchronous socket tutorial http://msdn.microsoft.com/en-us/library/6y0e13d3.aspx, particularly one single line in the code below.
QUESTION:
I want to make sure I am understanding the program flow right . When does handler.Receive(bytes) return? Does it return and store the number of bytes received in int bytesRec**when it "overflows" and has received more than 1024bytes? **And if that is so, and this might sound silly, what happens if MORE bytes arrive as it is storing the 1024 bytes in the *data*variable and not listening for more bytes that might be arriving at that time? Or should I not worry about it and let .net take care of that?
Socket handler = listener.Accept();
data = null;
// An incoming connection needs to be processed.
while (true) {
bytes = new byte[1024];
int bytesRec = handler.Receive(bytes);
// My question is WHEN does the following line
// get to be executed
data += Encoding.ASCII.GetString(bytes,0,bytesRec);
if (data.IndexOf("<EOF>") > -1) {
break;
}
}
When does handler.Receive(bytes) return?
Documentation:
If no data is available for reading, the Receive method will block
until data is available, unless a time-out value was set by using
Socket.ReceiveTimeout. If the time-out value was exceeded, the Receive
call will throw a SocketException. If you are in non-blocking mode,
and there is no data available in the in the protocol stack buffer,
the Receive method will complete immediately and throw a
SocketException. You can use the Available property to determine if
data is available for reading. When Available is non-zero, retry the
receive operation.
Does it return and store the number of bytes received in int
bytesRec when it "overflows" and has received more than 1024 bytes?
No, it always returns the number of bytes that have been read. If it didn't, how could you know what part of bytes contains meaningful data and what part of it has remained unused?
It is very important to understand how sockets typically work: bytes may be arriving in packets, but as far as the receiver is concerned each byte should be considered independently. This means that there is no guarantee that you will get the bytes in the chunks the sender sent them, and of course there is no guarantee that there is enough data to fill up your buffer.
If you only want to process incoming data in 1024-byte chunks, it is your own responsibility to keep calling Receive until it has released 1024 bytes in total to you.
And if that is so, and this might sound silly, what happens if MORE
bytes arrive as it is storing the 1024 bytes in the variable and
not listening for more bytes that might be arriving at that time?
Let's reiterate that Receive will not store 1024 bytes in the buffer because that's the buffer's size. It will store up to 1024 bytes.
If there is more data buffered internally by the network stack than your buffer can hold then 1024 bytes will be given back to you and the rest will stay in the network stack's buffers until you call Receive again. If Receive has begun copying data to your buffer and at that moment more data is received from the network then most likely what's going to happen is that these will have to wait for the next Receive call.
After all, at no point did anyone provide a guarantee that Receive would give you all the data it can (although of course that is desirable and it is what happens most of the time).
I just read an article that says TCPClient.Read() may not get all the sent bytes in one read. How do you account for this?
For example, the server can write a string to the tcp stream. The client reads half of the string's bytes, and then reads the other half in another read call.
how do you know when you need to combine the byte arrays received in both calls?
how do you know when you need to combine the byte arrays received in both calls?
You need to decide this at the protocol level. There are four common models:
Close-on-finish: each side can only send a single "message" per connection. After sending the message, they close the sending side of the socket. The receiving side keeps reading until it reaches the end of the stream.
Length-prefixing: Before each message, include the number of bytes in the message. This could be in a fixed-length format (e.g. always 4 bytes) or some compressed format (e.g. 7 bits of size data per byte, top bit set for the final byte of size data). Then there's the message itself. The receiving code will read the size, then read that many bytes.
Chunking: Like length-prefixing, but in smaller chunks. Each chunk is length-prefixed, with a final chunk indicating "end of message"
End-of-message signal: Keep reading until you see the terminator for the message. This can be a pain if the message has to be able to include arbitrary data, as you'd need to include an escaping mechanism in order to represent the terminator data within the message.
Additionally, less commonly, there are protocols where each message is always a particular size - in which case you just need to keep going until you've read that much data.
In all of these cases, you basically need to loop, reading data into some sort of buffer until you've got enough of it, however you determine that. You should always use the return value of Read to note how many bytes you actually read, and always check whether it's 0, in which case you've reached the end of the stream.
Also note that this doesn't just affect network streams - for anything other than a local MemoryStream (which will always read as much data as you ask for in one go, if it's in the stream at all), you should assume that data may only become available over the course of multiple calls.
You should call read() in a loop. The condition of that loop would check if there is still any data available to be read.
That is kinda hard to answer, because you can never know when data will arrive, and thats why I usually use a thread for receiving data in my chat program. But you should be able to use something similar to this:
do{
numberOfBytesRead = myNetworkStream.Read(myReadBuffer,
0,
myReadBuffer.Length);
myCompleteMessage.AppendFormat("{0}",
Encoding.ASCII.GetString(myReadBuffer, 0, numberOfBytesRead));
}
while(myNetworkStream.DataAvailable);
Look at this source!
I'm one of those guys who come here to find answers to those questions that others have asked, and I think i newer asked anything myself, but after two days searching unsuccessfully I decided that it's time to ask something myself. So here it is...
I have a TCP server and client written in C#, .NET 4, asynchronous sockets using SocketAsyncEventArgs. I have a length-prefixed message framing protocol. Overall everything works just fine, but one issue keeps bugging me.
Situation is like this (I will use small numbers just as an example):
Lets say Server has a Send buffer length of 16 bytes.
It sends a message which is 6 bytes long, and prefixes it with 4 bytes long length prefix. Total message length is 6+4=10.
Client reads the data and receives a buffer of 16 bytes length (yes 10 bytes of data and 6 bytes equal to zero).
Received buffer looks like this: 6 0 0 0 56 21 33 1 5 7 0 0 0 0 0 0
So I read first 4 bytes which is my length prefix, I determine that my message is 6 bytes long, I read it as well and everything is fine so far. Then i have 16-10=6 bytes left to read. All of them are zeroes I read 4 of them, since it's my length prefix. So it's a zero length message which is allowed as keep-alive packet.
Remaining data to read: 0 0
Now the issue "kicks in". I got only 2 remaining bytes to read, they are not enough to complete a 4 byte-long length prefix buffer. So I read those 2 bytes, and wait for more incoming data. Now server is not aware that I'm still reading length prefix (I'm just reading all those zeroes in the buffer) and sends another message correctly prefixed with 4 bytes. And the client is assuming the server sends those missing 2 bytes. I receive the data on the client side, and read first two bytes to form a complete 4 byte length buffer. The results are something like that
lengthBuffer = new byte[4]{0, 0, 42, 0}
Which then translates into 2752512 message length. So my code will continue to read next 2752512 bytes to complete the message...
So in every single message framing example I have seen zero length messages are supported as keep-alive's. And every example I've seen doesn't do anything more than I do. The problem is that I do not know how much data I have to read when I receive it from the server. Since I have partially-filled buffer with zeroes, I have to read it all as those zeroes could be keep-alive's I sent from the other end of connection.
I could drop zero-length messages and stop reading the buffer after first empty message and it should fix this issue, and use custom messages for my keep-alive mechanism. But I want to know if I am missing something, or doing something wrong, since every code example I've seen seems to have same issue (?)
UPDATE
Marc Gravell, you sir pulled words out of my mouth. Was about to update that the issue is with sending the data. The problem is that initially when exploring .NET Sockets and SocketAsyncEventArgs I came across this sample: http://archive.msdn.microsoft.com/nclsamples/Wiki/View.aspx?title=socket%20performance
It uses reusable pool of buffers. Simply takes predefined number of maximum client connections allowed, for example 10, takes maximum single buffer size, for example 512, and creates one large buffer for all of them. So 512 * 10 * 2 (for send and receive) = 10240
So we have byte[] buff = new byte[10240];
Then for each client that connects it assigns a piece of this large buffer. First connected client gets first 512 bytes for Data Reading operations, and gets next 512 bytes (offset 512) for Data Sending operations. Therefore the code ended up having already allocated Send buffer which size is 512 (exactly the number the client later receives as BytesTransferred). This buffer is populated with data, and all remaining space out of these 512 bytes is sent as zeroes.
Strange enough this example is from msdn. The reason there is a single huge buffer is to avoid fragmented heap memory, when buffer gets pinned and GC cant collect it or something like that.
Comment from BufferManager.cs in the provided example (see link above):
This class creates a single large buffer which can be divided up and
assigned to SocketAsyncEventArgs objects for use with each socket I/O
operation. This enables bufffers to be easily reused and gaurds
against fragmenting heap memory.
So the issue is pretty much clear. Any suggestions on how I should resolve this are welcome :) Is it true what they say about fragmented heap memory, is it OK to create a data buffer "on the fly"? If so, will I have memory issues when the server scales to a few hundred or even thousands of clients?
I guess the problem is that you are treating the trailing zeros in the buffer you read as data. This is not data. It is garbage. No one ever sent it to you.
The Stream.Read call returns you the number of bytes actually read. You should not interpret the rest of the buffer in any way.
The problem is that I do not know how much data I have to read when I
receive it from the server.
Yes, you do: Use the return value from Stream.Read.
That sounds simply like a bug in either your send or receive code. You should only get BytesTransferred as the data that was actually sent, or some number smaller than that if arriving in fragments. The first thing I would wonder is: did you setup the send correctly? i.e. if you have an oversized buffer, a correct implementation might look like:
args.SetBuffer(buffer, 0, actualBytesToSend);
if (!socket.SendAsync(args)) { /* whatever */ }
where actualBytesToSend can be much less than buffer.Length. My initial suspicion is that
you are doing something like:
args.SetBuffer(buffer, 0, buffer.Length);
and therefore sending more data than you have actually populated.
I should emphasize: there is something wrong in either your send or receive; I do not believe, at least without an example, that there is some fundamental underlying bug in the BCL here - I use the async API extensively, and it works fine - but you do need to accurately track the data you are sending and receiving at all points.
"Now server is not aware that I'm still reading length prefix (I'm just reading all those zeroes in the buffer) and sends another message correctly prefixed with 4 bytes.".
Why? How does the server know what you are and aren't reading? If the server retransmits any part of a message it is in error. TCP already does that for you.
There seems to be something radically wrong with your server.