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How do I write a small gap in networkstream?

I have a server that sends telemetry data of varying size to receivers, or as I'll call them, clients, via NetworkStream. I aim for maximum speed, so I want minimal delays. The frequency is uncontrolled, so I'd like to use an infinite loop in my clients, that uses NetworkStream.Read to read the data, process it, then repeat.
My problem is that sometimes, if two packets are sent very quickly, and a client has a slow internet connection, the two packets will be received as a continous stream, resulting unprocessable data. A half-solution I found (mostly to confirm that this is indeed the error) is to have a small delay after/before each transmission, using System.Threading.Thread.Sleep(100), but not only I find Sleep a botchy solution, it's inconsistent, as it also slows down clients with a good connection, and the problem may persist with an even worse connection.
What I'd like to do is to have the server send a gap between each transmission, providing a separation regardless of internet speed, as NetworkStream.Read should finish after the current continous stream ends. I don't understand deeply the working of a NetworkStream, and have no idea what a few bytes of empty stream look like or how it could be implemented. Any ideas?

I would strong advise changing the protocol instead if you possibly can. TCP is a stream-based protocol, and any attempt to effectively ignore that is likely to be unreliable.
Instead, I'd suggest changing to make it a stream of messages where each message has a prefix indicating the length of the body of the message. That way it doesn't matter if a particular message is split across multiple packets or received in the same packet as other messages. It also makes reading easier for clients: they know exactly how much data to read, so can just do that in a simple loop.
If you're concerned that the length prefix will introduce too much overhead (if the data is often small) you could potentially make the protocol slightly more complicated with a single message containing a whole batch of information (multiple telemetry items).
But fundamentally, it's worth assuming that the data will be split into multiple packets, combined again etc. Don't assume that one write operation corresponds to one read operation.

You don't specifically mention the ProtocolType you're using on your NetworkStream but TCP is bound to fail your requirements. Intermediate routers/switches have no way to know your intent is to separate packets by time and will not respect that desire. Furthermore, TCP, being stream oriented, delivers packets in order, and it has error correction against dropped packets and corrupt packets. On any occurrence of one or the other it will hold all further packets until the error packet is retransmitted - and then you'll probably get them all in a bunch.
Use UDP and implement throttling on the receiving (i.e., client) side - throwing out data if you're falling behind.

Socket.Send has a delay?

I am building an application and sending data back to the other side,
In my application, I have:
System.Net.Sockets.Socket.Send(byte[]) function.
My customer told me that there is 530 ms delay of receiving this packet. However, I have logged every where until: System.Net.Sockets.Socket.Send(byte[])
I measured that it took about 15ms to get to send an array from Socket. My customer advised me to check:
To flush after sending, but I dont see a flush function in Socket?
Fill up the data buffer before sending out otherwise, if the data is short, then I have to force the transmission
Is one of this advice correct? I see there are also another parameter of the method Send, which is: SocketFlags <= Is there any help of using this SocketFlags?

There's common problem with Nagle algorithm that tries to 'glue' data sent together into single packet. It is possible that your code suffers from it as well.
Try to disable it as shown here or by setting SocketOptionName.NoDelay option with SetSocketOption method.

Minimum guaranteed bytes sent in one go via NetworkStream

I am using the NetworkStream of a TcpClient to send bytes from a .NET server to a Silverlight Client that receives them via a Socket. Sending from the client is done via Socket.SendAsync.
My question is, what is the minimum amount of bytes I can expect to be received "in one go" on both sides, without the need to send the message length too and put several byte messages together by myself.
Thanks,
Andrej

You absolutely should send the message length. Network streams are precisely that - streams of information. Basically there are three ways of recognising the end of a message:
Prefix the data with the length (don't forget that it's at least theoretically possible that you won't even get all of the length data in one packet)
Use a message delimiter/terminator
Only send one message each way per connection, and close the connection afterwards

It depends on your network settings, but the default length for network nodes is 1500 bytes and most modems take something off that. So most homes have 1460 bytes packet size.
The optimum size for your situation can be calculated
But people can always have their own settings, so there's no guarantee that you get the optimal packet size for all clients.

How to send large data using C# UdpClient?

I'm trying to send a large amount of data (more than 50 MB) using C# UdpClient.
So at first I split the data into 65507 byte blocks and send them in a loop.
for(int i = 0; i < packetCount; i++)
myUdpClient.Send(blocks[i], block[i].Length, remoteEndPoint);
My problem is that only the first packets can be received.
During sending the first packet the networkload increases rapidly to 100%, and then the other packets cannot be received.
I want to get as much data throughput as possible.
I'm sorry for my English!
Thanks for your help in advance.

For all those people saying to use TCP....are foolishly wrong. Although TCP is reliable and the window maintained by the kernel it's fairly "set and forget" protocol, but when it comes to the guy wanting to use 100% of his throughput, TCP will not do (it throttles too hard, and the wait for an ACK automatically puts at least 50% trashed because of the RTT).
To the original question, you are sending UDP packets nonstop in that for-loop, the window fills up and then any new data is dropped immediately and doesn't even attempt to go on the line. You are also splitting your data too large. I would recommend building your own throttle mechanism that starts off with 2k segments per second, and slowly ramps up. Each "segment" contains a SEQ (sequence identifier for acknowledgements or ACK) and OFF (offset inside the file for this data set). As the data is being tagged, let the server keep track of these tags. When the other side gets them, it stores the SEQ numbers in an ACK list, and any missing SEQ numbers are placed into a NACK timer list, when the timer runs out (if they haven't been received) it moves to a NACK list. The receiver should send 5 or so ACKs from the ACK list along with up to 5 NACKs in a single transmission every couple seconds or so. If the sender receives these messages and there are any NACKs, it should immediately throttle down and resend the missing fragment before continuing. The data that is ACKed can be freed from memory.
Good luck!

I don't know about .Net implementation specifically, it might be buffering your data, but UDP datagram is normally limited by the link MTU, which is 1500 on normal ethernet (subtract 20 bytes for IP header and 8 bytes of UDP header.)
UDP is explicitly allowed to drop and reorder the datagrams, and there's no flow control as in TCP.
Exceeding the socket send buffer on the sender side will have the network stack ignore following send attempts until the buffer space is available again (you need to check the return value of the send() for that.)
Edit:
I would strongly recommend going with TCP for large file transfers. TCP gives you sequencing (you don't have to keep track of dropped and re-ordered packets.) It has advanced flow control (so fast sender does not overwhelm a slow receiver.) It also does Path MTU discovery (i.e. finds out optimal data packetization and avoids IP fragmentation.) Otherwise you would have to re-implement most of these features yourself.

I hate to say it but you need to sleep the thread. You are overloading your throughput. UDP is not very good for lossless data transfer. UDP is for when you don't mind dropping some packets.

Reliably - no, you won't do it with UDP.
As far as I understand, this makes sense for sending to multiple computers at a time (broadcasting).
In this case,
establish a TCP connection with each of them,
split the data into blocks,
give each block an ID,
send list of IDs to each computer with TCP connection,
broadcast data with UDP,
inform clients (via TCP) that data transmission is over,
than clients should ask to resend the dropped packets

C# socket abnormal latency

I am working on a little online multiplayer pong game with C# and XNA.
I use sockets to transfer data between two computers on my personnal LAN. It works fine.
The issue is with speed : the transfer is slow.
When I ping the second computer, it shows a latency of 2 ms.
I set up a little timer inside my code, and it shows a latency of about 200 ms.
Even when the server and the client are on the same computer (using 127.0.0.1), the latency still about 15 ms. I consider this as slow.
Here are some fragment of my code :
server = new Socket(AddressFamily.InterNetwork, SocketType.Stream, ProtocolType.Tcp);
server.Bind(new IPEndPoint(IPAddress.Any, port));
server.Listen(1);
// Begin Accept
server.BeginAccept(new AsyncCallback(ClientAccepted), null);
in ClientAccepted, I set up a NetworkStream, a StreamReader and a StreamWriter.
This is how I send a message when I want to update the player's location :
string message = "P" + "\n" + player.Position + "\n";
byte[] data = Encoding.ASCII.GetBytes(message);
ns.BeginWrite(data, 0, data.Length, new AsyncCallback(EndUpdate), null);
the only thing EndUpdate do is to call EndWrite.
This is how I receive data :
message = sr.ReadLine();
It dosen't block my game since it's on a second thread.
These are the stuff I tried :
- Use IP instead of TCP
- Use binary message instead of text
- Use IPv6 instead of IPv4
Nothing really did help.
Any thoughts about how I can improve the latency?
Thank you

The latency you're seeing is most likely due to Nagle's algorithm which is used to improve efficiency when sending lots of small packets using TCP. Essentially, the TCP stack collects small packets together and coalesces them into a larger packet before transmitting. This obviously means delaying for a small interval - up to 200ms - to see if any more data is sent, before sending what's waiting in the output buffers.
So try switching off Nagle by setting the NoDelay property to true on the socket.
However, be aware that if you do a LOT of small writes to the socket, that you may lose performance because of the TCP header overhead and processing of each packet. In that case, you'd want to try to collect your writes together into batches if possible.

There are several other meta-issues to consider:
Timer accuracy: Many system timers only update every 15 ms or so. I can't remember the precise value, but if your timings are always multiples of about 15 ms, be suspicious that your timer is not high precision. This is 'probably' not your issue, but keep it in mind.
If you're testing on the same computer, and only running a single core machine, the thread/process switching frequency will dominate your ping times. Not much to do but try to add Sleep(0) calls to allow thread/process swaps after sending data.
TCP/IP transmission settings. As alluded earlier in SocketOptionName.NoDelay. Also as mentioned earlier, consider UDP if you're continuously updating state. Order isn't guaranteed, but for volatile data missing packets are acceptible, as the state will be overridden soon anyway. To avoid using stale packets, add an incrementing value to the packet and never use a packet labelled earlier than the current state.
The difference between tcp and udp should not account for 200 ms, but a combination of other factors can.
Polling vs. Select. If you're polling for received data, the frequency of polling will interfere with receive rate, naturally. I use Socket.Select in a dedicated network thread to wait for incoming data and handle it as soon as possible.

Most networked games don't use TCP but UDP for communications. The problem with that is that data can be easily dropped, which is something you have to account for.
With TCP, there are a number of interactions between the host/server to guarantee data in an ordered manner (another thing you have to account for using UDP, the fact that data is not ordered).
On top of that, the timer in your code has to wait until the bytes bubble up from the unmanaged socket layer through unmanaged code, etc, etc. There is going to be some overhead there that you aren't going to be able to overcome (the unmanaged to managed transition).

You said you tried "IP", and you presumably did that by specifying ProtocolType.Ip, but I don't know what that really means (i.e. I don't know what protocol is chosen 'under the hood') when you combine that with SocketType.Stream.
As well as what casperOne said, if you're using TCP then see whether setting SocketOptionName.NoDelay makes any difference to you.