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.
Related
Here I am troubleshooting a theoretical problem about HOW servers and clients are working on machines. I know all NET Processes, but I am missing something referring to code. I was unable to find something related about this.
I code in Visual C# 2008, i use regular TCPClient / TCPListener with 2 different projects:
Project1 (Client)
Project2 (Server)
My issues are maybe so simple:
1-> About how server receives data, event handlers are possible?
In my first server codes i used to make this loop:
while (true)
{
if (NetworkStream.DataAvailable)
{
//stuff
}
Thread.Sleep(200);
}
I encounter this as a crap way to control the incoming data from a server. BUT server is always ready to receive data.
My question: There is anything like...? ->
AcceptTcpClient();
I want a handler that waits until something happen, in this case a specific socket data receiving.
2-> General networking I/O methods.
The problem is (beside I'm a noob) is how to handle multiple data writing.
If I use to send a lot of data in a byte array, the sending can break if I send more data. All data got joined and errors occurs when receiving. I want to handle multiple writes to send and receive.
Is this possible?
About how server receives data, event handlers are possible?
If you want to write call-back oriented server code, you may find MSDN's Asynchronous Server Socket Example exactly what you're looking for.
... the sending can break if I send more data. All data got joined and errors occurs when receiving.
That is the nature of TCP. The standardized Internet protocols fall into a few categories:
block oriented stream oriented
reliable SCTP TCP
unreliable UDP ---
If you really want to send blocks of data, you can use SCTP, but be aware that many firewalls DROP SCTP packets because they aren't "usual". I don't know if you can reliably route SCTP packets across the open Internet.
You can wrap your own content into blocks of data with your own headers or add other "synchronization" mechanisms to your system. Consider an HTTP server: it must wait until it reads an entire request like:
GET /index.html HTTP/1.1␍␊
Host: www.example.com␍␊
␍␊
Until the server sees the CRLFCRLF sequence, it must keep the partially-read data in a buffer. The bytes might come in one at a time in a dozen or more packets. Or, if the client is sending multiple requests in a single stream, a dozen requests might come in a single packet.
You just have to handle this.
I intend to receive packets of data over a socket but since they are sent from the sender with a high frequency a number of them get packed into a single byte array. SocketAsyncEventArgs.Buffer then holds multiple packets, even though they were sent separately (verified using wireshark).
I have already tried to queue the incoming packets and asynchronously work them off, but still I get the same result.
What could be reason for this behaviour?
This is how TCP works. TCP connection is a bi-directional stream of bytes, and you have to treat it as such. Single send from one end might result in multiple reads on the receiving side, and vice versa, multiple sends might end up in a single read, and application message boundaries are not preserved by the transport.
You have to buffer the input until you know you have a complete application message. Common approaches are:
fixed length messages,
pre-pending length in front of the message,
delimiting the stream with special "end-of-message" separator.
I might be mistaking, but isn't this the Naggle algorithm kicking in?
Your sockets should have a flag that disables this.
Im writing a server application for my iPhone app. The section of the server im working on is the relay server. This essentially relays messages between iPhones, through a server, using TCP sockets. The server reads the length of the header from the stream, then reads that number of bytes from the stream. It deserializes the header, and checks to see if the message is to be relayed on to another iPhone (rather than being processed on the server).
If it has to be relayed, it begins reading bytes from the sender's socket, 1024 bytes at a time. After each 1024 bytes are received, it adds those bytes (as a "packet" of bytes) to the outgoing message queue, which is processed in order.
This is all fine, however, but what happens if the sender gets interrupted, so it hasn't sent all its bytes (say, out of the 3,000 bytes it had to send, the sending iPhone goes into a tunnel after 2,500 bytes)?
This means that all the other devices are waiting on the remaining 500 bytes, which dont get relayed to them. Then if the sender (or anyone else for that matter) sends data to these sockets, they think the start of the new message is the end of the last one, corrupting the data.
Obviously from the description above, im using message framing, but I think im missing something. From what I can see, message framing only seems to allow the receiver to know the exact amount of bytes to read from the socket, before assembling them into an object. Wont things start to get hairy once a byte or two goes astray at some point, throwing everything out of sync? Is there a standard way of getting back in sync again?
Wont things start to get hairy once a byte or two goes astray at some point, throwing everything out of sync? Is there a standard way of getting back in sync again?
TCP/IP itself ensures that no bytes go "missing" over a single socket connection.
Things are a bit more complex in your situation, where (if I understand correctly) you're using a server as a sort of multiplexer.
In this case, here's some options off the top of my head:
Have the server buffer the entire message from point A before sending it to point B.
Close the B-side sockets if an abnormal close is detected from the A side.
Change the receiving side of the protocol so that a B-side client can detect and recover from a partial A-stream without killing and re-establishing the socket. e.g., if the server gave a unique id to each incoming A-stream, then the B client would be able to detect if a different stream starts. Or have an additional length prefix, so the B client knows both the entire length to expect and the length for that individual message.
Which option you choose depends on what kind of data you're transferring and how easy the different parts are to change.
Regardless of the solution, be sure to include detection of half-open connections.
I'm having an issue using Sockets in c#. Heres an example. Say I send the number 1, then I immediately send the number 2. The problem I run into sometimes is the client that is supposed to receive it will receive a packet containing '12'. I was wondering if there was a built in way to distinguish packets without using characters or something to separate the data.
To sum it up again, I said two packets. One with the number '1', one with the number '2'.
Server receives 1 packet with the data '12'.
I don't want to separate the packets with characters, like this ':1::2:' or anything like that, as I don't always have control over the format of the incoming data.
Any ideas?
Like if I do this
client.Send(new byte[1]{'1'}, 1,SocketFlags.None);
client.Send(new byte[1]{'2'}, 1,SocketFlags.None);
then on the server end
byte[] data = new byte[1024];
client.Receive(data);
data sometimes comes back with "12" even though I do two separate sends.
TCP/IP works on a stream abstraction, not a packet abstraction.
When you call Send, you are not sending a packet. You are only appending bytes to a stream.
When you call Receive, you are not receiving a packet. You are only receiving bytes from a stream.
So, you must define where a "message" begins and ends. There are only three possible solutions:
Every message has a known size (e.g., all messages are 42 bytes). In this case, you just keep Receiveing the data until you get that many bytes.
Use delimiters, which you're already aware of.
Use length prefixing (e.g., prefix each message with a 4-byte length-of-message). In this case, you keep Receiveing the length prefix until it arrives, decode it to get the actual length of the message, and then keep Receiveing the message itself.
You must do the message framing yourself. TCP/IP cannot do it for you.
TCP is a streaming protocol, so you will always receive whatever data has arrived since the last read, up to the streaming window size on the recipient's end. This buffer can be filled with data received from multiple packets of any given size sent by the sender.
TCP Receive Window Size and Scaling # MSDN
Although you have observed 1 unified blob of data containing 2 bytes on the receiving end in your example, it is possible to receive sequences of 1 byte by 1 byte (as sent by your sender) depending on network conditions, as well as many possible combinations of 0, 1, and 2 byte reads if you are doing non-blocking reads. When debugging on a typical uncongested LAN or a loopback setup, you will almost never see this if there is no delay on the sending side. There are ways, at lower levels of the network stack, to detect per-packet transmission but they're not used in typical TCP programming and are out of application scope.
If you switch to UDP, then every packet will be received as it was sent, which would match your expectations. This may suit you better, but keep in mind that UDP has no delivery promises and network routing may cause packets to be delivered out of order.
You should look into delimiting your data or finding some other method to detect when you have reached the end of a unit of data as defined by your application, and stick with TCP.
Its hard to answer your question without knowing the context. In contrast, by default, TCP/IP handles packet management for you automaticly (although you receive it in a streambased fashion). However when you have a very specific (bad?) implementation, you can send multiple streams over 1 socket at the same time making it impossible for lower level TCP/IP to detect the difference. Thereby making it very hard for yourself to identify different streams on the client. The only solution for this would be to send 2 totally unique streams (e.g. stream 1 only sends bytes lower then 127 and stream 2 only sends bytes higher or equal to 127). Yet again, this is terrible behavior
You must put into your TCP messages delimiters or use byte counts to tell where one message starts and another begins.
Your code has another serious error. TCP sockets may not give you all the data in a single call to Receive. You must loop on receiving of data until your application-specific records in the data stream indicate the entire message has been received. Your call to client.Receive(data) returns the count of bytes received. You should capture that number.
Likewise, when you send data, all of your data might not be sent in a single call. You must loop on sending of data until the count of bytes sent is equal to what you intended to send. The call to client.Send returns the actual count of bytes sent, which may not be all you tried to send!
The most common error I see people make with sockets is that they don't loop on the Send and Receive. If you understand why you need to do the looping, then you know why you need to have a delimiter or a byte count.
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