I am writing a server-client application using Winsock in c++ for sending a file line by line and I have a problem in sending huge string. The line size is very huge.
For getting the message from the client by the server I use the code below.
int result;
char message[200];
while (true)
{
recv(newSd, (char*)&message, sizeof(message), 0);
cout << "The Message from client: " << message << ";";
}
The above code working fine if I send small length of the message. But, what I wanted is to send an unknown size of lines in a file.
How to send a big unknown string instead of char message[200];
TCP is a byte stream, it knows nothing about messages or lines or anything like that. When you send data over TCP, all it knows about is raw bytes, not what the bytes represent. It is your responsibility to implement a messaging protocol on top of TCP to delimit the data in some meaningful way so the receiver can know when the data is complete. There are two ways to do that:
send the data length before sending the actual data. The receiver reads the length first, then reads however many bytes the length says.
send a unique terminator after sending the data. Make sure the terminator never appears in the data. The receiver can then read until the terminator is received.
You are not handling either of those in your recv() code, so I suspect you are not handling either of them in your send() code, too (which you did not show).
Since you are sending a text file, you can either:
send the file size, such as in a uint32_t or uint64_t (depending on how large the file is), then send the raw file bytes.
send each text line individually as-is, terminated by a CRLF or bare-LF line break after each line, and then send a final terminator after the last line.
You are also ignoring the return value of recv(), which tells you how many bytes were actually received. It can, and usually does, return fewer bytes than requested, so you must be prepared to call recv() multiple times, usually in a loop, to receive data completely. Same with send().
Related
I call a function multiple times using a for loop like this:
for ( int con=0; con < this->controller_info.size(); con++ ) {
try {
this->pi.home_axis( this->controller_info.at(con).addr );
}
catch( std::out_of_range &e ) { ... }
}
where the home_axis() function is defined as:
long ServoInterface::home_axis( int addr ) {
std::stringstream cmd;
if ( addr > 0 ) cmd << addr << " ";
cmd << "FRF";
cmd << "\n";
int bytes=this->controller.Write( cmd.str() );
return NO_ERROR;
}
and the controller.Write() function is just a wrapper for the standard write(2) which writes the characters in the string to a socket file descriptor.
You can see that each time home_axis() is called it should have its own, fresh, std::stringstream cmd buffer. But what is happening is that, for the first time the for loop executes, the host that is receiving the bytes written by home_axis, is receiving a single string, once:
1 FRF2 FRF
but if I print the bytes written then it prints 6, twice. So the writer is writing correctly, 6 bytes two different times, but the host is receiving it apparently as a single buffer.
If I execute that for loop again, then the host receives (properly),
1 FRF
and then
2 FRF
handling the two received buffers each as they come in.
How can the std::stringstream cmd buffers be getting mixed like this?
There are no threads involved here.
In an effort to pick this apart a bit, if I insert just 1µsec of delay in that for loop, i.e. usleep(1); then it works properly. Also, if I call the home_axis() function manually, but equally rapid succession, without using a for loop like this,
this->pi.home_axis( this->controller_info.at(0).addr );
this->pi.home_axis( this->controller_info.at(1).addr );
then that also works.
So I'm wondering if it's possible there is a compiler optimization going on?
This has nothing to do with the compiler at all.
TCP is a byte stream. It has no concept of message boundaries. There is no 1:1 relationship between writes and reads. You can write 2 messages of 6 bytes each, and the receiver may receive all 12 bytes at a time, or 1 byte and then 11 bytes, or any combination in between. That is just the way TCP works. By default, it breaks up data packets as it sees fit to optimize transmissions.
What is important is that TCP guarantees the bytes will be delivered (unless the connection is lost), and it will deliver the bytes in the same order that they are written.
As such, the sender must indicate in the data itself where each message begins and ends. Either by sending a message's length before its content, or by separating each message with a unique delimiter (as you are).
On the receiving side, a single read may receive a partial message, or pieces of multiple messages, etc. It is the receiver's responsibility to buffer incoming bytes and extract only complete messages from that buffer as needed, regardless of however many reads it takes to complete them.
As you are delimiting your messages with a trailing \n, the receiver should buffer all bytes and extract only messages that have received their \n, leaving any incomplete message at the end of the buffer for subsequent reads to finish.
This way, message boundaries are preserved and handled correctly.
I have been reading some socket guides such as Beej's guide to network programming. It is quite clear now that there is no guarantee on how many bytes are received in a single recv() call. Therefore a mechanism of e.g. first two bytes stating the message length should be sent and then the message. So the receiver receives the first two bytes and then receives in a loop until the whole message has been received. All good and dandy!?
I was asked by a colleague about messages going out of sync. E.g. what if, somehow, I receive two bytes in once recv() call that are actually in the middle of the message itself and it would appear as a integer of some value? Does that mean that the rest of the data sent will be out of sync? And what about receiving the header partially, i.e. one byte at a time?
Maybe this is overthinking, but I can't find this mentioned anywhere and I just want to be sure that I would handle this if it could be a possible threat to the integrity of the communication.
Thanks.
It is not overthinking. TCP presents a stream so you should treat it this way. A lot of problems concerning TCP are due to network issues and will probably not happen during development.
Start a message with a (4 byte) magic that you can look for followed by a (4 byte) length in an expected order (normally big endian). When receiving, read each byte of the header at the time, so you can handle it anyway the bytes were received. Based on that you can accept messages in a lasting TCP connection.
Mind you that when starting a new connection per message, you know the starting point. However, it doesn't hurt sending a magic either, if only to filter out some invalid messages.
A checksum is not necessary because TCP shows a reliable stream of bytes which was already checked by the receiving part of TCP, and syncing will only be needed if there was a coding issue with sending/receiving.
On the other hand, UDP sends packets, so you know what to expect, but then the delivery and order is not guaranteed.
Your colleague is mistaken. TCP data cannot arrive out of order. However you should investigate the MSG_WAITALL flag to recv() to overcome the possibility of the two length bytes arriving separately, and to eliminate the need for a loop when receiving the message body.
Its your responsibility to make you client and server syncing together, how ever in TCP there is no out of order delivery, if you got something by calling recv() you can think there isn't anything behind that that you doesn't received.
So the question is how to synchronize sender and receiver ? its easy, as stefaanv said, sender and receiver are knowing their starting point. so you can define a protocol for your network communication. for example a protocol could be defined this way :
4 bytes of header including message type and payload length
Rest of message is payload length
By this, you have to send 4 byte header before sending actual payload, then sending actual payload followed.
Because TCP has garauntied Inorder reliable delivery, you can make two recv() call for each pack. one recv() call with length of 4 bytes for getting next payload size, and another call to recv() with size specified in header. Its necessary to make both recv() blocking to getting synchronized all the time.
An example would be like this:
#define MAX_BUF_SIZE 1024 // something you know
char buf[MAX_BUF_SIZE];
int recvLen = recv(fd, buff, 4, MSG_PEEK);
if(recvLen==4){
recvLen = recv(fd, buff, 4);
if(recvLen != 4){
// fatal error
}
int payloadLen = extractPayloadLenFromHeader(buf);
recvLen = recv(fd, buff, payloadLen, MSG_PEEK);
if(recvLen == payloadLen){
recvLen = recv(fd, buff, payloadLen); // actual recv
if(recvLen != payloadLen){
// fatal error
}
// do something with received payload
}
}
As you can see, i have first called recv with MSG_PEEK flag to ensure is there really 4 bytes available or not, then received actual header. same for payload
I've been reading through Beej's Guide to Network Programming to get a handle on TCP connections. In one of the samples the client code for a simple TCP stream client looks like:
if ((numbytes = recv(sockfd, buf, MAXDATASIZE-1, 0)) == -1) {
perror("recv");
exit(1);
}
buf[numbytes] = '\0';
printf("Client: received '%s'\n", buf);
close(sockfd);
I've set the buffer to be smaller than the total number of bytes that I'm sending. I'm not quite sure how I can get the other bytes. Do I have to loop over recv() until I receive '\0'?
*Note on the server side I'm also implementing his sendall() function, so it should actually be sending everything to the client.
See also 6.1. A Simple Stream Server in the guide.
Yes, you will need multiple recv() calls, until you have all data.
To know when that is, using the return status from recv() is no good - it only tells you how many bytes you have received, not how many bytes are available, as some may still be in transit.
It is better if the data you receive somehow encodes the length of the total data. Read as many data until you know what the length is, then read until you have received length data. To do that, various approaches are possible; the common one is to make a buffer large enough to hold all data once you know what the length is.
Another approach is to use fixed-size buffers, and always try to receive min(missing, bufsize), decreasing missing after each recv().
The first thing you need to learn when doing TCP/IP programming: 1 write/send call might take
several recv calls to receive, and several write/send calls might need just 1 recv call to receive. And anything in-between.
You'll need to loop until you have all data. The return value of recv() tells you how much data you received. If you simply want to receive all data on the TCP connection, you can loop until recv() returns 0 - provided that the other end closes the TCP connection when it is done sending.
If you're sending records/lines/packets/commands or something similar, you need to make your own protocol over TCP, which might be as simple as "commands are delimited with \n".
The simple way to read/parse such a command would be to read 1 byte at a time, building up a buffer with the received bytes and check for a \n byte every time. Reading 1 byte is extremely inefficient, so you should read larger chunks at a time.
Since TCP is stream oriented and does not provide record/message boundaries it becomes a bit more tricky - you'd
have to recv a piece of bytes, check in the received buffer for a \n byte, if it's there - append the bytes to previously received bytes and output that message. Then check the remainder of the buffer after the \n - which might contain another whole message or just the start of another message.
Yes, you have to loop over recv() until you receive '\0' or an
error happen (negative value from recv) or 0 from recv().
For the first option: only if this zero is part of your
protocol (the server sends it). However from your code it seems that
the zero is just to be able to use the buffer content as a
C-string (on the client side).
The check for a return value of 0 from recv:
this means that the connection was closed (it could be part
of your protocol that this happens.)
I wrote a simple client-server program. Network.h is a header file which uses Winsock2.h (TCP/IP mode) to create socket, accept/connect in blocking mode, send/recv in non-blocking mode. I made it so that the function string TNetwork::Recv(int size) will return the string "Nothing" if it gets WSAWOULDBLOCK error (no data is received yet)
Here is my main function:
int main(){
string Ans;
TNetwork::StartUp(); //WSA start up, etc
cin >> Ans;
if (Ans == "0"){ // 0 --> server
TNetwork::SetupAsServer(); //accept connection (in blocking mode!)
while (true){
TNetwork::Send("\nAss" + '\0'); //without null terminator, the client may read extra bytes, causing undefined behavior (?)
TNetwork::Send("embly" + '\0');
cin >> Ans;
}
}
else{ // others --> regard Ans as IP address. e.g. I can type "127.0.0.1"
TNetwork::SetupAsClient(Ans);
string Rec;
while (true){
Rec = TNetwork::Recv(1000);
if (Rec != "Nothing"){
cout << Rec;
}
}
}
system("PAUSE");
}
Supposedly, the client would print "Assembly" when connected, and when the server enters anything to its console window. Sometimes, though, the client would only print out "\nAss" in the console without the "embly.
To my understanding, TCP/IP ensures all data to be sent and in the correct order, so I guess what happens is that both packets arrive at the same time, which happen quite often over the unstable internet. And due to this null terminator, the client would ignore the "embly", since the Recv() function stopped reading when it hits a null terminator.
So, how can I ensure that the client will always read all data packets correctly?
Yes, the network stack will send the data in the correct order and doesn't care what termination type you use. This has to do with how you're receiving and processing the data stream (note: not packets, stream). If you receive all 11 bytes and print it to the screen, the print function will stop when it reaches the zero, but the rest of the data is still there.
Note: since it's a stream, what happens if you received only 10 bytes of data from the stream? You need to scan what you receive for the zero to know if you've received a full "zero-terminated string" if that's how you want to communicate your data.
EDIT: Also, I don't think "\nAss" + '\0' is doing what you think it is. Instead of adding a 0 character to the end of the string (which already has one, by the way), it's adding 0 to your string pointer.
As #mark points out, TCP is all about streams, not packets. TCP takes care of ensuring that data is reliably transmitted from A to B and that the data is delivered to the consumer in the order in which it was transmitted. Yes, the data is packetized on the wire, but the TCP stack on the system takes those packets and builds the stream which it makes available to you through the recv() function. The TCP stack handles out-of-order data, missing data, and duplicated data such that by the time your application sees it, the stream is a mirror-copy of when the sender sent.
To properly receive TCP data, you will typically need some kind of loop that reads data from the socket when it becomes available. The way I normally do this is to have a thread that is dedicated to servicing the socket. In the thread function is a loop that reads data from the socket when it becomes available and is idle otherwise. This loop reads data into a buffer of, say, 1 KB. Once the data is received from the socket into this buffer, the buffer is copied to another thread for processing. In the thread function for the processing thread is a loop that receives the 1 KB buffers from the socket thread and adds them to the back end of a master buffer of, say, 1 MB. The processing thread then processes the messages out of this master buffer and makes them available to the application.
For a simple demo application, two threads may be overkill. The two threads I've described could be certainly be combined into one, but for my application, it is more efficient to have two threads and take advantage of the multiple cores on my system. The point is, if you're going to have a front-end UI, there's not going to be a way around using at least one thread and still have the UI be responsive.
One other thing. There are two commonly-used mechanisms for protocol design. You're using one, namely, a marker (e.g., a null terminator, etc.) to signal the begin/end of a message. I don't prefer this mechanism mainly because the marker may actually need to be part of the message at some point. The other mechanism is to have a header on each message that tells, at a minimum, how long the message is. I prefer this mechanism and include in my headers a sync word and the message type as well. For example,
struct Header
{
__int16 _sync; // a hex pattern, e.g., 0xABCD
__int16 _type;
__int32 _length;
}
That's a total of 8 bytes. So when processing from the master buffer, I read the first 8 bytes, verify the sync word, and get the length. I determine if there are 'length' bytes available in the master buffer. If not, I have to wait until the socket thread provides me more data before checking again. If so, I extract 'length' bytes from the master buffer and pass that to an object created according to the specified type, which knows how to interpret that particular message. Then repeat.
As I mentioned, I use a master buffer of 1 MB or so. As messages are processed, it is important to remove them from the master buffer so there is additional space available for new data on the back end. This involves simply copying the unprocessed data, if any, to the beginning of the buffer. In cases where data comes in faster than you can process it, the master buffer may need the ability to resize itself to accommodate the additional data.
I hope that's not overwhelming. Start simple and add as you go.
I've been reading through Beej's Guide to Network Programming to get a handle on TCP connections. In one of the samples the client code for a simple TCP stream client looks like:
if ((numbytes = recv(sockfd, buf, MAXDATASIZE-1, 0)) == -1) {
perror("recv");
exit(1);
}
buf[numbytes] = '\0';
printf("Client: received '%s'\n", buf);
close(sockfd);
I've set the buffer to be smaller than the total number of bytes that I'm sending. I'm not quite sure how I can get the other bytes. Do I have to loop over recv() until I receive '\0'?
*Note on the server side I'm also implementing his sendall() function, so it should actually be sending everything to the client.
See also 6.1. A Simple Stream Server in the guide.
Yes, you will need multiple recv() calls, until you have all data.
To know when that is, using the return status from recv() is no good - it only tells you how many bytes you have received, not how many bytes are available, as some may still be in transit.
It is better if the data you receive somehow encodes the length of the total data. Read as many data until you know what the length is, then read until you have received length data. To do that, various approaches are possible; the common one is to make a buffer large enough to hold all data once you know what the length is.
Another approach is to use fixed-size buffers, and always try to receive min(missing, bufsize), decreasing missing after each recv().
The first thing you need to learn when doing TCP/IP programming: 1 write/send call might take
several recv calls to receive, and several write/send calls might need just 1 recv call to receive. And anything in-between.
You'll need to loop until you have all data. The return value of recv() tells you how much data you received. If you simply want to receive all data on the TCP connection, you can loop until recv() returns 0 - provided that the other end closes the TCP connection when it is done sending.
If you're sending records/lines/packets/commands or something similar, you need to make your own protocol over TCP, which might be as simple as "commands are delimited with \n".
The simple way to read/parse such a command would be to read 1 byte at a time, building up a buffer with the received bytes and check for a \n byte every time. Reading 1 byte is extremely inefficient, so you should read larger chunks at a time.
Since TCP is stream oriented and does not provide record/message boundaries it becomes a bit more tricky - you'd
have to recv a piece of bytes, check in the received buffer for a \n byte, if it's there - append the bytes to previously received bytes and output that message. Then check the remainder of the buffer after the \n - which might contain another whole message or just the start of another message.
Yes, you have to loop over recv() until you receive '\0' or an
error happen (negative value from recv) or 0 from recv().
For the first option: only if this zero is part of your
protocol (the server sends it). However from your code it seems that
the zero is just to be able to use the buffer content as a
C-string (on the client side).
The check for a return value of 0 from recv:
this means that the connection was closed (it could be part
of your protocol that this happens.)