I am doing some test with TCP client application in a Raspberry Pi (server in the PC), with PPP (Point to Point Protocol) using a LTE Modem. I have used C program with sockets, checking system call's response. I wanted to test how socket works in a bad coverage area so I did some test removing the antenna.
I have followed the next steps:
Connect to server --> OK
Start sending data (write system call) --> OK (I also check in the server)
I removed the LTE modem's antenna (There is no network, it can't do ping)
Continue sending data (write system call) --> OK (server does not receive anything!!!)
It finished sending data and closed socket --> OK (connection is still opened and there is no data since the antenna was removed)
Program was finished
I put the antenna again
Some time later, the data has been uploaded and the connection closed. But I did another test following this steps but with more data, and it did not upload this data...
I do not know if there any way to ensure that the data written to TCP server is received by the server (I thought that TCP layer ensured this..). I could do it manually using an ACK but I guess that it has to be a better way to do.
Sending part code:
while(i<100)
{
sprintf(buf, "Message %d\n", i);
Return = write(Sock_Fd, buf, strlen(buf));
if(Return!=strlen(buf))
{
printf("Error sending data to TCP server. \n");
printf("Error str: %s \n", strerror(errno));
}
else
{
printf("write successful %d\n", i);
i++;
}
sleep(2);
}
Many thanks for your help.
The write()-syscall returns true, since the kernel buffers the data and puts it in the out-queue of the socket. It is removed from this queue when the data was sent and acked from the peer. When the OutQueue is full, the write-syscall will block.
To determine, if data has not been acked by the peer, you have to look at the size of the outqueue. With linux, you can use an ioctl() for this:
ioctl(fd, SIOCOUTQ, &outqlen);
However, it would be more clean and portable to use an inband method for determining if the data has been received.
TCP/IP is rather primitive technology. Internet may sound newish, but this is really antique stuff. TCP is needed because IP gives almost no guarantees, but TCP doesn't actually add that many guarantees. Its chief function is to turn a packet protocol into a stream protocol. That means TCP guarantees a byte order; no bytes will arrive out of order. Don't count on more than that.
You see that protocols on top of TCP add extra checks. E.g. HTTP has the famous HTTP error codes, precisely because it can't rely on the error state from TCP. You probably have to do the same - or you can consider implementing your service as a HTTP service. "RESTful" refers to an API design methodology which closely follows the HTTP philosophy; this might be relevant to you.
The short answer to your 4th and 5th topics was taken as a shortcut from this answer (read the whole answer to get more info)
A socket has a send buffer and if a call to the send() function succeeds, it does not mean that the requested data has actually really been sent out, it only means the data has been added to the send buffer. For UDP sockets, the data is usually sent pretty soon, if not immediately, but for TCP sockets, there can be a relatively long delay between adding data to the send buffer and having the TCP implementation really send that data. As a result, when you close a TCP socket, there may still be pending data in the send buffer, which has not been sent yet but your code considers it as sent, since the send() call succeeded. If the TCP implementation was closing the socket immediately on your request, all of this data would be lost and your code wouldn't even know about that. TCP is said to be a reliable protocol and losing data just like that is not very reliable. That's why a socket that still has data to send will go into a state called TIME_WAIT when you close it. In that state it will wait until all pending data has been successfully sent or until a timeout is hit, in which case the socket is closed forcefully.
The amount of time the kernel will wait before it closes the socket,
regardless if it still has pending send data or not, is called the
Linger Time.
BTW: that answer also refers to the docs where you can see more detailed info
Related
I'm learning about C socket programming and I came across this piece of code in a online tutorial
Server.c:
//some server code up here
recv(sock_fd, buf, 2048, 0);
//some server code below
Client.c:
//some client code up here
send(cl_sock_fd, buf, 2048, 0);
//some client code below
Will the server receive all 2048 bytes in a single recv call or can the send be be broken up into multiple receive calls?
TCP is a streaming protocol, with no message boundaries of packets. A single send might need multiple recv calls, or multiple send calls could be combined into a single recv call.
You need to call recv in a loop until all data have been received.
Technically, the data is ultimately typically handled by the operating system which programs the physical network interface to send it across a wire or over the air or however else applicable. And since TCP/IP doesn't define particulars like how many packets and of which size should compose your data, the operating system is free to decide as much, which results in your 2048 bytes of data possibly being sent in fragments, over a period of time.
Practically, this means that by calling send you may merely be causing your 2048 bytes of data be buffered for sending, much like an e-mail in a queue, except that your 2048 bytes aren't even a single piece of anything to the system that sends it -- it's just 2048 more bytes to chop into packets the network will accept, marked with a destination address and port, among other things. The job of TCP is to only make sure they're the same bytes when they arrive, in same order with relation to each other and other data sent through the connection.
The important thing at the receiving end is that, again, the arriving data is merely queued and there is no information retained as to how it was partitioned when requested sent. Everything that was ever sent through the connection is now either part of a consumable stream or has already been consumed and removed from the stream.
For a TCP connection a fitting analogy would be the connection holding an open water keg, which also has a spout (tap) at the bottom. The sender can pour water into the keg (as much as it can contain, anyway) and the receiver can open the spout to drain the water from the keg into say, a cup (which is an analogy to a buffer in an application that reads from a TCP socket). Both sender and receiver can be doing their thing at the same time, or either may be doing so alone. The sender will have to wait (send call will block) if the keg is full, and the receiver will have to wait (recv call will block) if the keg is empty.
Another, shorter analogy is that sender and receiver sit each at their own end of a opaque pipe, with the former pushing stuff in one end and the latter removing pushed stuff out of the other end.
I have a socket programming situation where the client shuts down the writing end of the socket to let the server know input is finished (via receiving EOF), but keeps the reading end open to read back a result (one line of text). It would be useful for the server to know that the client has successfully read the result and closed the socket (or at least shut down the reading end). Is there a good way to check/wait for such status?
No. All you can know is whether your sends succeeded, and some of them will succeed even after the peer read shutdown, because of TCP buffering.
This is poor design. If the server needs to know that the client received the data, the client needs to acknowledge it, which means it can't shutdown its write end. The client should:
send an in-band termination message, as data.
read and acknowledge all further responses until end of stream occurs.
close the socket.
The server should detect the in-band termination message and:
stop reading requests from the socket
send all outstanding responses and read the acknowledgements
close the socket.
OR, if the objective is only to ensure that client and server end at the same time, each end should shutdown its socket for output and then read input until end of stream occurs, then close the socket. That way the final closes will occur more or less simultaneously on both ends.
getsockopt with TCP_INFO seems the most obvious choice, but it's not cross-platform.
Here's an example for Linux:
import socket
import time
import struct
import pprint
def tcp_info(s):
rv = dict(zip("""
state ca_state retransmits probes backoff options snd_rcv_wscale
rto ato snd_mss rcv_mss unacked sacked lost retrans fackets
last_data_sent last_ack_sent last_data_recv last_ack_recv
pmtu rcv_ssthresh rtt rttvar snd_ssthresh snd_cwnd advmss reordering
rcv_rtt rcv_space
total_retrans
pacing_rate max_pacing_rate bytes_acked bytes_received segs_out segs_in
notsent_bytes min_rtt data_segs_in data_segs_out""".split(),
struct.unpack("BBBBBBBIIIIIIIIIIIIIIIIIIIIIIIILLLLIIIIII",
s.getsockopt(socket.IPPROTO_TCP, socket.TCP_INFO, 160))))
wscale = rv.pop("snd_rcv_wscale")
# bit field layout is up to compiler
# FIXME test the order of nibbles
rv["snd_wscale"] = wscale >> 4
rv["rcv_wscale"] = wscale & 0xf
return rv
for i in range(100):
s = socket.socket(socket.AF_INET, socket.SOCK_STREAM)
s.connect(("localhost", 7878))
s.recv(10)
pprint.pprint(tcp_info(s))
I doubt a true cross-platform alternative exists.
Fundamentally there are quite a few states:
you wrote data to socket, but it was not sent yet
data was sent, but not received
data was sent and losts (relies on timer)
data was received, but not acknowledged yet
acknowledgement not received yet
acknowledgement lost (relies on timer)
data was received by remote host but not read out by application
data was read out by application, but socket still alive
data was read out, and app crashed
data was read out, and app closed the socket
data was read out, and app called shutdown(WR) (almost same as closed)
FIN was not sent by remote yet
FIN was sent by remote but not received yet
FIN was sent and got lost
FIN received by your end
Obviously your OS can distinguish quite a few of these states, but not all of them. I can't think of an API that would be this verbose...
Some systems allow you to query remaining send buffer space. Perhaps if you did, and socket was already shut down, you'd get a neat error?
Good news is just because socket is shut down, doesn't mean you can't interrogate it. I can get all of TCP_INFO after shutdown, with state=7 (closed). In some cases report state=8 (close wait).
http://lxr.free-electrons.com/source/net/ipv4/tcp.c#L1961 has all the gory details of Linux TCP state machine.
TL;DR:
Don't rely on the socket state for this; it can cut you in many error cases. You need to bake the acknowledgement/receipt facility into your communications protocol. First character on each line used for status/ack works really well for text-based protocols.
On many, but not all, Unix-like/POSIXy systems, one can use the TIOCOUTQ (also SIOCOUTQ) ioctl to determine how much data is left in the outgoing buffer.
For TCP sockets, even if the other end has shut down its write side (and therefore will send no more data to this end), all transmissions are acknowledged. The data in the outgoing buffer is only removed when the acknowledgement from the recipient kernel is received. Thus, when there is no more data in the outgoing buffer, we know that the kernel at the other end has received the data.
Unfortunately, this does not mean that the application has received and processed the data. This same limitation applies to all methods that rely on socket state; this is also the reason why fundamentally, the acknowledgement of receipt/acceptance of the final status line must come from the other application, and cannot be automatically detected.
This, in turn, means that neither end can shut down their sending sides before the very final receipt/acknowledge message. You cannot rely on TCP -- or any other protocols' -- automatic socket state management. You must bake in the critical receipts/acknowledgements into the stream protocol itself.
In OP's case, the stream protocol seems to be simple line-based text. This is quite useful and easy to parse. One robust way to "extend" such a protocol is to reserve the first character of each line for the status code (or alternatively, reserve certain one-character lines as acknowledgements).
For large in-flight binary protocols (i.e., protocols where the sender and receiver are not really in sync), it is useful to label each data frame with an increasing (cyclic) integer, and have the other end respond, occasionally, with an update to let the sender know which frames have been completely processed, and which ones received, and whether additional frames should arrive soon/not-very-soon. This is very useful for network-based appliances that consume a lot of data, with the data provider wishing to be kept updated on the progress and desired data rate (think 3D printers, CNC machines, and so on, where the contents of the data changes the maximum acceptable data rate dynamically).
Okay so I recall pulling my hair out trying to solve this very problem back in the late 90's. I finally found an obscure doc that stated that a read call to a disconnected socket will return a 0. I use this fact to this day.
You're probably better off using ZeroMQ. That will send a whole message, or no message at all. If you set it's send buffer length to 1 (the shortest it will go) you can test to see if the send buffer is full. If not, the message was successfully transferred, probably. ZeroMQ is also really nice if you have an unreliable or intermittent network connection as part of your system.
That's still not entirely satisfactory. You're probably even better off implementing your own send acknowledge mechanism on top of ZeroMQ. That way you have absolute proof that a message was received. You don't have proof that a message was not received (something can go wrong between emitting and receiving the ack, and you cannot solve the Two Generals Problem). But that's the best that can be achieved. What you'll have done then is implement a Communicating Sequential Processes architecture on top of ZeroMQ's Actor Model which is itself implemented on top of TCP streams.. Ultimately it's a bit slower, but your application has more certainty of knowing what's gone on.
In C, to receive/send data you usually do(roughly):
Server:
Create socket
Bind socket to port
listen
Accept
Receive Send data
On client side:
Create socket
Connect
Receive send
My question comes after server has done accept.
Imagine after accept on the server side there are three separate lines
to send data:
connfd = accept(listenfd, (struct sockaddr*)NULL ,NULL);
write(connfd, var1, var1Size);
write(connfd, var2, var2Size);
write(connfd, var3, var3Size);
Does this mean on the client side I need to have three reads?
Like this:
read(sockfd, &x, size1);
read(sockfd, &y, size2);
read(sockfd, &z, size3);
In other words how should send and receive calls correspond
on server and client side? Should for each send be a corresponding receive on the client side?
What if on client side, after 3 read calls(like above), I want to send data to server?
Shall I just add one new send and one new receive on client and server side respectively?
Should all these send/receives be happening within a single accept call context?
Here is a image to better illustrate what kind of scenario I could be interested in:
Pseudo code explaining how to handle this kind of connections would be welcome.
Unless you are working with a protocol which has a concept of "messages", e.g. UDP, all you have is a stream of bytes. You can send and receive them any way you wish.
You could, for example, send two 16-bit integers and receive them as one 32-bit integer. This is probably not what you intended but it's perfectly legal and used all the time in situations where it is needed. You can compose data structures on either side (sending and receiving) independandly, as long as it makes sense to your application.
Your bytes are sent in the order of your write()'s and you WILL receive them in the same order. I.e.
send(var1) ---> recv(var1)
send(var2) ---> recv(var2)
There is no way in normal TCP (barring unused edge cases which I'll not even specify because nobody should use them) that you will receive var2 before var1.
TCP communication is bi-directional: each end-point (client and server) can send at the same time. It is up to you and your application to decide when to send and when to receive. The sending and receiving buffers are independant: you can send a few bytes, receive a few, send some more... and there will be no interference between them (i.e. you will not "overwrite" the receive buffer by sending some data nor vice versa).
I'll repeat it again: ALL you have in TCP is a stream of bytes. TCP doesn't know and doesn't care how these bytes are structured, neither on the sending nor on the receiving side. It's ALL up to you. When you send an integer or a data structure, you are sending a memory dump of those, as bytes.
For example, there's a common error where you attempt to send() a data structure and because the sending buffers are full, the system will make a partial write. If you do not check the return status of the send() call to detect this situation and then send the remainder of bytes by yourself, in another send() call, your client WILL be stuck in recv() when it expects the full structure and receives only a part of it, if you specify MSG_WAITALL.
TCP is a stream protocol, In the receiver side you cannot determine how many times the send has been called. Whenever recv is called it will give the number of bytes asked to read, if the requested number of bytes are not available then it will return the number of bytes currently in the socket buffer.
In case of UDP it will work as you mentioned, It is a datagram protocol. (use recvfrom to recv the data)
I have worked in non blocking TCP, in that both read and write can fail in non blocking case. TCP non blocking read can fail if there is no data available and TCP write can fail if peer side`s TCP buffer is full (I hope TCP buffer size is 64K).
Similarly UDP read (recvfrom) can fail if no data available. But what is the failure case for UDP write (sendto). I think in UDP write there will not be any non block error. Because TCP write sends data and wait for the ACK from other side. But this is not the case for UDP write it will just send and comes out and it doesnt wait for any ACK from peer side. If its not send to other side means its packet loss.
Whether my understanding of UDP non blocking write is correct ? Please explain ?
The most likely reason why a UDP non-blocking send would fail is that the UDP socket's in-kernel outgoing-data buffer is full. In this case, send()/sendto() would return -1 and errno would be set to EWOULDBLOCK.
Note that a non-blocking send()/sendto() doesn't actually send the data out the network device before it returns; rather it copies the data into an in-kernel buffer and returns immediately, and thereafter it is the kernel's responsibility to move that data out to the network as quickly as it can. The outgoing-data buffer can become full if your program tries to send a lot of data at once, because the CPU can add your new data to the buffer much faster than the network hardware can forward the buffer's data out to the network.
If you get a -1/EWOULDBLOCK error, usually the most graceful way to handle it is to stop trying to send on that socket until the socket select()'s (or poll()'s, or etc) as ready-for-write. When that happens, you know that the in-kernel buffer has been at least partially drained, and you can try the send()/sendto() call again.
Another (less likely) cause of an error from send() would be if the IP address you are trying to send to is invalid. In any case, you should check errno and find out what the errno value is, as that will give you better insight into what is going wrong.
Btw the behavior described above is not unique to UDP, either; you can and will have the same problem with a non-blocking TCP socket (even if the remote peer's receive window is not full) if you try to send() data on the socket faster than the local network card can drain the socket's in-kernel buffer.
Because TCP write sends data and wait for the ACK from other side.
No it doesn't. It copies your data into the socket send buffer, and if that is full it either blocks or returns -1/EWOULDBLOCK/EAGAIN.
But this is not the case for UDP write it will just send and comes out and it doesnt wait for any ACK from peer side.
No it doesn't. It copies your data into the socket send buffer, and if that is full it either blocks or returns -1/EWOULDBLOCK/EAGAIN.
In both cases the actual putting of bytes onto the wire is asynchronous to your program.
I use blocking C sockets on Windows.
I use them to send updates of a data from the server to the client and vice versa. I send updates at a high frequency (every 100ms). Does the send() function will wait for the recipient recv() to receive the data before ending ?
I assume not if I understand well the man page:
"Successful completion of send() does not guarantee delivery of the message."
So what will happen if one is running 10 send() occurences while the other has only complete 1 recv() ?
Do I need to use so some sort of acknowledgement system ?
Lets assume you are using TCP. When you call send, the data that you are sending is immediately placed on the outgoing queue and send then completes successfully. If however, send is unable to place the data on the outgoing queue, send will return with an error.
Since Tcp is a guaranteed delivery protocol, the data on the outgoing queue can only be removed once acknowledgement has been received by the remote end. This is because the data may need to be resent if no ack has been received in time.
If the remote end is sluggish, the outgoing queue will fill up with data and send will then block until there is space to place the new data on the outgoing queue.
The connection can however fail is such a way that there is no way any further data can be sent. Although once a TCP connection has been closed, any further sends will result in an error, the user has no way of knowing how much data did actually make it to the other side. (I know of no way of retrieving TCP bookkeeping from a socket to the user application). Therefore, if confirmation of receipt of data is required, you should probably implement this on application level.
For UDP, I think it goes without saying that some way of reporting what has or has not been received is a must.
send() blocks until the operating system (kernel) has taken the data and put it into a buffer of outgoing data. It does not wait until the other end has received the data.
If you're sending by TCP, you get guaranteed delivery1 and the other end will receive the data in the order sent. That might, however, be coalesced together so what you sent as 10 separate updates could be received as a single large packet (or vice versa -- a single update could be broken up across an arbitrary number of packets). This means, among other things, that any ACK of any data implicitly acknowledges receipt of all previous data.
If you're using UDP, none of that is true -- data can arrive out of order, or be dropped and never delivered at all. If you care about all the data being received, you just about need to build some sort of acknowledgement system of your own on top of UDP itself.
1 Of course, there's a limit on the guarantee -- if a network cable gets cut (or whatever) packets won't be delivered, but you'll at least get an error message telling you that the connection was lost.
If you're using TCP, you get the acknowledgements for free as that is part of what the protocol does under the hood. But sounds like for this type of application you would probably want to use UDP. In either case though send() will not block until the client has successfully recv().
If it's crucial that the client receive every message, then use TCP. If it's ok for the client to miss one or more messages, then use UDP.
TCP guarantees delivery at a lower TCP stack level. It retries delivery until the receiving part acknowledges that the data was received, but your application may never know about that fact.
Let's say that you are sending chunks of data and you need to place those chunks of data somewhere according to some logic. If your application is not prepared to know where each individual block has to be placed, receiving it at the TCP level may be useless. The original post was about the application level logic.