I have an app where a single client talks to a single server. Normally, the client does a single connect, and then calls send repeatedly, and there's no problem.
However, I need to do a version where the client sets up a connection for each individual send (a bit like HTTP with and without keep-alive). In this version, the client calls socket, connect, send once, and then close.
The problem with this is that I very quickly run out of ephemeral client ports, and the connect fails. To get around this I call setsockopt with SO_REUSEADDR, and then bind to port 0, before calling connect (see here, for example).
This works, except that the TCP connection is no longer reliable. I get occasional incorrect data, presumably because there's still data around when the TCP connection is closed.
Is there any way to make this reliable (and fast)? shutdown before close doesn't help. Maybe I can get select to tell me if the socket is ready for output, but that seems like overkill.
Do you have to use TCP? If so, you will probably have to maintain an open connection and route your messages over that one connection.
There is SCTP, which may be a good fit for your use case - a reliable datagram protocol:
Like TCP, SCTP provides reliable, connection oriented data delivery with congestion control. Unlike TCP, SCTP also provides message bound‐ ary preservation, ordered and unordered message delivery, multi-stream‐ ing and multi-homing. Detection of data corruption, loss of data and duplication of data is achieved by using checksums and sequence numbers. A selective retransmission mechanism is applied to correct loss or corruption of data.
Related
I have a fault tolerant application, where an X Server requests to start an Application on a remote client (by some other mechanism) and receive and display its X-window. Fault tolerance means that the server needs to detect loss of the connection to the client and then call a different back-up-client and start the application there and show the window.
My question is whether there exists a mechanism in the X11 protocol that allows to reliably detect in an X11-Server whether the connection has been broken or not.
Experiments show that when unplugging a cable connection it needs some TCP-Timeout to detect the connection loss on socket level. This is very OS-dependent. In our case it was abut 30 minutes after which the X-Server eventually closed the window.
So another assumption could be that the X11-stream constantly delivers some commands and the server could implement some logic like this: If the X11-stream does not deliver any X11 traffic for a timeout y (e.g. 3 seconds), we assume the connection is lost and actively close the window and establish the connection to the fall-back-client.
Is the assumption true? I did not see any such statement in the X11-protocol about how to detect connection loss. Is there any explicit lifesign that is regularly transmitted? Or is the assumption valid that there is constant traffic? Or could there be longer periods of inactivity where nothing is transmitted at all while the connection is perfectly up and running?
There is a NoOperation command from the client that could be used for such purpose. But do clients usually implement something like that as a lifesign?
I have a fault tolerant application, where an X Server needs to start an Application...
I don't think that an X server can "start an application". May be that some setup allows something similar to that, but normally is not so.
...whether there exists a mechanism in the X11 protocol that allows to reliably detect in an X11-Server whether the connection has been broken or not.
No, it does not exist. The X11 protocol is based on TCP/IP, which does not provide directly this "heartbeat". I think the assumption is that, if you click or otherwise stimulate an X11 window, the TCP layer will timeout or throw another error if the client application is gone.
I did not see any statement in the X11-protocol about how to detect connection loss.
There is a NoOperation command from the client that could be used for such purpose. But do clients usually implement something like that as a lifesign?
Maybe that some application uses that NoOperation, but the purpose would be different from what you need. I mean, the X11 server is like an extension from the point of view of an application; the application can have interest to know whether the server is up and working, but it is not true the contrary. And, anyway, even if the server could detect that the application is gone, probably there is no way to tell the server to launch another application.
Probably a special proxy could be deployed; it could launch the application and monitor the connection (in both ways) and take the required steps in case the application goes away. But then again, who would monitor the proxy application?
First of all, X Protocol relies completely on TCP to send/receive information.
You cannot safely put a timeout capable transaction in order to detect a timeout in TCP. TCP is designed to retransmit only those segments that have already been sent but no acknowledged. It is completely asynchronous, in the sense that you send a command, and you can receive many responses or events unrelated to that command, before you receive the response. There's no heartbeat mechanism on XProtocol (except that the NOOP command is sent to synchronize operations with the server, and you receive a response for it, but you cannot overuse it, as that slows down severely the X connection, just launch any client with the -synchronous option to see it, see X(7)). You can even have TCP connections alive for years without interchanging a single packet. There's some mechanism, activated by option SO_KEEPALIVE that makes tcp to employ such heartbeat on TCP for a connection that has no data to transmit, but the X11 protocol normally doesn't make use of it. You don't post any code, nor a description of how the system is configured. The standard XServer never starts a connection by itself, except when launched specifically to negotiate with an XDMCP server (and this is done on UDP protocol) to serve as an XTerminal.
From your words probably you don't know that the roles of server and client are exchanged in X Protocol (the client is the remote application that connects to the server to display its output, and the server is the application that controls your display, mouse and keyboard) There's no means for the server to create a new client, so you need to be creating this connection in other means (probably through SSH, but not described).
By the way, when you say:
Experiments show that when unplugging a cable connection it needs some TCP-Timeout to detect the connection loss on socket level. This is very OS-dependent. In our case it was abut 30 minutes after which the X-Server eventually closed the window.
That is not OS-dependent. It is precisely the standard behaviour when you don't have traffic to send, there's no packet exchanged, so no detection is made (except if your client ---remember, this is the remote application program that wants to show its data in your local server--- activates the SO_KEEPALIVE option, and it requires several losses before declaring a lost connection) In your case the amount of time is variable because timers don't start until there's some data sent over the unplugged connection, and this makes it variable (not OS dependant)
On other side, you cannot pretend the server is going to turn on your monitor in case you leave the office and turn it off by mistake or by accident. What is the fault tolerance specification in that case?
IMHO, in regard of the presentation protocol, the application should be ready to show you as much information about the system as soon as you activate the connection (but the connection must be something allowed to fail). What is important is the means you develop for the application to be fault tolerant, even in the case you are not there to see the display. Will be somebody be advised that no one is looking at the screen? Are you going to detect the absence of operators in that case? Don't take this as a flame, but common sense should imperate in this case.
In case you need to ensure the connectivity to the remote host is available, you need to use another means to check for it. I recommend you to have a simple application pinging the remote host and alerting in case you don't get a positive result. Or you can open a connection to the server and then close it as soon as you get a positive response from the server (the first packet, for example) This will lead us to the next step, that is to ensure that some human is looking at the (turned on) screen of the display :)
For example, you can run a client in parallel to the one you are interested in, and force a heartbeat by asking for some server atom name (or a root window property value) in a loop with some delay. This will make the connection fail or your client can alert in case it doesn't receive the answer in some configurable time.
I have a TCP client sending data to a server continuously . After successful connection of client with the server , client sends data continuously with some intervals in terms of few seconds .
When the link between the client and server got disconnected after sending few data ,I came to know that TCP retransmits the data according to the value in TCP_retries2 , I configured this value to be 8 , such that I get write error after 100 secs .
But there will be some unacknowledged packets in send-q .
Is there way to read the content of this unacknowledged packets in send-q in my program before closing this socket or should i remember the send data and resend it after connecting again ? Is there any other way to implement this ?
You can get the size of sendq with an ioctl:
SIOCOUTQ
Returns the amount of unsent data in the socket send queue.
The socket must not be in LISTEN state, otherwise an error
(EINVAL) is returned. SIOCOUTQ is defined in
<linux/sockios.h>. Alternatively, you can use the synonymous
TIOCOUTQ, defined in <sys/ioctl.h>.
Note that sendq only tells you what the kernel of the remote system accepted, it does not guarantee that the application running on that host handled it. Most failures exist in the network between the communicating parties, but this metric can't be used for definite proof as successful transmission.
Once the application has given its data to TCP, it is the responsibility of TCP to keep track of the acknowledgement of the packets. If ACKs are not forthcoming, it tries its best to get the packet delivered based on RTO algorithm. Now until ACK is received, the data is kept in TCP_SEND_Q. I do not think there is any control from the application to determine current state of TCP_SEND_Q.
//should i remember the send data and resend it after connecting again//
How do you do this? The previous connection status is gone, isn't? Until the client and the server applications maintain some understanding as to what was received and sent offline, you have to start fresh with new connection.
No there isn't.
If you need to know that the peer application has received the data, you need to have the peer application acknowledge it back to your application via your application protocol, and treat any unacknowledged data as needing re-sending from your application somehow. This also brings in the question of transactional idempotence, so that you can resend with impunity.
It takes two to tango. You can close your end of the connection and it waits for the other end of the connection to drop, too. Think 3-way handshake in reverse.
How long do you wait between closing the connectiion and re-opening it? You must wait at least the TIME_WAIT before trying to reconnect using the same connection info.
How do I find out from a socket client program that the remote connection is down (e.g. the server is down). When I do a recv and the server is down it blocks if I do not set any timeout. However in my case I cannot put any reliable timeout value to get around it since otherwise the recv times out even when the server is up but the response really takes longer than the timeout value that I have set.
Unfortunately, ZeroMQ just passes this on to the next layer. So the protocol you are implementing on top of ZeroMQ will have to handle this.
Heartbeats are recommended. Basically, just have one side send a message if the connection is otherwise idle. The other side can treat the absence of such messages as a failure condition and close the connection.
You may wish to modify your higher level protocols to be more robust. For example, you can submit a command, query its status, and allow the other side to forget about the command. That way, if the connection is lost, you can reconnect and query any outstanding commands. Any it doesn't have, you know didn't get through and can resubmit. Once you get a reply with the result of a command, you can tell the other side that it can now forget the response.
This allows you to keep the connection active while a long-running command is ongoing. Every so often you ask, "is everything okay". The other side responds, "yes". You can use long polling where the other side delays responding for a second or so while the command is in process. This allows it to return the results immediately rather than having to wait a second for your next query.
The specifics depend on your exact requirements, but you must design this correctly into your protocol.
If the remote host goes down without sending you a tcp FIN package then you have no chance to detect that. You can test that behaviour by firewalling a port after a connection has been established on that port. Your program will "hang" forever.
However, the Linux kernel supports a mechanism called TCP keep alives which are meant to close a tcp connection after a given timeout. If you can't specify a timeout for your application, than there isn't a reliable chance to use that. Last chance might be to use features of the application protocol (can you name it?), if that protocol does not support features for connection handling you may invent something on your own on top of that.
I'm new to socket programming and I need to implement a UDP based rateless file transmission system to verify a scheme in my research. Here is what I need to do:
I want a server S to send a file to a group of peers A, B, C.., etc. The file is divided into a number of packets. At the beginning, peers will send a Request message to the server to initialize transmission. Whenever S receives a request from a client, it ratelessly transmit encoded packets(how to encode is done by my design, the encoding itself has the erasure-correction capability, that's why I can transmit ratelessly via UDP) to that client. The client keeps collecting packets and try to decode them. When it finally decodes all packets and re-construct the file successfully, it sends back a Stop message to the server and S will stop transmitting to this client.
Peers request the file asynchronously (they may request the file at different time). And the server will have to be able to concurrently serve multiple peers. The encoded packets for different clients are different (they are all encoded from the same set source packets, though).
Here is what I'm thinking about the implementation. I have not much experience with unix network programming though, so I'm wondering if you can help me assess it, and see if it is possible or efficient.
I'm gonna implement the server as a concurrent UDP server with two socket ports(similar to TFTP according to the UNP book). One is to receive controlling messages, as in my context it is for the Request and Stop messages. The server will maintain a flag (=1 initially) for each request. When it receives a Stop message from the client, the flag will be set to 0.
When the serve receives a request, it will fork() a new process that use the second socket and port to send encoded packets to the client. The server keeps sending packets to the client as long as the flag is 1. When it turns to 0, the sending ends.
The client program is easy to do. Just send a Request, recvfrom() the server, progressively decode the file and send a Stop message in the end.
Is this design workable? The main concerns I have are: (1), is that efficient by forking multiple processes? Or should I use threads? (2), If I have to use multiple processes, how can the flag bit be known by the child process? Thanks for your comments.
Using UDB for file transfer is not best idea. There is no way for server or client to know if any packet has been lost so you would only know that during reconstruction assuming you have some mechanism (like counter) to detect lost packes. It would then be hard to request just one of those packets that got lost. And in the end you would have a code that would do what TCP sockets do. So I suggest to start with TCP.
Typical design of a server involves a listener thread that spawns a worker thread whenever there is a new client request. That new thread would handle communication with that particular client and then end. You should keep a limit of clients (threads) that are served simultaneously. Do not spawn a new process for each client - that is inefficient and not needed as this will get you nothing that you can't achieve with threads.
Thread programming requires carefulness so do not cut corners. Otherwise you will have hard time finding and diagnosing problems.
File transfer with UDP wil be fun :(
Your struct/class for each message should contain a sequence number and a checksum. This should enable each client to detect, and ask for the retransmission of, any missing blocks at the end of the transfer.
Where UDP might be a huge winner is on a local LAN. You could UDP-broadcast the entire file to all clients at once and then, at the end, ask each client in turn which blocks it has missing and send just those. I wish Kaspersky etc. would use such a scheme for updating all my local boxes.
I have used such a broadcast scheme on a CANBUS network where there are dozens of microControllers that need new images downloaded. Software upgrades take minutes instead of hours.
I should state that I'm not asking about specific implementation details (yet), but just a general overview of what's going on. I understand the basic concept behind a socket, and need clarification on the process as a whole. My (probably very wrong) understanding is currently this:
A socket is constantly listening for clients that want to connect (in its own thread). When a connection occurs, an event is raised that spawns another thread to perform the connection process. During the connection process the client is assigned it's own socket in which to communicate with the server. The server then waits for data from the client and when data arrives an event is raised which spawns a thread to read the data from a stream into a buffer.
My questions are:
How off is my understanding?
Does each client socket require it's own thread to listen for data on?
How is data routed to the correct client socket? Is this something taken care of by the guts of TCP/UDP/kernel?
In this threaded environment, what kind of data is typically being shared, and what are the points of contention?
Any clarifications and additional explanation would be greatly appreciated.
EDIT:
Regarding the question about what data is typically shared and points of contention, I realize this is more of an implementation detail than it is a question regarding general process of accepting connections and sending/receiving data. I had looked at a couple implementations (SuperSocket and Kayak) and noticed some synchronization for things like session cache and reusable buffer pools. Feel free to ignore this question. I've appreciated all your feedback.
One thread per connection is bad design (not scalable, overly complex) but unfortunately way too common.
A socket server works more or less like this:
A listening socket is setup to accept connections, and added to a socketset
The socket set is checked for events
If the listening socket has pending connections, new sockets are created by accepting the connections, and then added to the socket set
If a connected socket has events, the relevant IO functions are called
The socket set is checked for events again
This happens in one thread, you can easily handle thousands of connected sockets in a single thread, and there's few valid reasons for making this more complex by introducing threads.
while running
select on socketset
for each socket with events
if socket is listener
accept new connected socket
add new socket to socketset
else if socket is connection
if event is readable
read data
process data
else if event is writable
write queued data
else if event is closed connection
remove socket from socketset
end
end
done
done
The IP stack takes care of all the details of which packets go to what "socket" in which order. Seen from the applications point of view, a socket represents a reliable ordered byte stream (TCP) or an unreliable unordered sequence of packets(UDP)
EDIT: In response to updated question.
I don't know either of the libraries you mention, but on the concepts you mention:
A session cache typically keeps data associated with a client, and can reuse this data for multiple connections. This makes sense when your application logic requires state information, but it's a layer higher than the actual networking end. In the above sample, the session cache would be used by the "process data" part.
Buffer pools are also an easy and often effective optimization of a high-traffic server. The concept is very easy to implement, instead of allocating/deallocating space for storing data you read/write, you fetch a preallocated buffer from a pool, use it, then return it to a pool. This avoids the (sometimes relatively expensive) backend allocation/deallocation mechanisms. This is not directly related to networking, you can just as well use buffer pools for e.g. something that reads chunks of files and process them.
How off is my understanding?
Pretty far.
Does each client socket require it's own thread to listen for data on?
No.
How is data routed to the correct client socket? Is this something taken care of by the guts of TCP/UDP/kernel?
TCP/IP is a number of layers of protocol. There's no "kernel" to it. It's pieces, each with a separate API to the other pieces.
The IP Address is handled in on place.
The port # is handled in another place.
The IP addresses are matched up with MAC addresses to identify a particular host. The port # is what ties a TCP (or UDP) socket to a particular piece of application software.
In this threaded environment, what kind of data is typically being shared, and what are the points of contention?
What threaded environment?
Data sharing? What?
Contention? The physical channel is the number one point of contention. (Ethernet, for example depends on collision-detection.) After that, well, every part of the computer system is a scarce resource shared by multiple applications and is a point of contention.