I have seen a number of examples of paho clients reading sensor data then publishing, e.g., https://github.com/jamesmoulding/motion-sensor/blob/master/open.py. None that I have seen have started a network loop as suggested in https://eclipse.org/paho/clients/python/docs/#network-loop. I am wondering if the network loop is unnecessary for publishing? Perhaps only needed if I am subscribed to something?
To expand on what #hardillb has said a bit, his point 2 "To send the ping packets needed to keep a connection alive" is only strictly necessary if you aren't publishing at a rate sufficient to match the keepalive you set when connecting. In other words, it's entirely possible the client will never need to send a PINGREQ and hence never need to receive a PINGRESP.
However, the more important point is that it is impossible to guarantee that calling publish() will actually complete sending the message without using the network loop. It may work some of the time, but could fail to complete sending a message at any time.
The next version of the client will allow you to do this:
m = mqttc.publish("class", "bar", qos=2)
m.wait_for_publish()
But this will require that the network loop is being processed in a separate thread, as with loop_start().
The network loop is needed for a number of things:
To deal with incoming messages
To send the ping packets needed to keep a connection alive
To handle the extra packets needed for high QOS
Send messages that take up more than one network packet (e.g. bigger than local MTU)
The ping messages are only needed if you have a low message rate (less than 1 msg per keep alive period).
Given you can start the network loop in the background on a separate thread these days, I would recommend starting it regardless
Related
I currently have a game server with a customizable tick rate, but for this example let's suggest that the server is only going to tick once per second or 1hz. I'm wondering what's the best way to handle incoming packets if the client send rate is faster than the server's as my current setup doesn't seem to work.
I have my udp blocking receive with a timeout inside my tick function, and it works, however if the client tick rate is higher than the server, all of the packets are not received; only the one that is being read at the current time. So essentially the server is missing packets being sent by clients. The image below demonstrates my issue.
So my question is, how is this done correctly? Is there a separate thread where packets are read constantly, queued up and then the queue is processed when the server ticks or is there a better way?
Image was taken from a video https://www.youtube.com/watch?v=KA43TocEAWs&t=7s but demonstrates exactly what I'm explaining
There's a bunch going on with the scenario you describe, but here's my guidance.
If you are running a server at 1hz, having a blocking socket prevent your main logic loop from running is not a great idea. There is a chance you won't be receiving messages at the rate you expect (due to packet loss, network lag spike or client app closing)
A) you certainly could create another thread, continue to make blocking recv/recvfrom calls, and then enqueue them onto a thread safe data structure
B) you could also just use a non-blocking socket, and keep reading packets until it returns -1. The OS will buffer (usually configurable) a certain number of incoming messages, until it starts dropping them if you aren't reading.
Either way is fine, however for individual game clients, I prefer the second simple approaches when knowing I'm on a thread that is servicing the socket at a reasonable rate (5hz seems pretty low, but may be appropriate for your game). If there's a chance you are stalling the servicing thread (level loading, etc), then go with the first approach, so you don't detect it as a disconnection if you miss sending/receiving a periodic keepalive message.
On the server side, if I'm planning on a large number of clients/data, I go to great lengths to efficiently read from sockets - using IO Completion Ports on Windows, or epoll() on Linux.
Your server could have to have a thread to tick every 5 seconds just like the client to receive all the packets. Anything not received during that tick would be dropped as the server was not listening for it. You can then pass the data over from the thread after 5 ticks to the server as one chunk. The more reliable option though is to set the server to 5hz just like the client and thread every packet that comes in from the client so that it does not lock up the main thread.
For example, if the client update rate is 20, and the server tick rate is 64, the client might as well be playing on a 20 tick server.
I have been testing a program which has simple communication between two machines over a 1Gbps line. While running TCP communications over the line I occasionally receive write errors on the client side (due to a timeout) when the network is totally flooded (running at or close to 100% usage). This generally happens when I am running multiple instances of the same program going to different ports.
My question is, is it possible to get a write error but still receive the message on the server side. It appears that is what is happening, and I am not quite sure why. Could it be that the ACK coming back to the client is what is timing out?
Yes, that is possible. TCP does not guarantee you that data you sent successfully is received and that data that is sent unsuccessfully is not received. This problem is unsolvable. It is called the Generals Problem. There is always a way to loose messages/packets such that the sender comes to the wrong conclusion. TCP guarantees that the receiver receives the same stream of bytes that the sender sent, but possibly cut off at an arbitrary point.
This unreliability has performance reasons, too. TCP data is buffered on both hosts as well as on the network. Acknowledgement is delayed.
You have to live with this. If you make your scenario more concrete I can suggest some strategies of dealing with this.
send puts data into the TCP send buffer.
If the send buffer has no enough space, send will block util the data is completely or partly copied into the send buffer, or the designed timeout arrives.
Read timeout and write timeout is OK. You should check and process them. The way is restarting read/write operation after timeout. You also pay attention to other read/write error except timeout.
I'm not sure if this is the correct place to ask, so forgive me if it isn't.
I'm writing computer monitoring software that needs to connect to a server. The server may send out relatively urgent messages, such as sound or cancel an alarm, and the client may send out data about the computer, such as screenshots. The data that the client sends isn't too critical on timing, but shouldn't be more than a two minutes late.
It is essential to the software that portforwarding need not be set up, and it is assumed that the internet connection will be done through a wireless router that has NAT almost all the time.
My idea is to have a TCP connection initiated from the client, and use that to transfer data. Ideally, I would have no data being sent when it is not needed, but I believe this to be impossible. Would sending the equivalent of a ping every now and again keep the connection alive, and what sort of bandwidth would it use if this program was running all the time on the computer? In addition, would it be possible to reduce the header size for these keep-alives?
Before I start designing the communication and programming, is this plan for connection flawed? Are there better alternatives?
Thanks!
1) You do not need to send 'ping' data to keep the connection alive, the TCP stack does this automatically; one reason for sending 'ping' data would be to detect a connection close on the client side - typically you only find out something has gone wrong when you try and read/write from the socket. There may be a way to change various time-outs so you can detect this condition faster.
2) In general while TCP provides a stream-oriented error free channel, it makes no guarantees about timeliness, if you are using it on the internet it is even more unpredictable.
3) For applications such as this (I hope you are making it for ethical purposes) - I would tend to use TCP, since you don't want a situation where the client receives a packet to raise an alarm but misses that one that turns it off again.
I'm making a multiplayer game and often i want to test out if it perfectly works on global network, because sometime it's just works locally, so how could i do that without sending my client to friend to test it out.
If you want to test it for the "global network" - you have to test it that way. There are multiple things that can go wrong which are not an issue on a local network. Just off the top of my head
latency (important for a game)
NAT (common cause of problems depending on your game architecture - more so if P2P)
security
connection errors (Wifi/3G intermittent loss)
There are many aspects of networking that are often subtly different when you're talking over the internet rather than running on either localhost or a local network.
All manner of delays can occur, and this can throw out some poor assumptions in your code.
The TCP flow can be affected by TCP's flow control (when the TCP Window fills up the sender will stop sending and report the fact to you (or maybe not report it, if you're using async APIs)).
TCP reads that return 'complete messages' on localhost and your own network may start to return pieces of a message.
UDP datagrams may go missing and never arrive or may arrive multiple times or in any sequence.
So you're right to think that you need to test your code for these edge cases which rarely show up on your own network. You're also right to think that simply sending a client to a friend, or running it on a remote machine, is not enough.
One approach is to build a dedicated test client which sends known game play and checks that it gets expected responses (how hard this is depends on your protocol and your game). Once you have that working you then have the test client deliberately send data in such a way that the items above are tested. So, if you're using UDP, you might put some code in your test client so that it sometimes doesn't bother to send a UDP datagram at all. The client should think it sent it. The networking layer simply ditches it. This tests your UDP protocol for missing datagrams. Then send some datagrams multiple times, then send some out of sequence, etc. For TCP add delays, break logical "messages" into separate network sends with large delays between them; ideally send each distinct message type as a sequence of single bytes to check that the server's 'message accumulation code' works correctly.
Once you have done this you need to do the same for your client code, perhaps by adding a "fuzzing" option to your server's network code to do the same kind of thing...
Personally I tend to try and take a step back and do as much of this as possible in dedicated 'unit tests' (I know that some people will say that these aren't unit tests, call them what you like, just write them!). These tests exercise your networking layer using real networking (talking to a dummy server/client that the test creates) and validate the horrible edge cases.
I am designing and testing a client server program based on TCP sockets(Internet domain). Currently , I am testing it on my local machine and not able to understand the following about SIGPIPE.
*. SIGPIPE appears quite randomly. Can it be deterministic?
The first tests involved single small(25 characters) send operation from client and corresponding receive at server. The same code, on the same machine runs successfully or not(SIGPIPE) totally out of my control. The failure rate is about 45% of times(quite high). So, can I tune the machine in any way to minimize this.
**. The second round of testing was to send 40000 small(25 characters) messages from the client to the server(1MB of total data) and then the server responding with the total size of data it actually received. The client sends data in a tight loop and there is a SINGLE receive call at the server. It works only for a maximum of 1200 bytes of total data sent and again, there are these non deterministic SIGPIPEs, about 70% times now(really bad).
Can some one suggest some improvement in my design(probably it will be at the server). The requirement is that the client shall be able to send over medium to very high amount of data (again about 25 characters each message) after a single socket connection has been made to the server.
I have a feeling that multiple sends against a single receive will always be lossy and very inefficient. Shall we be combining the messages and sending in one send() operation only. Is that the only way to go?
SIGPIPE is sent when you try to write to an unconnected pipe/socket. Installing a handler for the signal will make send() return an error instead.
signal(SIGPIPE, SIG_IGN);
Alternatively, you can disable SIGPIPE for a socket:
int n = 1;
setsockopt(thesocket, SOL_SOCKET, SO_NOSIGPIPE, &n, sizeof(n));
Also, the data amounts you're mentioning are not very high. Likely there's a bug somewhere that causes your connection to close unexpectedly, giving a SIGPIPE.
SIGPIPE is raised because you are attempting to write to a socket that has been closed. This does indicate a probable bug so check your application as to why it is occurring and attempt to fix that first.
Attempting to just mask SIGPIPE is not a good idea because you don't really know where the signal is coming from and you may mask other sources of this error. In multi-threaded environments, signals are a horrible solution.
In the rare cases were you cannot avoid this, you can mask the signal on send. If you set the MSG_NOSIGNAL flag on send()/sendto(), it will prevent SIGPIPE being raised. If you do trigger this error, send() returns -1 and errno will be set to EPIPE. Clean and easy. See man send for details.