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Seems my question was asked nearly 10 years ago here...
Emulating accept() for UDP (timing-issue in setting up demultiplexed UDP sockets)
...with no clean and scalable solution. I think this could be solved handily by supporting listen() and accept() for UDP, just as connect() is now.
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In a followup to this question...
Can you bind() and connect() both ends of a UDP connection
...is there any mechanism to simultaneously bind() and connect()?
The reason I ask is that a multi-threaded UDP server may wish to move a new "session" to its own descriptor for scalability purposes. The intent is to prevent the listener descriptor from becoming a bottleneck, similar to the rationale behind SO_REUSEPORT.
However, a bind() call with a new descriptor will take over the port from the listener descriptor until the connect() call is made. That provides a window of opportunity, albeit briefly, for ingress datagrams to get delivered to the new descriptor queue.
This window is also a problem for UDP servers wanting to employ DTLS. It's recoverable if the clients retry, but not having to would be preferable.
connect() on UDP does not provide connection demultiplexing.
connect() does two things:
Sets a default address for transmit functions that don't accept a destination address (send(), write(), etc)
Sets a filter on incoming datagrams.
It's important to note that the incoming filter simply discards datagrams that do not match. It does not forward them elsewhere. If there are multiple UDP sockets bound to the same address, some OSes will pick one (maybe random, maybe last created) for each datagram (demultiplexing is totally broken) and some will deliver all datagrams to all of them (demultiplexing succeeds but is incredibly inefficient). Both of these are "the wrong thing". Even an OS that lets you pick between the two behaviors via a socket option is still doing things differently from the way you wanted. The time between bind() and connect() is just the smallest piece of this puzzle of unwanted behavior.
To handle UDP with multiple peers, use a single socket in connectionless mode. To have multiple threads processing received packets in parallel, you can either
call recvfrom on multiple threads which process the data (this works because datagram sockets preserve message boundaries, you'd never do this with a stream socket such as TCP), or
call recvfrom on a single thread, which doesn't do any processing, just queues the message to the thread responsible for processing it.
Even if you had an OS that gave you an option for dispatching incoming UDP based on designated peer addresses (connection emulation), doing that dispatching inside the OS is still not going to be any more efficient than doing it in the server application, and a user-space dispatcher tuned for your traffic patterns is probably going to perform substantially better than a one-size-fits-all dispatcher provided by the OS.
For example, a DNS (DHCP) server is going to transact with a lot of different hosts, nearly all running on port 53 (67-68) at the remote end. So hashing based on the remote port would be useless, you need to hash on the host. Conversely, a cache server supporting a web application server cluster is going to transact with a handful of hosts, and a large number of different ports. Here hashing on remote port will be better.
Do the connection association yourself, don't use socket connection emulation.
The issue you described is the one I encountered some time ago doing TCP-like listen/accept mechanism for UDP.
In my case the solution (which turned out to be bad as I will describe later) was to create one UDP socket to receive any incoming datagrams and when one arrives making this particular socket connected to sender (via recvfrom() with MSG_PEEK and connect()) and returning it to new thread. Moreover, new not connected UDP socket was created for next incoming datagrams. This way the new thread (and dedicated socket) did recv() on the socket and was handling only this particular channel from now on, while the main one was waiting for new datagrams coming from other peers.
Everything had worked well until the incoming datagram rate was higher. The problem was that while the main socket was transitioning to connected state, it was buffering not one but a few more datagrams (coming from many peers) and thus thread created to handle the particular sender was reading in effect a few more datagrams not intended to it.
I could not find solution (e.g. creating new connected socket (instead connecting the main one) and pass the received datagram on main socket to its receive buffer for futher recv()). Eventually, I ended up with N threads, each one having one "listening" socket (with use of SO_REUSEPORT) with datagram scattering done on OS level.
Related
I have a UDP server implemented using the template in the documentation, which can be found here: https://docs.python.org/3/library/asyncio-protocol.html#udp-echo-server-protocol
I would like to know the addr of the client which lost connection. The connection_lost callback only has a single parameter, exc for the exception.
Edit: Following the downvotes I want to highlight that its not a very noob-friendly part of the module naming a callback in the datagram ServerProtocol class 'connection_made'.
The Python API designers need to document this properly.
It looks like connection_made() is called when you create the socket and connect it, which in turn only happens if you specify a non-None Remote_addr.
To understand all that, first you need to understand what connect() does to a UDP socket at the Berkeley Sockets API level:
It conditions the socket so that write() andsend()can be used as well assendto()`, both of which will only send to the connected target address.
It conditions the socket to filter out all datagrams that did not originate at the connect target.
It does not create a wire connection of any kind. Nothing is received by the peer or sent on the wire in any way.
You can connect() a UDP socket multiple times, either to a different address or to null, which completely undoes (1) and (2).
So, I can only imagine that the connection_lost() callback is called when (4) happens, which it isn't in your code.
Whatever it does, if anything, it certainly can't be used to detect when a client disconnects, as there is no such event in UDP.
I have 1 server and several (maybe up to 20) clients. All clients are sending UDP datagram at random time. Each datagram is quite short (about 10B), but I must make sure all the data from each client is received correctly.
If I let all clients send datagram to the same port, and client B sends it datagram at the exact time when the server is receiving data from client A, it seems the server will miss the data from client A.
So what's the correct method to do this job? Do I need to create a listener for each of the 20 clients?
When you bind a UDP socket to a port, the networking stack will allocate a buffer for a finite number of incoming UDP packets for you, so that (assuming you call recv() in a relatively timely manner), no incoming packets should get lost.
If you want see your buffer size in terminal, you can take a look at:
/proc/sys/net/core/rmem_default for recv
and
/proc/sys/net/core/wmem_default for send
I think the default buffer size on Linux is 131071B.
On Linux, you can change the UDP buffer size (e.g. to 26214400) by (as root):
sysctl -w net.core.rmem_max=26214400
You can also make it permanent by adding this line to /etc/sysctl.conf:
net.core.rmem_max=26214400
Since each packet is only 10B, shouldnt be a problem.
If you are still worried about packet loss you could implement a protocol where your client waits for a ACK from the server or it will resend. Many protocols use such a feature, but this is only possible if timing allows it. For example in streaming data it is not useful because there is no time to resend.
or consider using tcp ( if it is an option)
I have a server that at the minute that creates a new thread for each client connecting securely. If I use a thread pool this will mean that I will have a finite number of clients at once. However this means that I can not be listening on ports for all clients.
My idea is to have the client send a UDP packet with some ID linked to there connection so that they can re-establish the connect rather than lock up a thread for 10-60 seconds (server will keep the SSLsockets in memory). Is that a good way to solve the problem? - I don't see any security security vulnerabilities.
The server is java and the client is C++ not that effects the question.
Your question doesn't make sense. If the client wants to reconnect it should just open a new socket. You are positing at least one extra thread to listen to the UDP port and then ... what? It still has to use the thread pool to handle that client, if that is your self-imposed constraint, or else start a new thread, in which case you may as well not have had the thread pool constraint in the first place.
However this means I cannot be listening on ports for all clients.
No it doesn't. It just means that some clients will get delayed service while the thread pool is full, and a very few clients will get connection failure while the backlog queue is full. It doesn't impair your ability to listen for clients at all.
What if the only port you have say TCP/443 (HTTPS)? What if UDP is firewalled (very much possible)? In other words, you should NOT introduce UDP into this picture.
Even in thread-pool scenario, you can still know the difference between multiple clients who connected to the same server port.
Typical solution for this is to create set of sockets you are going to be watching for at once (in one thread) - in C/C++ it is typically done using select()/poll()/epoll(), and in Java you can use java.nio.
This way, if any client(s) have something to say to you as a server, your select loop will instantly notice that, serve these clients and go back to select(), which consumes very little (effectively 0) CPU usage.
This is an example how to do select loop in C and similar example in Java.
On Redhat Linux, I have a multicast listener listening to a very busy multicast data source. It runs perfectly by itself, no packet losses. However, once I start the second instance of the same application with the exactly same settings (same src/dst IP address, sock buffer size, user buffer size, etc.) I started to see very frequent packet losses from both instances. And they lost exact the same packets. If I stop the one of the instances, the remaining one returns to normal without any packet loss.
Initially, I though it is the CPU/kernel load issue, maybe it could not get the packets out of buffer quickly enough. So I did another test. I still keep one instance of the application running. But then started a totally different multicast listener on the same computer but use the second NIC card and listen to a different but even busier multicast source. Both applications run fine without any packet loss.
So it looks like one NIC card is not powerful enough to support two multicast applications, even though they listen to exact the same thing. The possible cause to the packet loss problem might be that, in this scenario, the NIC card driver needs to copy the incoming data to two sock buffers, and this extra copy task is too much for the ether card to handle so it drops packets. Any deeper analysis on this issue and any possible solutions?
Thanks
You are basically finding out that the kernel is inefficient at fan-out of multicast packets. Worst case scenario the code is for every incoming packet allocating two new buffers, the SKB object and packet payload, and copying the NIC buffer twice.
Pick the best case scenario, for every incoming packet a new SKB is allocated but the packet payload is shared between the two sockets with reference counting. Now imagine what happens when two applications, each on their own core and on separate sockets. Every reference to the packet payload is going to cause the memory bus to stall whilst both core caches have to flush and reload, and above that each application is having to kernel context switch back and forth to pass the socket payload. The result is terrible performance.
You aren't the first to encounter such a problem and many vendors have created solutions to it. The basic design is to limit the incoming data to one thread on one core on one socket, then have that thread distribute the data to all other interested threads, preferably using user space code building upon shared memory and lockless data structures.
Examples are TIBCO's Rendezvous and 29 West's Ultra Messaging showing a 660ns IPC bus:
http://www.globenewswire.com/newsroom/news.html?d=194703
I have situation where I have to handle multiple live UDP streams in the server.
I have two options (as I think)
Single Socket :
1) Listen at single port on the server and receive the data from all clients on the same port and create threads for each client to process the data till the client stop sending.
Here only one port is used to receive the data and number of threads used to process the data.
Multiple Sockets :
2) Client will request open port from the server to send the data and the application will send the open port to the client and opens a new thread listening at the port to receive and process the data.Here for each client will have unique port to send the data.
I already implemented a way to know which packet is coming from which client in UDP.
I have 1000+ clients and 60KB data per second I am receiving.
Is there any performance issues using the above methods
or Is here any efficient way to handle this type of task in C ?
Thanks,
Raghu
With that many clients, having one thread per client is very inefficient since lots and lots of context switches must be performed.
Also, the number of ports you can open per IP is limited (port is a 16 bit number).
Therefore "Single Socket" will be far more efficient. But you can also use "Multipe Sokets" with just a single thread using the asynchronous API. If you can identify the client using the package's payload, then there is no need to have a port per client.