If the sender uses sendto() for a few times, and receiver uses recvfrom() in a while loop, when the 2nd datagram arrives before the receiver finish processing the 1st one, will it get lost?
It depends on your architecture,driver.
The packets that arrive at receiver are normally placed in driver's receive queue and processed accordingly like passing to IP layer's queue(based on architecture). If there is any overflow here, then it will be dropped at driver itself or IP layer. However, if has passed those layers but if the receive socket buffer(the amount of space for UDP queued in UDP socket) is full, then the UDP packets will be dropped else it will not be dropped.
netstat is one of the handy tool in such scenarios. netstat -s shall list statistics protocol-wise. The -p option can be used to display for specific protocol.
If you want a proper synchronization between sender and receiver, it shall be feasible only if receiver intimates sender by some means (or some kind of flow control mechanism). UDP protocol does not support it. If you are looking for such synchronization, then you need to select TCP protocol.
Related
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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.
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)
It seems that a UDP packet can be sent without a payload.
The only thing I can think of that doesn't need a payload is for NAT hole punching.
What else could this be used for?
This relates to my previous question Under Linux, can recv ever return 0 on UDP?
I suppose more to the point is that if it's been specified as part of some standard, then it's been thought to be useful somewhere right?
Anything! The UDP packet isn't empty -- it comes with the sender's identity. Therefore, such a packet could be used as a primitive kind of signal: maybe a hello, a goodbye, or a keep-alive.
With interfaces like sendmsg, an empty packet might be used in order to send auxiliary data, like a cmsg structure (which can be used for things like transferring file descriptors between two processes on Linux).
EDIT: One more use: NAT traversal algorithms such as STUN or UDP hole punching.
To answer the question of "why would a protocol do this": the old Daytime protocol just uses the arrival of a UDP packet to send back a reply packet. Similarly, it replies with time value as soon as a TCP connection happens regardless of any actual data that the TCP connection contains.
a UDP packet without payload may be sent to detect if a UDP port is closed. if closed, an ICMP-unreach is replied.
Is it doable to listen a device with 2 different filters and capture packets? For example i start listening a device with a filter and dumping the packets to a pcap file, after 15min can i start another listen on the same device with different filter and dump the packets to another pcap file without stopping the old one?
Does pcap_open or pcap_next_ex block the incoming packets? What i mean if a packet arrives while listening from two different threads one of them will get the packet and control it for filter can the other thread access the packet?
I hope im clear sorry for bad english.
can i start another listen on the same device with different filter and dump the packets to another pcap > file without stopping the old one?
Yes. Albeit, you'd better start that listener in a different thread/process to process it.
Does pcap_open or pcap_next_ex block the incoming packets?
It does not. It drops the packets to your pcap listener if one part (you or the OS) can't keep up with the incoming packets.
pcap will also duplicate the packets(speaking at least of the *nix pcap, assuming winpcap works the same) so if you have several pcap listners, filtering for the same packets , they will all get a copy.
For each open device handle you get, you have a seperate filter, and packet buffer.
say handle 'A' and handle 'B'
now lets say both handles are on the same network device.
Now lets say that the network device gets 4 packets.
each packet gets to the hardware driver, then to winpcap.
at that point winpcap applys each handles filter one at a time.
if a match is made, the packet will be copied to that handles packet buffer.
after all handles have been processed, the packet is handed off to the OS.
Does pcap_open or pcap_next_ex block the incoming packets? No.
The fact is that the operating system, will most likely see the packet before your application will process it.
I may be wrong but I don't think winpcap has any standard method for blocking a packet.
I have an application that transmits some data in a loop.
Underlying protocol is UDP on WinSock. If I don't add sleep(1ms) after each transmit operation most of the data is not sent (or wireshark can not capture it) Have you experienced such a behavour that UDP does not handle repetitive sending in a loop ?
Regards
Tugrul
First thing you should check the return values when you send data to check if data is successfully sent or not.
Second thing, This can happen internal buffer of UDP cannot accommodate more data because previous data is yet not transmitted. So the simplest solution is that each time before send the data you should check if your UDP socket is writable or not. You can do it by calling "select" or "poll" on that UDP socket.