Most articles are about how to increase a UDP socket's receive buffer size to handle more packets, but I need a solution to decrease the UDP receive buffer to accept only 1 packet at a time and discard/drop all other packets until that packet is read.
I'm trying to do this for Linux, and did some network stack tuning, like setting the RCVBUFF and RCVBUFFFORCE socket options, but that didn't work. I cannot reduce RCVBUFF lower than 2046B (maybe 1 memory page), even when setting the udp_rmem_min to 0.
Why i can’t set UDP RCVBUF lower than 2046?
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Description
I'm busy writing a high frequency UDP server with Go. I'd estimate at least 1000 packets/second both ways.
However as the size of data I'm sending over the UDP socket grew, I eventually ran into the follow error: read udp 127.0.0.1:1541->127.0.0.1:9737: wsarecv: A message sent on a datagram socket was larger than the internal message buffer or some other network limit, or the buffer used to receive a datagram into was smaller than the datagram itself.
I eventually just grew the size of the buffers I was reading from and writing into as follows:
buffer := make([]byte, 64 * 1024 * 1024) // used to just be 1024
l, err := s.socketSim.Read(buffer)
This worked fine and I stopped getting the error... However then I can across two functions inside the net package:
s.socketSim.SetWriteBuffer(64 * 1024 * 1024)
s.socketSim.SetReadBuffer(64 * 1024 * 1024)
I learned that these two act on the operating system's transmit buffer
Question
Do I even care to set the operating system buffer size and why? How does the size on the application buffer impact the size of the operating system buffer? Should they always be the same and how big should/can they become?
First, not only do you have an MTU size for each interface on your device and whatever destination you're send/recving from, but there is also an MTU size for each device in between. For this reason, as others have mentioned, you might want to use what is generally accepted for MTU since you might not control every device in the data route. In the case of UDP, MTU really just means how big a datagram can be before fragmenting.
Second, you almost certainly want your SND/RCV buffers to be larger than the MTU. These are kernel buffers which hold on to data when you're not ready to receive them. A larger UDP RCV buffer means that the kernel will buffer more packets for you instead before dropping them into the abyss. Maybe you have some non-trivial work to do for each packet. Depending on the bitrate, you might want a larger or smaller kernel buffer.
Finally, you're using UDP. There is no guarantee that you'll receive packets in order or at all. Any router in between you and a peer could decide to drop the packet for any reason. Since you're using UDP, you should prepare for dropped and out-of-order packets. You also might need some sort of retransmission mechanism, which further complicates things.
Or you might consider using TCP if dropped packets are unacceptable, knowing that timing is indeterminate.
If you're on linux, you can see current buffer sizes in /proc/sys/net. Usually the kernel will double what you ask for.
Also, you can tune your buffer size by watching for packet drops in /proc/net/udp. If you see drops, you might want to make your rcv buffer bigger, especially if the data is bursty and the processing intensive. If you're data is coming in at a consistent rate and you're still dropping packets, then you aren't processing them fast enough.
Does UDP packet loss percentage might increase considering packet size? For example if I send 100'000 packets, in first try byte[] size is 30, but second 300. Could packet size play role in it's drop ability or packet loss percentage is not its size dependent?
The packet loss is depending on the size of the packet. This has several reasons.
IP packets can go up to 64k approximately, but they are fragmented up to the MTU of ethernet and if one of those packets gets lost , the whole IP packet is dropped. For larger packets if the traffic is high the probability is higher that the larger packet will be dropped. The MTU is around 1500 bytes.
There is more to it than just that. Internally a protocol stack is implemented using internal buffers that are a lot smaller than the mtu, this can vary from 300 bytes and larger. But the point is that these buffers are also a limited resource. If the network device runs out of buffers, then the packet will be dropped as well.
If you don't know the MTU on the network in question according to the link below a 512-byte UDP payload is considered reasonable to allow a margin for other header information that you may not have anticipated.
What is the largest Safe UDP Packet Size on the Internet
Because you're sending larger packets, yes it could increase the chances that packets are dropped.
Now if you compare sending 100000 packets of 30 bytes or 10000 packets of 300 bytes, even though the user data is the same the total size of the packets is larger due to the headers.
I'm experimenting on an LTE connection for checking the maximum rate of bandwidth can be achieved in the uplink.While creating iperf sessions i observed that i'm not able to go beyond 100Kbps in the uplink when the UDP packet size is set as 1400.Apparently when i increased the packet size to 50000 i was able to achieve 2 Mbps in the same link.
Can someone guide me why this performance difference is observed ?When i tried this in a wired channel there i was able to achieve 10Mbps with UDP packet size set as 1400 itself.
What could be the reason for this?
Will trying TCP/IP instead of UDP increase the data throughput?
It probably matters where fragmentation is done - application or IP stack. Your observations show you that IP stack is more efficient.
TCP will be slower. TCP's built-in congestion control will not allow you to send packets until some of already sent have been ACK-ed. That adds round-trip time to performance considerations.
UDP has no such restrictions. It can (ab)use the network to its full potential.
I want to know the maximum size of receive buffer of network layer or TCP/IP layer. Can anyone help on this?
What is the socket type?
If the socket is TCP then I would like to prefer you to set the buffer size to 8K.
For UDP you can also set the buffer size to 8k. It is not actually important for UDP. Because in UDP a whole packet is transmitted at a time. For this reason, you do not need to save much data in the socket for longer period of time.
But in TCP, data comes as a stream. You cannot afford data loss here because it will result in several parsing related issues.
I'm wondering if there are tweaks I can do to a TCP socket, except disabling Nagle, in order to get the lowest possible latency for a client-server protocol with predominantly small packets.
Client packet are mostly smaller than 100 bytes, server packets 100-300 bytes in size.
I'm using java on the server end and (objective-) c on the client side.
You may want to consider reducing delayed ack timeout (if possible). Even though Nagle is turned off, in a situation where you send packets infrequently and packet loss has occurred, delayed ack could cause delay in packet loss detection then retransmission delay.