I know, I know. This question has been asked many times before. But I've spent an hour googling now without finding what I am looking for so I will ask it again and mention my context along with what makes the decision hard for me:
I am writing the server for a game where the response time is very important and a packet loss every now and then isn't a problem.
Judging by this and the fact that I as a server mostly have to send the same data to many different clients, the obvious answer would be UDP.
I had already started writing the code when I came across this:
In some applications TCP is faster (better throughput) than UDP.
This is the case when doing lots of small writes relative to the MTU size. For example, I read an experiment in which a stream of 300 byte packets was being sent over Ethernet (1500 byte MTU) and TCP was 50% faster than UDP.
In my case the information units I'm sending are <100 bytes, which means each one fits into a single UDP packet (which is quite pleasant for me because I don't have to deal with the fragmentation) and UDP seems much easier to implement for my purpose because I don't have to deal with a huge amount of single connections, but my top priority is to minimize the time between
client sends something to server
and
client receives response from server
So I am willing to pick TCP if that's the faster way.
Unfortunately I couldn't find more information about the above quoted case, which is why I am asking: Which protocol will be faster in my case?
UDP is still going to be better for your use case.
The main problem with TCP and games is what happens when a packet is dropped. In UDP, that's the end of the story; the packet is dropped and life continues exactly as before with the next packet. With TCP, data transfer across the TCP stream will stop until the dropped packet is successfully retransmitted, which means that not only will the receiver not receive the dropped packet on time, but subsequent packets will be delayed also -- most likely they will all be received in a burst immediately after the resend of the dropped packet is completed.
Another feature of TCP that might work against you is its automatic bandwidth control -- i.e. TCP will interpret dropped packets as an indication of network congestion, and will dial back its transmission rate in response; potentially to the point of dialing it down to near zero, in cases where lots of packets are being lost. That might be useful if the cause really was network congestion, but dropped packets can also happen due to transient network errors (e.g. user pulled out his Ethernet cable for a couple of seconds), and you might not want to handle those problems that way; but with TCP you have no choice.
One downside of UDP is that it often takes special handling to get incoming UDP packets through the user's firewall, as firewalls are often configured to block incoming UDP packets by default. For an action game it's probably worth dealing with that issue, though.
Note that it's not a strict either/or option; you can always write your game to work over both TCP and UDP, and either use them simultaneously, or let the program and/or the user decide which one to use. That way if one method isn't working well, you can simply use the other one, and it only takes twice as much effort to implement. :)
In some applications TCP is faster (better throughput) than UDP. This
is the case when doing lots of small writes relative to the MTU size.
For example, I read an experiment in which a stream of 300 byte
packets was being sent over Ethernet (1500 byte MTU) and TCP was 50%
faster than UDP.
If this turns out to be an issue for you, you can obtain the same efficiency gain in your UDP protocol by placing multiple messages together into a single larger UDP packet. i.e. instead of sending 3 100-byte packets, you'd place those 3 100-byte messages together in 1 300-byte packet. (You'd need to make sure the receiving program is able to correctly intepret this larger packet, of course). That's really all that the TCP layer is doing here, anyway; placing as much data into the outgoing packets as it has available and can fit, before sending them out.
Related
how can I transfer large data without splitting. Am using tcp socket. Its for a game. I cant use udp and there might be 1200 values in an array. Am sending array in json format. But the server receiving it like splitted.
Also is there any option to send http request like tcp? I need the response in order. Also it should be faster.
Thanks,
You can't.
HTTP may chunk it
TCP will segment it
IP will packetize it
routers will fragment it ...
and TCP will reassemble it all at the other end.
There isn't a problem here to solve.
You do not have much control over splitting packets/datagrams. The network decides about this.
In the case of IP, you have the DF (don't fragment) flag, but I doubt it will be of much help here. If you are communicating over Ethernet, then 1200 element array may not fit into an Ethernet frame (payload size is up to the MTU of 1500 octets).
Why does your application depend on the fact that the whole data must arrive in a single unit, and not in a single connection (comprised potentially of multiple units)?
how can I transfer large data without splitting.
I'm interpreting the above to be roughly equivalent to "how can I transfer my data across a TCP connection using as few TCP packets as possible". As others have noted, there is no way to guarantee that your data will be placed into a single TCP packet -- but you can do some things to make it more likely. Here are some things I would do:
Keep a single TCP connection open. (HTTP traditionally opens a separate TCP connection for each request, but for low-latency you can't afford to do that. Instead you need to open a single TCP connection, keep it open, and continue sending/receiving data on it for as long as necessary).
Reduce the amount of data you need to send. (i.e. are there things that you are sending that the receiving program already knows? If so, don't send them)
Reduce the number of bytes you need to send. (The easiest way to do this is to zlib-compress your message-data before you send it, and have the receiving program decompress the message after receiving it. This can give you a size-reduction of 50-90%, depending on the content of your data)
Turn off Nagle's algorithm on your TCP socket. That will reduce latency by 200mS and discourage the TCP stack from playing unnecessary games with your data.
Send each data packet with a single send() call (if that means manually copying all of the data items into a separate memory buffer before calling send(), then so be it).
Note that even after you do all of the above, the TCP layer will still sometimes spread your messages across multiple packets, etc -- that's just the way TCP works. And even if your local TCP stack never did that, the receiving computer's TCP stack would still sometimes merge the data from consecutive TCP packets together inside its receive buffer. So the receiving program is always going to "receive it like splitted" sometimes, because TCP is a stream-based protocol and does not maintain message boundaries. (If you want message boundaries, you'll have to do your own framing -- the easiest way is usually to send a fixed-size (e.g. 1, 2, or 4-byte) integer byte-count field before each message, so the receiver knows how many bytes it needs to read in before it has a full message to parse)
Consider the idea that the issue may be else where or that you may be sending too much unnecessary data. In example with PHP there is the isset() function. If you're creating an internet based turn based game you don't (need to send all 1,200 variables back and forth every single time. Just send what changed and when the other player receives that data only change the variables are are set.
I want to transfer a large number of messages. The messages do not need to be reliable. UDP comes to mind as a protocol choice.
Latency is important as well. I do not want to suffer from TCP head-of-line blocking.
I'm concerned I might overload the network when I just start sending messages at maximum speed (e.g. while (messagesRemaining != 0) Send(...);). If I send more than some middle-box can relay then, I think, large numbers of messages might be dropped. Some messages being dropped is fine but most of them should arrive.
How can I address this issue? How can I find out how fast I can send? I want to maintain reasonable packet loss (a few percent) and otherwise maximize bandwidth.
Whether you will overload the network or not depends on what is between the sender and the receiver hosts. The iperf utility has a UDP option that could help you determine the max rate you could send for a certain level of acceptable packet loss.
That said, from personal experience:
If it's local Gigabit network and client/server on same subnet, I highly doubt you would lose any packets. I've done tests before with iperf in this type of environment and never lost any packets; iperf is going to be one of the fastest and most efficient ways to put UDP packets on to the wire from a PC and we still never lost packets. We were even running the packets through an intel Atom-based Linux host with bridged ports setup while doing tcpdump at the same time and still never lost packets (note that even cheapo switches would perform as good or better than a bridge setup in a PC). The only way we were ever able to get packets to be lost was when we used FPGA/ASIC test devices that could put packets onto the wire at true line-speed for long periods of time. Even at that, the test setup was only losing packets when the packets were less than around 500 bytes.
If you aren't on a local network though (i.e. going over internet or routers) you will just need to do some testing with iperf to see what is acceptable for your environment. Problem is though that what rate can be sustained one day isn't guaranteed to be the same the next day. UDP doesn't have any sort of congestion/flow control algorithms like TCP does so you will have to figure out on your own how fast you can send.
I am writing an GUI application which receives UDP packets from a FPGA board of 4Gb data continuously (application is a data retrieval system).
I created my own class inherited from CAyncSocket and on receive message I am reading packets through ReceiveFrom API and writing data to file.
As packets are sent continuously from FPGA (about 400k packets of 1KB data) my application is missing the packets. I am receiving only 200k packets. but when I am monitoring with Wireshark all packets are received.
Can anyone suggest any technique or algorithm to solve this problem, so that I can receive large number of UDP packets without loss.
The first thing to understand and accept is that you cannot guarantee that no UDP packets will be dropped. It is part of the nature of the UDP transport layer that any step in the transmission is allowed to drop a UDP packet for any reason, and that this is something that will happen from time to time. In your case, it sounds like the Windows networking stack is dropping the incoming UDP packets after receiving them from the network card, probably because the incoming-UDP-packets buffer associated with your socket is too full and does not have room to store them. This could happen for example if your write-to-disk calls occasionally take a number of milliseconds to return, during which time your app is unable to read more data from the UDP socket.
That said, there are a few things you can do to make the dropping of packets somewhat less likely.
The first (and easiest) thing to do is to increase the size of your socket's incoming-packets-buffer, using setsockopt(SO_RCVBUF). This helps because the larger the buffer is, the more time your program will have to read packets out of the buffer before the networking stack fills the buffer up entirely and starts dropping packets because it has no place to put them.
If that isn't sufficient for your purposes, the other thing you can do is spawn a separate thread that does nothing but receive incoming UDP packets and add them to a queue (for another thread to process later). Because this thread does nothing else besides receive UDP packets, it will be able to respond quickly when new packets have arrived, and thus the incoming-sockets-buffer will be less likely to ever fill up and overflow. You'll probably want to run this thread at a high priority if possible, so that there is less chance of it being held off of the CPU in the case where other threads or programs are competing for CPU time.
If you've implemented both of the above and the rate of packet loss still isn't acceptable, then you may have to step back and re-evaluate your approach. This might include switching from UDP protocol to TCP, or rewriting your code as an in-kernel driver, or switching to a real-time OS that can make better guarantees about response times.
I'd like to know the general cost of creating a new connection, compared to UDP. I know TCP requires an initial exchange of packets (the 3 way handshake). What would be other costs? For instance is there some sort of magic in the kernel needed for setting up buffers etc?
The reason I'm asking is I can keep an existing connection open and reuse it as needed. However if there is little overhead reconnecting it would reduce complexity.
Once a UDP packet's been dumped onto the wire, the UDP protocol stack is free to completely forget about it. With TCP, there's at bare minimum the connection details (source/dest port and source/dest IP), the sequence number, the window size for the connection etc... It's not a huge amount of data, but adds up quickly on a busy server with many connections.
And then there's the 3-way handshake as well. Some braindead (and/or malicious systems) can abuse the process (look up 'syn flood'), or just drop the connection on their end, leaving your system waiting for a response or close notice that'll never come. The plus side is that with TCP the system will do its best to make sure the packet gets where it has to. With UDP, there's no guarantees at all.
Compared to the latency of the packet exchange, all other costs such as kernel setup times are insignificant.
OPTION 1: The general cost of creating a TCP connection are:
Create socket connection
Send data
Tear down socket connection
Step 1: Requires an exchange of packets, so it's delayed by to & from network latency plus the destination server's service time. No significant CPU usage on either box is involved.
Step 2: Depends on the size of the message.
Step 3: IIRC, just sends a 'closing now' packet, w/ no wait for destination ack, so no latency involved.
OPTION 2: Costs of UDP:*
Create UDP object
Send data
Close UDP object
Step 1: Requires minimal setup, no latency worries, very fast.
Step 2: BE CAREFUL OF SIZE, there is no retransmit in UDP since it doesn't care if the packet was received by anyone or not. I've heard that the larger the message, the greater probability of data being received corrupted, and that a rule of thumb is that you'll lose a certain percentage of messages over 20 MB.
Step 3: Minimal work, minimal time.
OPTION 3: Use ZeroMQ Instead
You're comparing TCP to UDP with a goal of reducing reconnection time. THERE IS A NICE COMPROMISE: ZeroMQ sockets.
ZMQ allows you to set up a publishing socket where you don't care if anyone is listening (like UDP), and have multiple listeners on that socket. This is NOT a UDP socket - it's an alternative to both of these protocols.
See: ZeroMQ.org for details.
It's very high speed and fault tolerant, and is in increasing use in the financial industry for those reasons.
One of our customers is having trouble submitting data from our application (on their PC) to a server (different geographical location). When sending packets under 1100 bytes everything works fine, but above this we see TCP retransmitting the packet every few seconds and getting no response. The packets we are using for testing are about 1400 bytes (but less than 1472). I can send an ICMP ping to www.google.com that is 1472 bytes and get a response (so it's not their router/first few hops).
I found that our application sets the DF flag for these packets, and I believe a router along the way to the server has an MTU less than/equal to 1100 and dropping the packet.
This affects 1 client in 5000, but since everybody's routes will be different this is expected.
The data is a SOAP envelope and we expect a SOAP response back. I can't justify WHY we do it, the code to do this was written by a previous developer.
So... Are there any benefits OR justification to setting the DF flag on TCP packets for application data?
I can think of reasons it is needed for network diagnostics applications but not in our situation (we want the data to get to the endpoint, fragmented or not). One of our sysadmins said that it might have something to do with us using SSL, but as far as I know SSL is like a stream and regardless of fragmentation, as long as the stream is rebuilt at the end, there's no problem.
If there's no good justification I will be changing the behaviour of our application.
Thanks in advance.
The DF flag is typically set on IP packets carrying TCP segments.
This is because a TCP connection can dynamically change its segment size to match the path MTU, and better overall performance is achieved when the TCP segments are each carried in one IP packet.
So TCP packets have the DF flag set, which should cause an ICMP Fragmentation Needed packet to be returned if an intermediate router has to discard a packet because it's too large. The sending TCP will then reduce its estimate of the connection's Path MTU (Maximum Transmission Unit) and re-send in smaller segments. If DF wasn't set, the sending TCP would never know that it was sending segments that are too large. This process is called PMTU-D ("Path MTU Discovery").
If the ICMP Fragmentation Needed packets aren't getting through, then you're dealing with a broken network. Ideally the first step would be to identify the misconfigured device and have it corrected; however, if that doesn't work out then you add a configuration knob to your application that tells it to set the TCP_MAXSEG socket option with setsockopt(). (A typical example of a misconfigured device is a router or firewall that's been configured by an inexperienced network administrator to drop all ICMP, not realising that Fragmentation Needed packets are required by TCP PMTU-D).
The operation of Path-MTU discovery is described in RFC 1191, https://www.rfc-editor.org/rfc/rfc1191.
It is better for TCP to discover the Path-MTU than to have every packet over a certain size fragmented into two pieces (typically one large and one small).
Apparently, some protocols like NFS benefit from avoiding fragmentation (link text). However, you're right in that you typically shouldn't be requesting DF unless you really require it.