I am using SDL and Net2 lib for a client-server application.
The problem I am facing is that I am not receiving all of my TCP packets from my client unless I place a delay before sending each packet from client.
Removing the delay I get only one packet.
A TCP connection is a stream of bytes. Your client could send 20 packets of 5 bytes each, and the server read it as one 100-byte sequence. You'll need to split the data up yourself.
Well you're not guaranteed (in regular sockets) to receive all packets at one time, you may have to call your receive function more than once, to receive all data. This is of course depends on your definition of a "packet" are you receiving all of your data?
+1 erik
Although it is not guaranteed to be reliable, you most likely want to use UDP, not TCP. Net2 handles UDP very well. UDP is actually very reliable. UDP is message oriented. UDP messages tend to get sent quickly and get special treatment by routers (not always a good thing :-). UDP is often used in games.
BTW, if you had asked this question on the SDL mailing list, or sent it to me directly, you would have gotten this advice many months ago.
I wrote Net2 and I hang out on the SDL list. I do not hang out here because this place is an infinite time sink.
Bob Pendleton
Related
I've been using boost asio sockets (UDP and TCP) to handle a custom protocol between my client server program. Its been working great until I discovered that on TCP async_send/async_recieve calls that data can arrived in combined chunks.
For example, if I make two send calls each with it's own packet, they can arrive combined at a single receive call. I wrongly assumed that every send corresponds to a receive, but I'm obviously wrong. It however has worked well for the longest time until I found the issue running the client for a different OS.
So my question is: are there any guarantees to the completeness of the data on arrival for every receive call? (e.g. async_send 128 bytes arrive in multiples of 128 bytes, or how it arrives must always be treated as random, like 1 bytes arrives then 127 bytes is possible)
More specifically, does this mean that:
Data can arrive concatenated or partial for every send call, and I
have to always handle the concatenated/partial data manually
Is this true for both UDP and TCP asio sockets?
I searched around and couldn't find any documentation on this so I was wondering if anyone have any idea.
First its important to understand that boost asio socket receive and sends methods just mean that they ordered the underlying network stack to receive or send data. By network stack this could be the windows socket API.
If you are sending data right to the same computer, via so called loopback addresses, the operating system (if there is any) can just "give" it to the listening i.e. receiving program. Thats the scenario where you would be most lucky to get things in order and always complete for all cases.
However if you want you are addressing another computer or because the operating system is in the mood, you will have different behaviour:
TCP was designed that you will get you data in the order you have send it. But the chunks or packet size if will be sent differs even on the same connection and is a key feature of TCP. Your OS or hardware network adapter might do some send or receive buffering too, before informing you. However things won't get lost.
So in short for TCP: You can make sure the data is complete by waiting for a certain point in your data async_read_until is just there for this case. Data from multiple send calls might be in one receive or many
UDP was designed to have a low latency in contrast to TCP, but without its ordering and completeness guarantees. So when you send a UDP datagram i.e. packet, usually the OS and network adapter will try to send it out ASAP. However on the way to the other computer, the internet might loose it, or hold one packet back until the one you send after the first, so that data you send later, could be received later, while you can also get the sent first, later, or might not. But when you receive a datagram it's complete in it self.
So in short for UDP: Data will arrive in datagram chunks, but some datagrams might be missing, or might arrive in another order than sent. The data from one send might be in one receive, might not, or later
So after some more testing here's what I concluded: the answer is no. Boost Asio sockets does not have magic that can enforce data completeness beyond what the TCP/UDP protocols enforces.
Edit:
So here's more of my research:
For TCP, it acts like a data stream. So packets may arrive partial or combined and is complete. So the user application need to handle deserialization of combined or partial data.
For UDP, because it is a datagram packet, if the packet arrives, it is guaranteed to be independent and complete. So there is no need to handle partial or combined packets.
I'm writing a client that connects to a server
Sometimes the server send dome special packets to the client with the size of 30kb but on client side I get that 30kb in multiple smaller packets (1kb) so my OnReceive callback calls almost 30 times until I get the full packet.
I managed to solve this problem by putting an Sleep(1000) just right before CAsyncSocket::Receive and then I get all 30kb in one single packet but putting Sleep() is a very bad idea.
Is there any configuration that I should do for CAsyncSocket ?
No. There is no way to do this because that's not how TCP sockets work. TCP does not deliver packets, it delivers streams of data.
It may be that your packet will be delivered in one piece because it's convenient to do so or it may split the packets or combine them with other packets.
The only reliable way to do this is for the receiving end to know how many bytes it needs and keep reading until it's got that many. That may mean sending a packet length in front of each packet for example.
I have a C++ application with GUI that runs (on PC 1) just like a network game, and receives data packets from another computer (2) via WiFi (ad-hoc, so it's quite reliable) at fairly regular intervals (like 40ms), once per loop on program (2). I use send/read.
Here is the problem:
- Packets are not always fully sent (but apparently you can simply keep send()ing the remaining data until all is sent, and thats works well)
- More importantly, packets are stacked in the socket during (1)'s loop until the read() occurs, and then there is no way to distinguish packets in the big stream of data, or know if you were already in the middle of a packet.
I tried to fix this with ID headers (you find an ID as first bytes and you know the length of the packet), but I often get lost (unknown ID : we are not at the beginning of the packet) and am forced to ignore all the remaining data.
So my question is:
Why do packets stack? (generally I have 400B of data whereas my packets are <100B long and fps (1) and (2) are not very different)
How can I have a more reliable way to receive actual packets, say, 80% of packets (discarding packet loss, it's not a question of UDP/TCP)?
Would a separate thread for receiving packets work? (on (1), the server)
How do real-time network games to that (including multiple client management)?
Thanks in advance.
(Sorry I do not have the code here, but I tried to be as clear as I could)
Well:
1) UDP transfers MESSAGES, but is unreliable.
2) TCP transfers BYTE STREAMS, and is reliable.
UDP cannot reliably transfer messages. Anything more reliable requires a protocol on top of UDP.
TCP cannot transfer messages unless they are one byte long. Anything more complex requires a protocol on top of TCP.
Why do packets stack? (generally I have 400B of data whereas my packets are <100B long and fps (1) and (2) are not very different)
Because the time to send packets across the net varies, it typically does not make sense to send packets at a high rate, so most networking libraries (e.g. RakNet) will queue up packets and do a send every 10 ms.
In the case of TCP, there is Nagle's algorithm which is a more principled way of doing the same thing. You can turn Nagle's off by setting the NO_DELAY TCP flag.
How can I have a more reliable way to receive actual packets, say, 80% of packets (discarding packet loss, it's not a question of UDP/TCP)?
If you use TCP, you will receive all of the packets and in the right order. The penalty for using TCP is if a packet is dropped, the packets after it wait until that packet can be resent before they are processed. This results in a noticeable delay, so any games that use TCP have sophisticated prediction techniques to hide this delay and other techniques to smoothly "catch up" once the missing packet arrives.
If you use UDP, you can implement a layer on top that gives you reliability but without the ordering if the order of the packets doesn't matter by sending a counter with each packet and having the receiver repeatedly notify the sender of gaps in the counts. You can also implement ordering by doing something similar. Of course, if you enforce both, then you are creating your own TCP layer. See http://www.jenkinssoftware.com/raknet/manual/reliabilitytypes.html for more details.
What you describe is what would happen if you are using TCP without a protocol on top of it to structure your transmitted data. Your idea of using an ID header and packet length is one such protocol. If you send a 4-byte ID followed by a 4-byte length followed by X number of bytes, then the receiver knows that it has to read 4 bytes followed by 4 bytes followed by X bytes to receive a complete packet. It doesn't get much simplier than that. The fact that you are still having problems reading packets with such a simple protocol suggests that your underlying socket reading code is flawed to begin with. Without seeing your actual code, it is difficult to tell you what you are doing wrong.
I have a client application using winsock's sendto() method to send data to a server application with UDP. On my client application I make, say, 5 quick sendto(). On my server application, I wait for, say, 10 secs, and then do a select() and recvfrom(). Will the recvfrom() give me the first packet sent by the client or will it be an arbitrary one (whichever one arrived first)? Will I still be able to get the other 4 data packets or does winsock's UDP framework only buffer one?
Will the recvfrom() give me the first packet sent by the client or
will it be an arbitrary one
Since UDP doesn't handle reordering, you can get any of the messages. You could get less than 4 messages or even more (but that's rare today).
UDP gives no guarantee to the received ordering of packets, so basically, the first packet you recvfrom() might be the first packet you sent, but must not be - that's what TCP is for (which guarantees ordering of received data). You might not receive a part of the packets (or any, for that matter) at all if they were lost in transit.
For the second part: generally, the operating system will buffer a certain amount of packets for you, this depends on the socket buffer set up for UDP sockets - the buffer is specific to every socket, and not shared between them. On Windows, I'm not sure at how to get at the size of the buffer, on Linux, check out "/proc/sys/net/ipv4/udp_mem"; generally, you'll easily fit five UDP packets in there.
With 5 packets of reasonable size, you will probably get all of the packets and you will probably get the first one sent first. But they might be out of order, might not arrive, and might not contain the original data if they do arrive. You have to handle all of that yourself with UDP. (But depending on your application and requirements and the stability of your network, it might not be a real problem; it is certainly possible for certain situations to exist where receiving 99% of the data is perfectly fine).
I don't mean how to connect to a socket. What should I know about UDP programming?
Do I need to worry about bad data in my socket?
I should assume if I send 200bytes I may get 120 and 60 bytes separately?
Should I worry about another connection sending me bad data on the same port?
If data doesnt arrive typically how long may I (typically) not see data for (250ms? 1 second? 1.75sec?)
What do I really need to know?
"i should assume if i send 200bytes i
may get 120 and 60bytes separately?"
When you're sending UDP datagrams your read size will equal your write size. This is because UDP is a datagram protocol, vs TCP's stream protocol. However, you can only write data up to the size of the MTU before the packet could be fragmented or dropped by a router. For general internet use, the safe MTU is 576 bytes including headers.
"i should worry about another
connection sending me bad data on the
same port?"
You don't have a connection, you have a port. You will receive any data sent to that port, regardless of where it's from. It's up to you to determine if it's from the right address.
If data doesnt arrive typically how
long may i (typically) not see data
for (250ms? 1 second? 1.75sec?)
Data can be lost forever, data can be delayed, and data can arrive out of order. If any of those things bother you, use TCP. Writing a reliable protocol on top of UDP is a very non trivial task and there is no reason to do so for almost all applications.
Should I worry about another
connection sending me bad data on the
same port?
Yes you should worry about it. Any application can send data to your open UDP port at any time. One of the big uses of UDP is many to one style communications where you multiplex communications with several peers on a single port using the addressed passed back during the recvfrom to differentiate between peers.
However, if you want to avoid this and only accept packets from a single peer you can actually call connect on your UDP socket. This cause the IP stack to reject packets coming from any host:port combo ( socket ) other than the one you want to talk to.
A second advantage of calling connect on your UDP socket is that in many OS's it gives a significant speed / latency improvement. When you call sendto on an unconnected UDP socket the OS actually temporarily connects the socket, sends your data and then disconnects the socket adding significant overhead.
A third advantage of using connected UDP sockets is it allows you to receive ICMP error messages back to your application, such as routing or host unknown due to a crash. If the UDP socket isn't connected the OS won't know where to deliver ICMP error messages from the network to and will silently discard them, potentially leading to your app hanging while waiting for a response from a crashed host ( or waiting for your select to time out ).
Your packet may not get there.
Your packet may get there twice or even more often.
Your packets may not be in order.
You have a size limitation on your packets imposed by the underlying network layers. The packet size may be quite small (possibly 576 bytes).
None of this says "don't use UDP". However you should be aware of all the above and think about what recovery options you may want to take.
Fragmentation and reassembly happens at the IP level, so you need not worry about that (Wikipedia). (This means that you won't receive split or truncated packets).
UDP packets have a checksum for the data and the header, so receiving bogus data is unlikely, but possible. Lost or duplicate packets are also possible. You should check your data in any case anyway.
There's no congestion control, so you may wish to consider that, if you plan on clogging the tubes with a lot of UDP packets.
UDP is a connectionless protocol. Sending data over UDP can get to the receiver, but can also get lost during transmission. UDP is ideal for things like broadcasting and streaming audio or video (i.e. a dropped packet is never a problem in those situations.) So if you need to ensure your data gets to the other side, stick with TCP.
UDP has less overhead than TCP and is therefore faster. (TCP needs to build a connection first and also checks data packets for data corruption which takes time.)
Fragmented UDP packets (i.e. packets bigger than about half a Kb) will probably be dropped by routers, so split your data into small chuncks before sending it over. (In some cases, the OS can take care of that.) Note that it is allways a packet that might make it, or not. Half packets aren't processed.
Latency over long distances can be quite big. If you want to do retransmission of data, I would go with something like 5 to 10 times the agerage latency time over the current connection. (You can measure the latency by sending and receiving a few packets.)
Hope this helps.
I won't follow suit with the other people who answered this, they all seem to push you toward TCP, and that's not for gaming at all, except maybe for login/chat info. Let's go in order:
Do I need to worry about bad data in my socket?
Yes. Even though UDP contains an extremely simple checksum for routers and such, it is not 100% efficient. You can add your own checksum device, but most of the time UDP is used when reliability is already not an issue, so data that doesn't conform should just be dropped.
I should assume if I send 200bytes I may get 120 and 60 bytes separately?
No, UDP is direct data write and read. However, if the data is too large, some routers will truncate and you lose part of the data permanently. Some have said roughly 576 bytes with header, I personally wouldn't use more than 256 bytes (nice round log2 number).
Should I worry about another connection sending me bad data on the same port?
UDP listens for any data from any computer on a port, so on this sense yes. Also note that UDP is a primitive and a raw format can be used to fake the sender, so you should use some sort of "key" in order for the listener to verify the sender against their IP.
If data doesnt arrive typically how long may I (typically) not see data for (250ms? 1 second? 1.75sec?)
Data sent on UDP is usually disposable, so if you don't receive data, then it can easily be ignored...however, sometimes you want "semi-reliable" but you don't want 'ordered reliable' like TCP uses, 1 second is a good estimate of a drop. You can number your packets on a rotation and write your own ACK communication. When a packet is received, it records the number and sends back a bitfield letting the sender know which packets it received. You can read this unfinished document for more information (although unfinished, it still yields valiable info):
http://gafferongames.com/networking-for-game-programmers/
The big thing to know when attempting to use UDP is:
Your packets might not all make it over the line, which means there is going to be possible data corruption.
If you're working on an application where 100% of the data needs to arrive reliably to provide functionality, use TCP. If you're working on an application where some loss is allowable (streaming media, etc.) then go for UDP but don't expect everything to get from one of the pipe to the other intact.
One way to look at the difference between applications appropriate for UDP vs. TCP is that TCP is good when data delivery is "better late than never", UDP is good when data delivery is "better never than late".
Another aspect is that the stateless, best-effort nature of most UDP-based applications can make scalability a bit easier to achieve. Also note that UDP can be multicast while TCP can't.
In addition to don.neufeld's recommendation to use TCP.
For most applications TCP is easier to implement. If you need to maintain packet boundaries in a TCP stream, a good way is to transmit a two byte header before the data to delimit the messages. The header should contain the message length. At the receiving end just read two bytes and evaluate the value. Then just wait until you have received that many bytes. You then have a complete message and are ready to receive the next 2-byte header.
This gives you some of the benefit of UDP without the hassle of lost data, out-of-order packet arrival etc.
And don't assume that if you send a packet it got there.
If there is a packet size limitation imposed by some router along the way, your UDP packets could be silently truncated to that size.
Two things:
1) You may or may not received what was sent
2) Whatever you receive may not be in the same order it was sent.