When a server has only 1 UDP socket, and many clients are sending UDP packets to it, what would be the best approach to handle all of the incoming packets?
I think this can also be a problem with TCP packets, since there's a limited thread count, which cannot cover all client TCP socket receive events.
But things are better in this situation because there's 1 TCP socket per client, and even if the network buffer is full, packet receiving is blocked until the queue has space (let me know if I'm wrong).
UDP packets, however, are discarded when the buffer is full, and there's only 1 socket, so the chances of that happening are higher.
How can I solve this problem? I've searched for a while, but I couldn't get a clear answer. Should I implement my own queueing system? Or just maximize the network buffer size?
There is no way to guarantee you won't drop UDP messages. No matter what you do, if the rate of packets being sent is too large, you will drop some, either on the receiving host or somewhere in the network.
Some things that can help include:
Implementing an internal queue for messages in your Java app, and handing them over to a thread pool to process.
Increasing the kernel's message buffering.
But neither of these can deal with the case where the average message arrival rate is higher that the receiver's ability to process them or the network capacity. This will inevitably lead to lost messages (requests).
I've searched for a while, but I couldn't get a clear answer.
That is because there isn't one! Some problems are fundamentally unsolvable. For others, the best answer depends on factors that are too hard to measure or predict.
(If you want certainty ... don't use networking!)
In the TCP case, what you should do is use a (long-term) socket for each client. Depending on the number of sockets you need to support, you could either:
Dedicate a server-side thread to each socket (and client).
Use java.nio.channels.Selector and a thread pool.
You will still get problems if the rate of requests exceeds your server's ability to process them. However, the TCP connections will ensure that requests are not lost, and that the clients get some "back pressure".
The Wikipedia article on TCP indicates that the IP packets transporting TCP segments can sometimes go lost, and that TCP "requests retransmission of lost data".
What exactly are the rules for requesting retransmission of lost data? At what time frequency are the retransmission requests performed? Is there an upper bound on the number? Is there functionality for the client to indicate to the server to forget about the whole TCP segment for which part went missing when the IP packet went missing?
What exactly are the rules for requesting retransmission of lost data?
The receiver does not request the retransmission. The sender waits for an ACK for the byte-range sent to the client and when not received, resends the packets, after a particular interval.
This is ARQ (Automatic Repeat reQuest). There are several ways in which this is implemented.
Stop-and-wait ARQ
Go-Back-N ARQ
Selective Repeat ARQ
are detailed in the RFC 3366.
At what time frequency are the retransmission requests performed?
The retransmissions-times and the number of attempts isn't enforced by the standard. It is implemented differently by different operating systems, but the methodology is fixed. (One of the ways to fingerprint OSs perhaps?)
The timeouts are measured in terms of the RTT (Round Trip Time) times. But this isn't needed very often due to Fast-retransmit which kicks in when 3 Duplicate ACKs are received.
Is there an upper bound on the number?
Yes there is. After a certain number of retries, the host is considered to be "down" and the sender gives up and tears down the TCP connection.
Is there functionality for the client to indicate to the server to forget about the whole TCP segment for which part went missing when the IP packet went missing?
The whole point is reliable communication. If you wanted the client to forget about some part, you wouldn't be using TCP in the first place. (UDP perhaps?)
There's no fixed time for retransmission. Simple implementations estimate the RTT (round-trip-time) and if no ACK to send data has been received in 2x that time then they re-send.
They then double the wait-time and re-send once more if again there is no reply. Rinse. Repeat.
More sophisticated systems make better estimates of how long it should take for the ACK as well as guesses about exactly which data has been lost.
The bottom-line is that there is no hard-and-fast rule about exactly when to retransmit. It's up to the implementation. All retransmissions are triggered solely by the sender based on lack of response from the receiver.
TCP never drops data so no, there is no way to indicate a server should forget about some segment.
Is TCP not responsible for making sure that a stream is sent intact over the wire by doing whatever may become necessary as losses etc. occur during a transfer?
Does it not do a proper job of it?
Why do higher application-layer protocols and their applications still perform checksums?
While TCP does contain its own checksum, it is only a 16-bit checksum and it is certainly possible for a multi-bit transmission error to slip by the TCP checksum mechanism. This is quite rare, but it is still possible and I have in fact seen it happen (once or twice in a couple of decades).
A robust protocol will want to use a higher-level hash function to assure integrity of transmitted data. Having said that, not many applications that transmit a small amount of data go to this trouble. Bulk transfer applications (such as a package manager or auto-update mechanism) will usually use a cryptographic hash function to increase the assurance of data integrity.
TCP ensures that TCP packets are delivered reliably, using checksums to trap errors introduced during transmission, and retransmitting lost or damaged packets as required. When a packet is transmitted it is retained in a retransmission queue until the peer host acknowledges receipt; if no acknowledgement is received within a certain timeout period then the packet is retransmitted. But the host won't keep retransmitting a packet forever - if a packet repeatedly fails then TCP eventually gives up and closes the connection.
Higher-level protocols assume that TCP works reliably (a fair assumption) and use their own checksums or whatever to check that the higher-level data stream arrived safely. I've written lots of buggy sockets applications that screwed up their own higher-level buffers and mangled the application data stream!
In any production-grade TCP/IP stack with a robust application I think you can be confident that the problem is that your connection is dropping out. Or you might have a buggy application, but I doubt that your fetch/wget is buggy.
Why is data not transferred during the 3rd part of TCP 3-way handshake?
e.g.
(A to B)SYN
(B to A)ACK+SYN
(A to B) ACK.... why cant data be transferred along with this ACK?
I've always believed it was to keep the session establishment phase separate from the data transfer phase so that no real data is transferred until both ends of the session have agreed on the sequence numbers and session options, especially since packets arriving may be from a totally different, previous, session that just happens to have the same endpoints.
However, on further investigation, I'm not entirely certain that transmitting data with the handshake packets is disallowed. The section on TCP connection establishment in my Internetworking with TCP/IP1 book contains the following snippet:
Because of the protocol design, it is possible to send data along with the initial sequence numbers in the handshake segments. In such cases, the TCP software must hold the data until the handshake completes. Once a connection has been established, the TCP software can release data being held and deliver it to a waiting application program quickly.
Since it's certainly possible to construct a TCP packet with SYN (or ACK) and data, this may well be allowed. I've never seen it happen in the wild but, then again, I've never seen a hairy-eared dwarf lemur in the wild either, though I'm assured they exist.
It may be that it's the sockets software that prevents data going out before the session is fully established but TCP appears to consider it valid. It appears you can send data with a SYN-ACK packet (phase 2 of the connection establishment) since you have the other end's sequence number and options. Likewise, sending data with the phase 3 ACK packet appears to be possible as well.
The reason the TCP software holds on to the data until the handshake is fully complete is probably due to the reason mentioned above - only once both ends have agreed on the sequence numbers can you be sure that the data is not from a previous session.
1 Internetworking with TCP/IP Volume 1 Principles, Protocols and Architecture, 3rd edition, Douglas E. Comer, ISBN 0-13-216987-8.
I wanted to know why UDP is used in RTP rather than TCP ?. Major VoIP Tools used only UDP as i hacked some of the VoIP OSS.
As DJ pointed out, TCP is about getting a reliable data stream, and will slow down transmission, and re-transmit corrupted packets, in order to achieve that.
UDP does not care about reliability of the communication, and will not slow down or re-transmit data.
If your application needs a reliable data stream, for example, to retrieve a file from a webserver, you choose TCP.
If your application doesn't care about corrupted or lost packets, and you don't need to incur the additional overhead to provide the additional reliability, you can choose UDP instead.
VOIP is not significantly improved by reliable packet transmission, and in fact, in some cases things in TCP like retransmission and exponential backoff can actually hurt VOIP quality. Therefore, UDP was a better choice.
A lot of good answers have been given, but I'd like to point one thing out explicitly:
Basically a complete data stream is a nice thing to have for real-time audio/video, but its not strictly necessary (as others have pointed out):
The important fact is that some data that arrives too late is worthless. What good is the missing data for a frame that should have been displayed a second ago?
If you were to use TCP (which also guarantees the correct order of all data), then you wouldn't be able to get to the more up-to-date data until the old one is transmitted correctly. This is doubly bad: you have to wait for the re-transmission of the old data and the new data (which is now delayed) will probably be just as worthless.
So RTP does some kind of best-effort transmission in that it tries to transfer all available data in time, but doesn't attempt to re-transmit data that was lost/corrupted during the transfer (*). It just goes on with life and hopes that the more important current data gets there correctly.
(*) actually I don't know the specifics of RTP. Maybe it does try to re-transmit, but if it does then it won't be as aggressive as TCP is (which won't ever accept any lost data).
The others are correct, however the don't really tell you the REAL reason why. Saua kind of hints at it, but here's a more complete answer.
Audio and Video is real-time. If you are listening to a radio, or watching TV, and the signal is interrupted, it doesn't pick up where you left off.. you're just "observing" the signal as it streams, and if you can't observe it at any given time, you lose it.
The reason, is simple. Delay. VOIP tries very hard to minimize the amount of delay from the time someone speaks into one end and you get it on your end, and your response back. Otherwise, as errors occured, the amount of delay between when the person spoke and when the signal was received would continuously grow until it became useless.
Remember, each delay from a retransmission has to be replayed, and that causes further data to be delayed, then another error causes an even greater delay. The only workable solution is to simply drop any data that can't be displayed in real-time.
A 1 second delay from retransmission would mean it would now be 1 second from the time I said something until you heard it. A second 1 second delay now means it's 2 seconds from the time i say something until you hear it. This is cumulative because data is played back at the same rate at which it is spoken, and so on...
RTP could be connection oriented, but then it would have to drop (or skip) data to keep up with retransmission errors anyways, so why bother with the extra overhead?
Technically RTP packets can be interleaved over a TCP connection. There are lots of great answers given here. Two additional minor points:
RFC 4588 describes how one could use retransmission with RTP data. Most clients that receive RTP streams employ a buffer to account for jitter in the network that is typically 1-5 seconds long and which means there is time available for a retransmit to receive the desired data.
RTP traffic can be interleaved over a TCP connection. In practice when this is done, the difference between Interleaved RTP (i.e. over TCP) and RTP sent over UDP is how these two perform over a lossy network with insufficient bandwidth available for the user. The Interleaved TCP stream will end up being jerky as the player continually waits in a buffering state for packets to arrive. Depending on the player it may jump ahead to catch up. With an RTP connection you will get artifacts (smearing/tearing) in the video.
UDP is often used for various types of realtime traffic that doesn't need strict ordering to be useful. This is because TCP enforces an ordering before passing data to an application (by default, you can get around this by setting the URG pointer, but no one seems to ever do this) and that can be highly undesirable in an environment where you'd rather get current realtime data than get old data reliably.
RTP is fairly insensitive to packet loss, so it doesn't require the reliability of TCP.
UDP has less overhead for headers so that one packet can carry more data, so the network bandwidth is utilized more efficiently.
UDP provides fast data transmission also.
So UDP is the obvious choice in cases such as this.
Besides all the others nice and correct answers this article gives a good understanding about the differences between TCP and UDP.
The Real-time Transport Protocol is a network protocol used to deliver streaming audio and video media over the internet, thereby enabling the Voice Over Internet Protocol (VoIP).
RTP is generally used with a signaling protocol, such as SIP, which sets up connections across the network. RTP applications can use the Transmission Control Protocol (TCP), but most use the User Datagram protocol (UDP) instead because UDP allows for faster delivery of data.
UDP is used wherever data is send, that does not need to be exactly received on the target, or where no stable connection is needed.
TCP is used if data needs to be exactly received, bit for bit, no loss of bits.
For Video and Sound streaming, some bits that are lost on the way do not affect the result in a way, that is mentionable, some pixels failing in a picture of a stream, nothing that affects a user, on DVDs the lost bit rate is higher.
just a remark:
Each packet sent in an RTP stream is given a number one higher than its predecessor.This allows thr destination to determine if any packets are missing.
If a packet is mising, the best action for the destination to take is to approximate the missing vaue by interpolation.
Retranmission is not a proctical option since the retransmitted packet would be too late to be useful.
I'd like to add quickly to what Matt H said in response to Stobor's answer. Matt H mentioned that RTP over UDP packets can be checksum'ed so that if they are corrupted, they will get resent. This is actually an optional feature on most PBXs. In Asterisk, for example, you can enable / disable checksums on your RTP over UDP traffic in the rtp.conf configuration file with the following line:
rtpchecksums=yes ; or no if you prefer
Cheers!