UDP的“不可靠性”是一种形式主义。传播并不能绝对保证。实际上，他们几乎总是能通过。它们只是在暂停后不被确认和重试。

协商TCP套接字和握手TCP数据包的开销是巨大的。真的很大。没有明显的UDP开销。

最重要的是，您可以轻松地用一些可靠的传输握手来补充UDP，开销比TCP要少。读这个:http://en.wikipedia.org/wiki/Reliable_User_Datagram_Protocol

UDP对于在发布-订阅类型的应用程序中广播信息非常有用。IIRC, TIBCO大量使用UDP来通知状态变化。

任何其他类型的单向“重要事件”或“日志记录”活动都可以用UDP包很好地处理。您希望在不构造整个套接字的情况下发送通知。你不期望从不同的听众那里得到任何回应。

系统“心跳”或“我还活着”消息也是一个不错的选择。错过一个不是危机。(连续)少了半打就是。

当TCP可以工作时，我有点不情愿建议使用UDP。问题是，如果TCP由于某种原因不能工作，因为连接太延迟或拥塞，将应用程序更改为使用UDP不太可能有帮助。一个坏的连接对UDP也不好。TCP在减少拥塞方面已经做得很好了。

我能想到的唯一需要UDP的情况是广播协议。在应用程序涉及两个已知主机的情况下，UDP可能只会提供边际的性能优势，而代码复杂性的成本则会大幅增加。

在某些情况下，保证数据包的到达并不重要，因此使用UDP是可以的。在其他情况下，UDP比TCP更可取。

你想要使用UDP而不是TCP的一个独特情况是你在另一个协议(例如隧道，虚拟网络等)上建立TCP隧道。如果您在TCP上建立隧道，则每个TCP的拥塞控制将相互干扰。因此，人们通常更喜欢在UDP(或其他无状态协议)上传输TCP。参见TechRepublic文章:理解TCP Over TCP: TCP隧道对端到端吞吐量和延迟的影响。

There are already many good answers here, but I would like to add one very important factor as well as a summary. UDP can achieve a much higher throughput with the correct tuning because it does not employ congestion control. Congestion control in TCP is very very important. It controls the rate and throughput of the connection in order to minimize network congestion by trying to estimate the current capacity of the connection. Even when packets are sent over very reliable links, such as in the core network, routers have limited size buffers. These buffers fill up to their capacity and packets are then dropped, and TCP notices this drop through the lack of a received acknowledgement, thereby throttling the speed of the connection to the estimation of the capacity. TCP also employs something called slow start, but the throughput (actually the congestion window) is slowly increased until packets are dropped, and is then lowered and slowly increased again until packets are dropped etc. This causes the TCP throughput to fluctuate. You can see this clearly when you download a large file.

因为UDP没有使用拥塞控制，它可以更快，并且经历更少的延迟，因为它不会寻求最大化缓冲区直到丢弃点，也就是说，UDP数据包在缓冲区中花费的时间更少，到达那里的速度更快，延迟更少。由于UDP不采用拥塞控制，但TCP采用拥塞控制，因此它可以从TCP中占用生成UDP流的容量。

UDP仍然容易受到拥塞和数据包丢失的影响，所以你的应用程序必须准备好以某种方式处理这些问题，可能使用重传或错误纠正代码。

结果是UDP可以:

Achieve higher throughput than TCP as long as the network drop rate is within limits that the application can handle. Deliver packets faster than TCP with less delay. Setup connections faster as there are no initial handshake to setup the connection Transmit multicast packets, whereas TCP have to use multiple connections. Transmit fixed size packets, whereas TCP transmit data in segments. If you transfer a UDP packet of 300 Bytes, you will receive 300 Bytes at the other end. With TCP, you may feed the sending socket 300 Bytes, but the receiver only reads 100 Bytes, and you have to figure out somehow that there are 200 more Bytes on the way. This is important if your application transmit fixed size messages, rather than a stream of bytes.

总之，UDP可以用于TCP可以使用的任何类型的应用程序，只要您还实现了适当的重传输机制。UDP可以非常快，有更少的延迟，在连接的基础上不受拥塞的影响，传输固定大小的数据报，并可用于组播。

我们知道UDP是一种无连接协议，的确如此

适用于需要简单请求-响应通信的流程。 适用于有内部流动、误差控制的工艺 适用于广泛铸造和多播

具体的例子:

用于SNMP 用于RIP等路由更新协议

关于这个问题，我所知道的最好的答案之一来自Hacker News的用户zAy0LfpBZLC8mAC。这个答案太好了，我就原原本本地引用它吧。

TCP has head-of-queue blocking, as it guarantees complete and in-order delivery, so when a packet gets lost in transit, it has to wait for a retransmit of the missing packet, whereas UDP delivers packets to the application as they arrive, including duplicates and without any guarantee that a packet arrives at all or which order they arrive (it really is essentially IP with port numbers and an (optional) payload checksum added), but that is fine for telephony, for example, where it usually simply doesn't matter when a few milliseconds of audio are missing, but delay is very annoying, so you don't bother with retransmits, you just drop any duplicates, sort reordered packets into the right order for a few hundred milliseconds of jitter buffer, and if packets don't show up in time or at all, they are simply skipped, possible interpolated where supported by the codec. Also, a major part of TCP is flow control, to make sure you get as much througput as possible, but without overloading the network (which is kinda redundant, as an overloaded network will drop your packets, which means you'd have to do retransmits, which hurts throughput), UDP doesn't have any of that - which makes sense for applications like telephony, as telephony with a given codec needs a certain amount of bandwidth, you can not "slow it down", and additional bandwidth also doesn't make the call go faster. In addition to realtime/low latency applications, UDP makes sense for really small transactions, such as DNS lookups, simply because it doesn't have the TCP connection establishment and teardown overhead, both in terms of latency and in terms of bandwidth use. If your request is smaller than a typical MTU and the repsonse probably is, too, you can be done in one roundtrip, with no need to keep any state at the server, and flow control als ordering and all that probably isn't particularly useful for such uses either. And then, you can use UDP to build your own TCP replacements, of course, but it's probably not a good idea without some deep understanding of network dynamics, modern TCP algorithms are pretty sophisticated. Also, I guess it should be mentioned that there is more than UDP and TCP, such as SCTP and DCCP. The only problem currently is that the (IPv4) internet is full of NAT gateways which make it impossible to use protocols other than UDP and TCP in end-user applications.