如果TCP数据包丢失，将重新发送。这对于依赖于以特定顺序实时处理数据的应用程序来说并不方便。

例如视频流，特别是VoIP(例如Skype)。然而，在这些情况下，一个掉落的包并不是什么大问题:我们的感官并不完美，所以我们甚至可能不会注意到。这就是为什么这些类型的应用程序使用UDP而不是TCP。

关键的问题是“在什么情况下UDP是更好的选择[而不是tcp]”

上面有很多很好的答案，但是缺少的是对传输不确定性对TCP性能影响的正式、客观的评估。

随着移动应用程序的大量增长，以及“偶尔连接”或“偶尔断开”的范式，在很难获得连接的情况下，TCP试图维持连接的开销肯定会导致UDP及其“面向消息”的性质的强烈情况。

现在我没有数学/研究/数字，但我制作的应用程序使用ACK/NAK和UDP上的消息编号比使用TCP更可靠，当时连接通常很差，可怜的旧TCP只是花费了时间和客户的金钱来尝试连接。在许多西方国家的地区和农村地区都有这种情况....

请参阅Steven的Unix网络编程的22.4节，“何时使用UDP而不是TCP”。

另外，请参阅关于UDP总是比TCP更快的误解的其他SO回答。

史蒂文的话可以总结如下:

使用UDP广播和组播，因为这是你唯一的选择(任何新的应用程序使用组播) 你可以在简单的请求/回复应用中使用UDP，但你需要构建自己的ack、超时和重传 不要使用UDP进行批量数据传输。

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.

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。

视频流是使用UDP的一个很好的例子。