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。

UDP是完美的VoIP地址，其中数据包必须发送不考虑其可靠性… 视频聊天是UDP的一个例子(你可以在任何视频聊天期间通过wireshark网络捕获来检查它)。 而且TCP不能与DNS和SNMP协议一起使用。 UDP没有任何开销，而TCP有很多开销

比较TCP和UDP，像UDP这样的无连接协议可以保证速度，但不能保证数据包传输的可靠性。 例如，电子游戏通常不需要可靠的网络，但速度是最重要的，在游戏中使用UDP具有减少网络延迟的优势。

视频流是使用UDP的一个很好的例子。

电子游戏的网络通信几乎都是通过UDP完成的。

速度是最重要的，如果错过更新也无关紧要，因为每次更新都包含玩家所能看到的完整当前状态。

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.