具有容错功能

你是说“容忍损失”吗?

基本上，UDP不是“容错”的。你可以发送100个包给某人，他们可能只收到其中的95个包，有些包的顺序可能是错误的。

对于视频流媒体和多人游戏之类的东西，错过一个数据包总比延迟它后面的所有其他数据包要好，这是显而易见的选择

然而，对于大多数其他事情，丢失或“重新排列”的数据包是至关重要的。你必须编写一些额外的代码来运行在UDP之上，以便在遗漏内容时重试，并强制执行正确的顺序。这在某些地方会增加一点开销。

值得庆幸的是，一些非常非常聪明的人已经做到了这一点，他们称之为TCP。

可以这样想:如果一个数据包丢失了，您是希望尽快获得下一个数据包并继续(使用UDP)，还是您实际上需要丢失的数据(使用TCP)。开销并不重要，除非你在一个真正的边缘情况下。

Which protocol performs better (in terms of throughput) - UDP or TCP - really depends on the network characteristics and the network traffic. Robert S. Barnes, for example, points out a scenario where TCP performs better (small-sized writes). Now, consider a scenario in which the network is congested and has both TCP and UDP traffic. Senders in the network that are using TCP, will sense the 'congestion' and cut down on their sending rates. However, UDP doesn't have any congestion avoidance or congestion control mechanisms, and senders using UDP would continue to pump in data at the same rate. Gradually, TCP senders would reduce their sending rates to bare minimum and if UDP senders have enough data to be sent over the network, they would hog up the majority of bandwidth available. So, in such a case, UDP senders will have greater throughput, as they get the bigger pie of the network bandwidth. In fact, this is an active research topic - How to improve TCP throughput in presence of UDP traffic. One way, that I know of, using which TCP applications can improve throughput is by opening multiple TCP connections. That way, even though, each TCP connection's throughput might be limited, the sum total of the throughput of all TCP connections may be greater than the throughput for an application using UDP.

当谈到“什么更快”时，至少有两个非常不同的方面:吞吐量和延迟。

如果说到吞吐量- TCP的流控制(在其他答案中提到)，是非常重要的，在UDP上做任何类似的事情，虽然肯定有可能，但会是一个大头痛(tm)。因此，当你需要吞吐量时使用UDP，很少被认为是一个好主意(除非你想获得比TCP更不公平的优势)。

然而，如果谈论延迟-整个事情是完全不同的。在没有丢包的情况下，TCP和UDP的行为非常相似(任何差异，如果有的话，都是边缘的)——在丢包之后，整个模式发生了巨大的变化。

After any packet loss, TCP will wait for retransmit for at least 200ms (1sec per paragraph 2.4 of RFC6298, but practical modern implementations tend to reduce it to 200ms). Moreover, with TCP, even those packets which did reach destination host - will not be delivered to your app until the missing packet is received (i.e., the whole communication is delayed by ~200ms) - BTW, this effect, known as Head-of-Line Blocking, is inherent to all reliable ordered streams, whether TCP or reliable+ordered UDP. To make things even worse - if the retransmitted packet is also lost, then we'll be speaking about delay of ~600ms (due to so-called exponential backoff, 1st retransmit is 200ms, and second one is 200*2=400ms). If our channel has 1% packet loss (which is not bad by today's standards), and we have a game with 20 updates per second - such 600ms delays will occur on average every 8 minutes. And as 600ms is more than enough to get you killed in a fast-paced game - well, it is pretty bad for gameplay. These effects are exactly why gamedevs often prefer UDP over TCP.

However, when using UDP to reduce latencies - it is important to realize that merely "using UDP" is not sufficient to get substantial latency improvement, it is all about HOW you're using UDP. In particular, while RUDP libraries usually avoid that "exponential backoff" and use shorter retransmit times - if they are used as a "reliable ordered" stream, they still have to suffer from Head-of-Line Blocking (so in case of a double packet loss, instead of that 600ms we'll get about 1.5*2*RTT - or for a pretty good 80ms RTT, it is a ~250ms delay, which is an improvement, but it is still possible to do better). On the other hand, if using techniques discussed in http://gafferongames.com/networked-physics/snapshot-compression/ and/or http://ithare.com/udp-from-mog-perspective/#low-latency-compression , it IS possible to eliminate Head-of-Line blocking entirely (so for a double-packet loss for a game with 20 updates/second, the delay will be 100ms regardless of RTT).

顺便说一句——如果你碰巧只能访问TCP而不能访问UDP(比如在浏览器中，或者你的客户端位于阻止UDP的丑陋防火墙的6-9%之一)——似乎有一种方法可以在不引起太多延迟的情况下实现UDP- in -TCP，请参阅这里:http://ithare.com/almost-zero-additional-latency-udp-over-tcp/(也请确保阅读注释(!))。

根据我的经验，UDP稍微快一点，但也快不了多少。选择不应该基于性能，而应该基于消息内容和压缩技术。

如果它是一种带有消息交换的协议，我建议使用TCP所带来的轻微性能损失是值得的。你得到了两个端点之间的连接它能提供你所需要的一切。不要尝试在UDP之上创建自己可靠的双向协议，除非你对自己的工作非常非常有信心。

网络的设置对于任何测量都是至关重要的。如果您通过本地机器上的套接字或与世界的另一端进行通信，则会产生巨大的差异。

我想在讨论中补充三点:

您可以在这里找到一篇关于TCP与UDP的非常好的文章 游戏开发的背景。 此外，iperf (jperf增强iperf与GUI)是一个 这是一个很好的工具，可以通过测量来回答你的问题。 我用Python实现了一个基准测试(参见这个SO问题)。在平均10^6次迭代中，UDP发送8字节的差异大约是1-2微秒。

我们已经做了一些工作，让程序员可以同时享受这两个世界的好处。

SCTP

它是一个独立的传输层协议，但它可以用作在UDP上提供附加层的库。通信的基本单位是消息(映射到一个或多个UDP包)。有内置的拥塞控制。该协议有许多旋钮和旋钮可以打开

按顺序传递信息 自动重传丢失的消息，与用户定义的参数

如果您的特定应用程序需要其中任何一个。

这样做的一个问题是建立连接是一个复杂的(因此是缓慢的过程)

其他类似的东西

https://en.wikipedia.org/wiki/Reliable_User_Datagram_Protocol

还有一个类似的专利实验

https://en.wikipedia.org/wiki/QUIC

这也试图改进TCP的三重握手，并改变拥塞控制以更好地处理快速线路。

更新2022:Quic和HTTP/3

QUIC(上面提到的)已经通过rfc标准化了，甚至在最初的答案写出来之后就成为了HTTP/3的基础。有各种各样的库，如lucas-clemente/quic-go或microsoft/msquic或谷歌/quiche或mozilla/neqo (web浏览器需要实现这个)。

这些库在UDP传输之上向程序员公开可靠的类tcp流。RFC 9221 (QUIC的不可靠数据报扩展)增加了处理单个不可靠数据包的功能。