UDP比TCP快，原因很简单，因为它不存在允许连续数据包流的确认数据包(ACK)，而不是使用TCP窗口大小和往返时间(RTT)来确认一组数据包的TCP。

要了解更多信息，我推荐简单但非常容易理解的Skullbox解释(TCP vs. UDP)

如果你需要在两个还没有通话的IP之间快速发送消息，那么UDP到达的速度至少要快3倍，通常是5倍。

如果不考虑网络情况，只讨论TCP或UDP是没有意义的。 如果两点之间的网络质量非常高，UDP绝对比TCP快，但在其他一些情况下，如GPRS网络，TCP可能比UDP更快，更可靠。

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。

请记住，TCP通常在网络上保存多条消息。如果你想在UDP中实现这一点，如果你想可靠地做到这一点，你将有相当多的工作。你的解决方案要么不太可靠，要么速度较慢，要么工作量巨大。有有效的UDP应用程序，但如果你问这个问题，你的可能不是。