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(!))。

在一些应用程序中，TCP比UDP更快(更好的吞吐量)。

当执行大量相对于MTU大小的小写操作时，就会出现这种情况。例如，我读到一个实验，在这个实验中，一个300字节的数据包流通过以太网发送(1500字节MTU)， TCP比UDP快50%。

这是因为TCP将尝试缓冲数据并填充整个网段，从而更有效地利用可用带宽。

另一方面，UDP将数据包立即放到网络上，从而使大量的小包阻塞网络。

你可能不应该使用UDP，除非你有非常具体的原因这样做。特别是因为你可以通过禁用Nagle算法来让TCP和UDP具有相同的延迟(例如，如果你正在传输实时传感器数据，而你不担心网络被大量的小包阻塞)。

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。

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.

人们说TCP给你的主要东西是可靠性。但事实并非如此。TCP提供给您的最重要的东西是拥塞控制:您可以在DSL链路上运行100个TCP连接，所有的连接都以最大速度运行，并且所有的100个连接都将是高效的，因为它们都“感知”到可用带宽。用100个不同的UDP应用程序尝试一下，所有的应用程序都尽可能快地推送数据包，看看事情对你有多好。

在更大的范围内，这种TCP行为可以防止Internet陷入“拥塞崩溃”。

倾向于将应用程序推向UDP的事情:

Group delivery semantics: it's possible to do reliable delivery to a group of people much more efficiently than TCP's point-to-point acknowledgement. Out-of-order delivery: in lots of applications, as long as you get all the data, you don't care what order it arrives in; you can reduce app-level latency by accepting an out-of-order block. Unfriendliness: on a LAN party, you may not care if your web browser functions nicely as long as you're blitting updates to the network as fast as you possibly can.

但即使你关心性能，你可能也不想使用UDP:

现在，您要考虑的是可靠性，您为实现可靠性所做的许多事情最终可能比TCP已经实现的要慢。 现在您对网络不友好，这可能会在共享环境中引起问题。 最重要的是，防火墙会阻止你。

通过将多个TCP连接“集群化”在一起，可以潜在地克服一些TCP性能和延迟问题;iSCSI这样做是为了绕过局域网上的拥塞控制，但是你也可以这样做来创建一个低延迟的“紧急”消息通道(TCP的“紧急”行为完全被破坏)。

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。