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。

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。

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(!))。

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。

每个TCP连接都需要在数据传输之前进行初始握手。此外，TCP报头包含大量用于不同信号和消息传递检测的开销。对于消息交换，如果失败的可能性很小，UDP可能就足够了。如果必须验证收据，TCP是最好的选择。