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(!))。

I will just make things clear. TCP/UDP are two cars are that being driven on the road. suppose that traffic signs & obstacles are Errors TCP cares for traffic signs, respects everything around. Slow driving because something may happen to the car. While UDP just drives off, full speed no respect to street signs. Nothing, a mad driver. UDP doesn't have error recovery, If there's an obstacle, it will just collide with it then continue. While TCP makes sure that all packets are sent & received perfectly, No errors , so , the car just passes obstacles without colliding. I hope this is a good example for you to understand, Why UDP is preferred in gaming. Gaming needs speed. TCP is preffered in downloads, or downloaded files may be corrupted.

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.

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。

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)

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(!))。