We have web service that has thousands of winforms client in as many PCs. The PCs have no connection with DB backend, all access is via the web service. So we decided to develop a central logging server that listens on a UDP port and all the clients sends an xml error log packet (using log4net UDP appender) that gets dumped to a DB table upon received. Since we don't really care if a few error logs are missed and with thousands of client it is fast with a dedicated logging service not loading the main web service.

UDP具有较低的开销，如前所述，它对视频和音频等流媒体很好，最好只是丢失一个数据包，然后尝试重新发送和追赶。

TCP传输没有保证，你只是被告知套接字是否断开，或者数据是否没有到达。否则它会在到达目的地的时候到达目的地。

A big thing that people forget is that udp is packet based, and tcp is bytestream based, there is no guarantee that the "tcp packet" you sent is the packet that shows up on the other end, it can be dissected into as many packets as the routers and stacks desire. So your software has the additional overhead of parsing bytes back into usable chunks of data, that can take a fair amount of overhead. UDP can be out of order so you have to number your packets or use some other mechanism to re-order them if you care to do so. But if you get that udp packet it arrives with all the same bytes in the same order as it left, no changes. So the term udp packet makes sense but tcp packet doesnt necessarily. TCP has its own re-try and ordering mechanism that is hidden from your application, you can re-invent that with UDP to tailor it to your needs.

UDP更容易在两端编写代码，基本上是因为你不需要建立和维护点到点连接。我的问题是在什么情况下你需要TCP开销?如果你走捷径，比如假设接收到的tcp“数据包”是发送的完整数据包，你会更好吗?(如果你费心检查长度/内容，你可能会扔掉两个包)

UDP是一种无连接协议，用于SNMP和DNS等协议，在这些协议中，无序到达的数据包是可以接受的，数据包的即时传输很重要。

在SNMP中使用它是因为网络管理通常必须在网络处于压力状态时进行，即当可靠的、拥塞控制的数据传输难以实现时。

它用于DNS，因为它不涉及连接建立，从而避免了连接建立延迟。

干杯

关于这个问题，我所知道的最好的答案之一来自Hacker News的用户zAy0LfpBZLC8mAC。这个答案太好了，我就原原本本地引用它吧。

TCP has head-of-queue blocking, as it guarantees complete and in-order delivery, so when a packet gets lost in transit, it has to wait for a retransmit of the missing packet, whereas UDP delivers packets to the application as they arrive, including duplicates and without any guarantee that a packet arrives at all or which order they arrive (it really is essentially IP with port numbers and an (optional) payload checksum added), but that is fine for telephony, for example, where it usually simply doesn't matter when a few milliseconds of audio are missing, but delay is very annoying, so you don't bother with retransmits, you just drop any duplicates, sort reordered packets into the right order for a few hundred milliseconds of jitter buffer, and if packets don't show up in time or at all, they are simply skipped, possible interpolated where supported by the codec. Also, a major part of TCP is flow control, to make sure you get as much througput as possible, but without overloading the network (which is kinda redundant, as an overloaded network will drop your packets, which means you'd have to do retransmits, which hurts throughput), UDP doesn't have any of that - which makes sense for applications like telephony, as telephony with a given codec needs a certain amount of bandwidth, you can not "slow it down", and additional bandwidth also doesn't make the call go faster. In addition to realtime/low latency applications, UDP makes sense for really small transactions, such as DNS lookups, simply because it doesn't have the TCP connection establishment and teardown overhead, both in terms of latency and in terms of bandwidth use. If your request is smaller than a typical MTU and the repsonse probably is, too, you can be done in one roundtrip, with no need to keep any state at the server, and flow control als ordering and all that probably isn't particularly useful for such uses either. And then, you can use UDP to build your own TCP replacements, of course, but it's probably not a good idea without some deep understanding of network dynamics, modern TCP algorithms are pretty sophisticated. Also, I guess it should be mentioned that there is more than UDP and TCP, such as SCTP and DCCP. The only problem currently is that the (IPv4) internet is full of NAT gateways which make it impossible to use protocols other than UDP and TCP in end-user applications.

这并不总是明确的。然而，如果您需要保证数据包以正确的顺序无丢失地传递，那么TCP可能是您想要的。

另一方面，UDP适用于传输信息的短数据包，其中信息的顺序不太重要，或者数据可以放入单个数据包中 包。

当您希望向许多用户广播相同的信息时，这种方法也很合适。

其他时候，当您正在发送序列数据时，它是合适的，但如果有些数据丢失了 错过你不太关心的(例如VOIP应用程序)。

有些协议更复杂，因为需要的是TCP的一些(但不是全部)功能，但比UDP提供的功能更多。这就是应用层必须做到的 实现附加功能。在这些情况下，UDP也是合适的(例如，互联网广播，顺序很重要，但不是每个数据包都需要通过)。

它在哪里/可以被使用的例子 1)时间服务器向局域网上的一堆机器广播正确的时间。 2) VOIP协议 3) DNS查找 4)请求局域网服务，例如:where are you? 5)网络电台 还有许多其他的……

在unix上，您可以输入grep udp /etc/services以获得实现的udp协议列表 今天……有几百个。

这是我最喜欢的问题之一。UDP被误解了。

当你真的想快速地向另一个服务器得到一个简单的答案时，UDP是最好的选择。通常，您希望答案在一个响应包中，并准备实现自己的协议以提高可靠性或重新发送。DNS是这个用例的完美描述。连接设置的成本太高了(然而，DNS 不支持TCP模式以及)。

另一种情况是，当您交付的数据可能会丢失，因为新的数据将取代之前的数据/状态。天气数据、视频流、股票报价服务(不用于实际交易)或游戏数据浮现在脑海中。

另一种情况是，当您正在管理大量的状态时，您希望避免使用TCP，因为操作系统无法处理那么多会话。这在今天是一个罕见的案例。事实上，现在可以使用用户专用的TCP堆栈，以便应用程序编写人员可以对该TCP状态所需的资源进行更细粒度的控制。在2003年之前，UDP是唯一的游戏。

另一种情况是多播流量。UDP可以多播到多个主机，而TCP根本不能这样做。