因为TCP保证数据包的传递,因此可以被认为是“可靠的”,而UDP不保证任何东西,数据包可能会丢失。在应用程序中使用UDP而不是TCP流传输数据的优势是什么?在什么情况下UDP是更好的选择,为什么?

我假设UDP更快,因为它没有创建和维护流的开销,但如果一些数据从未到达目的地,这不是无关紧要的吗?


当前回答

我们知道UDP是一种无连接协议,的确如此

适用于需要简单请求-响应通信的流程。 适用于有内部流动、误差控制的工艺 适用于广泛铸造和多播

具体的例子:

用于SNMP 用于RIP等路由更新协议

其他回答

视频流是使用UDP的一个很好的例子。

UDP确实有更少的开销,适合做一些事情,比如流式实时数据,如音频或视频,或者在任何情况下,如果数据丢失是ok的。

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.

关键的问题是“在什么情况下UDP是更好的选择[而不是tcp]”

上面有很多很好的答案,但是缺少的是对传输不确定性对TCP性能影响的正式、客观的评估。

随着移动应用程序的大量增长,以及“偶尔连接”或“偶尔断开”的范式,在很难获得连接的情况下,TCP试图维持连接的开销肯定会导致UDP及其“面向消息”的性质的强烈情况。

现在我没有数学/研究/数字,但我制作的应用程序使用ACK/NAK和UDP上的消息编号比使用TCP更可靠,当时连接通常很差,可怜的旧TCP只是花费了时间和客户的金钱来尝试连接。在许多西方国家的地区和农村地区都有这种情况....

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.