Network | TCP

Transmission Control Protocol, TCP是一种面向连接的、可靠的、基于字节流的传输层通信协议.

应用层向TCP层发送用于网间传输的、用8位字节表示的数据流,然后TCP把数据流分区成适当长度的报文段(通常受该计算机连接的网络的数据链路层的最大传输单元(MTU:Maximum Transmission Unit)的限制)。之后TCP把结果包传给IP层,由它来通过网络将包传送给接收端实体的TCP层。TCP为了保证不发生丢包,就给每个包一个序号,同时序号也保证了传送到接收端实体的包的按序接收。然后接收端实体对已成功收到的包发回一个相应的确认(ACK);如果发送端实体在合理的往返时延(RTT:Round-Trip Time)内未收到确认,那么对应的数据包就被假设为已丢失将会被进行重传。TCP用一个校验和函数来检验数据是否有错误;在发送和接收时都要计算校验和。

TCP连接包括三个状态:连接创建、数据传送和连接终止。

MSS

The maximum segment size (MSS) is the largest amount of data, specified in bytes, that TCP is willing to receive in a single segment. For best performance, the MSS should be set small enough to avoid IP fragmentation, which can lead to packet loss and excessive retransmissions. To try to accomplish this, typically the MSS is announced by each side using the MSS option when the TCP connection is established, in which case it is derived from the maximum transmission unit (MTU) size of the data link layer of the networks to which the sender and receiver are directly attached. 

MSS announcement is also often called "MSS negotiation". Strictly speaking, the MSS is not "negotiated" between the originator and the receiver, because that would imply that both originator and receiver will negotiate and agree upon a single, unified MSS that applies to all communication in both directions of the connection. In fact, two completely independent values of MSS are permitted for the two directions of data flow in a TCP connection.

Connection establishment连接创建

TCP用三路握手(three-way handshake)过程创建一个连接。在连接创建过程中,很多参数要被初始化,例如序号被初始化以保证按序传输和连接的强壮性。

一对终端同时初始化一个它们之间的连接是可能的。但通常是由一端打开一个套接字(socket)然后监听来自另一方的连接,这就是通常所指的被动打开(passive open)。服务器端被被动打开以后,用户端就能开始创建主动打开(active open)。

  1. SYN: The active open is performed by the client sending a SYN to the server. The client sets the segment's sequence number to a random value A.
  2. SYN-ACK: In response, the server replies with a SYN-ACK. The acknowledgment number is set to one more than the received sequence number i.e. A+1, and the sequence number that the server chooses for the packet is another random number, B.
  3. ACK: Finally, the client sends an ACK back to the server. The sequence number is set to the received acknowledgement value i.e. A+1, and the acknowledgement number is set to one more than the received sequence number i.e. B+1.

Network | TCP_第1张图片

Data transfer数据传输

在TCP的数据传送状态,很多重要的机制保证了TCP的可靠性和强壮性。它们包括:使用序号,对收到的TCP报文段进行排序以及检测重复的数据;使用校验和来检测报文段的错误;使用确认和计时器来检测和纠正丢包或延时。

在TCP的连接创建状态,两个主机的TCP层间要交换初始序号(ISN:initial sequence number)。这些序号用于标识字节流中的数据,并且还是对应用层的数据字节进行记数的整数。通常在每个TCP报文段中都有一对序号和确认号。TCP报文发送者认为自己的字节编号为序号,而认为接收者的字节编号为确认号。TCP报文的接收者为了确保可靠性,在接收到一定数量的连续字节流后才发送确认。这是对TCP的一种扩展,通常称为选择确认(Selective Acknowledgement)。选择确认使得TCP接收者可以对乱序到达的数据块进行确认。每一个字节传输过后,ISN号都会递增1。

通过使用序号和确认号,TCP层可以把收到的报文段中的字节按正确的顺序交付给应用层。序号是32位的无符号数,在它增大到232-1时,便会回绕到0。对于ISN的选择是TCP中关键的一个操作,它可以确保强壮性和安全性。

Network | TCP_第2张图片

TCP数据传输不同于UDP之处

  1. Ordered data transfer — the destination host rearranges according to sequence number
  2. Retransmission of lost packets — any cumulative stream not acknowledged is retransmitted
  3. Error-free data transfer
  4. Flow control — limits the rate a sender transfers data to guarantee reliable delivery. The receiver continually hints the sender on how much data can be received (controlled by the sliding window). When the receiving host's buffer fills, the next acknowledgment contains a 0 in the window size, to stop transfer and allow the data in the buffer to be processed.
  5. Congestion control

Connection termination通路的终结

连接终止使用了四路握手过程(four-way handshake),在这个过程中每个终端的连接都能独立地被终止。因此,一个典型的拆接过程需要每个终端都提供一对FIN和ACK。

Network | TCP_第3张图片

由于TCP连接是全双工的,因此每个方向都必须单独进行关闭。原则是主动关闭的一方(如已传输完所有数据等原因)发送一个FIN报文来表示终止这个方向的连接,收到一个FIN意味着这个方向不再有数据流动,但另一个方向仍能继续发送数据,直到另一个方向也发送FIN报文。四次挥手的具体过程如下:
客户端发送一个FIN报文给服务器,表示我将关闭客户端到服务器端这个方向的连接。
服务器收到后,发送一个ACK报文给客户端,序号为FIN报文的序号加1。
服务器发送一个FIN报文给客户端,表示自己也将关闭服务器端到客户端这个方向的连接。
客户端收到报文后,发回一个ACK报文给服务器,序号为服务器FIN报文的序号加1。
至此,一个TCP连接就关闭了。

TCP连接状态

下面是每一个TCP连接在任意时刻可能处于的状态,在Linux下可以在netstat命令的最后一列(State列)里看到。
各个状态的含义如下:
CLOSED :初始状态,表示TCP连接是“关闭着的”或“未打开的”。
LISTEN :表示服务器端的某个SOCKET处于监听状态,可以接受客户端的连接。
SYN_RCVD :表示接收到了SYN报文。在正常情况下,这个状态是服务器端的SOCKET在建立TCP连接时的三次握手会话过程中的一个中间状态,很短暂,基本上用netstat很难看到这种状态,除非故意写一个监测程序,将三次TCP握手过程中最后一个ACK报文不予发送。当TCP连接处于此状态时,再收到客户端的ACK报文,它就会进入到ESTABLISHED 状态。
SYN_SENT :这个状态与SYN_RCVD 状态相呼应,当客户端SOCKET执行connect()进行连接时,它首先发送SYN报文,然后随即进入到SYN_SENT 状态,并等待服务端的发送三次握手中的第2个报文。SYN_SENT 状态表示客户端已发送SYN报文。
ESTABLISHED :表示TCP连接已经成功建立。
FIN_WAIT_1 :这个状态得好好解释一下,其实FIN_WAIT_1 和FIN_WAIT_2 两种状态的真正含义都是表示等待对方的FIN报文。而这两种状态的区别是:FIN_WAIT_1状态实际上是当SOCKET在ESTABLISHED状态时,它想主动关闭连接,向对方发送了FIN报文,此时该SOCKET进入到FIN_WAIT_1 状态。而当对方回应ACK报文后,则进入到FIN_WAIT_2 状态。当然在实际的正常情况下,无论对方处于任何种情况下,都应该马上回应ACK报文,所以FIN_WAIT_1 状态一般是比较难见到的,而FIN_WAIT_2 状态有时仍可以用netstat看到。
FIN_WAIT_2 :上面已经解释了这种状态的由来,实际上FIN_WAIT_2状态下的SOCKET表示半连接,即有一方调用close()主动要求关闭连接。注意:FIN_WAIT_2 是没有超时的(不像TIME_WAIT 状态),这种状态下如果对方不关闭(不配合完成4次挥手过程),那这个 FIN_WAIT_2 状态将一直保持到系统重启,越来越多的FIN_WAIT_2 状态会导致内核crash。
TIME_WAIT :表示收到了对方的FIN报文,并发送出了ACK报文。 TIME_WAIT状态下的TCP连接会等待2*MSL(Max Segment Lifetime,最大分段生存期,指一个TCP报文在Internet上的最长生存时间。每个具体的TCP协议实现都必须选择一个确定的MSL值,RFC 1122建议是2分钟,但BSD传统实现采用了30秒,Linux可以cat /proc/sys/net/ipv4/tcp_fin_timeout看到本机的这个值),然后即可回到CLOSED 可用状态了。如果FIN_WAIT_1状态下,收到了对方同时带FIN标志和ACK标志的报文时,可以直接进入到TIME_WAIT状态,而无须经过FIN_WAIT_2状态。
CLOSING :这种状态在实际情况中应该很少见,属于一种比较罕见的例外状态。正常情况下,当一方发送FIN报文后,按理来说是应该先收到(或同时收到)对方的ACK报文,再收到对方的FIN报文。但是CLOSING 状态表示一方发送FIN报文后,并没有收到对方的ACK报文,反而却也收到了对方的FIN报文。什么情况下会出现此种情况呢?那就是当双方几乎在同时close()一个SOCKET的话,就出现了双方同时发送FIN报文的情况,这是就会出现CLOSING 状态,表示双方都正在关闭SOCKET连接。
CLOSE_WAIT :表示正在等待关闭。怎么理解呢?当对方close()一个SOCKET后发送FIN报文给自己,你的系统毫无疑问地将会回应一个ACK报文给对方,此时TCP连接则进入到CLOSE_WAIT状态。接下来呢,你需要检查自己是否还有数据要发送给对方,如果没有的话,那你也就可以close()这个SOCKET并发送FIN报文给对方,即关闭自己到对方这个方向的连接。有数据的话则看程序的策略,继续发送或丢弃。简单地说,当你处于CLOSE_WAIT 状态下,需要完成的事情是等待你去关闭连接。
LAST_ACK :当被动关闭的一方在发送FIN报文后,等待对方的ACK报文的时候,就处于LAST_ACK 状态。当收到对方的ACK报文后,也就可以进入到CLOSED 可用状态了。

Network | TCP_第4张图片

Network | TCP_第5张图片

Network | TCP_第6张图片

端口

TCP使用了端口号(Port number)的概念来标识发送方和接收方的应用层。对每个TCP连接的一端都有一个相关的16位的无符号端口号分配给它们。

Port numbers are categorized into three basic categories: well-known, registered, and dynamic/private. The well-known ports are assigned by the Internet Assigned Numbers Authority (IANA) and are typically used by system-level or root processes. Well-known applications running as servers and passively listening for connections typically use these ports. Some examples include: FTP (20 and 21), SSH (22), TELNET (23), SMTP (25), SSL (443) and HTTP (80). Registered ports are typically used by end user applications as ephemeral source ports when contacting servers, but they can also identify named services that have been registered by a third party. Dynamic/private ports can also be used by end user applications, but are less commonly so. Dynamic/private ports do not contain any meaning outside of any particular TCP connection.

Flow control流量控制

TCP uses an end-to-end flow control protocol to avoid having the sender send data too fast for the TCP receiver to receive and process it reliably. Having a mechanism for flow control is essential in an environment where machines of diverse network speeds communicate. For example, if a PC sends data to a smartphone that is slowly processing received data, the smartphone must regulate the data flow so as not to be overwhelmed.

TCP uses a sliding window flow control protocol. In each TCP segment, the receiver specifies in the receive window field the amount of additionally received data (in bytes) that it is willing to buffer for the connection. The sending host can send only up to that amount of data before it must wait for an acknowledgment and window update from the receiving host.

If a receiver is processing incoming data in small increments, it may repeatedly advertise a small receive window. This is referred to as the silly window syndrome, since it is inefficient to send only a few bytes of data in a TCP segment, given the relatively large overhead of the TCP header.

Congestion control拥塞控制

Modern implementations of TCP contain four intertwined algorithms: Slow-start, congestion avoidance, fast retransmit, and fast recovery.

总共只有两种模式:Slow-start, congestion avoidance.

Basic slow-start

The algorithm begins in the exponential growth phase initially with a Congestion Window Size (CWND) of 1, 2 or 10 segments and increases it by one Segment Size (SS) for each new ACK received. If the receiver sends an ACK for every segment, this behavior effectively doubles the window size each round trip of the network. If the receiver supports delayed ACKs, the rate of increase is lower, but still increases by a minimum of one MSS each round-trip time. This behavior continues until the congestion window size (CWND) reaches the size of the receiver's advertised window or until a loss occurs.

When a loss occurs, half of the current CWND is saved as a Slow Start Threshold (SSThresh) and slow start begins again from its initial CWND. Once the CWND reaches the SSThresh, TCP goes into congestion avoidance mode where each new ACK increases the CWND by SS × SS / CWND. This results in a linear increase of the CWND.

慢启动->loss occur->set ssthresh -> 慢启动->congestion avoidance,线性增

通过half threshold来实现乘性减。

Fast recovery

There is a variation to the slow-start algorithm known as Fast Recovery, which uses fast retransmit followed by Congestion Avoidance. In the Fast Recovery algorithm, during Congestion Avoidance mode, when packets (detected through 3 duplicate ACKs) are not received, the congestion window size is reduced to the slow-start threshold, rather than the smaller initial value.

Fast Recovery也是一种慢启动->loss occur->set ssthresh

这个快一点,就是直接half。

congestion avoidance

When the congestion window exceeds SSThresh the algorithm enters a new state, called congestion avoidance.

Transmission Control Protocol (TCP) uses a network congestion-avoidance algorithm that includes various aspects of an additive increase/multiplicative decrease (AIMD) scheme, with other schemes such as slow-start to achieve congestion avoidance.

AIMD有许多变种实现。

As long as non-duplicate ACKs are received, the congestion window is additively increased by one MSS every round trip time. When a packet is lost, the likelihood of duplicate ACKs being received is very high (it's possible though unlikely that the stream just underwent extreme packet reordering, which would also prompt duplicate ACKs). The behavior of Tahoe and Reno differ in how they detect and react to packet loss:

Tahoe: Triple duplicate ACKS are treated the same as a timeout. Tahoe will perform "fast retransmit", set the slow start threshold to half the current congestion window, reduce congestion window to 1 MSS, and reset to slow-start state. (同Basic slow-start)
Reno: If three duplicate ACKs are received (i.e., four ACKs acknowledging the same packet, which are not piggybacked on data, and do not change the receiver's advertised window), Reno will halve the congestion window (instead of setting it to 1 MSS like Tahoe), set the slow start threshold equal to the new congestion window, perform a fast retransmit, and enter a phase called Fast Recovery. If an ACK times out, slow start is used as it is with Tahoe.
Fast Recovery. (Reno Only) In this state, TCP retransmits the missing packet that was signaled by three duplicate ACKs, and waits for an acknowledgment of the entire transmit window before returning to congestion avoidance. If there is no acknowledgment, TCP Reno experiences a timeout and enters the slow-start state.

Both algorithms reduce congestion window to 1 MSS on a timeout event.

这两种方式的区别在于怎么处理loss。slow start是一种状态,fast recovery是Reno在处理loss时的策略。

 MSS与MTU之间存在什么关系?

MTU最大传输单元,以太网传输电气方面的限制,每个以太网帧都有最小的大小64bytes最大不能超过1518bytes,刨去以太网帧的帧头(DMAC目的MAC地址48bit=6Bytes+SMAC源MAC地址48bit=6Bytes+Type域2bytes)14Bytes和帧尾CRC校验部分4Bytes(这个部门有时候大家也把它叫做FCS),那么剩下承载上层协议的地方也就是Data域最大就只能有1500Bytes这个值我们就把它称之为MTU。

MSS最大传输大小的缩写,是TCP协议里面的一个概念。MSS就是TCP数据包每次能够传输的最大数据分段。为了达到最佳的传输效能TCP协议在建立连接的时候通常要协商双方的MSS值,这个值TCP协议在实现的时候往往用MTU值代替(需要减去IP数据包包头的大小20Bytes和TCP数据段的包头20Bytes)所以往往MSS为1460。通讯双方会根据双方提供的MSS值得最小值确定为这次连接的最大MSS值。

 

以上。

部分摘自:http://blog.163.com/pandalove@126/blog/static/98003245201221441436687/

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