【计算机网络第六版(谢希仁)】网络要点总结

一、网络分层

               OSI 七层模型、TCP/IP 四层模型、五层模型图:

     OSI 七层模型:

          1、 物理层:主要定义物理标准,如网线的接口类型、光纤的接口类型、各种介质的传输速率等。它的主要作用是传输比特流(就是由1、0转化为电流强弱来进行传输,

到大目的地后在转化为 1、0,也就是我们常说的数模转换与模数转换)。这一层的数据叫比特。

          2、数据链路层:定义了如何让格式化数据以数据帧为单位进行传输,以及如何让控制对物理介质的访问。这一层通常还提供错误检测和纠正,以确保数据的可靠传输。

(即 封装成帧、透明传输、差错检测)如:串口通信中使用到的 115200、8、N、1。

          3、网络层:在位于不同地理位置的网络中的两个主机系统之间提供连接和路径选择。Internet 的发展使得从世界各个站点访问信息的用户数大大增加,而网络层正是管理

这种连接的层。即 网络层负责数据包从源到宿的传递和网际互连(包PackeT)

           4、传输层:定义了一些传输数据的协议和端口(www 端口 80 等),如:TCP(传输控制协议,传输效率低,可靠性强,用于传输可靠性要求高,数据量大的数据),

UDP(用户数据报协议,与TCP特性恰恰相反,用于传输可靠性要求高,数据量小的数据,如QQ聊天就是通过这种方式传输的)。主要是将从下层接收的数据进行分段和传

输,到大目的地址后再进行重组。常常把这一层数据叫做段。传输层,提供端到端的可靠报文传递和错误恢复(段Segment)

           5、会话层:通过传输层(端口号:传输端口与接收端口)建立数据传输的通路。主要在你的系统之间发起会话或者接收会话请求。(设备之间需要相互认识可以是 IP 也

可以是 MAC 或者是主机名)。即,会话层:建立、管理和终止会话(会话协议数据单元SPDU)

           6、表示层:可确保一个系统的应用层所发送的信息可以被另一个系统的应用层读取。例如,PC 程序员与另一台计算机进行通信,其中一台计算机使用扩展二一十进制

交换码(EBCDIC),而另一台则使用美国信息交换标准码(ASCII)来表示相同的字符。如有必要,表示层会通过使用一种通格式来实现多种数据格式之间的转换。即,表示

:对数据进行翻译、加密和压缩(表示协议数据单元PPDU)

          7、应用层:是最靠近用户的OSI层。这一层为用户的应用程序(例如电子邮件、文件传输和终端仿真)提供网络服务。

      

     TCP/IP 四层模型:

    

二、典型协议

               传输层:         常见的协议有  TCP/UDP 协议

               应用层:         常见的协议有 HTTP,FTP 协议

               网络层:         常见的协议有 IP 协议,ICMP 协议,IGMP 协议

               网络接口层:  常见的协议有 ARP 协议,RARP 协议

               TCP      传输控协议(TransmissionControl Protocol)是一种面向连接的,可靠的,基于字节流的传输层通信协议

               UDP      用户数据包协议(UserDatagram Protocol)是 OSI 参考模型中一种无连接的传输层协议,提供面向事物的简单不可靠信息传送服务

               HTTP    超文本传输系协议(HyperText Transfer Protocol)是互联网上应用最为广泛的一种协议

               FTP       文件传输协议(File Transfer Protocol)

               IP          协议是英特网互联协议(Internet Protocol)

               ICMP     协议是 Internet 控制报文协议(Internet Control Message Protocol),它是 TCP/IP 协议族的一个子协议,用于在 IP 主机、路由器之间传递控制消息

               IGMP     协议是 Internet 网际组管理协议,是英特网协议家族中的一个组播协议。该协议运行在主机和组播路由之间

               ARP       协议是正向地址解析协议(Address Resolution Protocol),通过已知的 IP,寻找对应主机的 MAC 地址

               RARP    协议是方向地址解析协议,通过 MAC 地址确定 IP地址

               TFTP      协议是 TCP/IP 协议族中的一个用来在客户机和服务器之间进行简单文件传输的协议,提供不复杂、开销不大

的文件传输协议

               DHCP     协议,动态主机配置协议,是一种让系统得以连接到网络上的,并获取 所需要的配置参数的手段

               NAT       协议,网络地址转换属接入广域网(WAN),是将一种私有(保留)地址转换为合法IP地址的转换技术

三、TCP 和 UDP对应的协议

TCP 对应的协议:

(1)、FTP:定义了文件传输协议,使用 21 端口。

(2)、Telnet:远程登录协议。使用 23 号端口,用户可以以自己的身份远程登录到计算机上,可提供基于DOS模式下的通信服务。

(3)、SMTP:邮件传送协议,用于发送邮件。服务器开放的端口是 25 号端口。

(4)、POP3 :它和 SMTP 对应,POP3用于接收协议。所用的端口是 110。

(5)、HTTP:是从 Web 服务器传送超文本到本地浏览器的传送协议。

UDP对应的协议:

(1)、DNS:用于域名解析服务,将域名地址转换为 IP 地址。DNS 用的是 53号端口。

(2)、SNMP:简单网络管理协议,使用 161 号端口,是用来管理网络设备。

(3)、TFTP:简单文件传输协议,在 69 号端口上使用 UDP 服务。


四、IP地址的分类:

A类地址:以 0 开头,第一个字节范围:0~127(1.0.0.0-126.255.255.255)

B类地址:以10开头,第一个字节范围:128~191(128.0.0.0-191.255.255.255)

C类地址:以 110 开头,第一个字节范围:192~233(192.0.0.0-223.255.255.255)

10.0.0.0-10.255.255.255,172.16.0.0-172.31.255.255,192.168.0.0-192.168.255.255(Internet 保留地址,用于内部)


五、ARP是地址解析协议,简单语言解释一下工作原理。

1:首先,每个主机都会在自己的ARP缓冲区中建立一个ARP列表,以表示IP地址和MAC地址之间的对应关系。

2:当源主机要发送数据时,首先检查ARP列表中是否有对应IP地址的目的主机的MAC地址,如果有,则直接发送数据,如果没有,就向本网段

的所有主机发送ARP数据包,该数据包包括的内容有:源主机 IP地址,源主机MAC地址,目的主机的IP 地址

3:当本网络的所有主机收到该ARP数据包时,首先检查数据包中的IP地址是否是自己的IP地址,如果不是,则忽略该数据包,如果是,则首先从

数据包中取出源主机的IP和MAC地址写入到ARP列表中,如果已经存在,则覆盖,然后将自己的MAC地址写入ARP响应包中,告诉源主机自己是它想要找的MAC地址。

4:源主机收到ARP响应包后。将目的主机的IP和MAC地址写入ARP列表,并利用此信息发送数据。如果源主机一直没有收到ARP响应数据包,表示ARP查询失败。

广播发送ARP请求,单播发送ARP响应。

六、描述:RARP

RARP是逆地址解析协议,作用是完成硬件地址到IP地址的映射,主要用于无盘工作站,因为给无盘工作站配置的IP地址不能保存。工作流程:在

网络中配置一台RARP服务器,里面保存着IP地址和MAC地址的映射关系,当无盘工作站启动后,就封装一个RARP数据包,里面有其MAC地址,然后

广播到网络上去,当服务器收到请求包后,就查找对应的MAC地址的IP地址装入响应报文中发回给请求者。因为需要广播请求报文,因此RARP只能用

于具有广播能力的网络。


七、浏览器中输入www.baidu.com后执行的全部过程

1、客户端浏览器通过DNS解析到www.baidu.com的IP地址220.181.27.48,通过这个IP地址找到客户端到服务器的路径。客户端浏览器发起一个

HTTP会话到220.161.27.48,然后通过TCP进行封装数据包,输入到网络层。

2、在客户端的传输层,把HTTP会话请求分成报文段,添加源和目的端口,如服务器使用80端口监听客户端的请求,客户端由系统随机选择一个端口

如5000,与服务器进行交换,服务器把相应的请求返回给客户端的5000端口。然后使用IP层的IP地址查找目的端。

3、客户端的网络层不用关系应用层或者传输层的东西,主要做的是通过查找路由表确定如何到达服务器,期间可能经过多个路由器,这些都是由

路由器来完成的工作,我不作过多的描述,无非就是通过查找路由表决定通过那个路径到达服务器。

4、客户端的链路层,包通过链路层发送到路由器,通过邻居协议查找给定IP地址的MAC地址,然后发送ARP请求查找目的地址,如果得到回应后就可

以使用ARP的请求应答交换的IP数据包现在就可以传输了,然后发送IP数据包到达服务器的地址。

八、DNS域名系统,简单描述其工作原理。

当DNS客户机需要在程序中使用名称时,它会查询DNS服务器来解析该名称。客户机发送的每条查询信息包括三条信息:包括:指定的DNS域名,

指定的查询类型,DNS域名的指定类别。基于UDP服务,端口53. 该应用一般不直接为用户使用,而是为其他应用服务,如HTTP,SMTP等在其中需

要完成主机名到IP地址的转换。

九、面向连接和非面向连接的服务的特点是什么?

面向连接的服务,通信双方在进行通信之前,要先在双方建立起一个完整的可以彼此沟通的通道,在通信过程中,整个连接的情况一直可以被实

时地监控和管理。非面向连接的服务,不需要预先建立一个联络两个通信节点的连接,需要通信的时候,发送节点就可以往网络上发送信息,让信息

自主地在网络上去传,一般在传输的过程中不再加以监控。

十、了解交换机、路由器、网关的概念、并知道各自的用途

(1)、交换机

在计算机网络系统中,交换机是针对共享工作模式的弱点推出的。交换机拥有一条高带宽的背部总线和内部交换矩阵。交换机的所有端口都挂接在这条背部总线上,当控制电路 收到数据包以后,处理端口会查找内存中的地址对照表以确定目的 MAC (网卡的硬件地址)的 NIC(网卡)

挂接在哪个端口上,通过内部交换矩阵迅速将数据包传送到目的端口。目的 MAC 若不存在,交换机才广播到所有的端口,接收端口回应后交换机会”学习“ 新的地址。并把它添加到内部地址表或中。

交换机工作于 OSI 参考模型的第二层,即数据链路层。交换机内部的CPU 会在每个端口成功连接时,通过 ARP 协议学习它的 MAC 地址,保存一张 ARP 表。在今后的通讯中,发往该 MAC 地址的数据包将仅送往其对应的端口,而不是所有的端口。因此,交换机可用划分数据链路层广播,即冲突域;但它不能划分网络层广播,即广播域。

交换机被广泛用于二层 网络交换,俗称“二层交换机”。

交换机的种类有:二层交换机,三层交换机、四层交换机、七层交换机分别工作 OSI 七层模型中的第二层、第三层、第七层,并因此得名。

(2)、路由器

路由器是一种计算机网络设备,提供了路由与转送两种重要机制,可以决定数据包从来源段到目的端所经过的路由路径(host 到 host 之间的传输路径),这个过程称之为路由;将路由器输入端的数据包移送至适当的路由器输出端(在路由器内部进行),这称为传送。路由工作在 OSI 模型的第三层 — 即网络层,例如,网际协议。

路由器的一个作用是联通一个网络,另一个作用是选择信息传送的路径。路由器与交换器的差别,路由器是属于 OSI 第三层产品,交换器是属于OSI第二层产品(这里指的是二层交换机)。

(3)、网关

网关,顾名思义就是连接两个网络的设备,却别在于路由器(由于历史的原因,许多有关 TCP/IP 的文献曾经把网络层使用的路由器称为网关,在今天很多局域网采用的都是路由来接入网络,因此,现在通常指的网关就是路由器的 IP),经常在家庭中或者小型企业中使用,用于连接局域网和 Internet。网关经常指把一种协议转换成另一种协议的设备,比如语音网关。

在传统 TCP/IP 术语中,网络设备只分为两种,一种网关,另一种主机。网关能在网络之间传送数据包,但主机不能传送数据包。在主机中,数据包经过 TCP/IP 四层协议处理,但是在网关(又称中介系统)只需到达网际层,决定路径之后就可以传送。在当时,网关与路由器还没有区别。

在现代网络术语中,网关(gateway)与路由器(router)的定义不同。网关(gateway)能在不同协议间移动数据,而路由器(router)

是在不同网络间移动数据,相当于传统所说的IP网关(IP gateway)。

网关是连接两个网路的设备,对于语音网关来说,它可以连接 PSTN 网路和以太网,这就相当于 VOIP,把不同电话中的模拟信号通过网关转换成数字信号,而且加入协议再去传输。在到了接收端的时候在通过网关还原成模拟的电话信号,最后才能在主机上听到。

对于以太网中的网关只能转发三层以上数据包,这一点和路由是一样的。而不同的是网关中并没有路由表,他只能按照预先设定的不同

网段来进行转发。网关最重要的一点就是端口映射,子网内用户在外网看来只是外网的IP地址对应着不同的端口,这样看来就会保护子网内的用户。

十一、端口及对应的服务?

服务

端口号

服务

端口号

FTP

21

SSH

22

telnet

23

SMTP

25

Domain(域名服务器)

53

HTTP

80

POP3

110

NTP(网络时间协议)

123

MySQL数据库服务

3306

Shell或 cmd

514

POP-2

109

SQL Server

1433












十二、IP数据包的格式

IP数据报由首部 和数据  两部分组成。首部由固定部分和可选部分  组成。首部的固定部分有 20 字节。可选部分的长度变化范围为1——40字节。固定部分的字段:

字段名

位数(bit)

字段名

位数

版本

4  Ipv4

首部长度

4(表示的最大数为15个单位,一个单位表示4字节)

服务类型

8  以前很少用

总长度

16 (首部和数据部分的总长度,因此数据报的最大长度为65535字节,即64KB,但是由于链路层的MAC都有一定的最大传输单元,因此IP数据报的长度一般都不会有理论上的那么大,如果超出了MAC的最大单元就会进行分片)

标识

16 (相同的标识使得分片后的数据报片能正确的重装成原来的数据报)

标志

3 (最低位MF=1表示后面还有分片,MF=0表示这是若干个数据报片的最后一个中间位DF=0才允许分片)

片偏移

片偏移指出较长的分组在分片后,某片在原分组中的相对位置,都是8字节的偏移位置

生存时间

数据报在网络中的生存时间,指最多经过路由器的跳数

协议

8 (指出该数据报携带的数据是何种协议,以使得目的主机的IP层知道应将数据部分上交给哪个处理程序)如ICMP=1 IGMP=2 TCP=6 EGP=8 IGP=9 UDP=17 Ipv6=41 OSPF=89

首部校验和

这个部分只校验首部,不包括数据部分,计算方法:将首部划分为多个16位的部分,然后每个16位部分取反,然后计算和,再将和取反放到首部校验和。接收方收到后按同样的方法划分,取反,求和,在取反,如果结果为零,则接收,否则就丢弃

源地址

32

目的地址

32
























十三、TCP数据报的格式?

一个TCP报文段分为首部和数据两部分。首部由固定部分和选项部分组成,固定部分是20字节。TCP首部的最大长度为60。首部固定部分字段:


字段名

字节(Byte)

字段名

字节(Byte)

源端口

2

目的端口

2

序号

4

确认号

4,是期望收到对方的下一个报文段的数据的第一个字节的序号

数据偏移

4bit 指出TCP报文段的数据起始处距离TCP报文段的起始有多远

保留

6bit

紧急比特URG

确认比特ACK

只有当ACK=1时,确认号字段才有效

推送比特PSH

复位比特RST

同步比特SYN

终止比特FIN

窗口

2

检验和

2 (包括首部和数据两部分,同时还要加12字节的伪首部进行校验和计算)

选项

长度可变(范围1——40)















TCP的12字节伪首部:

源IP地址(4)

目的IP地址(4)

0 (1)

6(1) 代表这是TCP,IP协议中提到过

TCP长度(2)

TCP数据报的格式?

用户数据报UDP由首部和数据部分组成。首部只有8个字节,由4个字段组成,每个字段都是两个字节。


字段名

字节

字段名

字节

源端口

2

目的端口

2

长度

2

检验和

2 (检验首部和数据,加12字节的伪首部)



UDP的12字节伪首部:

源IP地址(4)

目的IP地址(4)

0 (1)

17(1) 代表这是UDP

UDP长度(2)

以太网MAC帧格式?

前导码 前定界符 目的地址 源目的地址 长度字段 数据字段 校验字段
7B 1B 6B 6B 2B 46-1500 4B







十四、TCP的12字节伪首部:

源IP地址(4)

目的IP地址(4)

0 (1)

6(1) 代表这是TCP,IP协议中提到过

TCP长度(2)



十五、TCP数据报的格式?

用户数据报UDP由首部和数据部分组成。首部只有8个字节,由4个字段组成,每个字段都是两个字节。


字段名

字节

字段名

字节

源端口

2

目的端口

2

长度

2

检验和

2 (检验首部和数据,加12字节的伪首部)







UDP的12字节伪首部:

源IP地址(4)

目的IP地址(4)

0 (1)

17(1) 代表这是UDP

UDP长度(2)



十六、以太网MAC帧格式?

前导码 前定界符 目的地址 源目的地址 长度字段 数据字段 校验字段
7B 1B 6B 6B 2B 46-1500 4B


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Computer Networking: A Top-Down Approach, 6th Edition Solutions to Review Questions and Problems Version Date: May 2012 This document contains the solutions to review questions and problems for the 5th edition of Computer Networking: A Top-Down Approach by Jim Kurose and Keith Ross. These solutions are being made available to instructors ONLY. Please do NOT copy or distribute this document to others (even other instructors). Please do not post any solutions on a publicly-available Web site. We’ll be happy to provide a copy (up-to-date) of this solution manual ourselves to anyone who asks. Acknowledgments: Over the years, several students and colleagues have helped us prepare this solutions manual. Special thanks goes to HongGang Zhang, Rakesh Kumar, Prithula Dhungel, and Vijay Annapureddy. Also thanks to all the readers who have made suggestions and corrected errors. All material © copyright 1996-2012 by J.F. Kurose and K.W. Ross. All rights reserved Chapter 1 Review Questions There is no difference. Throughout this text, the words “host” and “end system” are used interchangeably. End systems include PCs, workstations, Web servers, mail servers, PDAs, Internet-connected game consoles, etc. From Wikipedia: Diplomatic protocol is commonly described as a set of international courtesy rules. These well-established and time-honored rules have made it easier for nations and people to live and work together. Part of protocol has always been the acknowledgment of the hierarchical standing of all present. Protocol rules are based on the principles of civility. Standards are important for protocols so that people can create networking systems and products that interoperate. 1. Dial-up modem over telephone line: home; 2. DSL over telephone line: home or small office; 3. Cable to HFC: home; 4. 100 Mbps switched Ethernet: enterprise; 5. Wifi (802.11): home and enterprise: 6. 3G and 4G: wide-area wireless. HFC bandwidth is shared among the users. On the downstream channel, all packets emanate from a single source, namely, the head end. Thus, there are no collisions in the downstream channel. In most American cities, the current possibilities include: dial-up; DSL; cable modem; fiber-to-the-home. 7. Ethernet LANs have transmission rates of 10 Mbps, 100 Mbps, 1 Gbps and 10 Gbps. 8. Today, Ethernet most commonly runs over twisted-pair copper wire. It also can run over fibers optic links. 9. Dial up modems: up to 56 Kbps, bandwidth is dedicated; ADSL: up to 24 Mbps downstream and 2.5 Mbps upstream, bandwidth is dedicated; HFC, rates up to 42.8 Mbps and upstream rates of up to 30.7 Mbps, bandwidth is shared. FTTH: 2-10Mbps upload; 10-20 Mbps download; bandwidth is not shared. 10. There are two popular wireless Internet access technologies today: Wifi (802.11) In a wireless LAN, wireless users transmit/receive packets to/from an base station (i.e., wireless access point) within a radius of few tens of meters. The base station is typically connected to the wired Internet and thus serves to connect wireless users to the wired network. 3G and 4G wide-area wireless access networks. In these systems, packets are transmitted over the same wireless infrastructure used for cellular telephony, with the base station thus being managed by a telecommunications provider. This provides wireless access to users within a radius of tens of kilometers of the base station. 11. At time t0 the sending host begins to transmit. At time t1 = L/R1, the sending host completes transmission and the entire packet is received at the router (no propagation delay). Because the router has the entire packet at time t1, it can begin to transmit the packet to the receiving host at time t1. At time t2 = t1 + L/R2, the router completes transmission and the entire packet is received at the receiving host (again, no propagation delay). Thus, the end-to-end delay is L/R1 + L/R2. 12. A circuit-switched network can guarantee a certain amount of end-to-end bandwidth for the duration of a call. Most packet-switched networks today (including the Internet) cannot make any end-to-end guarantees for bandwidth. FDM requires sophisticated analog hardware to shift signal into appropriate frequency bands. 13. a) 2 users can be supported because each user requires half of the link bandwidth. b) Since each user requires 1Mbps when transmitting, if two or fewer users transmit simultaneously, a maximum of 2Mbps will be required. Since the available bandwidth of the shared link is 2Mbps, there will be no queuing delay before the link. Whereas, if three users transmit simultaneously, the bandwidth required will be 3Mbps which is more than the available bandwidth of the shared link. In this case, there will be queuing delay before the link. c) Probability that a given user is transmitting = 0.2 d) Probability that all three users are transmitting simultaneously = = (0.2)3 = 0.008. Since the queue grows when all the users are transmitting, the fraction of time during which the queue grows (which is equal to the probability that all three users are transmitting simultaneously) is 0.008. 14. If the two ISPs do not peer with each other, then when they send traffic to each other they have to send the traffic through a provider ISP (intermediary), to which they have to pay for carrying the traffic. By peering with each other directly, the two ISPs can reduce their payments to their provider ISPs. An Internet Exchange Points (IXP) (typically in a standalone building with its own switches) is a meeting point where multiple ISPs can connect and/or peer together. An ISP earns its money by charging each of the the ISPs that connect to the IXP a relatively small fee, which may depend on the amount of traffic sent to or received from the IXP. 15. Google's private network connects together all its data centers, big and small. Traffic between the Google data centers passes over its private network rather than over the public Internet. Many of these data centers are located in, or close to, lower tier ISPs. Therefore, when Google delivers content to a user, it often can bypass higher tier ISPs. What motivates content providers to create these networks? First, the content provider has more control over the user experience, since it has to use few intermediary ISPs. Second, it can save money by sending less traffic into provider networks. Third, if ISPs decide to charge more money to highly profitable content providers (in countries where net neutrality doesn't apply), the content providers can avoid these extra payments. 16. The delay components are processing delays, transmission delays, propagation delays, and queuing delays. All of these delays are fixed, except for the queuing delays, which are variable. 17. a) 1000 km, 1 Mbps, 100 bytes b) 100 km, 1 Mbps, 100 bytes 18. 10msec; d/s; no; no 19. a) 500 kbps b) 64 seconds c) 100kbps; 320 seconds 20. End system A breaks the large file into chunks. It adds header to each chunk, thereby generating multiple packets from the file. The header in each packet includes the IP address of the destination (end system B). The packet switch uses the destination IP address in the packet to determine the outgoing link. Asking which road to take is analogous to a packet asking which outgoing link it should be forwarded on, given the packet’s destination address. 21. The maximum emission rate is 500 packets/sec and the maximum transmission rate is 350 packets/sec. The corresponding traffic intensity is 500/350 =1.43 > 1. Loss will eventually occur for each experiment; but the time when loss first occurs will be different from one experiment to the next due to the randomness in the emission process. 22. Five generic tasks are error control, flow control, segmentation and reassembly, multiplexing, and connection setup. Yes, these tasks can be duplicated at different layers. For example, error control is often provided at more than one layer. 23. The five layers in the Internet protocol stack are – from top to bottom – the application layer, the transport layer, the network layer, the link layer, and the physical layer. The principal responsibilities are outlined in Section 1.5.1. 24. Application-layer message: data which an application wants to send and passed onto the transport layer; transport-layer segment: generated by the transport layer and encapsulates application-layer message with transport layer header; network-layer datagram: encapsulates transport-layer segment with a network-layer header; link-layer frame: encapsulates network-layer datagram with a link-layer header. 25. Routers process network, link and physical layers (layers 1 through 3). (This is a little bit of a white lie, as modern routers sometimes act as firewalls or caching components, and process Transport layer as well.) Link layer switches process link and physical layers (layers 1 through2). Hosts process all five layers. 26. a) Virus Requires some form of human interaction to spread. Classic example: E-mail viruses. b) Worms No user replication needed. Worm in infected host scans IP addresses and port numbers, looking for vulnerable processes to infect. 27. Creation of a botnet requires an attacker to find vulnerability in some application or system (e.g. exploiting the buffer overflow vulnerability that might exist in an application). After finding the vulnerability, the attacker needs to scan for hosts that are vulnerable. The target is basically to compromise a series of systems by exploiting that particular vulnerability. Any system that is part of the botnet can automatically scan its environment and propagate by exploiting the vulnerability. An important property of such botnets is that the originator of the botnet can remotely control and issue commands to all the nodes in the botnet. Hence, it becomes possible for the attacker to issue a command to all the nodes, that target a single node (for example, all nodes in the botnet might be commanded by the attacker to send a TCP SYN message to the target, which might result in a TCP SYN flood attack at the target). 28. Trudy can pretend to be Bob to Alice (and vice-versa) and partially or completely modify the message(s) being sent from Bob to Alice. For example, she can easily change the phrase “Alice, I owe you $1000” to “Alice, I owe you $10,000”. Furthermore, Trudy can even drop the packets that are being sent by Bob to Alice (and vise-versa), even if the packets from Bob to Alice are encrypted. Chapter 1 Problems Problem 1 There is no single right answer to this question. Many protocols would do the trick. Here's a simple answer below: Messages from ATM machine to Server Msg name purpose -------- ------- HELO Let server know that there is a card in the ATM machine ATM card transmits user ID to Server PASSWD User enters PIN, which is sent to server BALANCE User requests balance WITHDRAWL User asks to withdraw money BYE user all done Messages from Server to ATM machine (display) Msg name purpose -------- ------- PASSWD Ask user for PIN (password) OK last requested operation (PASSWD, WITHDRAWL) OK ERR last requested operation (PASSWD, WITHDRAWL) in ERROR AMOUNT sent in response to BALANCE request BYE user done, display welcome screen at ATM Correct operation: client server HELO (userid) --------------> (check if valid userid) <------------- PASSWD PASSWD --------------> (check password) <------------- AMOUNT WITHDRAWL --------------> check if enough $ to cover withdrawl (check if valid userid) <------------- PASSWD PASSWD --------------> (check password) <------------- AMOUNT WITHDRAWL --------------> check if enough $ to cover withdrawl <------------- BYE Problem 2 At time N*(L/R) the first packet has reached the destination, the second packet is stored in the last router, the third packet is stored in the next-to-last router, etc. At time N*(L/R) + L/R, the second packet has reached the destination, the third packet is stored in the last router, etc. Continuing with this logic, we see that at time N*(L/R) + (P-1)*(L/R) = (N+P-1)*(L/R) all packets have reached the destination. Problem 3 a) A circuit-switched network would be well suited to the application, because the application involves long sessions with predictable smooth bandwidth requirements. Since the transmission rate is known and not bursty, bandwidth can be reserved for each application session without significant waste. In addition, the overhead costs of setting up and tearing down connections are amortized over the lengthy duration of a typical application session. b) In the worst case, all the applications simultaneously transmit over one or more network links. However, since each link has sufficient bandwidth to handle the sum of all of the applications' data rates, no congestion (very little queuing) will occur. Given such generous link capacities, the network does not need congestion control mechanisms. Problem 4 Between the switch in the upper left and the switch in the upper right we can have 4 connections. Similarly we can have four connections between each of the 3 other pairs of adjacent switches. Thus, this network can support up to 16 connections. We can 4 connections passing through the switch in the upper-right-hand corner and another 4 connections passing through the switch in the lower-left-hand corner, giving a total of 8 connections. Yes. For the connections between A and C, we route two connections through B and two connections through D. For the connections between B and D, we route two connections through A and two connections through C. In this manner, there are at most 4 connections passing through any link. Problem 5 Tollbooths are 75 km apart, and the cars propagate at 100km/hr. A tollbooth services a car at a rate of one car every 12 seconds. a) There are ten cars. It takes 120 seconds, or 2 minutes, for the first tollbooth to service the 10 cars. Each of these cars has a propagation delay of 45 minutes (travel 75 km) before arriving at the second tollbooth. Thus, all the cars are lined up before the second tollbooth after 47 minutes. The whole process repeats itself for traveling between the second and third tollbooths. It also takes 2 minutes for the third tollbooth to service the 10 cars. Thus the total delay is 96 minutes. b) Delay between tollbooths is 8*12 seconds plus 45 minutes, i.e., 46 minutes and 36 seconds. The total delay is twice this amount plus 8*12 seconds, i.e., 94 minutes and 48 seconds. Problem 6 a) seconds. b) seconds. c) seconds. d) The bit is just leaving Host A. e) The first bit is in the link and has not reached Host B. f) The first bit has reached Host B. g) Want km. Problem 7 Consider the first bit in a packet. Before this bit can be transmitted, all of the bits in the packet must be generated. This requires sec=7msec. The time required to transmit the packet is sec= sec. Propagation delay = 10 msec. The delay until decoding is 7msec + sec + 10msec = 17.224msec A similar analysis shows that all bits experience a delay of 17.224 msec. Problem 8 a) 20 users can be supported. b) . c) . d) . We use the central limit theorem to approximate this probability. Let be independent random variables such that . “21 or more users” when is a standard normal r.v. Thus “21 or more users” . Problem 9 10,000 Problem 10 The first end system requires L/R1 to transmit the packet onto the first link; the packet propagates over the first link in d1/s1; the packet switch adds a processing delay of dproc; after receiving the entire packet, the packet switch connecting the first and the second link requires L/R2 to transmit the packet onto the second link; the packet propagates over the second link in d2/s2. Similarly, we can find the delay caused by the second switch and the third link: L/R3, dproc, and d3/s3. Adding these five delays gives dend-end = L/R1 + L/R2 + L/R3 + d1/s1 + d2/s2 + d3/s3+ dproc+ dproc To answer the second question, we simply plug the values into the equation to get 6 + 6 + 6 + 20+16 + 4 + 3 + 3 = 64 msec. Problem 11 Because bits are immediately transmitted, the packet switch does not introduce any delay; in particular, it does not introduce a transmission delay. Thus, dend-end = L/R + d1/s1 + d2/s2+ d3/s3 For the values in Problem 10, we get 6 + 20 + 16 + 4 = 46 msec. Problem 12 The arriving packet must first wait for the link to transmit 4.5 *1,500 bytes = 6,750 bytes or 54,000 bits. Since these bits are transmitted at 2 Mbps, the queuing delay is 27 msec. Generally, the queuing delay is (nL + (L - x))/R. Problem 13 The queuing delay is 0 for the first transmitted packet, L/R for the second transmitted packet, and generally, (n-1)L/R for the nth transmitted packet. Thus, the average delay for the N packets is: (L/R + 2L/R + ....... + (N-1)L/R)/N = L/(RN) * (1 + 2 + ..... + (N-1)) = L/(RN) * N(N-1)/2 = LN(N-1)/(2RN) = (N-1)L/(2R) Note that here we used the well-known fact: 1 + 2 + ....... + N = N(N+1)/2 It takes seconds to transmit the packets. Thus, the buffer is empty when a each batch of packets arrive. Thus, the average delay of a packet across all batches is the average delay within one batch, i.e., (N-1)L/2R. Problem 14 The transmission delay is . The total delay is Let . Total delay = For x=0, the total delay =0; as we increase x, total delay increases, approaching infinity as x approaches 1/a. Problem 15 Total delay . Problem 16 The total number of packets in the system includes those in the buffer and the packet that is being transmitted. So, N=10+1. Because , so (10+1)=a*(queuing delay + transmission delay). That is, 11=a*(0.01+1/100)=a*(0.01+0.01). Thus, a=550 packets/sec. Problem 17 There are nodes (the source host and the routers). Let denote the processing delay at the th node. Let be the transmission rate of the th link and let . Let be the propagation delay across the th link. Then . Let denote the average queuing delay at node . Then . Problem 18 On linux you can use the command traceroute www.targethost.com and in the Windows command prompt you can use tracert www.targethost.com In either case, you will get three delay measurements. For those three measurements you can calculate the mean and standard deviation. Repeat the experiment at different times of the day and comment on any changes. Here is an example solution: Traceroutes between San Diego Super Computer Center and www.poly.edu The average (mean) of the round-trip delays at each of the three hours is 71.18 ms, 71.38 ms and 71.55 ms, respectively. The standard deviations are 0.075 ms, 0.21 ms, 0.05 ms, respectively. In this example, the traceroutes have 12 routers in the path at each of the three hours. No, the paths didn’t change during any of the hours. Traceroute packets passed through four ISP networks from source to destination. Yes, in this experiment the largest delays occurred at peering interfaces between adjacent ISPs. Traceroutes from www.stella-net.net (France) to www.poly.edu (USA). The average round-trip delays at each of the three hours are 87.09 ms, 86.35 ms and 86.48 ms, respectively. The standard deviations are 0.53 ms, 0.18 ms, 0.23 ms, respectively. In this example, there are 11 routers in the path at each of the three hours. No, the paths didn’t change during any of the hours. Traceroute packets passed three ISP networks from source to destination. Yes, in this experiment the largest delays occurred at peering interfaces between adjacent ISPs. Problem 19 An example solution: Traceroutes from two different cities in France to New York City in United States In these traceroutes from two different cities in France to the same destination host in United States, seven links are in common including the transatlantic link. In this example of traceroutes from one city in France and from another city in Germany to the same host in United States, three links are in common including the transatlantic link. Traceroutes to two different cities in China from same host in United States Five links are common in the two traceroutes. The two traceroutes diverge before reaching China Problem 20 Throughput = min{Rs, Rc, R/M} Problem 21 If only use one path, the max throughput is given by: . If use all paths, the max throughput is given by . Problem 22 Probability of successfully receiving a packet is: ps= (1-p)N. The number of transmissions needed to be performed until the packet is successfully received by the client is a geometric random variable with success probability ps. Thus, the average number of transmissions needed is given by: 1/ps . Then, the average number of re-transmissions needed is given by: 1/ps -1. Problem 23 Let’s call the first packet A and call the second packet B. If the bottleneck link is the first link, then packet B is queued at the first link waiting for the transmission of packet A. So the packet inter-arrival time at the destination is simply L/Rs. If the second link is the bottleneck link and both packets are sent back to back, it must be true that the second packet arrives at the input queue of the second link before the second link finishes the transmission of the first packet. That is, L/Rs + L/Rs + dprop = L/Rs + dprop + L/Rc Thus, the minimum value of T is L/Rc  L/Rs . Problem 24 40 terabytes = 40 * 1012 * 8 bits. So, if using the dedicated link, it will take 40 * 1012 * 8 / (100 *106 ) =3200000 seconds = 37 days. But with FedEx overnight delivery, you can guarantee the data arrives in one day, and it should cost less than $100. Problem 25 160,000 bits 160,000 bits The bandwidth-delay product of a link is the maximum number of bits that can be in the link. the width of a bit = length of link / bandwidth-delay product, so 1 bit is 125 meters long, which is longer than a football field s/R Problem 26 s/R=20000km, then R=s/20000km= 2.5*108/(2*107)= 12.5 bps Problem 27 80,000,000 bits 800,000 bits, this is because that the maximum number of bits that will be in the link at any given time = min(bandwidth delay product, packet size) = 800,000 bits. .25 meters Problem 28 ttrans + tprop = 400 msec + 80 msec = 480 msec. 20 * (ttrans + 2 tprop) = 20*(20 msec + 80 msec) = 2 sec. Breaking up a file takes longer to transmit because each data packet and its corresponding acknowledgement packet add their own propagation delays. Problem 29 Recall geostationary satellite is 36,000 kilometers away from earth surface. 150 msec 1,500,000 bits 600,000,000 bits Problem 30 Let’s suppose the passenger and his/her bags correspond to the data unit arriving to the top of the protocol stack. When the passenger checks in, his/her bags are checked, and a tag is attached to the bags and ticket. This is additional information added in the Baggage layer if Figure 1.20 that allows the Baggage layer to implement the service or separating the passengers and baggage on the sending side, and then reuniting them (hopefully!) on the destination side. When a passenger then passes through security and additional stamp is often added to his/her ticket, indicating that the passenger has passed through a security check. This information is used to ensure (e.g., by later checks for the security information) secure transfer of people. Problem 31 Time to send message from source host to first packet switch = With store-and-forward switching, the total time to move message from source host to destination host = Time to send 1st packet from source host to first packet switch = . . Time at which 2nd packet is received at the first switch = time at which 1st packet is received at the second switch = Time at which 1st packet is received at the destination host = . After this, every 5msec one packet will be received; thus time at which last (800th) packet is received = . It can be seen that delay in using message segmentation is significantly less (almost 1/3rd). Without message segmentation, if bit errors are not tolerated, if there is a single bit error, the whole message has to be retransmitted (rather than a single packet). Without message segmentation, huge packets (containing HD videos, for example) are sent into the network. Routers have to accommodate these huge packets. Smaller packets have to queue behind enormous packets and suffer unfair delays. Packets have to be put in sequence at the destination. Message segmentation results in many smaller packets. Since header size is usually the same for all packets regardless of their size, with message segmentation the total amount of header bytes is more. Problem 32 Yes, the delays in the applet correspond to the delays in the Problem 31.The propagation delays affect the overall end-to-end delays both for packet switching and message switching equally. Problem 33 There are F/S packets. Each packet is S=80 bits. Time at which the last packet is received at the first router is sec. At this time, the first F/S-2 packets are at the destination, and the F/S-1 packet is at the second router. The last packet must then be transmitted by the first router and the second router, with each transmission taking sec. Thus delay in sending the whole file is To calculate the value of S which leads to the minimum delay, Problem 34 The circuit-switched telephone networks and the Internet are connected together at "gateways". When a Skype user (connected to the Internet) calls an ordinary telephone, a circuit is established between a gateway and the telephone user over the circuit switched network. The skype user's voice is sent in packets over the Internet to the gateway. At the gateway, the voice signal is reconstructed and then sent over the circuit. In the other direction, the voice signal is sent over the circuit switched network to the gateway. The gateway packetizes the voice signal and sends the voice packets to the Skype user.   Chapter 2 Review Questions The Web: HTTP; file transfer: FTP; remote login: Telnet; e-mail: SMTP; BitTorrent file sharing: BitTorrent protocol Network architecture refers to the organization of the communication process into layers (e.g., the five-layer Internet architecture). Application architecture, on the other hand, is designed by an application developer and dictates the broad structure of the application (e.g., client-server or P2P). The process which initiates the communication is the client; the process that waits to be contacted is the server. No. In a P2P file-sharing application, the peer that is receiving a file is typically the client and the peer that is sending the file is typically the server. The IP address of the destination host and the port number of the socket in the destination process. You would use UDP. With UDP, the transaction can be completed in one roundtrip time (RTT) - the client sends the transaction request into a UDP socket, and the server sends the reply back to the client's UDP socket. With TCP, a minimum of two RTTs are needed - one to set-up the TCP connection, and another for the client to send the request, and for the server to send back the reply. One such example is remote word processing, for example, with Google docs. However, because Google docs runs over the Internet (using TCP), timing guarantees are not provided. a) Reliable data transfer TCP provides a reliable byte-stream between client and server but UDP does not. b) A guarantee that a certain value for throughput will be maintained Neither c) A guarantee that data will be delivered within a specified amount of time Neither d) Confidentiality (via encryption) Neither SSL operates at the application layer. The SSL socket takes unencrypted data from the application layer, encrypts it and then passes it to the TCP socket. If the application developer wants TCP to be enhanced with SSL, she has to include the SSL code in the application. A protocol uses handshaking if the two communicating entities first exchange control packets before sending data to each other. SMTP uses handshaking at the application layer whereas HTTP does not. The applications associated with those protocols require that all application data be received in the correct order and without gaps. TCP provides this service whereas UDP does not. When the user first visits the site, the server creates a unique identification number, creates an entry in its back-end database, and returns this identification number as a cookie number. This cookie number is stored on the user’s host and is managed by the browser. During each subsequent visit (and purchase), the browser sends the cookie number back to the site. Thus the site knows when this user (more precisely, this browser) is visiting the site. Web caching can bring the desired content “closer” to the user, possibly to the same LAN to which the user’s host is connected. Web caching can reduce the delay for all objects, even objects that are not cached, since caching reduces the traffic on links. Telnet is not available in Windows 7 by default. to make it available, go to Control Panel, Programs and Features, Turn Windows Features On or Off, Check Telnet client. To start Telnet, in Windows command prompt, issue the following command > telnet webserverver 80 where "webserver" is some webserver. After issuing the command, you have established a TCP connection between your client telnet program and the web server. Then type in an HTTP GET message. An example is given below: Since the index.html page in this web server was not modified since Fri, 18 May 2007 09:23:34 GMT, and the above commands were issued on Sat, 19 May 2007, the server returned "304 Not Modified". Note that the first 4 lines are the GET message and header lines inputed by the user, and the next 4 lines (starting from HTTP/1.1 304 Not Modified) is the response from the web server. FTP uses two parallel TCP connections, one connection for sending control information (such as a request to transfer a file) and another connection for actually transferring the file. Because the control information is not sent over the same connection that the file is sent over, FTP sends control information out of band. The message is first sent from Alice’s host to her mail server over HTTP. Alice’s mail server then sends the message to Bob’s mail server over SMTP. Bob then transfers the message from his mail server to his host over POP3. 17. Received: from 65.54.246.203 (EHLO bay0-omc3-s3.bay0.hotmail.com) (65.54.246.203) by mta419.mail.mud.yahoo.com with SMTP; Sat, 19 May 2007 16:53:51 -0700 Received: from hotmail.com ([65.55.135.106]) by bay0-omc3-s3.bay0.hotmail.com with Microsoft SMTPSVC(6.0.3790.2668); Sat, 19 May 2007 16:52:42 -0700 Received: from mail pickup service by hotmail.com with Microsoft SMTPSVC; Sat, 19 May 2007 16:52:41 -0700 Message-ID: Received: from 65.55.135.123 by by130fd.bay130.hotmail.msn.com with HTTP; Sat, 19 May 2007 23:52:36 GMT From: "prithula dhungel" To: prithula@yahoo.com Bcc: Subject: Test mail Date: Sat, 19 May 2007 23:52:36 +0000 Mime-Version: 1.0 Content-Type: Text/html; format=flowed Return-Path: prithuladhungel@hotmail.com Figure: A sample mail message header Received: This header field indicates the sequence in which the SMTP servers send and receive the mail message including the respective timestamps. In this example there are 4 “Received:” header lines. This means the mail message passed through 5 different SMTP servers before being delivered to the receiver’s mail box. The last (forth) “Received:” header indicates the mail message flow from the SMTP server of the sender to the second SMTP server in the chain of servers. The sender’s SMTP server is at address 65.55.135.123 and the second SMTP server in the chain is by130fd.bay130.hotmail.msn.com. The third “Received:” header indicates the mail message flow from the second SMTP server in the chain to the third server, and so on. Finally, the first “Received:” header indicates the flow of the mail messages from the forth SMTP server to the last SMTP server (i.e. the receiver’s mail server) in the chain. Message-id: The message has been given this number BAY130-F26D9E35BF59E0D18A819AFB9310@phx.gbl (by bay0-omc3-s3.bay0.hotmail.com. Message-id is a unique string assigned by the mail system when the message is first created. From: This indicates the email address of the sender of the mail. In the given example, the sender is “prithuladhungel@hotmail.com” To: This field indicates the email address of the receiver of the mail. In the example, the receiver is “prithula@yahoo.com” Subject: This gives the subject of the mail (if any specified by the sender). In the example, the subject specified by the sender is “Test mail” Date: The date and time when the mail was sent by the sender. In the example, the sender sent the mail on 19th May 2007, at time 23:52:36 GMT. Mime-version: MIME version used for the mail. In the example, it is 1.0. Content-type: The type of content in the body of the mail message. In the example, it is “text/html”. Return-Path: This specifies the email address to which the mail will be sent if the receiver of this mail wants to reply to the sender. This is also used by the sender’s mail server for bouncing back undeliverable mail messages of mailer-daemon error messages. In the example, the return path is “prithuladhungel@hotmail.com”. With download and delete, after a user retrieves its messages from a POP server, the messages are deleted. This poses a problem for the nomadic user, who may want to access the messages from many different machines (office PC, home PC, etc.). In the download and keep configuration, messages are not deleted after the user retrieves the messages. This can also be inconvenient, as each time the user retrieves the stored messages from a new machine, all of non-deleted messages will be transferred to the new machine (including very old messages). Yes an organization’s mail server and Web server can have the same alias for a host name. The MX record is used to map the mail server’s host name to its IP address. You should be able to see the sender's IP address for a user with an .edu email address. But you will not be able to see the sender's IP address if the user uses a gmail account. It is not necessary that Bob will also provide chunks to Alice. Alice has to be in the top 4 neighbors of Bob for Bob to send out chunks to her; this might not occur even if Alice provides chunks to Bob throughout a 30-second interval. Recall that in BitTorrent, a peer picks a random peer and optimistically unchokes the peer for a short period of time. Therefore, Alice will eventually be optimistically unchoked by one of her neighbors, during which time she will receive chunks from that neighbor. The overlay network in a P2P file sharing system consists of the nodes participating in the file sharing system and the logical links between the nodes. There is a logical link (an “edge” in graph theory terms) from node A to node B if there is a semi-permanent TCP connection between A and B. An overlay network does not include routers. Mesh DHT: The advantage is in order to a route a message to the peer (with ID) that is closest to the key, only one hop is required; the disadvantage is that each peer must track all other peers in the DHT. Circular DHT: the advantage is that each peer needs to track only a few other peers; the disadvantage is that O(N) hops are needed to route a message to the peer that is closest to the key. 25. File Distribution Instant Messaging Video Streaming Distributed Computing With the UDP server, there is no welcoming socket, and all data from different clients enters the server through this one socket. With the TCP server, there is a welcoming socket, and each time a client initiates a connection to the server, a new socket is created. Thus, to support n simultaneous connections, the server would need n+1 sockets. For the TCP application, as soon as the client is executed, it attempts to initiate a TCP connection with the server. If the TCP server is not running, then the client will fail to make a connection. For the UDP application, the client does not initiate connections (or attempt to communicate with the UDP server) immediately upon execution Chapter 2 Problems Problem 1 a) F b) T c) F d) F e) F Problem 2 Access control commands: USER, PASS, ACT, CWD, CDUP, SMNT, REIN, QUIT. Transfer parameter commands: PORT, PASV, TYPE STRU, MODE. Service commands: RETR, STOR, STOU, APPE, ALLO, REST, RNFR, RNTO, ABOR, DELE, RMD, MRD, PWD, LIST, NLST, SITE, SYST, STAT, HELP, NOOP. Problem 3 Application layer protocols: DNS and HTTP Transport layer protocols: UDP for DNS; TCP for HTTP Problem 4 The document request was http://gaia.cs.umass.edu/cs453/index.html. The Host : field indicates the server's name and /cs453/index.html indicates the file name. The browser is running HTTP version 1.1, as indicated just before the first pair. The browser is requesting a persistent connection, as indicated by the Connection: keep-alive. This is a trick question. This information is not contained in an HTTP message anywhere. So there is no way to tell this from looking at the exchange of HTTP messages alone. One would need information from the IP datagrams (that carried the TCP segment that carried the HTTP GET request) to answer this question. Mozilla/5.0. The browser type information is needed by the server to send different versions of the same object to different types of browsers. Problem 5 The status code of 200 and the phrase OK indicate that the server was able to locate the document successfully. The reply was provided on Tuesday, 07 Mar 2008 12:39:45 Greenwich Mean Time. The document index.html was last modified on Saturday 10 Dec 2005 18:27:46 GMT. There are 3874 bytes in the document being returned. The first five bytes of the returned document are : <!doc. The server agreed to a persistent connection, as indicated by the Connection: Keep-Alive field Problem 6 Persistent connections are discussed in section 8 of RFC 2616 (the real goal of this question was to get you to retrieve and read an RFC). Sections 8.1.2 and 8.1.2.1 of the RFC indicate that either the client or the server can indicate to the other that it is going to close the persistent connection. It does so by including the connection-token "close" in the Connection-header field of the http request/reply. HTTP does not provide any encryption services. (From RFC 2616) “Clients that use persistent connections should limit the number of simultaneous connections that they maintain to a given server. A single-user client SHOULD NOT maintain more than 2 connections with any server or proxy.” Yes. (From RFC 2616) “A client might have started to send a new request at the same time that the server has decided to close the "idle" connection. From the server's point of view, the connection is being closed while it was idle, but from the client's point of view, a request is in progress.” Problem 7 The total amount of time to get the IP address is . Once the IP address is known, elapses to set up the TCP connection and another elapses to request and receive the small object. The total response time is Problem 8 . . Problem 9 The time to transmit an object of size L over a link or rate R is L/R. The average time is the average size of the object divided by R:  = (850,000 bits)/(15,000,000 bits/sec) = .0567 sec The traffic intensity on the link is given by =(16 requests/sec)(.0567 sec/request) = 0.907. Thus, the average access delay is (.0567 sec)/(1 - .907)  .6 seconds. The total average response time is therefore .6 sec + 3 sec = 3.6 sec. The traffic intensity on the access link is reduced by 60% since the 60% of the requests are satisfied within the institutional network. Thus the average access delay is (.0567 sec)/[1 – (.4)(.907)] = .089 seconds. The response time is approximately zero if the request is satisfied by the cache (which happens with probability .6); the average response time is .089 sec + 3 sec = 3.089 sec for cache misses (which happens 40% of the time). So the average response time is (.6)(0 sec) + (.4)(3.089 sec) = 1.24 seconds. Thus the average response time is reduced from 3.6 sec to 1.24 sec. Problem 10 Note that each downloaded object can be completely put into one data packet. Let Tp denote the one-way propagation delay between the client and the server. First consider parallel downloads using non-persistent connections. Parallel downloads would allow 10 connections to share the 150 bits/sec bandwidth, giving each just 15 bits/sec. Thus, the total time needed to receive all objects is given by: (200/150+Tp + 200/150 +Tp + 200/150+Tp + 100,000/150+ Tp ) + (200/(150/10)+Tp + 200/(150/10) +Tp + 200/(150/10)+Tp + 100,000/(150/10)+ Tp ) = 7377 + 8*Tp (seconds) Now consider a persistent HTTP connection. The total time needed is given by: (200/150+Tp + 200/150 +Tp + 200/150+Tp + 100,000/150+ Tp ) + 10*(200/150+Tp + 100,000/150+ Tp ) =7351 + 24*Tp (seconds) Assuming the speed of light is 300*106 m/sec, then Tp=10/(300*106)=0.03 microsec. Tp is therefore negligible compared with transmission delay. Thus, we see that persistent HTTP is not significantly faster (less than 1 percent) than the non-persistent case with parallel download. Problem 11 Yes, because Bob has more connections, he can get a larger share of the link bandwidth. Yes, Bob still needs to perform parallel downloads; otherwise he will get less bandwidth than the other four users. Problem 12 Server.py from socket import * serverPort=12000 serverSocket=socket(AF_INET,SOCK_STREAM) serverSocket.bind(('',serverPort)) serverSocket.listen(1) connectionSocket, addr = serverSocket.accept() while 1: sentence = connectionSocket.recv(1024) print 'From Server:', sentence, '\n' serverSocket.close() Problem 13 The MAIL FROM: in SMTP is a message from the SMTP client that identifies the sender of the mail message to the SMTP server. The From: on the mail message itself is NOT an SMTP message, but rather is just a line in the body of the mail message. Problem 14 SMTP uses a line containing only a period to mark the end of a message body. HTTP uses “Content-Length header field” to indicate the length of a message body. No, HTTP cannot use the method used by SMTP, because HTTP message could be binary data, whereas in SMTP, the message body must be in 7-bit ASCII format. Problem 15 MTA stands for Mail Transfer Agent. A host sends the message to an MTA. The message then follows a sequence of MTAs to reach the receiver’s mail reader. We see that this spam message follows a chain of MTAs. An honest MTA should report where it receives the message. Notice that in this message, “asusus-4b96 ([58.88.21.177])” does not report from where it received the email. Since we assume only the originator is dishonest, so “asusus-4b96 ([58.88.21.177])” must be the originator. Problem 16 UIDL abbreviates “unique-ID listing”. When a POP3 client issues the UIDL command, the server responds with the unique message ID for all of the messages present in the user's mailbox. This command is useful for “download and keep”. By maintaining a file that lists the messages retrieved during earlier sessions, the client can use the UIDL command to determine which messages on the server have already been seen. Problem 17 a) C: dele 1 C: retr 2 S: (blah blah … S: ………..blah) S: . C: dele 2 C: quit S: +OK POP3 server signing off b) C: retr 2 S: blah blah … S: ………..blah S: . C: quit S: +OK POP3 server signing off C: list S: 1 498 S: 2 912 S: . C: retr 1 S: blah ….. S: ….blah S: . C: retr 2 S: blah blah … S: ………..blah S: . C: quit S: +OK POP3 server signing off Problem 18 For a given input of domain name (such as ccn.com), IP address or network administrator name, the whois database can be used to locate the corresponding registrar, whois server, DNS server, and so on. NS4.YAHOO.COM from www.register.com; NS1.MSFT.NET from ww.register.com Local Domain: www.mindspring.com Web servers : www.mindspring.com 207.69.189.21, 207.69.189.22, 207.69.189.23, 207.69.189.24, 207.69.189.25, 207.69.189.26, 207.69.189.27, 207.69.189.28 Mail Servers : mx1.mindspring.com (207.69.189.217) mx2.mindspring.com (207.69.189.218) mx3.mindspring.com (207.69.189.219) mx4.mindspring.com (207.69.189.220) Name Servers: itchy.earthlink.net (207.69.188.196) scratchy.earthlink.net (207.69.188.197) www.yahoo.com Web Servers: www.yahoo.com (216.109.112.135, 66.94.234.13) Mail Servers: a.mx.mail.yahoo.com (209.191.118.103) b.mx.mail.yahoo.com (66.196.97.250) c.mx.mail.yahoo.com (68.142.237.182, 216.39.53.3) d.mx.mail.yahoo.com (216.39.53.2) e.mx.mail.yahoo.com (216.39.53.1) f.mx.mail.yahoo.com (209.191.88.247, 68.142.202.247) g.mx.mail.yahoo.com (209.191.88.239, 206.190.53.191) Name Servers: ns1.yahoo.com (66.218.71.63) ns2.yahoo.com (68.142.255.16) ns3.yahoo.com (217.12.4.104) ns4.yahoo.com (68.142.196.63) ns5.yahoo.com (216.109.116.17) ns8.yahoo.com (202.165.104.22) ns9.yahoo.com (202.160.176.146) www.hotmail.com Web Servers: www.hotmail.com (64.4.33.7, 64.4.32.7) Mail Servers: mx1.hotmail.com (65.54.245.8, 65.54.244.8, 65.54.244.136) mx2.hotmail.com (65.54.244.40, 65.54.244.168, 65.54.245.40) mx3.hotmail.com (65.54.244.72, 65.54.244.200, 65.54.245.72) mx4.hotmail.com (65.54.244.232, 65.54.245.104, 65.54.244.104) Name Servers: ns1.msft.net (207.68.160.190) ns2.msft.net (65.54.240.126) ns3.msft.net (213.199.161.77) ns4.msft.net (207.46.66.126) ns5.msft.net (65.55.238.126) d) The yahoo web server has multiple IP addresses www.yahoo.com (216.109.112.135, 66.94.234.13) e) The address range for Polytechnic University: 128.238.0.0 – 128.238.255.255 f) An attacker can use the whois database and nslookup tool to determine the IP address ranges, DNS server addresses, etc., for the target institution. By analyzing the source address of attack packets, the victim can use whois to obtain information about domain from which the attack is coming and possibly inform the administrators of the origin domain. Problem 19 The following delegation chain is used for gaia.cs.umass.edu a.root-servers.net E.GTLD-SERVERS.NET ns1.umass.edu(authoritative) First command: dig +norecurse @a.root-servers.net any gaia.cs.umass.edu ;; AUTHORITY SECTION: edu. 172800 IN NS E.GTLD-SERVERS.NET. edu. 172800 IN NS A.GTLD-SERVERS.NET. edu. 172800 IN NS G3.NSTLD.COM. edu. 172800 IN NS D.GTLD-SERVERS.NET. edu. 172800 IN NS H3.NSTLD.COM. edu. 172800 IN NS L3.NSTLD.COM. edu. 172800 IN NS M3.NSTLD.COM. edu. 172800 IN NS C.GTLD-SERVERS.NET. Among all returned edu DNS servers, we send a query to the first one. dig +norecurse @E.GTLD-SERVERS.NET any gaia.cs.umass.edu umass.edu. 172800 IN NS ns1.umass.edu. umass.edu. 172800 IN NS ns2.umass.edu. umass.edu. 172800 IN NS ns3.umass.edu. Among all three returned authoritative DNS servers, we send a query to the first one. dig +norecurse @ns1.umass.edu any gaia.cs.umass.edu gaia.cs.umass.edu. 21600 IN A 128.119.245.12 The answer for google.com could be: a.root-servers.net E.GTLD-SERVERS.NET ns1.google.com(authoritative) Problem 20 We can periodically take a snapshot of the DNS caches in the local DNS servers. The Web server that appears most frequently in the DNS caches is the most popular server. This is because if more users are interested in a Web server, then DNS requests for that server are more frequently sent by users. Thus, that Web server will appear in the DNS caches more frequently. For a complete measurement study, see: Craig E. Wills, Mikhail Mikhailov, Hao Shang “Inferring Relative Popularity of Internet Applications by Actively Querying DNS Caches”, in IMC'03, October 27­29, 2003, Miami Beach, Florida, USA Problem 21 Yes, we can use dig to query that Web site in the local DNS server. For example, “dig cnn.com” will return the query time for finding cnn.com. If cnn.com was just accessed a couple of seconds ago, an entry for cnn.com is cached in the local DNS cache, so the query time is 0 msec. Otherwise, the query time is large. Problem 22 For calculating the minimum distribution time for client-server distribution, we use the following formula: Dcs = max {NF/us, F/dmin} Similarly, for calculating the minimum distribution time for P2P distribution, we use the following formula: Where, F = 15 Gbits = 15 * 1024 Mbits us = 30 Mbps dmin = di = 2 Mbps Note, 300Kbps = 300/1024 Mbps. Client Server N 10 100 1000 u 300 Kbps 7680 51200 512000 700 Kbps 7680 51200 512000 2 Mbps 7680 51200 512000 Peer to Peer N 10 100 1000 u 300 Kbps 7680 25904 47559 700 Kbps 7680 15616 21525 2 Mbps 7680 7680 7680 Problem 23 Consider a distribution scheme in which the server sends the file to each client, in parallel, at a rate of a rate of us/N. Note that this rate is less than each of the client’s download rate, since by assumption us/N ≤ dmin. Thus each client can also receive at rate us/N. Since each client receives at rate us/N, the time for each client to receive the entire file is F/( us/N) = NF/ us. Since all the clients receive the file in NF/ us, the overall distribution time is also NF/ us. Consider a distribution scheme in which the server sends the file to each client, in parallel, at a rate of dmin. Note that the aggregate rate, N dmin, is less than the server’s link rate us, since by assumption us/N ≥ dmin. Since each client receives at rate dmin, the time for each client to receive the entire file is F/ dmin. Since all the clients receive the file in this time, the overall distribution time is also F/ dmin. From Section 2.6 we know that DCS ≥ max {NF/us, F/dmin} (Equation 1) Suppose that us/N ≤ dmin. Then from Equation 1 we have DCS ≥ NF/us . But from (a) we have DCS ≤ NF/us . Combining these two gives: DCS = NF/us when us/N ≤ dmin. (Equation 2) We can similarly show that: DCS =F/dmin when us/N ≥ dmin (Equation 3). Combining Equation 2 and Equation 3 gives the desired result. Problem 24 Define u = u1 + u2 + ….. + uN. By assumption us <= (us + u)/N Equation 1 Divide the file into N parts, with the ith part having size (ui/u)F. The server transmits the ith part to peer i at rate ri = (ui/u)us. Note that r1 + r2 + ….. + rN = us, so that the aggregate server rate does not exceed the link rate of the server. Also have each peer i forward the bits it receives to each of the N-1 peers at rate ri. The aggregate forwarding rate by peer i is (N-1)ri. We have (N-1)ri = (N-1)(usui)/u = (us + u)/N Equation 2 Let ri = ui/(N-1) and rN+1 = (us – u/(N-1))/N In this distribution scheme, the file is broken into N+1 parts. The server sends bits from the ith part to the ith peer (i = 1, …., N) at rate ri. Each peer i forwards the bits arriving at rate ri to each of the other N-1 peers. Additionally, the server sends bits from the (N+1) st part at rate rN+1 to each of the N peers. The peers do not forward the bits from the (N+1)st part. The aggregate send rate of the server is r1+ …. + rN + N rN+1 = u/(N-1) + us – u/(N-1) = us Thus, the server’s send rate does not exceed its link rate. The aggregate send rate of peer i is (N-1)ri = ui Thus, each peer’s send rate does not exceed its link rate. In this distribution scheme, peer i receives bits at an aggregate rate of Thus each peer receives the file in NF/(us+u). (For simplicity, we neglected to specify the size of the file part for i = 1, …., N+1. We now provide that here. Let Δ = (us+u)/N be the distribution time. For i = 1, …, N, the ith file part is Fi = ri Δ bits. The (N+1)st file part is FN+1 = rN+1 Δ bits. It is straightforward to show that F1+ ….. + FN+1 = F.) The solution to this part is similar to that of 17 (c). We know from section 2.6 that Combining this with a) and b) gives the desired result. Problem 25 There are N nodes in the overlay network. There are N(N-1)/2 edges. Problem 26 Yes. His first claim is possible, as long as there are enough peers staying in the swarm for a long enough time. Bob can always receive data through optimistic unchoking by other peers. His second claim is also true. He can run a client on each host, let each client “free-ride,” and combine the collected chunks from the different hosts into a single file. He can even write a small scheduling program to make the different hosts ask for different chunks of the file. This is actually a kind of Sybil attack in P2P networks. Problem 27 Peer 3 learns that peer 5 has just left the system, so Peer 3 asks its first successor (Peer 4) for the identifier of its immediate successor (peer 8). Peer 3 will then make peer 8 its second successor. Problem 28 Peer 6 would first send peer 15 a message, saying “what will be peer 6’s predecessor and successor?” This message gets forwarded through the DHT until it reaches peer 5, who realizes that it will be 6’s predecessor and that its current successor, peer 8, will become 6’s successor. Next, peer 5 sends this predecessor and successor information back to 6. Peer 6 can now join the DHT by making peer 8 its successor and by notifying peer 5 that it should change its immediate successor to 6. Problem 29 For each key, we first calculate the distances (using d(k,p)) between itself and all peers, and then store the key in the peer that is closest to the key (that is, with smallest distance value). Problem 30 Yes, randomly assigning keys to peers does not consider the underlying network at all, so it very likely causes mismatches. Such mismatches may degrade the search performance. For example, consider a logical path p1 (consisting of only two logical links): ABC, where A and B are neighboring peers, and B and C are neighboring peers. Suppose that there is another logical path p2 from A to C (consisting of 3 logical links): ADEC. It might be the case that A and B are very far away physically (and separated by many routers), and B and C are very far away physically (and separated by many routers). But it may be the case that A, D, E, and C are all very close physically (and all separated by few routers). In other words, a shorter logical path may correspond to a much longer physical path. Problem 31 If you run TCPClient first, then the client will attempt to make a TCP connection with a non-existent server process. A TCP connection will not be made. UDPClient doesn't establish a TCP connection with the server. Thus, everything should work fine if you first run UDPClient, then run UDPServer, and then type some input into the keyboard. If you use different port numbers, then the client will attempt to establish a TCP connection with the wrong process or a non-existent process. Errors will occur. Problem 32 In the original program, UDPClient does not specify a port number when it creates the socket. In this case, the code lets the underlying operating system choose a port number. With the additional line, when UDPClient is executed, a UDP socket is created with port number 5432 . UDPServer needs to know the client port number so that it can send packets back to the correct client socket. Glancing at UDPServer, we see that the client port number is not “hard-wired” into the server code; instead, UDPServer determines the client port number by unraveling the datagram it receives from the client. Thus UDP server will work with any client port number, including 5432. UDPServer therefore does not need to be modified. Before: Client socket = x (chosen by OS) Server socket = 9876 After: Client socket = 5432 Problem 33 Yes, you can configure many browsers to open multiple simultaneous connections to a Web site. The advantage is that you will you potentially download the file faster. The disadvantage is that you may be hogging the bandwidth, thereby significantly slowing down the downloads of other users who are sharing the same physical links. Problem 34 For an application such as remote login (telnet and ssh), a byte-stream oriented protocol is very natural since there is no notion of message boundaries in the application. When a user types a character, we simply drop the character into the TCP connection. In other applications, we may be sending a series of messages that have inherent boundaries between them. For example, when one SMTP mail server sends another SMTP mail server several email messages back to back. Since TCP does not have a mechanism to indicate the boundaries, the application must add the indications itself, so that receiving side of the application can distinguish one message from the next. If each message were instead put into a distinct UDP segment, the receiving end would be able to distinguish the various messages without any indications added by the sending side of the application. Problem 35 To create a web server, we need to run web server software on a host. Many vendors sell web server software. However, the most popular web server software today is Apache, which is open source and free. Over the years it has been highly optimized by the open-source community. Problem 36 The key is the infohash, the value is an IP address that currently has the file designated by the infohash.   Chapter 3 Review Questions Call this protocol Simple Transport Protocol (STP). At the sender side, STP accepts from the sending process a chunk of data not exceeding 1196 bytes, a destination host address, and a destination port number. STP adds a four-byte header to each chunk and puts the port number of the destination process in this header. STP then gives the destination host address and the resulting segment to the network layer. The network layer delivers the segment to STP at the destination host. STP then examines the port number in the segment, extracts the data from the segment, and passes the data to the process identified by the port number. The segment now has two header fields: a source port field and destination port field. At the sender side, STP accepts a chunk of data not exceeding 1192 bytes, a destination host address, a source port number, and a destination port number. STP creates a segment which contains the application data, source port number, and destination port number. It then gives the segment and the destination host address to the network layer. After receiving the segment, STP at the receiving host gives the application process the application data and the source port number. No, the transport layer does not have to do anything in the core; the transport layer “lives” in the end systems. For sending a letter, the family member is required to give the delegate the letter itself, the address of the destination house, and the name of the recipient. The delegate clearly writes the recipient’s name on the top of the letter. The delegate then puts the letter in an envelope and writes the address of the destination house on the envelope. The delegate then gives the letter to the planet’s mail service. At the receiving side, the delegate receives the letter from the mail service, takes the letter out of the envelope, and takes note of the recipient name written at the top of the letter. The delegate then gives the letter to the family member with this name. No, the mail service does not have to open the envelope; it only examines the address on the envelope. Source port number y and destination port number x. An application developer may not want its application to use TCP’s congestion control, which can throttle the application’s sending rate at times of congestion. Often, designers of IP telephony and IP videoconference applications choose to run their applications over UDP because they want to avoid TCP’s congestion control. Also, some applications do not need the reliable data transfer provided by TCP. Since most firewalls are configured to block UDP traffic, using TCP for video and voice traffic lets the traffic though the firewalls. Yes. The application developer can put reliable data transfer into the application layer protocol. This would require a significant amount of work and debugging, however. Yes, both segments will be directed to the same socket. For each received segment, at the socket interface, the operating system will provide the process with the IP addresses to determine the origins of the individual segments. For each persistent connection, the Web server creates a separate “connection socket”. Each connection socket is identified with a four-tuple: (source IP address, source port number, destination IP address, destination port number). When host C receives and IP datagram, it examines these four fields in the datagram/segment to determine to which socket it should pass the payload of the TCP segment. Thus, the requests from A and B pass through different sockets. The identifier for both of these sockets has 80 for the destination port; however, the identifiers for these sockets have different values for source IP addresses. Unlike UDP, when the transport layer passes a TCP segment’s payload to the application process, it does not specify the source IP address, as this is implicitly specified by the socket identifier. Sequence numbers are required for a receiver to find out whether an arriving packet contains new data or is a retransmission. To handle losses in the channel. If the ACK for a transmitted packet is not received within the duration of the timer for the packet, the packet (or its ACK or NACK) is assumed to have been lost. Hence, the packet is retransmitted. A timer would still be necessary in the protocol rdt 3.0. If the round trip time is known then the only advantage will be that, the sender knows for sure that either the packet or the ACK (or NACK) for the packet has been lost, as compared to the real scenario, where the ACK (or NACK) might still be on the way to the sender, after the timer expires. However, to detect the loss, for each packet, a timer of constant duration will still be necessary at the sender. The packet loss caused a time out after which all the five packets were retransmitted. Loss of an ACK didn’t trigger any retransmission as Go-Back-N uses cumulative acknowledgements. The sender was unable to send sixth packet as the send window size is fixed to 5. When the packet was lost, the received four packets were buffered the receiver. After the timeout, sender retransmitted the lost packet and receiver delivered the buffered packets to application in correct order. Duplicate ACK was sent by the receiver for the lost ACK. The sender was unable to send sixth packet as the send win

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