Kuvempu Univeristy BSc(IT) 3rd semester exercise answer (Computer Networks II 34)

©NIIT Coordinator Guide – Computer Networks 1

CHAPTER-SPECIFIC INPUTS

Chapter One

Objectives

In this chapter, the students have learned to:

Define computer networks Discuss the evolution of computer networks

Focus Areas

This chapter explains the evolution of computer networks. You can start the session by asking the students that what do they understand of the Internet. Then tell them that the Internet is a network of computers. Now tell them that earlier computers were stand-alone units. But gradually it was required to share resources and exchange information amongst the computers. This led to the evolution of computer networks.

After discussing the evolution of network, explain the topics given in the Additional Inputs section

Additional Inputs

The following section provides some extra inputs on the important topics covered in the SG:

Types of Network

The most common types of network are as follows:

Local Area Network (LAN): LAN consists of a group of computers and associated devices, such as printers, modem, fax machine, and scanners. It provides networking capability to a group of computers that are located within a single physical location such as a building. You can connect LAN to MAN and WAN using routers. The data transmission speeds normally ranges from 1 to 100 megabits per second.

Metropolitan Area Network (MAN): MAN connects computers over a large geographical area, such as a city or a state. The size of MAN is intermediate between LAN and WAN. In addition, the data transmission speeds ranges between the speeds of LAN and WAN.

Wide Area Network (WAN): WAN can spread over a large physical distance, such as the Internet that spans across countries. You can use routers to connect a LAN to WAN. Normally, the transmission rates are 2 Mbps, 34 Mbps, 45 Mbps, 155 Mbps, 625 Mbps. Organizations that support WAN and use the Internet Protocol are called NSP (Network Service Providers). NSP forms the core of the Internet.

Network Configurations

The network configurations can be categorized as follows:

Peer-to-peer network: You can design a peer-to-peer network if you have a small network, which does not require security. A peer-to- peer network is also called as a workgroup. In a peer-to-peer network, the computers communicate with each other directly. This network type does not have a server, which manages the resources on the network. A single device such as a hub can centrally connect and control all the nodes on the network. Each computer on the network can share files and peripherals that are connected to the network. Peer-to-peer network is a low-cost solution. However, if you have to share large and complex files, such as databases or graphics, then the peer-to-peer network is not an efficient solution. In this situation, client-server network will be efficient.

Client-server network: If your network requirements include more than ten nodes, and work with large files such as databases, then you need to design a client-server network. The computers in a client-server network are connected to a server. The server enables


you to manage data, share information, and secure the network. You can set up a client-server network to support more than thousand user accounts, which are managed by administrators. To set up a client-server network, you need more resources than a peer-to-peer network.

Components of Network Infrastructure

The components of a network infrastructure are as follows:

Intranet: Intranet can be a corporate LAN or MAN. A private network enables distribution of information within an organization. LAN offers services such as access to databases and software distribution. In addition to file and printing services, the intranet uses applications that are associated with the Internet such as, Web Browsers, Web pages, FTP (File Transfer Protocols) sites, and mailing lists that can be accessed only by the users in the organization. Intranets are set up to provide quick access to employees within an organization.The basic protocol that is used for connectivity in intranet is TCP/IP. It is a set of networking protocols that are based on industry standards. The scalability of the protocol includes all sizes of networks.

Extranet: Extranet refers to an intranet that is partially accessible to authorized users out side an organization. A collaborative network uses Internet Protocol and the public telecommunication systems to securely access resources from the organization. It uses Internet technology to connect the business of an organization to suppliers, customers, or other business organizations. An extranet can be accessed only if you have a username and a password. Permissions are used to set various levels of accessibility for the outsiders. By doing so, you can provide limited access to information on your intranet.

Internet: Internet is a public and global communication network that can be used to access a Web site, upload files in the FTP, or use an e-mail message. You can connect to Internet using the following two methods: Dial-up connection can be used to connect to an ISP. Small organizations or Internet

users at home normally use this method. A dedicated line such as T1 can be used to connect to LAN. LAN can be configured to

connect to Internet. This method is used in large organizations that have their own node on the Internet or connect to an ISP.

FAQ

1. What is the use of sharing computers over the network?

Ans:

Sharing computers over the network enable you to exchange information and resources among computers and their users. Information exchange can be electronic mail or file transfer. Resource sharing can be usage of peripheral device, such as printer, connected to other computers.

2. What is a workgroup?

Ans:

A group of users who share information in a multi-user environment is known as a workgroup.

3. What is the WWW? Who invented it?

Ans:

World Wide Web or the WWW is a bank of information that is available over the network. Tim Berners-Lee invented it in 1989.


4. What is a Web browser?

Ans:

Web browser is a software for viewing websites. Internet Explorer and Netscape Navigator are some popular Web browsers.

5. What is a modem?

Ans:

A modem (MOdulate DEmodulate) is a device used to connect to the Internet through telephone lines.


Chapter Two

Objectives


Identify the layers of the OSI protocol architecture Identify the layers of the TCP/IP protocol architecture Compare OSI and TCP/IP Discuss about the organizations that standardized network protocols

Focus Areas

This chapter explains protocol architecture. You can start the session by telling the students that rules are defined for many activities. For example, there are rules to conduct business, which should be followed to get effective results. Similarly, there is a set of rules that coordinates the exchange of information over the network. Then tell the students that these rules are called protocols.

Stress upon the OSI protocol architecture. Explain in detail the function of each layer.

Stress upon the fact that TCP/IP is a suite of protocols. Explain the functionality of each layer in TCP/IP. Refer to the Additional Inputs section while explaining TCP/IP.

After explaining the two protocol architectures, ask the students to compare the two.

Additional Inputs


TCP/IP

TCP/IP is a connection-based protocol that provides a reliable flow of data between two computers. It is a suite of protocols and is based on industry standards. TCP/IP is a routable protocol capable of switching packets of data between subnets by using the address of the destination that is contained in the packet.

Features of TCP/IP

The features of TCP/IP are as follows:

TCP/IP supports large window support. If large number of data packets is broadcasted and the session is lengthy, then the size of the window is dynamically changed. This increases the bandwidth and enables more data packets to be broadcasted in the network at one time.

A TCP/IP host can notify and request the sender if any data packets are missing or corrupted during the broadcast. This enables the sender to send only the corrupted or missing data packets to the receiver again.

The time taken for a round-trip communication between the sender and receiver is called RTT (Round Trip Time). Estimating RTT enables you to estimate the packet transit times and adjust for optimum retransmission time for packets.

Advantages of TCP/IP

The advantages offered by TCP/IP are as follows:

It connects disparate hosts. In addition, TCP/IP is a routable protocol, which can be connected to different networks by using gateways.

It provides a reliable, robust, and scalable cross-platform client-server framework. It can establish a virtual private network (VPN) or extranet with Internet for providing remote

access.


Utilities of TCP/IP

The utilities of TCP/IP are as follows:

Data transfer utilities: TCP/IP provides support for data transfer protocols such as, FTP (File Transfer Protocol), HTTP (Hypertext Transfer Protocol), and CIFS (Common Internet File System).

Printing utilities: Printing can be done directly to IP-based printers. Diagnostic utilities: TCP/IP utilities such as, ipconfig, ping, and Nslookup can be used for

diagnosing TPC/IP related problems.

TCP/IP Protocol Architecture

The four layers of the TCP/IP protocol architecture are as follows:

Network Interface Layer: This layer encompasses the data link layer and physical layer of the OSI model. This layer is responsible for placing TCP/IP packets in the network and for sending TCP/IP packets. This layer includes LAN technologies such as Ethernet, and Token Ring and WAN technologies such as Frame Relay.

Network Layer: This layer encompasses the network layer of the OSI model. The Internet layer performs addressing, packaging, and routing. The five protocols that are implemented in this layer are: Internet Protocol (IP): The IP protocol routes the data packets from the source to the

destination. Address Resolution Protocol (ARP): The ARP protocol determines the hardware

address of the hosts. Reverse Address Resolution Protocol (RARP): The RARP protocol determines the

IP address of disk-less host systems. Internet Control Message Protocol Resolution (ICMP): The ICMP protocol sends

error messages if there is any TCP/IP related problems. Internet Group Management Protocol (IGMP): The IGMP protocol sends information

to the routers regarding the availability of the multicast groups. Transport Layer: This layer encompasses the transport layer of the OSI model. TCP

provides connection-oriented reliable communications for applications that typically transfer large amounts of data at one time. This is very useful if an acknowledgement for data received is required. The protocols of the transport layer are TCP and UDP (User Datagram Protocol). TCP provides a connection-oriented service whereas UDP provides a connectionless service.

Application Layer: This layer encompasses the application layer, presentation layer, and session layer of the OSI model. This layer is the top most layer in the TCP/IP model. Most of the TCP/IP utilities such as HTTP, FTP and Telnet run at the Application level.

Chapter Two Questions

Part 1

1. OSI is a protocol model standardized by the ___________

Ans.

ISO

2. OSI has ________ layers.

Ans.

Seven


3. The ________ layer is responsible for delivering data frames from one station to next without error.

Ans.

Data Link

4. The __________ layer uses the services provided by the network layer.

Ans.

Transport

5. At the receiving end, the data is passed to transport layer from ________.

Ans.

Network

6. Each layer adds ________ to the data it received from the upper layer.

Ans.

Its own information

7. TCP/IP has ___________ layers.

Ans.

Five

8. The layers not present in TCP/IP model are __________ and ___________.

Ans.

Session and Presentation

9. At transport layer the protocols defined by TCP/IP model are _________ and _________.

Ans.

UDP and TCP

10. RFC stands for ______________.

Ans.

Request for Comments

Part 2

1. Briefly explain what is meant by protocol and protocol architecture.

Ans.

A protocol is software that resides either in a computer's memory or in the memory of a transmission device, such as a network interface card. Protocols govern format, timing, sequencing, and error control. A protocol is used for communication between entities in different systems. For two entities to communicate successfully, they must speak the same language. What is communicated, how it is communicated, and when it is communicated must conform to some mutually acceptable convention or protocol between the entities involved. The following are the key elements of a protocol:


Syntax: Includes such things as data format and signal levels Semantics: Includes control information for coordination and error handling Timing: Includes speed matching and sequencing

Protocol architecture is a structured set of modules that implements the communication function between computers.

2. Briefly explain the OSI architecture.

Ans.

Open System Interconnection (OSI) protocol architecture, defined by the International Standards Organization (ISO), includes a set of protocols to define and standardize the data communication process. It defines how data communications take place in the real world and what protocols should be used at each layer.

The following are the salient features of OSI protocol architecture:

Layered architecture: The OSI model is composed of seven ordered layers: Physical (layer 1), Data link (layer 2), Network (layer 3), Transport (layer 4), Session (layer 5), Presentation (layer 6), and Application (layer 7). Within a single computer system, each layer calls upon the services of the layer just below it. For example, Network layer (layer 3) uses the services provided by the Data link layer (layer 2) and it provides services to Transport layer (Layer 4).

Peer-to-peer process: The processes on each system that communicate at a given layer are called peer-to-peer processes. Communication between machines is therefore a peer-to-peer processes communicating to each other using protocols appropriate to the given layer. At the physical layer, communication is direct. However, at the higher layers, communication must move down through the layers on Device A, over to the device B, and then back up through the layers. Each layer in the sending device adds its own information to the message it receives from the layer just above it and passes the whole package to the layer just below it. Headers are added to the data at layers 6,5,4,3 and 2. Trailers are usually added only at layers 2.

Layer organization: The seven layers can be thought of as belonging to three subgroups. The layers 1, 2, and 3, called the Physical, Data Link, and Network layers respectively, are the network support layers. They deal with the physical aspects of moving data from one device to another device. The layers 5, 6, and 7, called Session, Presentation, Application respectively, are the user support layers. They allow interoperability among unrelated software systems. The layer 4 called the Transport layer links the two subgroups and ensures that the upper layers can use the data that the lower layers have transmitted. The upper OSI layers are implemented in software. The lower OSI layers are a combination of hardware and software, except for the physical layer, which is mostly hardware.

3. What are the functions of the network layer? Explain.

Ans.

The network layer is responsible for the source-to-destination delivery of a packet possibly across multiple networks (links). Whereas data link layer oversees the delivery of the packet between two systems on the same network (link), the network layer ensures that each packet gets from point of origin to its final destination. If the two systems are connected to the same link then there would be no requirement for the network layer. However if the two systems are attached to different networks with connecting devices (known as routers) between the networks, then this layer plays a crucial role in getting the packet from source to destination.

The Network layer provides for the transfer of data in the form of packets across the communication networks. It establishes, maintains, and terminates logical and physical connections across multiple interconnected networks. A key aspect of this transfer is the routing of packets from the source to the destination machine typically traversing a number of transmission links and network nodes where routing is carried out. Routing is the process by which a path is selected out of many available paths to the destination so that data packet


reaches the destination fast, efficiently, reliably as required. This function makes the network most complex layer in the reference model.


4. Explain the functionalities and the services offered by the transport layer of the OSI model.

Ans.

The transport layer is designed to provide the “transparent transfer of data from a source end open system to a destination end open system," according to the OSI Reference Model. The transport layer establishes, maintains, and terminates communications links between two machines.

The Transport layer ensures data is successfully sent and received between two end systems. If data is sent incorrectly, this layer has the responsibility to ask for retransmission of the data. Also it ensures data are passed onto the upper layers in the same order in which they were sent. Specifically, it provides a reliable, network-independent message-interchange service to the top three application-oriented layers. This layer acts as an interface between the bottom and top three layers. By providing the session layer with a reliable message transfer service, it hides the detailed operation of the underlying network from the session layer.

Some of the important services that are performed by the transport layer in order to meet the above requirements are:

Service-point addressing: Computers often run several programs at the same time. For this reason, source-to-destination delivery means delivery not only from one computer to the next but also from a specific process (a running program) to a specific process at the receiving end. The transport layer header must therefore include a type of address called service-point address (or port address). The network layer gets each packet to the correct computer; the transport layer gets the entire message to the correct process on that computer.

Segmentation and reassembly: If the message sent by the application program at the transmitting end is huge, there would be problem in moving that from the transmitting system to the receiving system as many of the physical network underneath would impose restriction on the maximum size of the data that can be transferred at one time. In order to over-come this restriction, transport layer splits the incoming message into segments (each of transferable size) at the transmitting system and reassembles them at the receiving end.

Apart from the above, other tasks performed by the transport layer are Connection control, Flow control, and Error control.

5. What are the functionalities of the session layer of OSI model?

Ans.

The session layer organizes and synchronizes the exchange of data between application processes. It works with the application layer to provide simple data sets called synchronization points that let an application know how the transmission and reception of data are progressing. In simplified terms, the session layer can be thought of as a timing and flow control layer.

The session layer is the network dialog controller. It establishes, maintains, and synchronizes the interaction between communicating systems. Specifically its responsibilities include:

Dialog control: This allows two systems to enter into a dialog. It allows the communication between two processes to take place either in half-duplex (one way at a time) or full duplex (both ways simultaneously).

Synchronization: The session layer allows a process to add checkpoints (synchronization points) into a stream of data. For example, if a system is sending a file of 2,000 pages, it is advisable to insert checkpoints after every 100 pages to ensure that each 100 page is received and acknowledged independently. The reception of acknowledgement for these checkpoints ensures to the sending system that data up to the corresponding checkpoint is received properly by the receiving end system.

An example where session layer plays a crucial role is the file transfer application when used to transfer huge files (downloading of huge files). When huge files are transferred across the network, if the network connection speed is not high, it would take very long time for transfer. There is every chance that network connection would break during this transfer and the user is required to start a fresh to do the file transfer, and there is no guarantee that the same would


not repeat, making it sometime practically impossible to download very huge files. If the checkpoints are used by the file transfer application then, this problem can be sorted out easily. Suppose the connection fails during the 1034th page, then after reconnecting we can proceed from the last checkpoint, i.e. from page 1001. During retransmission time the only pages that are resent for the second time are 1001 to 1034 as pages till 1000 are received and acknowledged. If checkpoints were not there, we have to start from page 1 every time we try to reconnect.

6. What are the layers present in TCP/IP model? Explain the layer, which provides the transport functionalities along with the protocols defined for that layer.

Ans.

The TCP/IP model is made up of five layers:

Physical layer Data Link layer Network layer (IP) Transport layer (UDP and TCP) Application layer

The layer that provides the transport functionalities along with the protocols defined for that layer is Network layer. The network layer is concerned with access to and routing data across a network for two end systems attached to multiple interconnected networks. The Internet Protocol (IP) is used at this layer to provide the routing function across multiple networks. This protocol is implemented not only in the end systems but also in routers. A router is a processor that connects two or more networks and whose primary function is to relay data from one network to the other on its route from the source to the destination end system.

The Internet Protocol (IP) is the transmission mechanism used by the TCP/IP protocols. It is an unreliable and connectionless datagram protocol – i.e. it provides a best-effort delivery service. The term best-effort means that IP provides no error checking or tracking. IP assumes the unreliability of the underlying layers and does its best to get the get a transmission through to its destination, but with no guarantees. IP transports data in packets called datagrams each of which is transported separately. Datagram can travel through different routes and can arrive out of sequence or be duplicated. IP does not keep track of routes and has no facility to reorganize datagrams once they arrive at their destination. These limitations should not be considered as the weakness of the protocol stack. It is intentional, to get the maximum efficiency. The purpose at this layer is to provide the bare-bone transmission functions that free the user to add only those facilities necessary for a given application.

At this network layer TCP/IP supports the Internetworking Protocol (IP). IP, in turn, contains four supporting protocols: ARP, RARP, ICMP and IGMP.

7. Make a brief comparison between OSI and TCP/IP models.

Ans.

The OSI architecture is a de-jure (according to law) standard. The focus in the OSI world has always been more on the standard than the implementation of the standard. The OSI model is used as a reference model to make comparisons. The OSI reference model was devised before the protocols were implemented. The ordering means that the model was not biased toward one particular set of protocols, which made it quite general.

The TCP/IP architecture is a de-facto (in reality) standard. With the TCP/IP model, the protocols came first, and the model was just a description of the existing protocols. The TCP/IP model is not used to describe other models.


FAQ

1. What is the difference between network protocol and protocol architecture?

Ans:

A network protocol is a set of rules for communication between computers. Protocol architecture is a structured set of modules that implements the communication function between computers.

2. What does OSI stands for?

Ans:

OSI stands for Open System Interconnection.

3. What is OSI protocol architecture?

Ans:

OSI protocol architecture, defined by the International Standards Organization (ISO), includes a set of rules that are used to define and standardize the data communication process. OSI protocol architecture defines how data communications take place in the real world and what protocols should be used at each layer.

4. How many layers are present in the OSI protocol architecture?

Ans:

The OSI model is composed of seven ordered layers: Physical (layer 1), Data link (layer 2), Network (layer 3), Transport (layer 4), Session (layer 5), Presentation (layer 6), and Application (layer 7).

5. What is TCP/IP protocol architecture? How many layers are present in the TCP/IP protocol architecture?

Ans:

TCP/IP is the most widely used architecture and is a result of protocol research and development work conducted on the experimental network, ARPANET. The TCP/IP model is made up of five layers: Physical layer, Data link layer, Network layer, Transport layer, and Application layer. The first four layers provide physical standards, network interface, internetworking and transport functions that correspond to the first four layers of the OSI model. However, the three top layers of the OSI model are represented in the TCP/IP by a single layer called Application layer.

6. Explain the data flow from device A to device B through different OSI layers.

Ans:

The sequence of data transfer from device A to device B through different OSI layers is:

Application layer of device A Presentation layer of device A Session layer of device A Transport layer of device ANetwork layer of device A Data link layer of device A Physical layer of device A Physical layer of device B Data link layer of device B Network layer of device B Transport layer of device B Session layer of device B Presentation layer of device B Application layer of device B.


The preceding concept is displayed in the following figure:

Data Flow from Device A to Device B through Different OSI Layers

7. What is the difference between physical layer and data link layer?

Ans:

The physical layer is the lowest layer of the OSI model and deals with the "mechanical, electrical, functional, and procedural means" required for transmission of data. The physical layer is responsible for data transmission from one host to another. The data link layer provides for the control of the physical layer, and detects and corrects errors that can occur during transmission. The data link layer adds header and trailer at the transmitting end and removes the same at the receiving end.

8. Which layer is responsible for encoding, encrypting, and compressing data?

Ans:

The presentation layer is responsible for encoding, encrypting, and compressing data.


9. What is the difference between OSI model and TCP/IP model?

Ans:

The OSI model is a de-jure (according to law) standard. The focus in the OSI world has always been more on the standard than the implementation of the standard. The OSI model is used as a reference model to make comparisons. The OSI reference model was devised before the protocols were implemented. The ordering means that the model was not biased toward one particular set of protocols, which made it quite general. The TCP/IP model is a de-facto (in reality) standard. With the TCP/IP model, the protocols came first, and the model was just a description of the existing protocols. The TCP/IP model is not used to describe other models.

10. What is the use of the session layer?

Ans:

The session layer organizes and synchronizes the exchange of data between application processes. It works with the application layer to provide simple data sets called synchronization points that let an application know how the transmission and reception of data are progressing. In simplified terms, the session layer can be thought of as a timing and flow control layer.


Chapter Three

Objectives


Define addressing techniques in TCP/IP Define Internet Protocol Define Internet addressing Define private networks

Focus Areas

This chapter explains IP addressing. You can start the session by telling the students that while requesting for some website, they must have seen some numbers separated with dots in the status bar of the browser. This address is known as the IP address. Tell the students that each system on the Internet has a unique IP address.

You should stress upon the various IP address classes. Explain the three levels of addressing used in TCP/IP, namely Physical address, Internetwork (IP) address, and Transport or Port address. Discuss the ways of determining the class of an address. Refer to IP addressing in the Additional Inputs section.

Explain the concepts of unicast, multicast, and broadcast addresses. Explain the reserved blocks of address for private networks. Discuss the need for private networks. If time permits, explain the other topics given in the Additional Inputs section.

Additional Inputs


IP Addressing

A TCP/IP host is identified by a logical and unique IP address. It identifies the physical location of the host in the network and is similar to a street address that identifies a house on a city block.

The IP address consists of two parts, netid and hostid:

Netid: The netid is also called network address. The netid identifies all hosts that are located on the same physical network. All the hosts of one physical network are assigned the same netid to facilitate communication among them. The netid must be unique in the internetwork.

Hostid: The hostid is also called host address. The hostid identifies a host in the network. A hostid is unique for every computer in the netid.

Note

You cannot assign 127 as a netid in the network. This ID is reserved for loopback and diagnostic functions. In addition, you cannot have netid and hostid as 255. This ID is reserved for IP broadcast functions.

The IP address is a 32-bit number that uniquely identifies a host such as computer, printer, and router. The IP address is expressed in a dotted-decimal format and the IP address consists of four octets. Each octet is separated by a dot. 191.168.3.24 is an example of IP address.


Each octet can be converted to decimal system using a base 10 numbering system. The following table lists the IP address in binary and dotted decimal format:

Binary Format Dotted Decimal Format

10101011 10101000 00000011 00011000

191.168.3.24

A bit that is set to one can be converted to a decimal value. The following graphic illustrates the conversion of the number 191 to binary digits:

Binary Octet of 191=1011111

The conversion of binary digits to a decimal number is illustrated in the following figure:

Conversion of Binary Digit to a Decimal Number

Note

You cannot have all bits within the netid set to 0. All 0’s in the netid are used to denote a specific host on the local network. The data from these hosts will not be routed.

Subnets

In a physical network, all the hosts that are bounded by IP routers share the same IP traffic. This is because all the hosts are a part of a single domain. It is not possible to have millions of hosts in a single broadcast domain. You can divide the network to smaller domains called subnets. This process is called subnetting. Each subnet has a unique subnetted netid. Using bits from the hostid portion of the original class-based netid create the subnetted ID.

However, if you use more bits than needed, then you can have more subnets and few hosts. If you use fewer bits than needed, then you can have less subnets and more hosts.


Consider an example. You are using Class B IP addresses in your network. Routers connect the different parts of the network. Class B can have up to 65,536 hosts. The network becomes saturated with IP traffic if you have numerous hosts on the network. The following figure displays the Class B network connected to the internetwork through a router:

Router Connecting the Class B Network to the Internetwork

The class B network of 124.105.0.0 can be subnetted by utilizing the first eight host bits for the new subnetted netid. When you subnet the 124.105.0.0, you create separate network segments called subnets. You specify unique subnet IDs for these subnets. The router present on the network identifies the subnet ID. Therefore, it can route IP traffic to the appropriate subnet instead of forwarding the IP traffic to the whole network. The following figure displays subnets of Class B network connected to the internetwork through a router:

Router Connecting the Subnets of Class B Network to the Internetwork

Subnet Mask

A subnet mask is a 32-bit address that is used for distinguishing the netid from a hostid in the IP address. In addition, it is used for specifying if the IP address of a host is located in a local network or in a remote network.

Note

A default subnet mask is specified for TCP/IP networks that do not have subnets. Although there is a single network, all TCP/IP hosts require a subnet mask. The default subnet mask depends on the type of class IP addresses used on the network.

The following table lists the bits used for subnet mask and the dotted decimal notation for each class IP address:

Class Binary FormatDotted

Decimal Format

Class A IP address

11111111 00000000 00000000 00000000

255.0.0.0

Class B IP address

11111111 11111111 00000000 00000000

255.255.0.0

Class C IP address

11111111 11111111 11111111 00000000

255.255.255.0


ANDing Process

ANDing is a process that is used by TCP/IP to determine if the destination of the data packet is to a local network or a remote network. Before a data packet is sent to the destination, the IP address of the destination is ANDed with the subnet mask. If the results match, then IP routes the packets to the appropriate host. If the results do not match, then the packet is routed to the IP router.

To perform the ANDing process, TCP/IP compares each bit in the IP address of the destination host to the corresponding bit in the subnet mask. If both the bits are set to one, then the resulting bit is one. In other cases, the resulting bit is zero. The following table lists the resulting bit after combining two binary bits:

Combining the bits Resulting Bit

1 AND 1 1

1 AND 0 0

0 AND 1 0

0 AND 0 0

Consider an example. The network of your organization consists of two subnets, Subnet A and Subnet B. The TCP/IP host in Subnet A needs to send a packet to another TCP/IP host in Subnet A. When TCP/IP is initialized, the IP address of the destination is ANDed with the subnet mask. Since the subnet mask of the destination is same as that of the source, the resulting bit is one. Therefore, TCP/IP determines that the packet is destined for the local network, Subnet A.

Defining a Subnet Mask

Consider an example. You have Class B IP addresses in your network. The netid that you have to use is 172.10.0.0. You are required to divide the network into subnets for reducing the network congestion. You decide to divide the network into five subnets.

To define the subnet mask for each subnet, you need to perform the following steps:

1. Count the number of bits that are required for representing the number of physical segments in binary format. You have five subnets on the network. To represent five in binary, you require three bits. Therefore, the binary value is 011.

2. In this scenario, you require three bits. Therefore, you configure the first three bits of the hostid as subnet ID. This would mean that the values for first three bits of the third octet would be one, as shown in the figure above. The decimal value for 11100000 is 224. Therefore, the subnet mask for the above-specified bits is 255.255.224.0.

Defining Subnet IDs

Consider the subnet mask 255.255.224.0. The binary value for this subnet mask is 11111111 11111111 11100000 00000000. The bits that were used for defining the subnet mask are the first three bits of the third octet. By varying the values of these three bits, you can have the following list of combinations:

00000000=0


00100000=32 01000000=64 01100000=96 10000000=128 10100000=160 11000000=192 11100000=224

The bits that use all 0s or all 1s cannot be used. This is because all 0s are invalid IP addresses and with all 1s are subnet IDs. All 0s indicate ‘this network only’ and all 1s match the subnet mask. Now, convert the bits to decimal for each subnet. You can use this value to define the range of hostids for a subnet. The following list displays some of the ranges of IP addressees that you can have using the 255.255.224.0 as the subnet mask:

172.10.32.1 to 172.10.63.254 172.10.64.1 to 172.10.95.254

Defining Hostids

The first three bits in the third octet of the hostid are used for defining the subnet. Therefore, you can calculate the range of hostids by changing the rest five bits to 11111. The decimal value of 00111111 is 32.

The third octet in the subnet mask indicates the starting value in the range of the values in the third octet of the IP address. The ending value in range of IP address is one less than the starting value.

The following table displays the range of hostids possible in a Class B network:

Bit values of the third octet in the

Subnet Mask

Decimal value

Starting range value

Ending range value

00100000 32 x.y.32.1 x.y.63.254

01000000 64 x.y.64.1 x.y.95.254

01100000 96 x.y.96.1 x.y.127.254

10000000 128 x.y.128.1 x.y.159.254

10100000 160 x.y.160.1 x.y.191.254

11000000 192 x.y.192.1 x.y.223.254

For a Class B Network, that uses 14 bits for the hostid, you can have 16,384 hosts on the network. The decimal value of 11111111111111 (14 bits that are used for host ID) is 16,384.

Supernetting

Supernetting enables you to have multiple IP networks on the same network interface. This is in contrast with subnetting in which you have a single IP network on multiple network interfaces. If the IP addresses are contiguous, then you can use a supernet.


In subnetting, you borrow bits from the hostid and give it to the netid. The following figure illustrates the netid borrowing a bit from the hostid:

Netid Borrowing A Bit from Hostid

In supernetting, you borrow bits from the netid and give it to the hostid. The following figure illustrates the hostid borrowing a bit from the netid:

Hostid Borrowing A Bit from Netid

To create a supernet by combining two Class C networks, you need to have the third octet in the first IP address divisible by two. For example, the networks 198.41.17.0 and 198.41.18.0 cannot be combined into a supernet. However, the networks 198.31.16.0 and 198.31.17.0 can be combined into a supernet.

Chapter Three Questions

1. What is an IP address?

Ans.

Each system on the Internet is recognized by a unique IP address. It is a 32 bit (4 bytes) long number. It uniquely identifies the host that is connected to the Internet. The Internet Authority grants these addresses. If the local network is not connected to the Internet then it is possible to have IP address of ones choice. However, there is a subset of addresses available for that purpose and it is recommended to use those addresses only. It is possible for a host to have multiple IP addresses.

An IP address has two parts. The one that identifies the network is known as netid. The other one that identifies the host is known as hostid.

2. What are the different classes of IP addresses available in IPV4?

Ans.

There are five different IP classes: A, B, C, D, and E. these are designed to cover the needs of different types of organization.

Class AIf the first bit is 0, then the given address is a class A address. Remaining portion of the first byte define the netid. Byte 2 to 4 determines the hostid. Class A addresses are designed for organizations that are having huge number of hosts connected to their network.

Class BIf the first two bits of the IP address are 10, then the given address is a Class B address. Here hostid is 16 bits long and the rest 14 bits are for the netid. These addresses are designed for midsize organizations.

Class CIf the first three bits of the IP address are 110,then the given address is a Class C address. The next 21 bits define the netid and the remaining 8 bits determine the hostid. These are addresses are designed for small organizations.

Class D


This address is meant for multicasting. This does not have netid or hosted. The first four bits (1110) define the class here.

Class EThis is a class reserved for special purposes by the Internet authority. The first four bits of this class are 1111. There is no netid or hostid in this class.

3. What is a subnet?

Ans.

A subnet is one of the networks that is created when a large physical network is divided into smaller networks.

4. What is a subnet mask? Why is it necessary?

Ans.

A subnet mask is a 32-bit mask used to distinguish the network ID from the host ID.

In the process of routing, when a TCP/IP host sends data to another TCP/IP host, the IP at the source host determines whether the destination host is local or remote. To determine this, IP uses the AND operation. IP ANDs the source IP address with the source subnet mask and the destination IP address with the source subnet mask. If the ANDing results of the source and destination addresses match, the destination host is local. If they do not match, then the destination host is remote.

5. What are private networks?

Ans.

Internet authorities have reserved three blocks of address from which an organization can choose the netid of their choice. These addresses are

Class A: (netid)10.0.0 Class B: (netid)172.16 to 172.31 Class C: (netid)192.168.0 to 192.168.255

Addresses with these netid are private addresses.

6. What is IP multicasting?

Ans.

When a data packet is sent from an individual source to a group of destinations, a multicast communication takes place. A multicast address is a Class D address. The whole address defines a multicast group id. The multicast address can never be a source IP address. It can only be a destination addresses in an IP datagram packet.

Some of the multicast addresses are:

224.0.0.1: All systems on this SUBNET 224.0.0.7: ST routers 224.0.1.7: Audio news

FAQ

1. What are IP addresses? Briefly explain them.

Ans:

IP addresses are necessary for universal communication services that are independent of underlying physical network. An IP address is currently a 32-bit (4 byte) address, which can


uniquely identify a host connected to the Internet. No two hosts on the Internet can have the same IP address. In addition, IP addresses are defined such a way so that one can easily identify the network to which it is connected so that routing becomes easy. For example, 128.11.3.31 is an IP address.


2. What is a port address?

Ans:

A port address identifies the correct process (a running program on the computer) on the source and destination host systems to which the data actually corresponds. For example, data sent by the FTP client process from system A should reach the FTP server process at the system B. It should not reach the MAIL server process running on the system B. So it is not only crucial to identify the end systems to which the data meant but the end processes has to be identified also. To achieve this, different processes are labeled uniquely. In TCP/IP, this labeling is called as port address. A port address is 16 bits long (2 byte).

3. Explain the IP datagram format.

Ans:

The unit of transfer in an IP network is called an IP datagram. IP datagram has a predefined format in which a datagram has to be filled so that all the systems that receive it, understand its content without any ambiguity. It consists of an IP header and data relevant to higher-level protocols. A datagram is a variable length packet consisting of two parts, header and data. The header can be from 20 to 60 bytes long and contains information essential for routing and delivery. The length of the data part varies from packet to packet but the total length of the IP datagram should be within 65,535 bytes.

4. What are IP address classes? Classify the IP address classes and briefly explain them.

Ans:

The InterNIC (The Internet's Network Information Center) allocates the IP addresses. The bits that correspond to the network ID are set to one. The bits that correspond to the host ID are set to zero. These IP addresses are divided into Class A, Class B, Class C, Class D, and Class E. End users do not use the Class D and Class E IP addresses.

Class A IP address: You can have Class A IP addresses in your network if you have large number of hosts. The high-order bit is always set to binary 0. The next seven bits complete the network ID. The last three octets represent the host ID. The default subnet mask used in the Class A IP address is 255.0.0.0. In addition, they have the first octet from 0-126. The number of networks possible in class A IP addresses is 128. The number of hosts possible per network is 16,777,216. For example, 11.52.36.8 is a Class A IP address.

Class B IP address: You can have Class B IP addresses if your organization has medium-sized networks. The two high-order bits are always set to binary 1 0. The next 14 bits complete the network ID. The last two octets represent the host ID. The default subnet mask used in the Class B IP address is 255.255.0.0. In addition, they have the first octet from 128-191. The number of networks possible in class B IP addresses is 16,384. The number of hosts possible per network is 65,536. For example, 172.52.36.8 is a Class B IP address.

Class C IP address: Class C IP addresses are used in small local area networks. The three high-order bits are always set to binary 1 1 0. The next 21 bits complete the network ID. The last octet represents the host ID. The default subnet mask used in the Class C IP address is 255.255.255.0. In addition, they have the first octet from 192-223. The number of networks possible in class A IP addresses is 2,097,152. The number of hosts possible per network is 256. For example, 192.52.56.8 is a Class C IP address.

Class D IP address: Class D IP addresses are reserved for IP multicast addresses. The four high-order bits are always set to binary 1 1 1 0. The remaining bits represent the specific group in which the client participates. Microsoft supports class D IP addresses for multicast applications to multicast data to hosts on an internetwork. These hosts are designed to receive multicast data.

Class E IP address: The class E IP addresses are for experimental purposes. They are not in use. The four high-order bits are always set to binary 1 1 1 1.


Chapter Four

Objectives


Define routing Identify the types of routing Define ARP Define RARP Define ICMP

Focus Areas

This chapter explains protocols and routing. You can start the session by telling the students that information over a network travels in the form of packets. Sending information through packets requires the knowledge of various protocols.

Explain the difference between direct and indirect routing. Refer to routing protocols given in the Additional Inputs section.

Describe the fields of an IP header. Explain checksum calculation.

Ask the students to differentiate between ARP and RARP. Discuss the format of an ARP packet.

Explain the two categories of ICMP messages. Explain the types of messages present in each category.

Additional Inputs


Routing Protocols

Routing is the process of transmitting information from one computer in a network to another computer in the same or a different network. The OSI model provides a conceptual framework about how communication takes place between computers. Internet protocols and routing protocols govern the actual process, which takes place at the Network layer of the Open System Interconnection (OSI) reference model.

Most networks are IP-based and use routing protocols for transmitting data. The routing of data packets, both within a network and between networks, is made possible by a device called the Router. A Router receives the data packet from a host device, looks at the IP address, and forwards the packet to another router or destination device.

What are Routing Protocols?

Routing protocols are network layer protocols that help routers identify the appropriate path for transmitting data packets between routers. Routers use routing tables to determine the network path in which a data packet can travel from a host computer to a destination computer. Routing tables are built using routing protocols.

Routing protocols can be classified based on the type of networks or systems in which they operate. Routing protocols can be used to connect devices within an autonomous system or between two or more autonomous systems.

Using this criterion, you can classify the routing protocols into the following two categories:

Interior Gateway Protocol (IGP) Exterior Gateway Protocol (EGP)


Static Routing Protocol

In static routing, you manually build the routing tables of all the routers in a network by configuring the routers to forward data packets in predetermined routes. Routers do not exchange any information regarding the network topology.

Static routing is used in small TCP/IP networks with two or more routers and a few subnets. Static routing can also be used if there is a predefined route over which data packets can travel. For example, if your ISP has provided a leased line, you can use static routing between the customer and ISP routers.

Static routing has the following advantages:

Easy to administer Easy to troubleshoot Provides a secure environment Provides efficient utilization of resources

However, static routing also has some disadvantages:

Requires substantial maintenance and coordination effort Cannot adapt to changing network topology

Dynamic Routing Protocol

In dynamic routing, routing protocols are deployed to automatically build and maintain the routing tables of all the routers in a network. It is used in large networks with multiple routers that can exchange information with each other to build routing tables. The information is exchanged using a routing protocol.

Chapter Four Questions

Part 1

1. Indicate the length of following addresses in terms of bytes:

a. Physical address (for Ethernet)b. IP addressc. Port address

Ans.

a. Physical address (for Ethernet): 6 bytes

b. IP address: 4 bytes

c. Port address: 2 bytes

2. Identify the class of the following IP addresses.

a. 229.15.76.110b. 191. 32.78.27c. 4.5.6.7

Ans.

a. 229.15.76.110: Class D

b. 191. 32.78.27: Class B

c. 4.5.6.7: Class A


3. A device having more than one IP address is known as _________ device.

Ans.

Multihomed

4. Change the following IP address from binary to decimal point notation:

011101001 01011101 00101101 10001010

Ans.

233.93.45.138

5. A device whose primary function is routing the IP datagrams over the physical network is called as ___________.

Ans.

Router

6. The two types of routing are _________ routing and _________ routing.

Ans.

Direct and indirect

7. The minimum length of IP datagram is __________ bytes.

Ans.

576

8. The IP datagram header has the IP addresses of _______ and _________ systems.

Ans.

Source, destination

9. In the IP datagram __________ field is used for the error detection purpose.

Ans.

Checksum

10. Protocol used to find out the physical address of a destination host from its IP address is known as _______.

Ans.

ARP

11. Protocol used for reporting the errors occurred during the datagram transfer is ________.

Ans.

ICMP


Part 2

1. Write in detail about the different addressing mechanisms used in the TCP/IP model.

Ans.

TCP/IP model has the following three different levels of addressing:

Physical address: This is the address of a node at the data link layer, as defined by the LAN or WAN. It is included in the frame sent by the data link layer. This address determines the host system on a particular network. The TCP/IP does not define the size and format of the physical address and it depends on the kind of the network. For example, Ethernet LAN uses 6 byte (48 bit) physical address that is imprinted into the Network Interface Card.Physical addresses can be either unicast (Single recipient), multicast (a group of recipients) or broadcast (received by all in the network). However not all networks supports these. Ethernet – one of the popular LAN – supports all of these.

Internet address: Internet addresses are necessary for universal communication services that are independent of underlying physical network. Physical networks have different addressing format depending upon the network technology used. In addition, the addressing does not have any component using which one can identify the network to which it is connected; which is essential for the routing purpose. The Internet addresses are designed for this purpose. An Internet address is currently a 32 bit (4 byte) address, which can uniquely identify a host, connected to the Internet. No two hosts on the Internet can have the same IP address. In addition, Internet addresses are defined such a way that given an IP address one can easily identify the network to which it is connected so that routing becomes easy.

Port address: The IP address and the physical address identify the source and the destination systems. They do not identify the process (a running program on the computer) on these systems to which the data actually corresponds. The final objective of the Internet communication is providing a communication link between two processes running on two different systems. For example, data sent by the FTP (File Transfer Protocol) client process from system A should reach the FTP server process at the system B. It should not reach the MAIL server process running on the system B. So it is not only crucial to identify the end systems to which the data meant, but also the end processes are also to be identified. To achieve this, different processes are labeled uniquely. In TCP/IP, this labeling is called as port address. A port address is 16 bits long (2 byte).

2. Explain the different classes of IP addresses. Write briefly about, given an IP address how to find its class.

Ans.

The InterNIC (The Internet's Network Information Center) allocates the IP addresses. The bits that correspond to the network ID are set to 1. The bits that correspond to the host ID are set to 0. These IP addresses are divided into Class A, Class B, Class C, Class D, and Class E. End users do not use the Class D and Class E IP addresses.

Class A IP address: You can have Class A IP addresses in your network if you have large number of hosts. The high-order bit is always set to binary 0. The next seven bits complete the network ID. The last three octets represent the host ID. The default subnet mask used in the Class A IP address is 255.0.0.0. In addition, they have the first octet from 0-127. The number of networks possible in class A IP addresses is 128. The number of hosts possible per network is 16,777,216. For example, 11.52.36.8 is a Class A IP address.

Class B IP address: You can have Class B IP addresses if your organization has medium-sized networks. The two high-order bits are always set to binary 1 0. The next 14 bits complete the network ID. The last two octets represent the host ID. The default subnet mask used in the Class B IP address is 255.255.0.0. In addition, they have the first octet from 128-191. The number of networks possible in class B IP addresses is 16,384. The number of hosts possible per network is 65,536. For example, 172.52.36.8 is a Class B IP address.

Class C IP address: Class C IP addresses are used in small local area networks. The three high-order bits are always set to binary 1 1 0. The next 21 bits complete the network


ID. The last octet represents the host ID. The default subnet mask used in the Class C IP address is 255.255.255.0. In addition, they have the first octet from 192-223. The number of networks possible in class A IP addresses is 2,097,152. The number of hosts possible per network is 256. For example, 192.52.56.8 is a Class C IP address.

Class D IP address: Class D IP addresses are reserved for IP multicast addresses. The four high-order bits are always set to binary 1 1 1 0. The remaining bits represent the specific group in which the client participates. Microsoft supports class D IP addresses for multicast applications to multicast data to hosts on an internetwork. These hosts are designed to receive multicast data.

Class E IP address: The class E IP addresses are for experimental purposes. They are not in use. The four high-order bits are always set to binary 1 1 1 1.

There are two ways of determining the class of an address depending on the format it is represented.

If the address is given in the form of binary then depending upon the first few bits one can identify the class.

If the first bit is 0 then it is Class A If the first 2 bits are 10 – Class B If the first 3 bits are 110 – Class C If the first 4 bits are 1110 – Class D If the first 4 bits are 1111 – Class E

However mostly the addresses are given in the decimal notation, which requires following method for determining the class.

Class A: first number is between 0 and 127 Class B: first number is between 128 and 191 Class C: first number is between 192 and 223 Class D: first number is between 224 and 239 Class E: first number is between 240 and 255

3. Write briefly about unicast, multicast and broadcast addresses

Ans.

A data packet can be sent to one system, more than one system, or to all the systems in the network. The Internet addressing mechanism has provision for all of these.

Unicast Addresses: Unicast communication is one-to-one. When a data packet is sent from a source system to an individual destination system, a unicast communication takes place. All system on the Internet should have one unique unicast address. Unicast addresses belong to class A, B or C.

Multicast Addresses: Multicast communication is one-to-many. When a data packet is sent from an individual source to group of destinations, a multicast communication takes place. A multicast address is a Class D address. The whole address defines a multicast group id. A system on the Internet can have one or more multicast addresses (in addition to its unicast address or addresses). If a system intends to participate in a particular multicast group then it should enable corresponding multicast address in its software. Note that the multicast addresses (or Class D) can never be a source IP address; it can only be a destination address in an IP datagram packet. Some of the multicast addresses are listed below:

224.0.0.1: All systems on this SUBNET 224.0.0.7: ST routers. 224.0.1.7: Audio news 224.0.1.11: IETF-1-Audio 224.0.1.12: IETF-1 Video

Broadcast Addresses: Broadcast communication is one-to-all. The Internet allows broadcasting only at the local network level. There are two types of broadcasting allowed.

Limited broadcast: all 1s in the IP address. Both netid and hostid in this case are all 1(in decimal notation 255.255.255.255). This identifies all the hosts connected to the local


network. An IP datagram packet with destination address as 255.255.255.255 should be received by all the systems connected to the local network.

Direct broadcast: This identifies all the hosts connected to a particular network (need not be local network as in the case of Limited broadcast). Here the netid part will identify the destination network (any valid netid) and hostid part will have all 1s. Example address is 63.255.255.255. (63 identifies the Class A network and rest implies all hosts on that network).


4. Write a brief note about direct and indirect routing.

Ans.

Direct Routing: If the destination host is attached to the same physical network as the source host, IP datagrams can be directly exchanged. This is called direct delivery and is referred to as direct routing. Direct routing occurs when both source and destination hosts are connected to the same physical network.

Indirect Routing: Indirect routing occurs when the destination host is not connected to a network directly attached to the source host. The only way to reach the destination is via one or more IP router. The address of the first router (the first hop) is called an indirect route in the IP routing algorithm. The address of the first router is the only information needed by the source host to send a packet to the destination host.

5. What are the types of routing tables? Explain briefly.

Ans.

The determination of routes is derived from the table known as routing table. The types of routing tables are as follows:

Static routing table Dynamic routing table

Static Routing Table: Routing tables can be a static one in which case the entries remain same unless someone changes it manually. If there is any change in the Internet topology, like some of the links going down temporarily, the entries will not be updated automatically resulting in the routing table which does not reflect the changed topology. However this simple mechanism is sufficient for many of the routers which interconnect small networks where changes in the topology is very unlikely or even if that happens the inconvenience caused is not much before the administrator updates the routing table manually.

Dynamic Routing Table: A dynamic routing table is updated periodically automatically. It does not need the manual intervention from the administrator for the updating process. However for this the routers must have implemented one of the dynamic routing protocols such as RIP, OSPF or BGP. Basically the routers participating in dynamic routing communicate with each other using one of the above-mentioned protocols informing about the status of the Internet by exchanging respective protocol packets. Through these communication whenever there is a change in the Internet topology, such as shutdown of a router or a link becoming inactive or even a link becoming active, the router which comes to know about such a change informs the rest of the router about the change using routing protocol, so that the remaining routers updates their routing tables appropriately. A change in the Internet may not result in the change in routing table at all the routers. It may affect only some of the routers. The routers in the Internet need to be updated dynamically for efficient delivery of the IP packets.

6.Write the IP header datagram. Explain the fields Service Type, Protocol, Checksum, Source IP address and Destination IP address.

Ans.

The unit of transfer in an IP network is called an IP datagram header. A datagram is a variable length packet consisting of two parts, header and data. The maximum length of an IP datagram is 65,535 bytes (octets). The header can be from 20 to 60 bytes long and contains information essential for routing and delivery. The length of the data part varies from packet to packet but the total length of the IP datagram should be within 65,535 bytes.

The following are some of the fields of the header:

Service Type: The service type is an indication of the quality of service requested for this IP datagram.


Protocol Checksum: This field is a checksum for the information contained in the header. If the header checksum does not match the contents, it implies that the datagram is corrupted and is discarded.

Source IP Address: This is the 32-bit IP address of the host, sending this datagram. Destination IP Address: This is the 32-bit IP address of the destination host, for this

datagram.


7. Why fragmentation is required? Explain how the IP header fields are used for the same purpose.

Ans.

When an IP datagram travels from one host to another, it may pass through different physical networks. Each physical network has a maximum frame size. This is called the maximum transmission unit (MTU). It limits the length of a datagram that can be placed in one physical frame.

The format and size of the received frame depends on the protocol used by the underlying physical network through which the frame passes through. Since each protocol used at the physical layer has its own MTU, there is every possibility that the incoming frame exceeds the MTU of the outgoing physical network. To enable forwarding the datagram in such cases, IP implements a process to fragment datagrams exceeding the MTU. The process creates a set of datagrams within the maximum size. The receiving host reassembles the original datagram.

The source or any router in the path can fragment a datagram. However, only the destination host does the reassembly of the datagram, as each fragment becomes an independent datagram. When a datagram is fragmented, required parts of the header must be copied by all the fragments. The host or the router that fragments the datagram must change the values of three fields: flags, fragmentation offset, total length. The checksum field for each of the fragments has to be recomputed and duly filled.

The following steps are performed to fragment the datagram:

The DF flag bit in the flag field is checked to see if fragmentation is allowed. If the bit is set which indicates not to fragment, the datagram will be discarded as it cannot be forwarded and an ICMP error returned to the originator.

Based on the MTU value, the data field of the datagram is split into two or more parts. All newly created data portions must have a length that is a multiple of 8 octets, with the exception of the last data portion.

Each data portion is placed in an IP datagram. The headers of these datagrams are minor modifications of the original: The more fragments flag bit is set in all fragments except the last. The fragment offset field in each is set to the location this data portion occupied in the

original datagram, relative to the beginning of the original unfragmented datagram. The offset is measured in 8-octet units.

The header length field of the new datagram is set. The total length field of the new datagram is set. The header checksum field is re-calculated. Each of these fragmented datagrams is now forwarded as a normal IP datagram. IP

handles each fragment independently. The fragments can traverse different routers to the intended destination. They can be subject to further fragmentation if they pass through networks specifying a smaller MTU.

8. Give an overview of ARP mechanism using diagrams.

Ans.

The ARP protocol is a network-specific standard protocol. The address resolution protocol is responsible for converting the higher-level IP addresses to physical network addresses.

IP layer provides a virtual or logical network view to the higher layer protocols where in each host is identified by the unique address known as IP address. However, when the datagram are sent on the physical network, this IP address cannot identify the destination system. At the physical or data link layer, it is the physical address, which identifies the host system. Therefore, there is a need to do address translation from IP address to physical address. The ARP mechanism, which resides at the network layer in the TCP/IP protocol, performs this address translation.

For this purpose, the ARP module will have a lookup table called ARP cache. This table will have entries for all the known IP addresses. For each IP address present in the cache,


corresponding physical address is stored against it. Therefore, when ARP module is requested to perform an address translation, it first looks into this cache. If it finds an entry for the IP address, it gets the corresponding physical address and returns it.

However, many times there will be no entry in the cache for requested IP address. In such cases ARP module broadcasts a message over the network requesting for the address translation and after getting the proper reply stores the physical address of the IP address requested in the cache table and passes the same to the upper layers that requested it.

Every host or the router on the network receives and processes the ARP query packet, but only the intended recipient recognizes the IP address and sends back the an ARP response packet. The response packet contains the recipient’s IP and physical addresses. The packet is unicast directly to the inquirer using the physical address received in the query packet.

9. Write a brief note about RARP.

Ans.

In some local area networks, there would be a powerful computer, which acts as a server. There would be several hosts, which will not have any disks (hard disk / floppy disk) that will be connected to the host. People would use these hosts as the front-end system and would be connected to the server over the network. Finally, they would be working on the server but using these hosts as the front-end systems. Having disk-less hosts has an advantage in some setups, where there would be a central powerful server computer on which many people can work simultaneously by connecting through such hosts. As these hosts are disk-less system, they would not need any configuration by the administrator. Maintenance would be minimal. These disk-less systems would be booted from the ROM (Read Only Memory chip), which is programmed by the manufacturer. It cannot include the IP address as the network administrator assigns them.

Each time these disk-less hosts are switched on, they will not be aware of their IP address, as they do not have any disk or storage device. However, they will be aware of their hardware address as encoded into the Network Interface Card, from which they can get it.

Since in a network each system is recognized by the logical or more commonly, IP address, it is vital for the system to be aware of its IP address. Any IP datagram to be sent should have the senders IP address duly filled in. Otherwise, the recipient of that packet would not know from whom the packet arriver and it would just discard.

To find out ones IP address these disk-less host systems, uses a protocol known as Reverse Address Resolution Protocol. This is a protocol, which does the function opposite of that of ARP i.e. given the physical address gets the IP address.

At boot up time, these disk-less hosts would broadcast a RARP request packet over the network after filling its physical address in it. The server which will have a reference table of physical address and their corresponding IP address would respond with the RARP response packet. This reference table has entries for each of the physical address (present in the local network) and their IP address would be created and maintained by the administrator.

10. What are the error reporting messages present in the ICMP? Explain any two of them.

Ans.

IP provides an unreliable and connectionless datagram delivery. IP protocol does not have any error-reporting or error-correcting mechanism. Therefore, if something goes wrong, the router does not find route for the IP packet and so discards it. The sender will be never aware of it, and he would keep resending the same. The ICMP was designed to compensate these deficiencies of IP. It is a companion to the IP protocol and is part of the network layer along with the IP.

ICMP messages are divided into two broad categories, error-reporting messages and query messages.


The error reporting messages report the problems that a router or a host may encounter when it processes an IP packet. They would report the problem back to the IP module at the sending system.

The query messages, which occur in pairs, help a host or a network manager get specific information from a router or another host. For example, this can be used by the hosts to discover the routers present in their network. The host would send an ICMP query asking for routers to respond. The routers present in the network will respond with an ICMP reply message. The host would get enough information about the router from this reply.


Under error-reporting type, the following messages are present:

Destination unreachable Source quench Time exceeded Parameter problem Redirection

The query messages type has following messages:

Echo request and reply Timestamp request and reply Address mask request and reply Router solicitation and advertisement

FAQ

1. What is the functionality of a network layer?

Ans:

The network layer is responsible for transferring the data from the source to the destination host across several physical networks. A key aspect of this transfer is the routing of packets fast, efficiently, and reliably from the source to the destination machine typically traversing a number of transmission links and network nodes where routing is carried out.

The network layer also provides the virtual network concept to the upper layers. As far as the upper layers are concerned, there is only one network, and each host is recognized by a unique IP address.

2. Identify the protocols used in the network layer.

Ans:

The following protocols are used in the network layer:

IP: Internet Protocol ARP: Address Resolution Protocol RARP: Reverse Address Resolution Protocol ICMP: Internet Control Message Protocol

3. Define routing.

Ans:

Routing is the process by which a path is selected out of many available paths to the destination host so that data packet reaches the destination fast, efficiently, reliably as required. It uses the destination IP address that is stored in the IP datagram packet to reach the correct destination.

4. Define ARP.

Ans:

ARP (Address Resolution Protocol) is a network-specific standard protocol. ARP is responsible for converting the higher-level protocol addresses (IP addresses) to physical network addresses.


5. Define RARP.

Ans:

RARP (Reverse Address Resolution Protocol) is responsible for converting the physical network addresses of disk-less host systems to IP addresses.

6. Explain the working of ICMP.

Ans:

ICMP (Internet Control Message Protocol) is a network layer protocol and it provides error-reporting or error-correcting mechanism. In addition, ICMP provides query messages, which occur in pairs and it helps a host or a network manager get specific information from a router or another host. Therefore, ICMP is designed to compensate the unreliable and connectionless deficiencies of IP.


Chapter Five

Objectives


Identify the features of UDP Define process-to-process communication Define UDP datagram List the steps for UDP checksum computation Define UDP operations Identify the uses of UDP Identify the services offered by TCP/IP Identify the connections in TCP Describe flow control mechanism in TCP Define the various TCP operations

Focus Areas

This chapter explains the transport layer. Explain the features of UDP and TCP. Explain the basic function of the transport layer. Refer to the topics given in the Additional Inputs section. List and explain the protocols used in the transport layer, namely, UDP (User Datagram Protocol) and TCP (Transmission Control Protocol).

When discussing UDP, explain the following features of UDP:a. Simple protocol b. Connectionless servicec. No flow controld. Process to process communicatione. Limited error checkingf. Provides unreliable service

Sensitize the students to the fact that UDP is used in applications that require high data transfer rate and can have low reliability. Explain the UDP datagram format. Also explain the operation of UDP.

When discussing TCP, explain the following features of TCP:a. Simple network communication from the application programmer's point of view.b. Reliable servicec. Connection-oriented protocold. Built-in flow-controle. Stream data transferf. Full-duplex service

Explain the TCP segment. Explain how connection is established and terminated in TCP using three-way handshake and four-way handshake respectively. Explain the flow control and error control mechanisms in TCP. Also explain the operation of TCP.

Additional Inputs


Transport Layer

There may be several processes running on a computer. Thus, source to destination delivery involves not only the delivery of a message from one computer system to another but also from one process to another across the two systems. The Transport layer does this by including a service-point address on its header, which ensures the delivery of the message to the right process.

The Transport layer also determines the type of service to provide to the Session layer and, ultimately, to the users of the network. The most popular type of transport connection is an


error-free end-to-end channel. This type of connection delivers messages in the order in which they were sent. The other type of transport messages is service- and transport-isolated messages, which do not have a guarantee about the order of delivery and broadcasting to multiple destinations.

Reliable Delivery

There are a number of aspects that the Transport layer needs to manage for reliable delivery of data between the computers on a network. These are error control, sequence control, loss control, and duplication control.

Error control: The Transport layer implements error control by ensuring that the data sent by the sending machine is delivered at the destination machine exactly as it originated from the source machine. The Data Link layer takes care of error control, but only for computer-to-computer delivery. However, the Transport layer ensures source-to-destination reliability while implementing error control. One method of implementing reliability is through acknowledgements. When a packet is delivered to a destination computer, an acknowledgement of receipt is returned to the source computer. In this way, the Transport layer tracks and validates the transmission of each packet.

Sequence control: The Transport layer is responsible for ensuring the correct sequence of data both at the sender’s end and at the receiver’s end. At the sender’s end, the Transport layer receives data from the upper layers and ensures that the data is usable by the lower layers. At the receiver’s end, the Transport layer ensures that the data is assembled correctly. The Transport layer makes use of the sequence numbers. It assigns a sequence number to each segment that it transfers. These sequence numbers enable the message to be reassembled in an ordered manner at the destination. The sequence numbers also help to identify and replace the packets that are lost during transmission.

Loss control: The Transport layer also ensures that all the segments of data are delivered to the destination machine. Some segments of data may be lost during transmission due to the segmentation of data. The Transport layer uses the sequence numbers to identify the segments that are missing and requests for the redelivery of those segments.

Duplication control: The Transport layer guarantees that none of the segments of data arrives duplicated at the destination machine. In addition to identifying the missing segments of data, the sequence numbers also help to identify and discard the segments that are duplicated.

Multiplexing

Generally, the Transport layer creates different network connections for each transport connection that the Session layer requires. However, if a particular transport connection requires a high throughput, the Transport layer might create multiple network connections and divide the data among the network connections. Though this improves throughput, it also increases the cost of creating and maintaining a network connection. Therefore, the Transport layer might transfer and merge several transport connections into the same network connection to reduce the cost. This process is called multiplexing. In all cases, the Transport layer is responsible for notifying the Session layer about the multiplexing process. Multiplexing can be of two types – upward and downward:

Upward: In upward multiplexing, the Transport layer uses virtual circuits depending on the services of the layers below it. Generally, the underlying networks charge all the virtual circuits for their established connections. By using upward multiplexing, the Transport layer sends multiple transmissions that are bound for the same destination along the same path or circuit. By doing this, the Transport layer makes use of the established circuit in a cost-effective manner.

Downward: In downward multiplexing, the Transport layer splits up a single connection into several paths to improve the speed of delivery. Downward multiplexing is helpful when the capacity of the underlying networks is low or slow.

In addition to multiplexing several transport connections to one network connection, the Transport layer is also responsible for establishing and deleting connections across a network. The Transport layer also implements a mechanism to regulate the flow of information, so that a fast host cannot overrun a slow one.


Flow Control

The Transport layer ensures end-to-end flow control by using the technique of a sliding window. This window stores a set of packets that must be transmitted without waiting to receive acknowledgements for the previously transmitted packets. In most cases, the receiver decides the size of the window based on the number of bytes that it can accommodate. The sender does not need to send data to completely fill the sliding window. Instead, the receiver, while sending the acknowledgement for a previously transmitted packet, can specify that the size of the window be increased or decreased.

The sliding window consists of three virtual areas. The top of the window stores packets that are waiting to be acknowledged. In the bottom of the window, the packets that are waiting to be transmitted are stored. When an acknowledgement is received for a packet, the window slides down to exclude the acknowledged packet and to include a new packet in the window.

Protocols used in the Transport Layer

Two core protocols that are used in the Transport layer are Transmission Control Protocol (TCP) and User Datagram Protocol (UDP). These protocols are with respect to the TCP/IP suite. TCP is a connection-oriented protocol that assures reliable packet delivery. In other words, TCP tracks packets and verifies that a packet has reached its destination. UDP can be described as a connectionless protocol that supports only packet transmission.

Connection in the Transport Layer

The Transport layer can be of two types – connectionless or connection-oriented. When it is connectionless, it delivers each packet of data to the destination machine individually and independent of other packets. When the Transport layer is connection-oriented, it first establishes a connection with the Transport layer of the destination machine and then delivers the packets. The connection is terminated only after all the packets are delivered. Connection-oriented transmission is considered more reliable. There are three stages in a connection-oriented transmission - connection establishment, data transfer, and connection termination.

Establishing connections follows a technique called a Three Way Handshake. Before moving ahead, it is important to know that computers communicate with each other through a software object called an endpoint. Each endpoint is a channel consisting of the IP address and the port number through which the data must flow in or out of the computer. To move on, consider the following procedure to understand how the Three Way Handshake works:

When an application needs to establish connection with another computer, the networking software on the sending computer notifies the OS that an endpoint must be created. The OS creates an endpoint through which the Transport layer sends an initial synchronization packet, also known as SYN. This step can be described as the first handshake.

On receiving the SYN packet, the destination computer initiates the creation of an endpoint through its OS. Next, the destination computer sends a SYN to the sending computer. The acknowledgement for the received SYN is added within the SYN packet sent from the destination computer. In addition, the sequence number of the SYN packet is also incremented. This synchronizes the sequence numbering between the sending and receiving computers. Completing this step indicates the end of the second handshake.

On receiving the SYN packet and the acknowledgement from the destination computer, the sending computer returns and acknowledgement of the destination computer’s SYN. This step is the third handshake completing the connection process. The computers can now start communicating with each other.


Chapter Five Questions

Part 1

1. UDP is an acronym for _____________________

Ans.

User Datagram Protocol

2. In the sending system, UDP receives data unit from the ______________ layer.

Ans.

Network

3. UDP needs the _________ address to deliver the data to the correct process.

Ans.

Port

4. UDP offers ______ to ______ data communication.

Ans.

Process-to-process

5. UDP has the fixed header size of ___ bytes.

Ans.

Eight

6. In TCP a unit of data is referred to as ________.

Ans.

Datagram

7. Flow control is achieved through the use of _____________ window mechanism.

Ans.

Sliding

8. _________ field in the TCP header is used to detect errors in the TCP segment.

Ans.

Checksum

9. In TCP, before sending data, a __________ is established between the sending and receiving processes.

Ans.

Virtual path


10. TCP offers ________ duplex data transfer service.

Ans.

Full

Part 2

1. Write a short note on, services offered by the UDP and what kind of applications UDP service is used.

Ans.

UDP is a connectionless, unreliable transport protocol. UDP provides process-to-process communication. In addition, it performs very limited error checking. UDP is very efficient because of its simplicity. Establishing a connection with the destination process is fast, and if reliability is not an important criterion then the transfer of data is faster for bulk transfer.

The services offered by UDP are as follows:

Connectionless Services: UDP provides a connectionless service. This means that each user datagram sent by the UDP is an independent datagram. There is no relationship between the different user datagrams even if they are coming from the same source process and going to the same destination program. The user datagrams are not numbered. Also, there is no connection establishment and no connection termination at the beginning and end of a transaction. This means that each user datagram can travel a different path.One of the ramifications of being connectionless is that the process that uses UDP cannot send a stream of data to UDP and expect to receive the same stream of data at the destination. Instead each request must be small enough to fit into one user datagram. Only those processes sending short messages should use UDP. Otherwise if needed, the end application should take care of section of data arriving out of order and reordering them to get the original stream.

Flow and Error Control: UDP is a very simple, unreliable transport protocol. There is no flow control, and hence no windowing mechanism. The receiver may overflow with incoming messages. Again it is left for the end application to take of it. The end applications which uses UDP either generates less data or they have inbuilt mechanism to take care of flow control.There is no error control mechanism in UDP except for the checksum. This means that the sender does not know if a message has been lost or duplicated. When the receiver detects an error using the checksum, the user datagram is silently discarded.

Encapsulation ad Decapsulation: To send a message from one process to another, the UDP protocol encapsulates and decapsulates the messages.

Multiplexing and Demultiplexing: In a host running a TCP/IP protocol, there is only one UDP but possibly several processes that may want to use the services of UDP. To handle this situation, UDP uses the concept called Multiplexing and Demultiplexing.

The applications at the Transport layer use the services of UDP. There are several multimedia applications with powerful desktop computer and high bandwidth communication network that uses UDP. The multimedia applications, which require bulk transfer of data, needs no fool-proof reliability. The reliability provided by the underlying networks is sufficient. Loss of data to some extent is tolerable, and in fact in some cases there are techniques available to recover the lost data with the source requiring sending it again.

Today’s multimedia applications, such as VoIP (Voice over IP) and Video conferencing applications use a protocol known as RTP (Real Time Protocol) for data transfer at the application layer. This RTP protocol uses the services of UDP.

2. Explain how port number concept is used to offer process-process data communication in UDP.

Ans.


In TCP/IP model, each process requiring the TCP/IP communication service is assigned a 16-bit (2 byte) number called port number. The TCP/IP standard has defined unique port numbers for some of the well-known network application process. This helps to uniquely identify a network application process running on a host machine.

Computer systems today support both multi-user and multiprogramming environments. Local and remote computers can run several server programs at the same time. For communication, the following hosts and processes must be defined:

Local host Local process Remote host Remote process

The local host and the remote host are identified using IP addresses. To define the processes, we need second identifiers, which are called port numbers. In the TCP/IP protocol suite, the port numbers are integers between 0 and 65,535.

The client program is assigned a port number, chosen randomly by the UDP software running on the client host. This is the ephemeral port number.

The server process must also be assigned a port number. This port number, however, cannot be chosen randomly. If the computer at the server site runs a server process and assigns a random number as the port number, the process at the client site that wants to access that server and use its services will not know the port number. Therefore, TCP/IP has decided to use universal port numbers for servers; these are called well-known port numbers. Every client process knows the well-known port number of the corresponding server process.

The destination IP address defines the host among the different hosts in the Internet. After the host has been selected, the port number selects one of the intended processes on this particular host.

3. Write the UDP datagram header. Explain each field.

Ans.

UDP packets, known as user datagrams, have a fixed-size header of eight bytes.

The fields of UDP packets are explained below:

Source port number: This is the port number of the source process sending the UDP datagram. It is 16 bits long (2 byte). If the source process is a client (a client sending a request), the port number, in most cases, is an ephemeral port number. If the source process is a server (a server sending a response), the port number, in most cases, is a well-known port number.

Destination port number: This is the port number of the destination process to which this UDP datagram is meant. It is also 16 bits long. Its characteristics are similar to the Source Port Number.

Length: This is a 16-bit field that defines the total length of the UDP datagram, header plus data. The 16 bits can define a total length of 0 to 65,535 bytes. However, the minimum length is eight bytes, which indicates a user datagram with only header and no data. However it should be noted that IP has a limitation of 65,535 bytes for its datagram. Therefore, the length of the data can be between 0 and 65,507 (65,535 – 20 – 8) bytes (twenty bytes for IP header and 8 bytes for UDP header).The UDP length can be computed using data length in the IP header. However, the designers of UDP protocol felt that it was more efficient for the destination UDP to calculate the length of the data from the information provided in the UDP datagram rather than asking to supply this information. We should remember that when the IP module delivers the UDP user datagram to the UDP layer, it has already dropped the IP header.

Checksum: This field is used to detect errors over the entire user datagram (header plus data). Unlike lower layer protocols like IP, which provide checksum only for their header, this provides the checksum for entire datagram, which includes the data sent by the user process.


4. Describe how checksum is calculated for the UDP datagram.

Ans.

UDP checksum calculation includes three sections:

Pseudo header: The pseudo header is part of the header of the IP packet in which the user datagram is to be encapsulated for transmission with some fields with 0s.

UDP header: Header for this UDP datagram. User Data: Data sent by the upper layer.

The pseudoheader is added to ensure that the user datagram reaches the intended process, which uses the intended transport protocol on the intended host. Since in no other layer, it is possible to check all these three, a pseudoheader derived from the IP header that has destination and source IP address and protocol numbers is used. The pseudoheader along with the UDP header uniquely identifies the destination process.

Checksum Calculation at the Source: At the source host system, the sender follows these steps to calculate the checksum:

1. Add the pseudoheader to the UDP user datagram.2. If the total number of bytes is not even, add one byte of padding (all 0s). The padding is

only for the purpose of calculating the checksum and will be discarded afterwards.3. Fill the checksum field with zeros.4. Divide the total bits into 16-bit (two-byte) words sections.5. Add all 16-bit sections using one’s complement arithmetic.6. Take ones complement of the result (change 0s to 1s and all original 1s to 0s), which is a

16-bit checksum number.

Once 16 bits checksum field is computed using the above method, it is filled into the original UDP header and the resulting UDP datagram is sent. It should be noted that the pseudoheader and the padding bits are not sent but only used during the computation of checksum.

Checksum Calculation at Destination Host: The receiver follows these steps to calculate the checksum:

7. Obtain the IP header, derive the pseudoheader from it, and add it to the UDP user datagram.

8. Add padding if needed.9. Divide the total bits into 16-bit sections.10. Add all 16-bit sections using one’s complement arithmetic.11. Take ones’ complement the result to get the checksum.

UDP datagram is accepted if the resulting checksum is all 0s. Otherwise it is discarded, as it indicates an error in it.

5. Explain the Encapsulation and Decapsulation mechanism used in UDP.

Ans.

Encapsulation: When a process has a message to send through UDP, it passes the message to UDP along with a pair of socket addresses and the length of data. UDP receives the data and adds the UDP header. UDP then passes the user datagram to the IP with the socket addresses. IP adds its own header, using the value 17 in the protocol field, indicating that the data has come from the UDP protocol. The IP datagram is then passed to the data link layer. The data link layer receives the IP datagram, adds its own header (and possibly a trailer), and passes it to the physical layer. The physical layer encodes the bits into electrical or optical signals and sends it to the remote machine.

Decapsulation: When the message arrives at the destination host, the physical layer decodes the signals into bits and passes it to the link layer. The data link uses the header (and the trailer) to check the data. If there is no error, the header and trailer are dropped and the datagram is passed to the IP. The IP software does its own checking. If there is no error, the header is dropped and the user datagram is passed to the UDP with the sender and receiver IP


addresses. UDP uses the checksum to check the entire user datagram. If there is no error, the header is dropped and the application data along with the sender socket address is passed t the process. The sender socket address is passed to the process in case it needs to respond to the message received.

6. Write a short note on importance of TCP.

Ans.

The protocol, which is responsible for widespread usage of TCP/IP model for computer-to-computer communication network is the TCP. It is the combination of TCP at the Transport layer and IP at the Network layer, which is largely responsible for success of this model.

The network application program, which needs to communicate over the network, expects a very simple, easy to implement communication channel. If a user wants to write an application program, which communicates with another over the network, he should not be expected to build logic into his program to take care of so many issues that are involved in the communication. He should not worry about things such as, some part of the message getting corrupted or lost. There are other issues which should not be handled by him, like rearranging the datagrams which arriver out of order, issues of congestion and resulting delay, issues of flow control by which the rate at which the source sends the data can be controlled.

TCP ensures easy to use communication channel, which provides the reliable, stream oriented communication service, which takes are of such error and flow control mechanisms.

IP provides an unreliable, best-effort, datagram service and UDP adds just the process-to-process communication facility apart from error checking mechanism to some extent. Therefore, there is a need for another protocol, which sits on top of the IP and provides the services as mentioned above to the end application. So the designer of TCP/IP model came out with TCP at the Transport layer.

TCP provides process-to-process communication channel, which is reliable and stream oriented in nature. In addition, it takes care of the error-control and flow control.

The process-to-process mechanism is very similar to the one used by UDP. Port numbers are used to identify an individual process. Association of IP address and port number is known as the socket. A socket identifies each communicating process and a pair of socket determines the communication channel. A port number can be shared by both UDP as well as TCP. I.e. there can be a port number, which is assigned to two processes at the same host, but which uses different protocols, one process uses UDP and another TCP. Port number does not identify the protocol. The protocol field in the IP header makes that distinction.

7. Explain the stream-oriented data transfer offered by the TCP.

Ans.

TCP provides stream data transfer service, which means the destination process receives the stream of data in exactly the same manner it is sent by the source process. Unlike in datagram service, there is no concept of unit of data transfer. In datagram service, all the data have to be sent as a single unit. This imposes two restrictions. One restriction is that all the data should be available at the time of sending. Otherwise, they will be sent in another datagram. Each datagram unit is a separate entity and there is no relation between them. Another restriction is on the size of the data transfer.

Source TCP accepts a stream of characters from the sending application program as and when they arrive, creates packets, called segments, of appropriate size extracted from the stream, and sends them across the network. The receiving TCP receives segments, extracts data from them, orders them if they have arrived out of order, and delivers them as a stream of characters to the receiving application program.

For stream delivery, the sending and receiving TCPs use buffers. The sending TCP uses a sending buffer to store the data coming from the sending application program. The sending application program delivers data at the rate it is created. For example, if the user is typing the


data on a keyboard, the data is delivered to the sending TCP character by character. If the data is coming from a file, data may be delivered to the sending TCP line-by-line, or block-by-block. The sending application program writes data to the buffer of the sending TCP. However, the sending TCP does not create a segment of data for each write operation issued from the sending application program. TCP may choose to combine the result of several write operations into one segment to make transmission more efficient.

The receiving TCP receives the segments and stores them in a receiving buffer. The receiving application program uses the read operation to read the data from the receiving buffer, but it does not have to read all of the data contained in one segment in one operation. Since the rate of reading can be slower than the rate of receiving, the data is kept in the buffer until the receiving application reads if completely.


8. Write a diagram of TCP header. Explain the fields, Sequence number, Acknowledgement number, Window size and Urgent pointer.

Ans.

In TCP, a unit of data transfer is known as TCP segment. Since TCP provides the stream-oriented data service, it uses the concept of segment for a unit of transfer.

The segment consists of a 20-to 60-byte header, followed by data from the application program. The header is 20 bytes if there are no options and up to 60 bytes if it contains some options. Some of the header fields are as follows:

Source port address: This is a 16-bit (2 bytes) field that defines the port number of the application program in the host that is sending the segment.

Destination port address: This is a 16-bit (2 bytes) field that defines the port number of the application program in the host that is receiving the segment.

Sequence number: This 32-bit (4 bytes) field defines the number assigned to the first byte of data contained in this segment. TCP is a stream transport protocol. To ensure connectivity, each byte to be transmitted is numbered. The sequence number informs the destination the position of the first byte of this segment in the original stream of data at the source.

Acknowledgement number: This 32-bit (4 bytes) field defines the byte number that the source of the segment is expecting to receive from the other end process. If host has received successfully till byte number n from the other host, then it defines n + 1 as the acknowledgement number, which indicates it is expecting data starting from location n+1 at the other hosts’ stream.

Header length: This four-bit field indicates the length of the TCP header. It is equal to number of four-byte words in the TCP header. The length of the header can be between 20 and 60 bytes. Therefore, the value of this field can be between 5 (5 x 4 = 20) and 15 (15 x 4 = 60).

Reserved: This is a six-bit field reserved for future use. Control: This field defines six different control bits or flags as shown in figure. One or more

of these bits can be set at a time. These bits enable flow control, connection establishment and termination, and the mode of data transfer in TCP. A brief description of each bit is given below: URG: The value of urgent pointer is valid. ACK: The value of acknowledgement field is valid. PSH: Request to push the data RST: Request to reset the connection. SYN: Request to synchronize the sequence number during connection FIN: Request to terminate the connection.

Window size: This field defines the size of the window, in bytes, that the other end must maintain. The length of this field is 16 bits, which means that the maximum size of the window is 65,535 bytes.

Checksum: This 16-bit (2 bytes) field contains the checksum, used for error checking purpose.

Urgent pointer: This 16-bit field, which is valid only if the urgent flag is set, is used when the segment contains urgent data. It defines the number that must be added to the sequence number to obtain the number of the last urgent byte in the data section of the segment.

Options: There can be up to 40 bytes of optional information in the TCP header.

9. Describe briefly how error-control mechanism is achieved in TCP when data is corrupted, and when acknowledgement is lost.

Ans.

TCP is a reliable transport layer protocol. This means that an application program that delivers a stream of data to TCP relies on TCP to deliver the entire stream to the application program on the other end in order, without error, and without any part lost or duplicated. TCP provides reliability using error control. Error control includes mechanisms for detecting corrupted


segments, lost segments, out-of-order segments, and duplicated segments. Error control also includes a mechanism for correcting errors after they are detected.

Error Detection and correction: Error detection in TCP is achieved through the use of three simple tools: checksum, acknowledgement, and time-out. Time-out is the duration for which the sending TCP waits for the acknowledgement, after which it considers whether the data sent is either corrupted or lost and retransmits the same.

Each segment includes the checksum field, which is used to check for a corrupted segment. By computing checksum, one can find whether the segment is corrupted or not. If it is corrupted, it is discarded by the destination TCP. TCP uses the acknowledgement method to confirm the receipt of those segments that have reached the destination TCP uncorrupted. If a segment is not acknowledged before the time-out, it is considered to be either corrupted or lost.

The error-correction mechanism used by TCP is also very simple. The source TCP starts one time-out counter for each segment sent. Each counter is checked periodically. When a counter expires, the corresponding segment is considered to be either corrupted or lost, and the segment will be retransmitted.

Consider an example where data becomes corrupted while it is sent from one host to another. In this example the source host sends segments 1 through 3, each 200 bytes. The sequence number begins at 1,201 on segment 1. The receiving TCP receives segments 1 and 2, using the checksum, finds them error free. It acknowledges the receipt of segments 1 and 2 using acknowledgement number 1,601, which means that it has received bytes 1,201 to 1,600 safe and sound, and is expecting to receive byte 1,601. However, it finds that segment 3 to be corrupted and discards segment 3. Note that although it has received bytes 1,601 to 1,800 in segment 3, the destination does not consider this as a receipt because this segment was corrupted. After the time-out for segment 3 expires, the source TCP will resend segment 3. After receiving segment 3, the destination sends the acknowledgement for byte 1,801, which indicates that it has received bytes 1,201 to 1,800 error free.

Consider an example where acknowledgement is lost while data is sent from one host to another. In the TCP acknowledgement mechanism, a lost acknowledgement may not even be noticed by the source TCP. TCP uses an accumulative acknowledgement system. Each acknowledgement is a confirmation that everything up to the byte specified by the acknowledgement number has been received. For example, if the destination sends an ACK segment with an acknowledgement number for byte 1,801, it is confirming that bytes 1,201 to 1,800 have been received. If the destination had previously sent an acknowledgement for byte 1,601, meaning it has received bytes 1,202 to 1,600, loss of the acknowledgement is totally irrelevant.

10. Explain why one needs the service of Urgent Data operation of TCP.

Ans.

TCP is a stream-oriented protocol. This means that the data is presented from the application program to the TCP as a stream of characters. Each byte of data has a position in the stream. However, there are occasions in which an application program needs to send urgent bytes. This means that the sending application program wants a piece of data to be read out of order by the receiving application program. Suppose that the sending application program is sending data to be processed by the receiving application program. When the result of processing comes back, the sending application program finds that everything is wrong. It wants to abort the process, but it has already sent a huge amount of data. If it issues an abort command (Control + C), these two characters will be stored at the end of the receiving TCP buffer. It will be delivered to the receiving application program after all the data has been processed.

The solution is to send a segment with the URG bit set. The sending application program tells the sending TCP that the piece of data is urgent. The sending TCP creates a segment and inserts the urgent data at the beginning of the segment. The rest of the segment can contain normal data from the buffer. The urgent pointer field in the header defines the end of the urgent data and the start of normal data.


When the receiving TCP receives a segment with the URG bit set, it extracts the urgent data from the segment, using the value of the urgent pointer, and delivers it, out of order, to the receiving application program.


11. Explain the concept of Multiplexing and Demultiplexing in TCP.

Ans.

The concept of multiplexing and demultiplexing operation is as follows:

Multiplexing: At the source host, there may be several processes that need to send user datagrams. However, there is only one TCP. This is a many-to-one relationship and requires multiplexing. TCP accepts messages from different processes; each message will have a port number assigned to it, which identifies the process, which generated the message. After adding the header, TCP passes the user datagram to IP.

Demultiplexing: At the receiver site, there is only one TCP. However, we may have many processes that can receive user datagrams. This is a one-to-many relationship and requires demultiplexing. TCP receives user datagrams from IP. After error checking and dropping of the header, TCP delivers each message to the appropriate process based on the port numbers.

12. Explain the need of Pushing Data operation in TCP.

Ans.

In certain situations, the sending TCP uses a buffer to store the stream of data coming from the sending application program. The sending TCP has the choice to create segments of any size from the stream. The receiving TCP also buffers the data when they arrive and delivers them to the application program when the application program is ready or when the receiving TCP feels that it is convenient. This type of flexibility increases the efficiency of TCP.

However, there are occasions in which the application program is not comfortable with this flexibility. For example, consider an application program that communicates interactively with another application program on the other end. The application program on one site wants to send a keystroke to the application at the other site and receive an immediate response. Delayed transmission and delayed delivery of data may not be acceptable by the application program.

TCP can handle such a situation. The application program on the sending site can request a push operation. This means that the sending TCP should not wait for the window to be filled. It should create a segment and send it immediately. The sending TCP should also set the push bit (PSH) to tell the receiving TCP that the segment includes data that must be delivered to the receiving application program as soon as possible and not to wait for more data to come.

Although the push operation can be dictated by the application program, today most implementations ignore such requests. TCP has the choice to use this operation or not.

13. Explain how connection is established in TCP using three-way hand shaking.

Ans.

The connection establishment in TCP is called three-way handshaking. In this procedure, an application program, called the client, wants to make a connection with another application program, called the server, using TCP as the transport layer protocol.

The three-way handshaking procedure starts with the server. The server program tells its TCP that it is ready to accept a connection. This is called a request for a passive open.

The client program makes a request for an active open. A client that wishes to connect to a server tells its TCP that it needs to be connected to a particular server. The clients TCP can now start the three-way handshaking process.

The steps of the process are as follows:

The client sends the first segment, with the SYN bit set, which is known as SYN segment. The segment includes the source and destination port numbers. The destination port number clearly defines the server to which the client wants to be connected. The segment also contains the client Initialization Sequence Number (ISN) used for numbering the bytes of data sent from


the client to the server. If the client needs a large window, it defines the window scale factor here using the appropriate option. This segment defines the wish of the client to make a connection with certain parameters.

The server sends the second segment, a SYN and ACK segment. This segment has a dual purpose. First, it acknowledges the receipt of the first segment using the ACK flag and acknowledgement number field. The acknowledgement is the client initialization sequence number plus one. The server must also define the client window size. Second, the segment is used as the initialization segment for the server. It contains the initialization segment for the server and initialization sequence number used to number the bytes sent from the server. As mentioned before, this is two segments combined into one.

The client sends the third segment. This is just an ACK segment. It acknowledges the receipt of the second segment using the ACK flag and acknowledgement number field. The acknowledgement number is the server initialization sequence number plus one. The client must also define the server window size. Note that data can be sent with the third packet.

14. Describe briefly how Flow-control is achieved in TCP using sliding window mechanism.

Ans.

To accomplish flow control, TCP uses a sliding window protocol. With this method, both hosts use a window for each connection. The window covers a portion of the buffer that a host can send before worrying about an acknowledgement from the other host. The window is called a sliding window because it slides over the buffer as the receiver sends acknowledgement of the bytes received without any error. Consider a sliding window of size 10. Before receiving any acknowledgement from the destination, the source can send up to 10 bytes. However, if it receives acknowledgement of the first three bytes, it can slide the window three bytes to the right. This means that now it can send 10 more bytes before worrying about an acknowledgement.

The previous example shown is a fixed size window. However flow-control needs the size of the window to vary to achieve the flow control. Hence in TCP window size is variable. The destination, in each acknowledgement segment, can define the size of the window. The advertised size is relative to the acknowledgement number. For example, if the receiver acknowledges the receipt of byte 3,000 and defines the size of the window to be 200, it means that the window now expands from byte 3001 to byte 3,200.

The destination can also increase or decrease the size of the window in an acknowledgement segment.

FAQ

1. What does process-to-process communication mean?

Ans:

There may be several processes running on a computer. Thus, source to destination delivery involves not only the delivery of a message from one computer system to another but also from one process to another across the two systems. The Transport layer does this by including a service-point address on its header, which ensures the delivery of the message to the right process.

2. What is a socket address?

Ans:

Socket address is a combination of an IP address and a port number.

3. What is checksum?

Ans:


The error detection method used by most of the TCP/IP protocols is called checksum. The checksum protects against the corruption that may occur during the transmission of a packet.


4. How does TCP ensure reliable data delivery?

Ans:

There are a number of aspects that TCP needs to manage for reliable delivery of data between the computers on a network. These are error control, sequence control, loss control, and duplication control:

Error control: TCP implements error control by ensuring that the data sent by the sending machine is delivered at the destination machine exactly as it originated from the source machine. The Data Link layer takes care of error control, but only for computer-to-computer delivery. However, TCP ensures source-to-destination reliability while implementing error control. One method of implementing reliability is through acknowledgements. When a packet is delivered to a destination computer, an acknowledgement of receipt is returned to the source computer. In this way, the Transport layer tracks and validates the transmission of each packet.

Sequence control: TCP is responsible for ensuring the correct sequence of data both at the sender’s end and at the receiver’s end. At the sender’s end, the Transport layer receives data from the upper layers and ensures that the data is usable by the lower layers. At the receiver’s end, the Transport layer ensures that the data is assembled correctly. TCP also makes use of the sequence numbers. It assigns a sequence number to each segment that it transfers. These sequence numbers enable the message to be reassembled in an ordered manner at the destination. The sequence numbers also help to identify and replace the packets that are lost during transmission.

Loss control: TCP also ensures that all the segments of data are delivered to the destination machine. Some segments of data may be lost during transmission due to the segmentation of data. TCP uses the sequence numbers to identify the segments that are missing and requests for the redelivery of those segments.

Duplication control: TCP guarantees that none of the segments of data arrive duplicated at the destination machine. In addition to identifying the missing segments of data, the sequence numbers also help to identify and discard the segments that are duplicated.

5. What is buffer?

Ans:

A buffer is a data area shared by hardware devices or program processes that operate at different speeds or with different sets of priorities. The buffer allows each device or process to operate without being held up by the other. In order for a buffer to be effective, the size of the buffer and the algorithms for moving data into and out of the buffer need to be considered by the buffer designer. For stream delivery, the sending and receiving TCPs use buffers. The sending TCP uses a buffer to store the data coming from the sending application program. The receiving TCP receives the segments and stores them in a receiving buffer. The receiving application program uses the read operation to read the data from the receiving buffer, but it does not have to read all of the data contained in one segment in one operation. Since the rate of reading can be slower than the rate of receiving, the data is kept in the buffer until the receiving application reads if completely.

6. What is full duplex service?

Ans:

TCP offers full-duplex service, where data can flow in both directions at the same time. After two application programs are connected to each other, they can both send and receive data. One TCP connection can carry data from application A to B and at the same time, from B to A. When a packet is going from A to B, it can also carry an acknowledgement of the packets received from B. Likewise, when a packet is going from B to A, it can also carry an acknowledgement of the packets received from A. This is called piggybacking because acknowledgements can be sent with data.


7. What is an out-of-order segment?

Ans:

TCP uses the services of IP, an unreliable, connectionless network layer protocol. The TCP segment is encapsulated in an IP datagram. Each datagram is an independent entity. The routers are free to send each datagram through any route they find suitable. One datagram may follow a route with a short delay; another may follow another route with a longer delay. Therefore, datagrams may arrive out of order. If datagrams arrive out of order, the TCP segments that are encapsulated in the datagrams will be out of order as well. This is called an out-of-order segment.

8. What is the difference between pushing data and urgent data?

Ans:

The difference between pushing data and urgent data is:

Pushing data: The application program on the sending site can request a push operation. This means that the sending TCP should not wait for the sliding window to be filled. It should create a segment and send it immediately. The sending TCP should also set the push bit (PSH) to inform the receiving TCP that the segment includes data that must be delivered to the receiving application program as soon as possible and not to wait for more data to come.

Urgent data: Consider a situation where an application program needs to send urgent bytes. This means that the sending application program wants a piece of data to be read out of order by the receiving application program. Suppose that the sending application program is sending data to be processed by the receiving application program. When the result of processing comes back, the sending application program finds that everything is wrong. It wants to abort the process, but it has already sent a huge amount of data. If it issues an abort command (Control + C), these two characters will be stored at the end of the receiving TCP buffer. It will be delivered to the receiving application program after all the data has been processed.

The solution is to send a segment with the URG bit set. The sending application program tells the sending TCP that the piece of data is urgent. The sending TCP creates a segment and inserts the urgent data at the beginning of the segment. The rest of the segment can contain normal data from the buffer. The urgent pointer field in the header defines the end of the urgent data and the start of normal data.

When the receiving TCP receives a segment with the URG bit set, it extracts the urgent data from the segment, using the value of the urgent pointer, and delivers it, out of order, to the receiving application program.

9. Identify the two protocols present in the Transport layer.

Ans:

The two protocols present in the Transport layer are UDP and TCP.

10. What is the length of following addresses in terms of bytes?

a. Physical address b. IP addressc. Port address

Ans:

a. Physical address: 6 bytes

b. IP address: 4 bytes

c. Port address: 2 bytes


Chapter Six

Objectives


Understand the client server model Define process Understand FTP

Focus Areas

This chapter explains the application layer. You can explain the client server model with the help of an example. You can say that in a market when you requesting for an item from the shopkeeper, you are the client. Similarly, there is the concept of client in a network. The Web browser is the client, which requests for websites from the server. The server is like the shopkeeper.

You should refer to application layer given in the Additional Inputs section.

Explain the issues involved in the application layer using a client server model. Explain how a client program operates. Explain the concepts of active open, passive open, and concurrency. Also explain the two ways in which a server program can operate, concurrently or iteratively. Explain the difference between a program and a process. Finally, explain how the FTP (File Transfer Protocol) network application operates. Discuss the various commands of FTP.

If time permits, you can list the various protocols in the application layer given in the Additional Inputs section.

Additional Inputs


The Application Layer

The Application layer is the layer closest to the end-user. This layer provides protocols that are commonly needed and provides an interface that applications use to access network resources. For example, a full-screen editor that is supposed to work over a network generally has to work with many different terminal types. Each terminal may have different screen layouts; escape sequences for inserting and deleting text, moving the cursor, etc. In this situation, it is very difficult for the screen editor to work with the incompatible terminal types.

One way to solve this problem is to define an abstract network virtual terminal. This virtual terminal is managed by using editors and other programs. To handle each terminal type, a code must be written to map the functions of the network virtual terminal with the real terminal. For example, when the editor moves the virtual terminal's cursor to the upper left-hand corner of the screen, the proper command sequence must be issued to the real terminal to get its cursor there too. The entire virtual terminal software is in the Application layer.

Thus, the main function of the Application layer is to provide network services to the end-user or the computer application. Computer applications can be divided into three groups:

Network applications: These applications are specifically designed for working on a network. For example, using the Telnet program, you can connect to a remote machine and operate it.

Standalone applications: These application programs are for individual access and are not designed for a network. Solitaire is an example of a standalone application.

Embedded applications: These applications have other network applications imbedded in them. Microsoft Word is an example of an imbedded application as it allows you to imbed and activate e-mail or Web site references within a document.

Another important function of the Application layer is file transfer. The Application layer handles the different file systems because they have different file naming conventions, different ways of


representing text lines, and so on. The Application layer also handles e-mail, remote job entries, directory lookups, and various other general-purpose and special-purpose facilities.

Protocols in the Application Layer

The Application Layer implements a host of protocols that provide network connectivity and data transfer services for users. Some Application layer protocols include File Transfer Protocol (FTP), Telnet, Simple Mail Transfer Protocol (SMTP), Hypertext Transfer Protocol (HTTP), Simple Network Management Protocol (SNMP), Domain Naming Service (DNS), and Dynamic Host Configuration Protocol (DHCP). The following table provides a brief description of these protocols:

Protocol Description

FTP FTP is used to transfer files from one system to another on a network.

Telnet Telnet is used to access a remote system on a network and perform operations on it.

SMTP SMTP is used to transfer e-mail messages between the mail servers on a network, or between mail clients and a mail server.

HTTP HTTP is used to exchange text, graphics, image, audio, and video files over the WWW.

SNMP SNMP is used to monitor and manage network devices and their functions.

DNS DNS is used primarily to map or resolve a hostname to an IP address.

DHCP DHCP is used to dynamically allocate IP addresses on a network.

Chapter Six Questions

Part 1

1. In TCP/IP model, the session and presentation layers are integrated with _________ layer.

Ans.

Application

2. In TCP/IP the two communicating process can be best described using the __________________ model.

Ans.

Client-server

3. In client-server, the _________ program is always running.

Ans.

Server


4. Connection-oriented concurrent server uses the service of ________ transport protocol.

Ans.

TCP

5. In client-server model, a user wanting a particular service runs the ________ program.

Ans.

Application

6. FTP is used to transfer ________ from one host to another.

Ans.

Files

7. FTP uses _________ protocol, for transmission purpose.

Ans.

File transfer

8. The number of connections required for FTP operation is ______.

Ans.

Two

9. Commands are sent over ________ connection.

Ans.

Control

10. Information regarding the file, such as file type, structure, and transmission mode are sent to the server _______ the data transfer takes place.

Ans.

Before

Part 2

1. Write a brief description about Client-server model.

Ans.

The purpose of a TCP/IP or Internet is to provide data communication services to users. The hurdles created because of the geographical distance while accessing information from another host should be minimized. If a user at a local host computer wishes to receive a service from a computer at a remote site. Computer executes specific programs to perform the specific job. There would be a program to do the Word processing, a program to browse the web site and so on. In other words for data communication to take place, a computer runs a program to request a service from another program residing on the destination computer. This means that two computers connected by an Internet should each run a program, one to provide a service and one to request a service.

It is important to determine beforehand, who makes the request for service and who offers it. Issue of whether both the application programs are able to request services and provide


services or should the application programs just do one or the other, must be resolved. One solution is to have an application program, called the client, running on the local host computer, request a service from another application program, called the server, running on the remote computer. In other words, the task of requesting program is either a requester (a client), or a provider (a server). If a machine needs to request a service and provide a service, two application programs must be installed. In other words, application programs come in pairs.

An application program in a client server model provides services to any application program installed anywhere in an Internet that requests this service. Therefore, a server provides service to any client. In other words, the client-server relationship is many-to-one. Many clients can use the services of one server.

In a client server model, a client program, which requests a service, should run only when it is needed. The server program, which provides a service, should run all the time because it does not know when its service is needed.

2. What is a connectionless-iterative server? Using a diagram explain the same.

Ans.

The servers that use UDP are normally iterative, which means that the server processes one request at a time. A server gets the request received in a datagram from UDP, processes the request, and gives the response to UDP to send to the client. The server pays no attention to the other datagrams. These datagrams are stored in a queue, waiting for service. They could all be from one client or from many clients. In either case they are processed one by one in order of arrival. The following figure illustrates the same:

Connectionless-Iterative Server

3. Write a brief description about the connection-oriented concurrent server.

Ans.

The servers that use TCP are normally concurrent. This means that the server can serve many clients at the same time. Communication is connection-oriented, which means that a request is a stream of bytes that can arrive in several segments and the response can occupy several segments. A connection is established between the server and each client, and


the connection remains open until the entire stream is processed and the connection is terminated.

This type of server cannot use only one well-known port because each connection needs a port and many connections may be open at the same time. Many ports are needed, but a server can use only one well-known port. The solution is to have one well-known port and many ephemeral ports. The server makes a passive open at the well-known port. A client can make its initial approach to this port to make the connection. After the connection is made, the server assigns a temporary port to this connection to free the well-known port. Data transfer can now take place between these two temporary ports, one at the client site and the other at the server site. The well-known port is now free for another client to make the connection. The idea is to push demultiplexing to TCP instead of the server.

The server must also have one buffer for each connection. The segments coming from the client are stored in the appropriate buffer, and will be served concurrently by the server.

To provide this service, most implementations use the concept of parent and child servers. A server running infinitely and accepting connections from the clients is called a parent server. The parent uses the well-known port. After it makes a connection, the parent server creates a child server and an ephemeral port and lets the child server handle he service. It thereby frees itself so that it can wait for another connection.

4. Explain the concept of process.

Ans.

A process is an instance of a program. When the operating system executes a program, an instance of the program, a process, is created. The operating system can create several processes from one program, which means several instances of the same program are running at the same time (concurrently). Although all processes have the same data types, memory is allocated for each process separately. In addition, the values stored in variables may be totally different from one process to another. The functions executed by each process of the same program may differ, as each may take different inputs.

5. Write a short note on FTP.

Ans.

File transfer protocol (FTP) is the standard mechanism provided by TCP/IP for copying a file from one host to another. Transferring files from one computer to another is one of the most common tasks expected from the networking environment.

Although transferring files from one system to another seems simple and straightforward, some problems must be dealt with first. For example, two systems may use different file name conventions. Two systems may have different ways to represent text and data. Two systems may have different directory structures. All of these problems have been solved by FTP in a very simple and elegant approach.

FTP differs from other client-server applications in that it establishes two connections between the hosts. One connection is used for data transfer, the other for control information (commands and responses). Separation of commands and data transfer makes FTP more efficient. The control connection uses very simple rules of communication. We need to transfer only a line of command or a line of response at a time. The data connection, on the other hand, needs more complex rules due to the variety of data type transferred.

FTP uses two well-known TCP ports, Port 21 is used for the control connection, and port 20 is used for the data connection.

The client has three components: user interface, client control process, and the client data transfer process. The server has two components: the server control process and the server data transfer process. The control connection is made between the control processes. The data connection is made between the data processes. The control connection remains


connected during the entire interactive FTP session. The data connection is opened and then closed for each file transferred. It opens each time commands that involve transferring files are used, and it closes when the file is transferred. Similarly, when a user starts an FTP session, the control connection is opened. While the control connection is open, the data connection can be opened and closed multiple times if several files are transferred.

The two FTP connections control and data use different strategies and different port numbers.


6. Write a brief description about the control connection used in FTP.

Ans.

The control connection is created in the same way as other application programs described so far.

There are two steps:

12. The server issues a passive open on the well-known port 21 and waits for a client.13. The client uses an ephemeral port and issues an active open connection to the port 21 of

server process.

The connection remains open during the entire process. The service type, used by the IP protocol, is to minimize delay because this is an interactive connection between a user (human) and a server. The user types commands and expects to receive responses without significant delay. After the initial connection, the server process creates a child process and assigns the duty of serving the client to the child process using an ephemeral port.

7. Write a brief description about the data connection used in FTP.

Ans.

The data connection uses the well-known port 20 at the server site. However, the creation of a data connection is different from what we have seen so far. The following shows how FTP creates a data connection:

14. The client, not the server, issues a passive open using an ephemeral port. The client must do this because it is the client that issues the commands for transferring files.

15. The client sends this port number to the server using the PORT command.16. The server receives the port number and issues an active open using the well-known port

20 and the received ephemeral port number.17. After the initial connection, the server process creates a child process and assigns the duty

of serving the client to the child process using an ephemeral port.

8. Write a brief note on the Commands processing. Describe the groups into which Commands (that are sent from client to server) can be divided.

Ans.

FTP uses the control connection to establish a communication between the client control process and the server control process. During this communication, the commands are sent from the client to the server and the responses are sent back from the server to the client.

Commands: We can roughly divide the commands into six groups: access commands, file management commands, data formatting commands, port defining commands, file transferring commands, and miscellaneous commands.

Access commands: These commands let the user access the remote system.

Commands are - USER, PASS, ACCT, REIN, QUIT, ABOR

File management commands: These commands let the user access the file system on the remote computer. They allow the user to navigate through the directory structure, create new directories, delete files, and so on.

Commands are – CWD, CDUP, DELE, LIST, ….Data formatting commands: These commands let the user define the data structure, file type and transmission mode. The defined format is then used by the file transfer commands.

Commands are – TYPE, STRU, and MODE.

Port defining commands: These commands define the port number for the data connection on the client site.

Commands are – PORT, PASV


File transfer commands: These commands actually let the user transfer files.

RETR, STOR, APPE, STOU, ….

Miscellaneous commands: These commands deliver information to the FTP user at the client site.Commands are - HELP, NOOP, …

9. Briefly explain the steps involved in storing an image file into the server from client side.

Ans.

The steps involved in storing an image file into the server from client side are as follows:

18. The control connection to port 21 is created.19. After the control connection to port 21 is created, the FTP server sends the 220 (service

ready) response on the control connection.20. The client sends the USE command (this USER command passes the user information to

server).21. The server responds with 331 (user name is OK, a password is required).22. The client sends the PASS command (sends the password for the user).23. The server responds with 230 (user login OK, if user id and password match).24. The client issues a passive open on an ephemeral port for data connection and sends the

PORT command (over the control connection) to give this port number to the server.25. The server does not open the connection at this time, but prepares itself for issuing an

active open on the data connection between port 20 (server side) and the ephemeral port received from the client. It sends the response 150 (data connection will be open shortly).

26. The client sends the TYPE command (to indicate the type of data as binary …)27. The server responds with response 200 (command OK)28. The client sends the STRU command (defines the data following as of file type.)29. The server responds with response 200 (command OK)30. The client sends the STOR command (to request the server to store the data)31. The server opens the data connection and sends the message 250.32. The client sends the file on the data connection. After the entire file is sent the data

connection is closed. Closing the data connection means end-of-file.33. The server sends the response 226 on the control connection.34. The client sends the QUIT command (or it can send the other commands to open another

data connection for transferring another file).35. The server responds with 221 (service closing) and it closes the control connection.

FAQ

1. What is concurrency?

Ans:

Concurrency means the ability to run simultaneously. Both clients and serves can run in concurrent mode. However, the concurrency in clients and server differ from each other.

Concurrency in Clients:

Clients can be run on a computer either iteratively or concurrently. Running clients iteratively means running them one by one; one client must start, run, and terminate before the computer can start another client. However today, most computers allow concurrent clients, that is, two or more clients running at the same time.

Concurrency in Servers

Because an iterative server can process only one request at a time, it receives a request, processes it, and sends the response to the requestor before it handles another request. If there is a request from another client, then it has to either reject the request or keep the request in waiting until the server finishes the first one. On the other hand, a concurrent server can process many requests at the same time and thus can share its time between many requests.


2. What are a connectionless iterative server and a connectionless concurrent server?

Ans:

Connectionless iterative server: The servers that use UDP, a connectionless transport layer protocol, are normally iterative. This means that the server processes one client request at a time in order of their arrival. A server receives the request in a datagram from UDP, processes the request, and gives the response to UDP to send to the client.

Connectionless concurrent server: The servers that use UDP, a connectionless transport layer protocol can also be iterative. This means that the server processes multiple client requests at the same time. This type of server will be found occasionally.

3. What are a connection-oriented iterative server and a connection-oriented concurrent server?

Ans:

Connection-oriented iterative server: The servers that use TCP, a connection-oriented transport layer protocol can be concurrent. This means that the server can serve one client request at a time. This type of server will be found occasionally.

Connection-oriented concurrent server: The servers that use TCP, a connection-oriented transport layer protocol, are normally concurrent. This means that the server can serve many clients at the same time. Communication is connection-oriented, which means that a request is a stream of bytes that can arrive in several segments and the response can occupy several segments. A connection is established between the server and each client, and the connection remains open until the entire stream is processed and the connection is terminated.

4. What is an active open connection?

Ans:

A client opens the communication channel using the IP address of the remote host and the well-known port address of the specific server program running on that machine. This is called an active open connection.

5. What is a passive open connection?

Ans:

A server opens the channel for incoming requests from clients, but it never initiates a service until it is requested to do so. This is called a passive open connection.

6. What is a control connection?

Ans:

FTP uses ASCII character set to communicate across the control channel. Communication is achieved through commands and responses. This simple method is adequate for the control connection because we send one command (response) at a time. Each command or response is only one short line so we need not worry about file format or file structure. Each line is terminated with a two-character (carriage return and line feed) end-of-line token.

7. What is data connection?

Ans:

Data connection allows you to transfer files after defining the types of file to be transferred, the structure of the data, and the transmission mode. Before sending the file through the data connection, you need to prepare for transmission through the control connection. The heterogeneity problem is resolved by defining three attributes of communications: file type, data


structure, and transmission mode. A common format for the above three attributes is agreed upon before actually transferring the data.


8. Write a brief description about client-server model.

Ans:

The purpose of a client-server model is to provide data communication services to users. In the client-server model, an application program, called the client, requests for a service from another application program, called the server. In other words, when a machine needs to request for a service, it is called a requester or a client. When a machine provides a service, it is called a provider or a server.

9. Write a short note on FTP.

Ans:

File transfer protocol (FTP) is the standard mechanism provided by TCP/IP to copy a file from one host to another. When two systems use different file name conventions, different ways to represent text and data, and different directory structures, you can use FTP in such a situation to transfer files. FTP establishes two connections between the hosts. One connection is used for data transfer and the other for control information (commands and responses). FTP uses two well-known TCP ports, port 20 is used for the data connection and port 21 is used for the control connection.

10. What are the common FTP commands for processing?

Ans:

FTP uses the control connection to establish a communication between the client control process and the server control process. During this communication, the FTP commands are sent from the client to the server and the responses are sent back from the server to the client. You can divide the FTP commands into six groups: access commands, file management commands, data formatting commands, port defining commands, file transferring commands, and miscellaneous commands.

Access Commands: These commands let the user access the remote system.

Commands are - USER, PASS, ACCT, REIN, QUIT, ABOR

File Management Commands: These commands let the user access the file system on the remote computer. They allow the user to navigate through the directory structure, create new directories, delete files, and so on.

Commands are – CWD, CDUP, DELE, LIST, …

Data Formatting Commands: These commands let the user define the data structure, file type and transmission mode. The defined format is then used by the file transfer commands.

Commands are – TYPE, STRU, and MODE.

Port Defining Commands: These commands define the port number for the data connection on the client site.

Commands are – PORT, PASV

File Transfer Commands: These commands actually let the user transfer files.

RETR, STOR, APPE, STOU, …

Miscellaneous Commands: These commands deliver information to the FTP user at the client site.

Commands are - HELP, NOOP, …


Kuvempu Univeristy BSc(IT) 3rd semester exercise answer (Computer Networks II 34)

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