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Computer Networks Computer Networks

UNIT-I

A Computer network to mean a collection of autonomous computers interconnected by a single technology.

Two computers are said to be interconnected if they are able to exchange information. The connection need
not be via a copper wire; fiber optics, microwaves, infrared, and communication satellites can also be used.
The computers are autonomous, which are not forcibly started or controlled by other one. A system with one
control unit and many slaves is not a computer network.
A computer network consists of end systems (or) nodes which are capable of transmitting of information,
and which communicate through transit system interconnecting them. The transit system is also called an
interconnection subsystem or simply a subnetwork. An end system comprises of terminals, software and
peripherally forming an autonomous system capable of performing information processing.
 The old model of a single computer serving all of the organization's computational needs has
been replaced by one in which a large number of separate but interconnected computers do the
In a multipoint environment, the capacity of the channel is shared, either spatially or temporally. If several
job. These systems are called computer networks.
devices can use the link simultaneously, it is a spatially shared connection. If users must take turns, it is a
Network Criteria: timeshared connection.
A network must be able to meet a certain number of criteria. The most important of these are performance,
reliability, and security. Physical Topology
Performance: It can be measured in many ways, including transit time and response time. The term physical topology refers to the way in which a network is laid out physically. A network
o Transit time is the amount of time required for a message to travel from one device to another. topology is the arrangement of a network, including its nodes and connecting lines. There are two ways
of defining network geometry: the physical topology and the logical (or signal) topology. There are four
o Response time is the elapsed time between an inquiry and a response.
The performance of a network depends on a number of factors, including the number of users, the type of basic topologies possible: mesh, star, bus, and ring.
transmission medium, the capabilities of the connected hardware, and the efficiency of the software. Mesh Topology
Performance is often evaluated by two networking metrics: throughput and delay. In a mesh topology, every device has a dedicated point-to-point link to every other device. The term
Reliability: dedicated means that the link carries traffic only between the two devices it connects.
Network reliability is measured by the frequency of failure, the time it takes a link to recover from a failure,  In a mesh topology, we need n (n – 1) / 2 duplex-mode links.
and the network’s robustness in a catastrophe. Every device on the network must have n – 1 input/output (I/O) ports to be connected to the other n – 1
station.
Security:
Network security issues include protecting data from unauthorized access, protecting data from damage and
development, and implementing policies and procedures for recovery from breaches and data losses.

Physical Structures:
Type of Connection
A network is two or more devices connected through links. A link is a communications pathway that
transfers data from one device to another. There are two possible types of connections: point-to-point and
multipoint.
Point-to-Point:
A point-to-point connection provides a dedicated link between two devices. The entire capacity of the link Fig: Mesh Topology
is reserved for transmission between those two devices.
Advantages:
Multipoint:  The use of dedicated links guarantees that each connection can carry its own data load, thus eliminating
A multipoint (also called multidrop) connection is one in which more than two specific devices share a the traffic problems that can occur when links must be shared by multiple devices.
single link.  A mesh topology is robust. If one link becomes unusable, it does not incapacitate the entire system.
 There is the advantage of privacy or security. When every message travels along a dedicated line, only
the intended recipient sees it. Physical boundaries prevent other users from gaining access to messages.
 Finally, point-to-point links make fault identification and fault isolation easy.
Disadvantages:
 Installation and reconnection are difficult.
 The sheer bulk of the wiring can be greater than the available space can accommodate.
 The hardware required to connect each link (I/O ports and cable) can be prohibitively expensive.
Ex: connection of telephone regional offices.
Star Topology

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Computer Networks Computer Networks

In a star topology, each device has a dedicated point-to-point link only to a central controller, usually
called a hub. The devices are not directly linked to one another. A star topology does not allow direct traffic
between devices.
The controller acts as an exchange: If one device wants to send data to another, it sends the data to the
controller, which then relays the data to the other connected device.

Fig: Ring Topology

Advantages:
 A ring is relatively easy to install and reconfigure.
Fig: Star Topology
Advantages:  To add or delete a device requires changing only two connections.
 Less expensive than a mesh topology.  Fault isolation is simplified.
 Easy to install and reconfigure. Disadvantages:
 Unidirectional traffic can be a disadvantage.
 Additions, moves, and deletions involve only one connection: between that device and the hub.
 Star topology is robust.  A break in the ring (such as a disabled station) can disable the entire network.
 If one link fails, only that link is affected. All other links remain active. This factor also lends itself to Network Types:
easy fault identification and fault isolation. One network can be distinguished from another network based on few criteria such as size, Geographical
 Hub can be used to monitor link problems and bypass defective links. area, and ownership. There are 2 basic types of Networks. They are Local area networks and Wide Area
Disadvantage: Networks.
 Star topology is the dependency of the whole topology on one single point, the hub. If the hub goes
down, the whole system is dead. Local Area Network (LAN): LAN’s, are privately-owned networks within a single building or campus of up
to a few kilometers in size. They are widely used to connect personal computers and workstations in
Bus Topology company offices and factories to share resources (e.g., printers) and exchange information.
A bus topology is example of multipoint Link. One long cable acts as a backbone to link all the devices in a
 Each host in a LAN has an identifier, an address that uniquely defines the host in the LAN.
network.
 A packet sent by a host to another host carries both the source host’s and the destination host’s
addresses.
LANs are distinguished from other kinds of networks by three characteristics:
1) Their size
2) Their transmission technology, and
3) Their topology.
Fig: Bus Topology
I. Size: LANs are restricted in size, which means that the worst-case transmission time is bounded and
Nodes are connected to the bus cable by drop lines and taps. A drop line is a connection running between
known in advance.
the device and the main cable. A tap is a connector that either splices into the main cable or punctures the
II. Transmission technology: LANs consisting of a cable to which all the machines are attached.
sheathing of a cable to create a contact with the metallic core.
Traditional LANs run at speeds of 10 Mbps to 100 Mbps, have low delay (microseconds or
Advantages:
nanoseconds), and make very few errors. NewerLANs operate at up to 10 Gbps. [(1 Mbps is 1,000,000
 Ease of installation.
bits/sec) and gigabits/sec (1 Gbps is 1,000,000,000 bits/sec)].
 A bus uses less cabling than mesh or star topologies. III. Topology: Various topologies are possible for broadcast LANs. Ex: Bus and Ring.
 Only the backbone cable stretches through the entire facility
Disadvantages:
 It includes difficult reconnection and fault isolation.
 Difficult to add new devices.
 Signal reflection at the taps can cause degradation in quality.
 Fault or break in the bus cable stops all transmission, even between devices on the same side of the
problem. The damaged area reflects signals back in the direction of origin, creating noise in both
directions.
Ring Topology
In a ring topology, each device has a dedicated point-to-point connection with only the two devices on
either side of it. A signal is passed along the ring in one direction, from device to device, until it reaches its
destination. Each device in the ring incorporates a repeater. Wide Area Network:
 A wide area network (WAN) is also an interconnection of devices capable of communication.
 A wide area network, or WAN, spans a large geographical area, often a country or continent.

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Computer Networks Computer Networks

 It contains a collection of machines intended for running user (i.e., application) programs. An internet is a switched network in which a switch connects at least two links together. The two most
 A WAN interconnects connecting devices such as switches, routers, or modems. common types of switched networks are circuit-switched and packet-switched networks.
 A WAN is normally created and run by communication companies and leased by an organization that i. Circuit-Switched Network
uses it. In a circuit-switched network, a dedicated connection, called a circuit, is always available between the two
Ex: point-to-point WANs and switched WANs. end systems; the switch can only make it active or inactive.
The collections of machines called as hosts. The hosts are connected by a communication subnet, or just In Figure, the four telephones at each side are connected to a switch. The switch connects a telephone set at
subnet for short. one side to a telephone set at the other side. The thick line connecting two switches is a high-capacity
 The job of the subnet is to carry messages from host to host. communication line that can handle four voice communications at the same time; the capacity can be shared
 In most wide area networks, the subnet consists of two distinct components: transmission lines and between all pairs of telephone sets.
switching elements.
 Transmission lines: move bits between machines. They can be made of copper wire, optical
fiber, or even radio links.
 Switching elements: These are specialized computers that connect three or more transmission
lines. When data arrive on an incoming line, the switching element must choose an outgoing
line on which to forward them. These switching computers have been called as Router.
 The collection of communication lines and routers (but not the hosts) form the subnet. Fig: A circuit-switched network
i. Point-to-Point WAN:  A circuit-switched network is efficient only when it is working at its full capacity; most of the time, it is
A point-to-point WAN is a network that connects two communicating devices through a transmission media inefficient because it is working at partial capacity.
(cable or air).
ii. Packet-Switched Network:
In a computer network, the communication between the two ends is done in blocks of data called packets.
This allows us to make the switches function for both storing and forwarding because a packet is an
independent entity that can be stored and sent later.

ii. Switched WAN:

A switched WAN is a network with more than two ends. Switched WAN is a combination of several point-
to-point WANs that are connected by switches.
Fig: A packet-switched network
A router in a packet-switched network has a queue that can store and forward the packet. If only two
computers need to communicate with each other, there is no waiting for the packets. However, if packets
arrive at one router when the thick line is already working at its full capacity, the packets should be stored
and forwarded in the order they arrived.
Packet-switched network is more efficient than a circuit switched network, but the packets may
encounter some delays.
Internetwork:
When two or more networks are connected, they make an internetwork, or internet. The Internet
A collection of interconnected networks is called an internetwork or internet. Internet is composed of thousands of interconnected networks. The figure shows the Internet as several
 Internet is a collection of LANs connected by a WAN. backbones, provider networks, and customer networks.

Fig: An internetwork made of two LANs and one point-to-point WAN

When a host in the west coast office sends a message to another host in the same office, the router blocks the
message, but the switch directs the message to the destination. On the other hand, when a host on the west
coast sends a message to a host on the east coast, router R1 routes the packet to router R2, and the packet
reaches the destination.
Switching:

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Computer Networks Computer Networks

ARPANET
In 1967, at an Association for Computing Machinery (ACM) meeting, ARPA presented its ideas for the
Advanced Research Projects Agency Network (ARPANET), a small network of connected computers.
The idea was that
Each host computer would be attached to a specialized computer, called an interface message
processor (IMP).
The IMPs, in turn, would be connected to each other.
Each IMP had to be able to communicate with other IMPs as well as with its own attached host.
By 1969, ARPANET was a reality.
Network Control Protocol (NCP) provided communication between the hosts.
Birth of the Internet
To link dissimilar networks, there were many problems to overcome: diverse packet sizes, diverse interfaces,
and diverse transmission rates, as well as differing reliability requirements. Cerf and Kahn devised the idea
o The backbones are large networks are connected through some complex switching systems, called
peering points. of a device called a gateway to serve as the intermediary hardware to transfer data from one network to
o Provider networks are smaller networks that use the services of the backbones for a fee. another.
o The customer networks use the services provided by the Internet. TCP/IP
Backbones and provider networks are also called Internet Service Providers (ISPs). The backbones are In 1973 Cerf and Kahn outlined the protocols transmission control protocol (TCP) included concepts such
often referred to as international ISPs; the provider networks are often referred to as national or regional as encapsulation, the datagram, and the functions of a gateway.
ISPs. A radical idea was the transfer of responsibility for error correction from the IMP to the host machine.
In TCP/IP, IP would handle datagram routing while TCP would be responsible for higher level functions
Accessing the Internet
such as segmentation, reassembly, and error detection.
Using Telephone Networks:
In 1981, under a Defence Department contract, UC Berkeley modified the UNIX operating system to include
One option for residences and small businesses to connect to the Internet is to change the voice line between
TCP/IP but it did much for the popularity of internetworking.
the residence or business and the telephone center to a point-to-point WAN. This can be done in two ways.
In 1983, TCP/IP became the official protocol for the ARPANET.
 Dial-up service: The first solution is to add to the telephone line a modem that converts data to
voice. MILNET
 DSL Service: The DSL service also allows the line to be used simultaneously for voice and data In 1983, ARPANET split into two networks: Military Network (MILNET) for military users and
communication. ARPANET for nonmilitary users.

Using Cable Networks: CSNET

A residence or a small business can be connected to the Internet by using this service. It provides a higher Computer Science Network (CSNET) was created in 1981 and it was sponsored by National Science
speed connection, but the speed varies depending on the number of neighbors that use the same cable. Foundation (NSF).
Using Wireless Networks:
NSFNET
Wireless connectivity has recently become increasingly popular. With the growing wireless WAN access, a
household or a small business can be connected to the Internet through a wireless WAN. With the success of CSNET, the NSF in 1986 sponsored the National Science Foundation Network
(NSFNET), a backbone that connected five supercomputer centers located throughout the United States.
Direct Connection to the Internet:
A large organization or a large corporation can itself become a local ISP and be connected to the Internet. ANSNET
This can be done if the organization or the corporation leases a high-speed WAN from a carrier provider and In 1991, the U.S. government decided that NSFNET was not capable of supporting the rapidly increasing
connects itself to a regional ISP. For example, a large university with several campuses can create an Internet traffic. Three companies, IBM, Merit, and Verizon, filled the void by forming a nonprofit
internetwork and then connect the internetwork to the Internet. organization called Advanced Network & Services (ANS) to build a new, high-speed Internet backbone
called Advanced Network Services Network (ANSNET).
INTERNET HISTORY
Early History
Before 1960, there were telegraph and telephone networks, suitable for constant-rate communication at that
time, which means that after a connection was made between two users, the encoded message (telegraphy) or Internet Today
voice (telephony) could be exchanged. To handle bursty data we needed to invent packet-switched network. The Internet today is a set of pier networks that provide services to the whole world.
Birth of Packet-Switched Networks World Wide Web
The Web was invented at CERN by Tim Berners-Lee. This invention has added the commercial applications
The theory of packet switching for bursty traffic was first presented by Leonard Kleinrock in 1961 at MIT.
to the Internet.

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Computer Networks Computer Networks

Multimedia RFCs are classified into five requirement levels: required, recommended, elective, limited use, and not
Recent developments in the multimedia applications such as voice over IP (telephony), video over IP recommended.
(Skype), view sharing (YouTube), and television over IP (PPLive) has increased the number of users and the  Required: An RFC is labeled required if it must be implemented by all Internet systems to achieve
amount of time each user spends on the network.
minimum conformance. For example, IP and ICMP (Chapter 19) are required protocols.
Peer-to-Peer Applications
Peer-to-peer networking is also a new area of communication with a lot of potential.  Recommended: An RFC labeled recommended is not required for minimum conformance; it is
recommended because of its usefulness. For example, FTP and TELNET.
STANDARDS AND ADMINISTRATION  Elective: An RFC labeled elective is not required and not recommended. However, a system can use it
An Internet standard is a thoroughly tested specification that is useful to and adhered to by those who work for its own benefit.
with the Internet. It is a formalized regulation that must be followed. There is a strict procedure by which a  Limited Use: An RFC labeled limited use should be used only in limited situations.
specification attains Internet standard status. A specification begins as an Internet draft.  Not Recommended. An RFC labeled not recommended is inappropriate for general use. Normally a
An Internet draft is a working document with no official status and a six-month lifetime. historic (deprecated) RFC may fall under this category.
A draft may be published as a Request for Comment (RFC). Each RFC is edited, assigned a number, and
made available to all interested parties. Internet Administration
General organization of Internet administration is:
ISOC
The Internet Society (ISOC) is an international, nonprofit organization formed in 1992 to provide support
for the Internet standards process. ISOC accomplishes this through maintaining and supporting other Internet
administrative bodies such as IAB, IETF, IRTF, and IANA. ISOC also promotes research and other scholarly
activities relating to the Internet.
IAB
The Internet Architecture Board (IAB) is the technical advisor to the ISOC. The main purposes of the IAB
are to oversee the continuing development of the TCP/IP Protocol Suite and to serve in a technical advisory
capacity to research members of the Internet community. IAB accomplishes this through its two primary
components, the Internet Engineering Task Force (IETF) and the Internet Research Task Force (IRTF).
IETF
The Internet Engineering Task Force (IETF) is a forum of working groups managed by the Internet
Engineering Steering Group (IESG). IETF is responsible for identifying operational problems and proposing
Maturity Levels
solutions to these problems. IETF also develops and reviews specifications intended as Internet standards.
An RFC, during its lifetime, falls into one of six maturity levels: proposed standard, draft standard, Internet
IRTF
standard, historic, experimental, and informational.
The Internet Research Task Force (IRTF) is a forum of working groups managed by the Internet Research
 Proposed Standard: A proposed standard is a specification that is stable, well understood, and of
Steering Group (IRSG). IRTF focuses on long-term research topics related to Internet protocols,
sufficient interest to the Internet community. At this level, the specification is usually tested and
applications, architecture, and technology.
implemented by several different groups.
 Draft Standard: A proposed standard is elevated to draft standard status after at least two successful PROTOCOL LAYERING
independent and interoperable implementations. Barring difficulties, a draft standard, with Protocol defines the rules that both the sender and receiver and all intermediate devices need to follow to be
modifications if specific problems are encountered, normally becomes an Internet standard. able to communicate effectively. When communication is simple, we may need only one simple protocol;
 Internet Standard: A draft standard reaches Internet standard status after demonstrations of successful when the communication is complex, we may need to divide the task between different layers, in which case
implementation. we need a protocol at each layer, or protocol layering.
 Historic: The historic RFCs are significant from a historical perspective. They either have been Scenarios
superseded by later specifications or have never passed the necessary maturity levels to become an First Scenario
Internet standard. In the first scenario, communication is so simple that it can occur in only one layer. Assume Maria and Ann
 Experimental: An RFC classified as experimental describes work related to an experimental situation is neighbors with a lot of common ideas. Communication between Maria and Ann takes place in one layer,
that does not affect the operation of the Internet. Such an RFC should not be implemented in any face to face, in the same language.
functional Internet service.
 Informational: An RFC classified as informational contains general, historical, or tutorial information
related to the Internet. It is usually written by someone in a non-Internet organization, such as a vendor.

Requirement Levels

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Computer Networks Computer Networks

Logical connection means that we have layer-to-layer communication. The concept of logical connection
will help us better understand the task of layering.

Second Scenario
In this scenario communication between Sender and Receiver takes place in three layers, as shown in Figure.
We assume that Ann and Maria each have three machines that can perform the task at each layer.

The OSI Reference Model:

The OSI model is based on a proposal developed by the International Standards Organization
(ISO) as a first step toward international standardization of the protocols used in the various layers. The
model is called the ISO OSI (Open Systems Interconnection) Reference Model because it deals with
connecting open systems—that is, systems that are open for communication with other systems.
The OSI model has seven layers. The principles that were applied to arrive at the seven layers can be
briefly summarized as follows:
1. A layer should be created where a different abstraction is needed.
2. Each layer should perform a well-defined function.
3. The function of each layer should be chosen with an eye toward defining internationally standardized
protocols.
4. The layer boundaries should be chosen to minimize the information flow across the interfaces.
5. The number of layers should be large enough that distinct functions need not be thrown together in the
same layer out of necessity and small enough that the architecture does not become unwieldy.
Let us assume that Maria sends the first letter to Ann. Maria talks to the machine at the third layer as though Functions of each layer
the machine is Ann and is listening to her. The third layer machine listens to what Maria says and creates the 1) The Physical Layer:
plaintext (a letter in English), which is passed to the second layer machine. The second layer machine takes The physical layer is concerned with transmitting raw bits over a communication channel. The design
issues have to do with making sure that when one side sends a 1 bit, it is received by the other side as a 1 bit,
the plaintext, encrypts it, and creates the ciphertext, which is passed to the first layer machine. The first layer
not as a 0 bit.
machine, presumably a robot, takes the ciphertext, puts it in an envelope, adds the sender and receiver The simple tasks of physical layer to be considered are,
addresses, and mails it. i. How many volts should be used to represent a 1 and how many for a 0?
Protocol layering enables us to divide a complex task into several smaller and simpler tasks. ii. How the initial connection is established and how it is torn down when both sides are finished?
Advantages of protocol layering: iii. How many pins the network connector has and what each pin is used for.
 It allows us to separate the services from the implementation. The design issues here largely deal with mechanical, electrical, and timing interfaces, and the physical
transmission medium, which lies below the physical layer.
 which cannot be seen in our simple examples but reveals itself when we discuss protocol layering in
the Internet, is that communication does not always use only two end systems; there are intermediate
systems that need only some layers, but not all layers.
Principles of Protocol Layering
There are two principles of protocol layering.
First Principle
If we want bidirectional communication, we need to make each layer so that it is able to perform two
opposite tasks, one in each direction.
For example, the third layer task is to listen (in one direction) and talk (in the other direction). The second
layer needs to be able to encrypt and decrypt. The first layer needs to send and receive mail.
Second Principle
The two objects under each layer at both sites should be identical.
For example, the object under layer 3 at both sites should be a plaintext letter. The object under layer 2 at
both sites should be a ciphertext letter. The object under layer 1 at both sites should be a piece of mail.
Logical Connections

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Computer Networks Computer Networks

 One widely-used application protocol is HTTP (HyperText Transfer Protocol), which is the basis for
the World Wide Web.
Other application protocols are used for file transfer, electronic mail, and network news.
Figure 1-20. The OSI reference model.
OSI vs TCP/IP:
2) The Data Link Layer: Session and Presentation layers were not added to the TCP/IP protocol suite after the publication of the OSI
The main task of the data link layer is to transform a raw transmission facility into a line that appears model. The application layer in the suite is usually considered to be the combination of three layers in the
free of undetected transmission errors to the network layer. It accepts the data unit from the network layer OSI model.
and adds meaningful bits at the beginning (header) and end (trailer) that contains address and control
information. The data unit with this additional information is called Frame. His layer is responsible for
station to station deliveries.
Functions of data link layer are:
i. It provides services to network layer and accepts services from physical layer.
ii. It is the responsible of the data link layer for node-to-node delivery.
iii. It keeps a fast transmitter from drowning a slow receiver in data.
iv. It regulates traffic mechanism is often needed to let the transmitter know how much buffer space the
receiver has at the moment.
v. A special sublayer of the data link layer, the medium access control sublayer, deals with how to control
access to the shared channel.
3) The Network Layer:
Two reasons were mentioned for this decision.
 The network layer controls the operation of the subnet. A key design issue is determining how
First, TCP/IP has more than one transport-layer protocol. Some of the functionalities of the session layer are
packets are routed from source to destination.
available in some of the transport-layer protocols.
 If too many packets are present in the subnet at the same time, they will get in one
Second, the application layer is not only one piece of software. Many applications can be developed at this
another's way, forming bottlenecks. The control of such congestion also belongs to the
network layer. layer. If some of the functionalities mentioned in the session and presentation layers are needed for a
 The network layer is responsible for Logical Addressing and Routing. particular application, they can be included in the development of that piece of software.
4) The Transport Layer: Lack of OSI Model’s Success
 The basic function of the transport layer is to accept data from above, split it up into smaller units if The OSI model appeared after the TCP/IP protocol suite. TCP/IP protocol cannot be fully replaced by the
need be, pass these to the network layer, and ensure that the pieces all arrive correctly at the other OSI for three reasons:
end. First, OSI was completed when TCP/IP was fully in place and a lot of time and money had been spent on the
 The transport layer also determines what type of service to provide to the session layer. suite; changing it would cost a lot.
The most popular type of transport connection is an error-free point-to-point channel that delivers Second, some layers in the OSI model were never fully defined.
messages or bytes in the order in which they were sent. However, other possible kinds of transport service Third, when OSI was implemented by an organization in a different application, it did not show a high
are the transporting of isolated messages, with no guarantee about the order of delivery, and the enough level of performance to entice the Internet authority to switch from the TCP/IP protocol suite to the
broadcasting of messages to multiple destinations. OSI model.
 The transport layer is a true end-to-end layer, all the way from the source to the destination.
 Transport layer is responsible for, service-point addressing, Segmentation and reassembly, TCP/IP PROTOCOL SUITE
Connection control and Error control. TCP/IP is a protocol suite (a set of protocols organized in different layers) used in the Internet today. It is a
5) The Session Layer: hierarchical protocol made up of interactive modules, each of which provides a specific functionality.
The session layer allows users on different machines to establish sessions between them. Sessions offer
Layers in the TCP/IP Protocol Suite
various services, they are:
 Dialog control, token management, and synchronization.
Dialog control: keeping track of whose turn it is to transmit.
Token management: preventing two parties from attempting the same critical operation at the same time.
Synchronization: Checkpointing long transmissions to allow them to continue from where they were after a
crash.
6) The Presentation Layer:
 The presentation layer is concerned with the syntax and semantics of the information transmitted.
 Example of the presentation service is, the data structures to be exchanged can be defined in an
abstract way, along with a standard encoding to be used ''on the wire.''
 This layer mainly concerned with Translation, Encryption & Compression.
7) The Application Layer:
 The application layer contains a variety of protocols that are commonly needed by users.

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As the figure shows, the duty of the application, transport, and network layers is end-to-end. However, the layer). To communicate, a process sends a request to the other process and receives a response. Process-to-
duty of the data-link and physical layers is hop-to-hop, in which a hop is a host. In other words, the domain process communication is the duty of the application layer. The application layer in the Internet includes
of duty of the top three layers is the internet, and the domain of duty of the two lower layers is the link. many predefined protocols, but a user can also create a pair of processes to be run at the two hosts.
Figure shows the second principle discussed previously for protocol layering. We show the identical objects
below each layer related to each device. The Hypertext Transfer Protocol (HTTP) is a vehicle for accessing the World Wide Web (WWW). The
Simple Mail Transfer Protocol (SMTP) is the main protocol used in electronic mail (e-mail) service.
Physical Layer
Physical layer is responsible for carrying individual bits in a frame across the link. Although the physical Encapsulation and Decapsulation
layer is the lowest level in the TCP/IP protocol suite, the communication between two devices at the physical
layer is still a logical communication because there is another, hidden layer, the transmission media, under
the physical layer. Two devices are connected by a transmission medium (cable or air).
Data-link Layer
The data-link layer is responsible for taking the datagram and moving it across the link. The link can be a
wired LAN with a link-layer switch, a wireless LAN, a wired WAN, or a wireless WAN. The data-link layer
takes a datagram and encapsulates it in a packet called a frame. Each link-layer protocol may provide a
different service. Some link-layer protocols provide complete error detection and correction, some provide
only error correction.
Network Layer
The network layer is responsible for creating a connection between the source computer and the destination
computer. The communication at the network layer is host-to-host. However, since there can be several
routers from the source to the destination, the routers in the path are responsible for choosing the best route
for each packet. Network layer is responsible for host-to-host communication and routing the packet through Encapsulation at the Source Host
possible routes. At the source, we have only encapsulation.
The network layer in the Internet includes the main protocol, Internet Protocol (IP), that defines the format of 1. At the application layer, the data to be exchanged is referred to as a message. A message normally does
the packet, called a datagram at the network layer. IP also defines the format and the structure of addresses not contain any header or trailer, but if it does, we refer to the whole as the message. The message is
used in this layer. IP is also responsible for routing a packet from its source to its destination, which is passed to the transport layer.
achieved by each router forwarding the datagram to the next router in its path.
Transport Layer 2. The transport layer takes the message as the payload, the load that the transport layer should take care of.
The logical connection at the transport layer is also end-to-end. The transport layer at the source host gets the It adds the transport layer header to the payload, which contains the identifiers of the source and
message from the application layer, encapsulates it in a transport layer packet (called a segment or a user destination application programs that want to communicate plus some more information that is needed for
datagram in different protocols) and sends it, through the logical (imaginary) connection, to the transport the end-toend delivery of the message, such as information needed for flow, error control, or congestion
layer at the destination host. In other words, the transport layer is responsible for giving services to the control. The result is the transport-layer packet, which is called the segment (in TCP) and the user
application layer: to get a message from an application program running on the source host and deliver it to datagram (in UDP). The transport layer then passes the packet to the network layer.
the corresponding application program on the destination host. 3. The network layer takes the transport-layer packet as data or payload and adds its own header to the
Transmission Control Protocol (TCP), is a connection-oriented protocol that first establishes a logical payload. The header contains the addresses of the source and destination hosts and some more
connection between transport layers at two hosts before transferring data. It creates a logical pipe between information used for error checking of the header, fragmentation information, and so on. The result is the
two TCPs for transferring a stream of bytes. TCP provides flow control (matching the sending data rate of network-layer packet, called a datagram. The network layer then passes the packet to the data-link layer.
the source host with the receiving data rate of the destination host to prevent overwhelming the destination), 4. The data-link layer takes the network-layer packet as data or payload and adds its own header, which
error control (to guarantee that the segments arrive at the destination without error and resending the contains the link-layer addresses of the host or the next hop (the router). The result is the link-layer
corrupted ones), and congestion control to reduce the loss of segments due to congestion in the network. packet, which is called a frame. The frame is passed to the physical layer for transmission.
The other common protocol, User Datagram Protocol (UDP), is a connectionless protocol that transmits user Decapsulation and Encapsulation at the Router
datagrams without first creating a logical connection. In UDP, each user datagram is an independent entity At the router, we have both decapsulation and encapsulation because the router is connected to two or more
without being related to the previous or the next one (the meaning of the term connectionless). UDP is a links.
simple protocol that does not provide flow, error, or congestion control. Its simplicity, which means small 1. After the set of bits are delivered to the data-link layer, this layer decapsulates the datagram from the
overhead, is attractive to an application program that needs to send short messages and cannot afford the frame and passes it to the network layer.
retransmission of the packets involved in TCP, when a packet is corrupted or lost. A new protocol, Stream
Control Transmission Protocol (SCTP) is designed to respond to new applications that are emerging in the 2. The network layer only inspects the source and destination addresses in the datagram header and consults
multimedia. its forwarding table to find the next hop to which the datagram is to be delivered. The contents of the
datagram should not be changed by the network layer in the router unless there is a need to fragment the
Application Layer datagram if it is too big to be passed through the next link. The datagram is then passed to the data-link
As Figure shows, the logical connection between the two application layers is end to end. The two layer of the next link.
application layers exchange messages between each other as though there were a bridge between the two
layers. Communication at the application layer is between two processes (two programs running at this

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3. The data-link layer of the next link encapsulates the datagram in a frame and passes it to the physical layer To be able to multiplex and demultiplex, a protocol needs to have a field in its header to identify to which
for transmission. protocol the encapsulated packets belong. At the transport layer, either UDP or TCP can accept a message
from several application-layer protocols. At the network layer, IP can accept a segment from TCP or a user
Decapsulation at the Destination Host datagram from UDP. IP can also accept a packet from other protocols such as ICMP, IGMP, and so on. At
At the destination host, each layer only decapsulates the packet received, removes the payload, and delivers the data-link layer, a frame may carry the payload coming from IP or other protocols such as ARP.
the payload to the next-higher layer protocol until the message reaches the application layer. It is necessary INTRODUCTION TO PHYSICAL LAYER
to say that decapsulation in the host involves error checking.
Addressing DATA AND SIGNALS:
It is worth mentioning another concept related to protocol layering in the Internet, addressing. As we
discussed before, we have logical communication between pairs of layers in this model. Any communication Figure 1.28 shows a scenario in which a scientist working in a research company, Sky Research, needs to
that involves two parties needs two addresses: source address and destination address. Although it looks as if order a book related to her research from an online bookseller, Scientific Books.
we need five pairs of addresses, one pair per layer, we normally have only four because the physical layer
does not need addresses; the unit of data exchange at the physical layer is a bit, which definitely cannot have We can think of five different levels of communication between Alice, the computer on which our scientist is
an address. Figure shows the addressing at each layer. As the figure shows, there is a relationship between working, and Bob, the computer that provides online service. Communication at application, transport,
the layer, the address used in that layer, and the packet name at that layer. At the application layer, we network, or data-link is logical; communication at the physical layer is physical.
normally use names to define the site that provides services, such as someorg.com, or the e-mail
For simplicity, we have shown only host-to-router, router-to-router, and router-to-host, but the switches are
also involved in the physical communication.

Although Alice and Bob need to exchange data, communication at the physical layer means exchanging
signals. Data need to be transmitted and received, but the media have to change data to signals. Both data
and the signals that represent them can be either analog or digital in form.

Analog and Digital Data:

Data can be analog or digital. The term analog data refers to information that is continuous; digital data
refers to information that has discrete states.

address, such as [email protected]. At the transport layer, addresses are called port numbers, and For example, an analog clock that has hour, minute, and second hands gives information in a continuous
these define the application-layer programs at the source and destination. Port numbers are local addresses form; the movements of the hands are continuous. On the other hand, a digital clock that reports the hours
that distinguish between several programs running at the same time. At the network-layer, the addresses are and the minutes will change suddenly from 8:05 to 8:06.
global, with the whole Internet as the scope. A network-layer address uniquely defines the connection of a
device to the Internet. The link-layer addresses, sometimes called MAC addresses, are locally defined Analog data, such as the sounds made by a human voice, take on continuous values. When someone speaks,
addresses, each of which defines a specific host or router in a network (LAN or WAN). We will come back
an analog wave is created in the air. This can be captured by a microphone and converted to an analog signal
to these addresses in future chapters.
or sampled and converted to a digital signal.
Multiplexing and Demultiplexing
Since the TCP/IP protocol suite uses several protocols at some layers, we can say that we have multiplexing Digital data take on discrete values. For example, data are stored in computer memory in the form of 0s and
at the source and demultiplexing at the destination. Multiplexing in this case means that a protocol at a layer 1s. They can be converted to a digital signal or modulated into an analog signal for transmission across a
can encapsulate a packet from several next-higher layer protocols (one at a time); demultiplexing means that medium.
a protocol can decapsulate and deliver a packet to several next-higher layer protocols (one at a time). Figure
shows the concept of multiplexing and demultiplexing at the three upper layers.

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FIGURE 1.29: COMPARISON OF ANALOG AND DIGITAL SIGNALS

Periodic and Nonperiodic: Both analog and digital signals can take one of two forms: periodic or
nonperiodic (Sometimes referred to as aperiodic; the prefix a in Greek means “non”).

A periodic signal completes a pattern within a measurable time frame, called a period, and repeats that
pattern over subsequent identical periods. The completion of one full pattern is called a cycle. A
nonperiodic signal changes without exhibiting a pattern or cycle that repeats over time. Both analog and
digital signals can be periodic or nonperiodic.

In data communications, we commonly use periodic analog signals and nonperiodic digital signals.

TRANSMISSION IMPAIRMENT:

Signals travel through transmission media, which are not perfect. The imperfection causes signal
impairment. This means that the signal at the beginning of the medium is not the same as the signal at the
end of the medium. What is sent is not what is received. Three causes of impairment are attenuation,
distortion, and noise (see Figure 1.32).

FIGURE 1.32: CAUSES OF IMPAIRMENT

Attenuation:

Attenuation means a loss of energy. When a signal, simple or composite, travels through a medium, it loses
FIGURE 1.28: COMMUNICATION AT THE PHYSICAL LAYER some of its energy in overcoming the resistance of the medium. That is why a wire carrying electric signals
ANALOG AND DIGITAL SIGNALS: Like the data they represent, signals can be either analog or digital. gets warm, if not hot, after a while. Some of the electrical energy in the signal is converted to heat. To
compensate for this loss, amplifiers are used to amplify (meaning enlarge on/go into
An analog signal has infinitely many levels of intensity (meaning strength/power) over a period of time. As detail/develop/expand/clarify/add details to) the signal.
the wave moves from value A to value B, it passes through and includes an infinite number of values along
its path. Decibel:

A digital signal, on the other hand, can have only a limited number of defined values. Although each value To show that a signal has lost or gained strength, engineers use the unit of the decibel. The decibel (dB)
can be any number, it is often as simple as 1 and 0. measures the relative strengths of two signals or one signal at two different points. Note that the decibel is
negative if a signal is attenuated and positive if a signal is amplified.
The simplest way to show signals is by plotting them on a pair of perpendicular axes. The vertical axis
represents the value or strength of a signal. The horizontal axis represents time.

Figure 1.29 illustrates an analog signal and a digital signal. The curve representing the analog signal passes Distortion:
through an infinite number of points. The vertical lines of the digital signal, however, demonstrate the
sudden jump that the signal makes from value to value.

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Distortion means that the signal changes its form or shape. Distortion can occur in a composite signal made In this formula, bandwidth is the bandwidth of the channel, SNR is the signal-to-noise ratio, and capacity is
of different frequencies. Each signal component has its own propagation speed through a medium and, the capacity of the channel in bits per second.
therefore, its own delay in arriving at the final destination. Differences in delay may create a difference in
phase if the delay is not exactly the same as the period duration. The Shannon capacity gives us the upper limit; the Nyquist formula tells us how many signal levels we
need.
Noise:
PERFORMANCE:
Noise is another cause of impairment. Several types of noise, such as thermal noise, induced noise,
crosstalk, and impulse noise, may corrupt the signal. One important issue in networking is the performance of the network—how good is it? There are certain
characteristics that measure the network performance which are given as follows:
Thermal noise is the random motion of electrons in a wire, which creates an extra signal not originally sent
by the transmitter. BANDWIDTH:

Induced noise comes from sources such as motors and appliances. These devices act as a sending antenna, One characteristic that measures network performance is bandwidth. However, the term can be used in two
and the transmission medium acts as the receiving antenna. different contexts with two different measuring values: bandwidth in hertz and bandwidth in bits per second.

Crosstalk is the effect of one wire on the other. One wire acts as a sending antenna and the other as the Bandwidth in Hertz: Bandwidth in hertz is the range of frequencies contained in a composite signal or the
receiving antenna. range of frequencies a channel can pass. For example, we can say the bandwidth of a subscriber telephone
line is 4 kHz.
Impulse noise is a spike (a signal with high energy in a very short time) that comes from power lines,
lightning, and so on. Bandwidth in Bits per Seconds: The term bandwidth can also refer to the number of bits per second that a
channel, a link, or even a network can transmit. For example, one can say the bandwidth of a Fast Ethernet
DATA RATE LIMITS: network (or the links in this network) is a maximum of 100 Mbps. This means that this network can send 100
Mbps.
A very important consideration in data communications is how fast we can send data, in bits per second, over
a channel. Data rate depends on three factors: Relationship: There is an explicit relationship between the bandwidth in hertz and bandwidth in bits per
second. Basically, an increase in bandwidth in hertz means an increase in bandwidth in bits per second.
1. The bandwidth available
THROUGHPUT:
2. The level of the signals we use
The throughput is a measure of how fast we can actually send data through a network. Although, at first
3. The quality of the channel (the level of noise) glance, bandwidth in bits per second and throughput seem the same, they are different. A link may have a
Two theoretical formulas were developed to calculate the data rate: one by Nyquist for a noiseless channel, bandwidth of B bps, but we can only send T bps through this link with T always less than B.
another by Shannon for a noisy channel. For example, we may have a link with a bandwidth of 1 Mbps, but the devices connected to the end of the
link may handle only 200 kbps. This means that we cannot send more than 200 kbps through this link.
Noiseless Channel: Nyquist Bit Rate:
Imagine a highway designed to transmit 1000 cars per minute from one point to another. However, if there is
For a noiseless channel, the Nyquist bit rate formula defines the theoretical maximum bit rate BitRate =
2 x bandwidth x log2L congestion on the road, this figure may be reduced to 100 cars per minute. The bandwidth is 1000 cars per
minute; the throughput is 100 cars per minute.
In this formula, bandwidth is the bandwidth of the channel, L is the number of signal levels used to represent
data, and BitRate is the bit rate in bits per second. According to the formula, we might think that, given a LATENCY (DELAY):
specific bandwidth, we can have any bit rate we want by increasing the number of signal levels. The latency or delay defines how long it takes for an entire message to completely arrive at the destination
Although the idea is theoretically correct, practically there is a limit. When we increase the number of signal from the time the first bit is sent out from the source. We can say that latency is made of four components:
levels, we impose a burden on the receiver. propagation time, transmission time, queuing time and processing delay.

Latency = propagation time + transmission time + queuing time + processing delay

Increasing the levels of a signal may reduce the reliability of the system.

Noisy Channel: Shannon Capacity: Propagation Time: Propagation time measures the time required for a bit to travel from the source to the
destination. The propagation time is calculated by dividing the distance by the propagation speed.
In reality, we cannot have a noiseless channel; the channel is always noisy. In 1944, Claude Shannon
introduced a formula, called the Shannon capacity, to determine the theoretical highest data rate for a noisy Propagation time = Distance / (Propagation Speed)
channel: Capacity = bandwidth x log2 (1 + SNR)

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Transmission Time: In data communications we don’t send just 1 bit, we send a message. The first bit may In data communications the definition of the information and the transmission medium is more
take a time equal to the propagation time to reach its destination; the last bit also may take the same amount specific. The transmission medium is usually free space, metallic cable, or fiber-optic cable.
of time. However, there is a time between the first bit leaving the sender and the last bit arriving at the
receiver. The information is usually a signal that is the result of a conversion of data from another form.

The first bit leaves earlier and arrives earlier; the last bit leaves later and arrives later. The transmission time The use of long-distance communication using electric signals started with the invention of the
of a message depends on the size of the message and the bandwidth of the channel. telegraph by Morse in the 19th century.

Transmission time = (Message size) / Bandwidth Communication by telegraph was slow and dependent on a metallic medium. Extending the
range of the human voice became possible when the telephone was invented in 1869.
Queuing Time: The third component in latency is the queuing time, the time needed for each intermediate
or end device to hold the message before it can be processed. The queuing time is not a fixed factor; it Telephone communication at that time also needed a metallic medium to carry the electric
changes with the load imposed on the network. signals that were the result of a conversion from the human voice.

When there is heavy traffic on the network, the queuing time increases. An intermediate device, such as a The communication was, however, unreliable due to the poor quality of the wires. The lines
router, queues they arrived messages and processes them one by one. If there are many messages, each were often noisy and the technology was unsophisticated.
message will have to wait. Wireless communication started in 1895 when Hertz was able to send high frequency signals.
Bandwidth-Delay Product Later, Marconi devised a method to send telegraph-type messages over the Atlantic Ocean.

Bandwidth and delay are two performance metrics of a link. The bandwidth-delay product defines the We have come a long way. Better metallic media have been invented (twisted-pair and coaxial
number of bits that can fill the link. cables, for example).

JITTER: The use of optical fibers has increased the data rate incredibly. Free space (air, vacuum, and water) is used
more efficiently, in part due to the technologies (such as modulation and multiplexing).
Another performance issue that is related to delay is jitter. We can roughly say that jitter is a problem if
different packets of data encounter different delays and the application using the data at the receiver site is Electromagnetic energy, a combination of electric and magnetic fields vibrating in relation to each other,
time-sensitive (audio and video data, for example). If the delay for the first packet is 20 ms, for the second is includes power, radio waves, infrared light, visible light, and ultraviolet light, and X, gamma, and cosmic
45 ms, and for the third is 40 ms, then the real-time application that uses the packets endures jitter. rays. Each of these constitutes a portion of the electromagnetic spectrum.

TRANSMISSION MEDIA In telecommunications, transmission media can be divided into two broad categories: guided and unguided.

INTRODUCTION: Guided media include twisted-pair cable, coaxial cable, and fiber-optic cable.

Transmission media are actually located below the physical layer and are directly controlled by the physical Unguided medium is free space. Figure 1.34 shows this taxonomy.
layer. We could say that transmission media belong to layer zero. Figure 1.33 shows the position of
transmission media in relation to the physical layer.

FIGURE 1.33: TRANSMISSION MEDIUM AND PHYSICAL LAYER

FIGURE 1.34: CLASSES OF TRANSMISSION MEDIA

A transmission medium can be broadly defined as anything that can carry information from a
source to a destination. For example, the transmission medium for two people having a dinner
conversation is the air.

The air can also be used to convey the message in a smoke signal or semaphore.
GUIDED MEDIA:
For a written message, the transmission medium might be a mail carrier, a truck, or an airplane.

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Guided media, which are those that provide a conduit (meaning medium) from one device to another, Categories: The Electronic Industries Association (EIA) has developed standards to classify unshielded
include twisted-pair cable, coaxial cable, and fiber-optic cable. twisted-pair cable into seven categories. Categories are determined by cable quality, with 1 as the lowest and
7 as the highest. Each EIA category is suitable for specific uses. Table 1.1 shows these categories.
A signal traveling along any of these media is directed and contained by the physical limits of the medium.
Twisted-pair and coaxial cable use metallic (copper) conductors that accept and transport signals in the form DATA
of electric current. Optical fiber is a cable that accepts and transports signals in the form of light. CATEGORY SPECIFICATION RATE USE
(Mbps)
TWISTED-PAIR CABLE:
1 Unshielded twisted-pair used in telephone <0.1 Telephone
A twisted pair consists of two conductors (normally copper), each with its own plastic insulation, twisted
together, as shown in Figure 1.35. 2 Unshielded twisted-pair originally used in T lines 2 T-1 lines

3 Improved CAT 2 used in LANs 10 LANs

4 Improved CAT 3 used in Token Ring networks 20 LANs

Cable wire is normally 24 AWG with a jacket

5 100 LANs
FIGURE 1.35: TWISTED-PAIR CABLE
and outside sheath (meaning case/cover)
One of the wires is used to carry signals to the receiver, and the other is used only as a ground reference. The An extension to category 5 that includes extra features to minimize the
receiver uses the difference between the two. crosstalk and
5E 125 LANs
In addition to the signal sent by the sender on one of the wires, interference (noise) and crosstalk may affect electromagnetic interference
both wires and create unwanted signals.
A new category with matched components coming from the same
6 200 LANs
If the two wires are parallel, the effect of these unwanted signals is not the same in both wires because they manufacturer. The cable must be tested at a 200-Mbps data rate.
are at different locations relative to the noise or crosstalk sources (e.g., one is closer and the other is farther).
Sometimes called SSTP (shielded screen twisted-pair). Each pair is
This results in a difference at the receiver.
individually wrapped in a helical metallic foil followed by a metallic foil
7 600 LANs
By twisting the pairs, a balance is maintained. For example, suppose in one twist, one wire is closer to the shield in addition to the outside sheath. The shield decreases the effect of
noise source and the other is farther; in the next twist, the reverse is true. Twisting makes it probable that crosstalk and increases the data rate.
both wires are equally affected by external influences (noise or crosstalk). This means that the receiver,
TABLE 1.1: CATEGORIES OF UNSHIELDED TWISTED-PAIR CABLES
which calculates the difference between the two, receives no unwanted signals.
Connectors: The most common UTP connector is RJ45 (RJ stands for registered jack), as shown in Figure
Unshielded Versus Shielded Twisted-Pair Cable The most common twisted-pair cable used in
1.37. The RJ45 is a keyed connector, meaning the connector can be inserted in only one way.
communications is referred to as unshielded twisted-pair (UTP). IBM has also produced a version of
twisted-pair cable for its use, called shielded twisted-pair (STP); STP cable has a metal foil or braided mesh
covering that encases each pair of insulated conductors.

Although metal casing improves the quality of cable by preventing the penetration of noise or crosstalk, it is
bulkier and more expensive. Figure 1.36 shows the difference between UTP and STP.

FIGURE 1.37: UTP CONNECTOR

Performance: One way to measure the performance of twisted-pair cable is to compare attenuation versus
frequency and distance. A twisted-pair cable can pass a wide range of frequencies.

Applications: Twisted-pair cables are used in telephone lines to provide voice and data channels. The local
FIGURE 1.36: UTP AND STP CABLES
loop—the line that connects subscribers to the central telephone office— commonly consists of unshielded
twisted-pair cables.

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The DSL lines that are used by the telephone companies to provide high-data-rate connections also use the The BNC connector is used to connect the end of the cable to a device, such as a TV set. The BNC T
high-bandwidth capability of unshielded twisted-pair cables. connector is used in Ethernet networks to branch out to a connection to a computer or other device. The BNC
terminator is used at the end of the cable to prevent the reflection of the signal.
COAXIAL CABLE:
FIBER-OPTIC CABLE:
Coaxial cable (or coax) carries signals of higher frequency ranges than those in twisted pair cable, in part
because the two media are constructed quite differently. Instead of having two wires, coax has a central core A fiber-optic cable is made of glass or plastic and transmits signals in the form of light. To understand
conductor of solid or stranded wire (usually copper) enclosed in an insulating sheath, which is, in turn, optical fiber, we first need to explore several aspects of the nature of light. Light travels in a straight line as
encased in an outer conductor of metal foil, braid, or a combination of the two. The outer metallic wrapping long as it is moving through a single uniform substance. If a ray of light is traveling through one substance
serves both as a shield against noise and as the second conductor, which completes the circuit; this outer suddenly enters another substance (of a different density), the ray changes direction.
conductor is also enclosed in an insulating sheath, and the whole cable is protected by a plastic cover (see
Figure 1.38). Figure 1.40 shows how a ray of light changes direction when going from a more dense to a less dense
substance.

FIGURE 1.38: COAXIAL CABLE

Coaxial Cable Standards: Coaxial cables are categorized by their Radio Government (RG) ratings. Each
FIGURE 1.40: BENDING OF LIGHT RAY
RG number denotes a unique set of physical specifications, including the wire gauge of the inner conductor,
the thickness and type of the inner insulator, the construction of the shield, and the size and type of the outer As the figure shows, if the angle of incidence I (the angle the ray makes with the line perpendicular to the
casing. Each cable defined by an RG rating is adapted for a specialized function, as shown in Table 1.2. interface between the two substances) is less than the critical angle, the ray refracts and moves closer to the
surface. If the angle of incidence is equal to the critical angle, the light bends along the interface. If the angle
Category Impedance Use
is greater than the critical angle, the ray reflects (makes a turn) and travels again in the denser substance.
RG-59 75 Ω Cable TV
Optical fibers use reflection to guide light through a channel. A glass or plastic core is surrounded by a
RG-58 50 Ω Thin Ethernet cladding of less dense glass or plastic. The difference in density of the two materials must be such that a
beam of light moving through the core is reflected off the cladding instead of being refracted into it. See
RG-11 50 Ω Thick Ethernet
Figure 1.41.
TABLE 1.2: CATEGORIES OF COAXIAL CABLES

Coaxial Cable Connectors:

To connect coaxial cable to devices, we need coaxial connectors. The most common type of connector used
today is the Bayonet Neill-Concelman (BNC) connector. Figure 1.39 shows three popular types of these
connectors: the BNC connector, the BNC T connector, and the BNC terminator. FIGURE 1.41: OPTICAL FIBER

Propagation Modes: Current technology supports two modes (multimode and single mode) for propagating
light along optical channels, each requiring fiber with different physical characteristics. Multimode can be
implemented in two forms: step-index or graded-index (see Figure 1.42).

FIGURE 1.39: BNC CONNECTORS

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FIGURE 1.42: PROPAGATION MODES Optical fibers are defined by the ratio of the diameter of their core to the diameter of their cladding, both
expressed in micrometers. The common sizes are shown in Table 1.3. Note that the last size listed is for
Multimode: Multimode is so named because multiple beams from a light source move through the core in single-mode only.
different paths. How these beams move within the cable depends on the structure of the core, as shown in
Figure 1.43. Type Core (μm) Cladding (μm) Mode

50/125 50.0 125 Multi mode, Graded index

62.5/125 62.5 125 Multi mode, Graded index

100/125 100.0 125 Multi mode, Graded index

7/125 7.0 125 Single mode

TABLE 1.3: FIBER TYPES

Cable Composition

Figure 1.44 shows the composition of a typical fiber-optic cable. The outer jacket is made of either PVC or
Teflon. Inside the jacket are Kevlar strands to strengthen the cable.

FIGURE 1.43: MODES

In multimode step-index fiber, the density of the core remains constant from the center to the edges. A
beam of light moves through this constant density in a straight line until it reaches the interface of the core
and the cladding. At the interface, there is an abrupt change due to a lower density; this alters the angle of the FIGURE 1.44: FIBER CONSTRUCTION
beam’s motion. The term step-index refers to the suddenness of this change, which contributes to the
distortion of the signal as it passes through the fiber. Kevlar is a strong material used in the fabrication of bulletproof vests. Below the Kevlar is another plastic
coating to cushion the fiber. The fiber is at the center of the cable, and it consists of cladding and core.
A second type of fiber, called multimode graded-index fiber, decreases this distortion of the signal through
the cable. The word index here refers to the index of refraction. As we saw above, the index of refraction is Fiber-Optic Cable Connectors:
related to density. A graded-index fiber, therefore, is one with varying densities. Density is highest at the
There are three types of connectors for fiber-optic cables, as shown in Figure 1.45. The subscriber channel
center of the core and decreases gradually to its lowest at the edge. Figure 1.43 shows the impact of this (SC) connector is used for cable TV. It uses a push/pull locking system. The straight-tip (ST) connector is
variable density on the propagation of light beams.
used for connecting cable to networking devices. It uses a bayonet locking system and is more reliable than
Single-Mode: Single-mode uses step-index fiber and a highly focused source of light that limits beams to a SC. MT-RJ is a connector that is the same size as RJ45.
small range of angles, all close to the horizontal.

The single-mode fibers itself is manufactured with a much smaller diameter than that of multimode fiber
and with substantially lower density (index of refraction). The decrease in density results in a critical angle
that is close enough to 90° to make the propagation of beams almost horizontal. In this case, propagation of
different beams is almost identical, and delays are negligible. All the beams arrive at the destination
“together” and can be recombined with little distortion to the signal (see Figure 1.43).

Fiber Sizes:

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ADVANTAGES: Fiber-optic cable has several advantages over metallic cable (twisted-pair or coaxial).

 Higher bandwidth. Fiber-optic cable can support dramatically higher bandwidths (and hence
data rates) than either twisted-pair or coaxial cable. Currently, data rates and bandwidth
utilization over fiber-optic cable are limited not by the medium but by the signal generation and
reception technology available.

 Less signal attenuation. Fiber-optic transmission distance is significantly greater than that of
other guided media. A signal can run for 50 km without requiring regeneration. We need
repeaters every 5 km for coaxial or twisted-pair cable.

 Immunity to electromagnetic interference. Electromagnetic noise cannot affect fiber-optic

FIGURE 1.45: FIBER-OPTIC CABLE CONNECTORS cables.
Performance  Resistance to corrosive materials. Glass is more resistant to corrosive materials than copper.
The plot of attenuation versus wavelength in Figure 1.46 shows a very interesting phenomenon in fiber-optic  Light weight. Fiber-optic cables are much lighter than copper cables.
cable. Attenuation is flatter than in the case of twisted-pair cable and coaxial cable. The performance is such
that we need fewer (actually one tenth as many) repeaters when we use fiber-optic cable.  Greater immunity to tapping. Fiber-optic cables are more immune to tapping than copper
cables. Copper cables create antenna effects that can easily be tapped.

Disadvantages: There are some disadvantages in the use of optical fiber.Installation and maintenance.
Fiber-optic cable is a relatively new technology. Its installation and maintenance require expertise that is not
yet available everywhere.

 Unidirectional light propagation. Propagation of light is unidirectional. If we need

bidirectional communication, two fibers are needed.

 Cost. The cable and the interfaces are relatively more expensive than those of other guided
media. If the demand for bandwidth is not high, often the use of optical fiber cannot be justified.

UNGUIDED MEDIA: WIRELESS:

Unguided medium transport electromagnetic waves without using a physical conductor. This type of
communication is often referred to as wireless communication. Signals are normally broadcast through free
space and thus are available to anyone who has a device capable of receiving them.

FIGURE 1.46: OPTICAL FIBER PERFORMANCE Unguided signals can travel from the source to the destination in several ways: ground propagation, sky
propagation, and line-of-sight propagation, as shown in Figure 1.47.
Applications:

Fiber-optic cable is often found in backbone networks because its wide bandwidth is cost-effective. Today,
with wavelength-division multiplexing (WDM), we can transfer data at a rate of 1600 Gbps.

Some cable TV companies use a combination of optical fiber and coaxial cable, thus creating a hybrid
network. Optical fiber provides the backbone structure while coaxial cable provides the connection to the
user premises. This is a cost-effective configuration since the narrow bandwidth requirement at the user end
does not justify the use of optical fiber.

Local-area networks such as 100Base-FX network (Fast Ethernet) and 1000Base-X also use fiber-optic
cable.
FIGURE 1.47: PROPAGATION METHODS
ADVANTAGES AND DISADVANTAGES OF OPTICAL FIBER:

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In ground propagation, radio waves travel through the lowest portion of the atmosphere, hugging the earth. Although there is no clear-cut demarcation between radio waves and microwaves, electromagnetic waves
These low-frequency signals emanate in all directions from the transmitting antenna and follow the curvature ranging in frequencies between 3 kHz and 1 GHz are normally called radio waves; waves ranging in
of the planet. Distance depends on the amount of power in the signal: The greater the power, the greater the frequencies between 1 and 300 GHz are called microwaves.
distance.
However, the behavior of the waves, rather than the frequencies, is a better criterion for classification. Radio
In sky propagation, higher-frequency radio waves radiate upward into the ionosphere (the layer of waves, for the most part, are omnidirectional. When an antenna transmits radio waves, they are propagated in
atmosphere where particles exist as ions) where they are reflected back to earth. This type of transmission all directions.
allows for greater distances with lower output power.
This means that the sending and receiving antennas do not have to be aligned. A sending antenna sends
In line-of-sight propagation, very high-frequency signals are transmitted in straight lines directly from waves that can be received by any receiving antenna. The omnidirectional property has a disadvantage, too.
antenna to antenna. Antennas must be directional, facing each other and either tall enough or close enough The radio waves transmitted by one antenna are susceptible to interference by another antenna that may send
together not to be affected by the curvature of the earth. Line-of-sight propagation is tricky because radio signals using the same frequency or band.
transmissions cannot be completely focused.
Radio waves, particularly those waves that propagate in the sky mode, can travel long distances. This makes
The section of the electromagnetic spectrum defined as radio waves and microwaves is divided into eight radio waves a good candidate for long-distance broadcasting such as AM radio.
ranges, called bands, each regulated by government authorities. These bands are rated from very low
frequency (VLF) to extremely high frequency (EHF). Table 1.4 lists these bands, their ranges, propagation Radio waves, particularly those of low and medium frequencies, can penetrate walls. This characteristic can
methods, and some applications. be both an advantage and a disadvantage.

BAND RANGE PROPAGATION APPLICATION It is an advantage because, for example, an AM radio can receive signals inside a building.

Very low frequency (VLF) 3-30kHz Ground Long-range radio It is a disadvantage because we cannot isolate a communication to just inside or outside a
building.
navigation
The radio wave band is relatively narrow, just under 1 GHz, compared to the microwave band. When this
Low frequency (LF) 30–300 kHz Ground Radio beacons and band is divided into subbands, the subbands are also narrow, leading to a low data rate for digital
navigational locators
communications.

Middle frequency (MF) 300 kHz–3 MHz Sky AM radio Omnidirectional Antenna: Radio waves use omnidirectional antennas that send out signals in all
directions. Based on the wavelength, strength, and the purpose of transmission, we can have several types of
High frequency (HF) 3–30 MHz Sky Citizens band (CB), antennas. Figure 1.48 shows an omnidirectional antenna.
ship/aircraft

Very high frequency (VHF) 30–300 MHz Sky and line-of- VHF TV, FM radio
sight

Ultrahigh frequency (UHF) 300 MHz–3 Line-of-sight UHF TV, cellular phones, paging,
GHz satellite

Superhigh frequency (SF) 3–30 GHz Line-of-sight Satellite

FIGURE 1.48: OMNIDIRECTIONAL ANTENNA

Applications: The omnidirectional characteristics of radio waves make them useful for multicasting, in
Extremely high frequency 30–300 GHz Line-of-sight Radar, satellite
which there is one sender but many receivers. AM and FM radio, television, maritime radio, cordless phones,
(EHF)
and paging are examples of multicasting.
TABLE 1.4: BANDS (CONTINUED)
Microwaves: Electromagnetic waves having frequencies between 1 and 300 GHz are called microwaves.
We can divide wireless transmission into three broad groups: radio waves, microwaves, and infrared waves. Microwaves are unidirectional. When an antenna transmits microwaves, they can be narrowly focused. This
means that the sending and receiving antennas need to be aligned. The unidirectional property has an obvious
RADIO WAVES: advantage. A pair of antennas can be aligned without interfering with another pair of aligned antennas.

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Unidirectional Antenna: Microwaves need unidirectional antennas that send out signals in one direction. bus, are ruled out because the distances between devices and the total number of devices increase beyond the
Two types of antennas are used for microwave communications: the parabolic dish and the horn (see Figure capacities of the media and equipment.
1.49).
A better solution is switching. A switched network consists of a series of interlinked nodes, called switches.
Switches are devices capable of creating temporary connections between two or more devices linked to the
switch. In a switched network, some of these nodes are connected to the end systems (computers or
telephones, for example). Others are used only for routing. Figure 1.50 shows a switched network.

FIGURE 1.49: UNIDIRECTIONAL ANTENNAS

Applications:

Microwaves, due to their unidirectional properties, are very useful when unicast (one-to- one)
communication is needed between the sender and the receiver. They are used in cellular phones, satellite
networks and wireless LANs.
FIGURE 1.50: SWITCHED NETWORK
Infrared:
The end systems (communicating devices) are labeled A, B, C, D, and so on, and the switches are labeled I,
Infrared waves, with frequencies from 300 GHz to 400 THz (wavelengths from 1 mm to 770 nm), can be II, III, IV, and V. Each switch is connected to multiple links.
used for short-range communication. Infrared waves, having high frequencies, cannot penetrate walls. This
advantageous characteristic prevents interference between one system and another; a short-range THREE METHODS OF SWITCHING:
communication system in one room cannot be affected by another system in the next room. Traditionally, three methods of switching have been discussed: circuit switching, packet switching, and
When we use our infrared remote control, we do not interfere with the use of the remote by our neighbors. message switching. The first two are commonly used today. The third has been phased out in general
However, this same characteristic makes infrared signals useless for long-range communication. In addition, communications but still has networking applications. Packet switching can further be divided into two
we cannot use infrared waves outside a building because the sun’s rays contain infrared waves that can subcategories—virtual circuit approach and datagram approach—as shown in Figure 1.51.
interfere with the communication. Note: we discuss only circuit switching and packet switching; message switching is more conceptual than
Applications: The infrared band, almost 400 THz, has an excellent potential for data transmission. Such a practical.
wide bandwidth can be used to transmit digital data with a very high data rate.

SWITCHING:

We have switching at the physical layer, at the data-link layer, at the network layer, and even logically at the
application layer (message switching).

INTRODUCTION:

A network is a set of connected devices. Whenever we have multiple devices, we have the problem of how to
connect them to make one-to-one communication possible. One solution is to make a point-to-point FIGURE 1.51: TAXONOMY OF SWITCHED NETWORKS
connection between each pair of devices (a mesh topology) or between a central device and every other
device (a star topology). These methods, however, are impractical and wasteful when applied to very large Switching and TCP/IP Layers:
networks.
Switching can happen at several layers of the TCP/IP protocol suite.
The number and length of the links require too much infrastructure to be cost-efficient, and the majority of
those links would be idle most of the time. Other topologies employing multipoint connections, such as a

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Switching at Physical Layer: At the physical layer, we can have only circuit switching. There are no packets Circuit switching takes place at the physical layer.
exchanged at the physical layer. The switches at the physical layer allow signals to travel in one path or
another. Before starting communication, the stations must make a reservation for the resources to be used
during the communication. These resources, such as channels (bandwidth in FDM and time slots
Switching at Data-Link Layer: At the data-link layer, we can have packet switching. However, the term in TDM), switch buffers, switch processing time, and switch input/output ports, must remain
packet in this case means frames or cells. Packet switching at the data-link layer is normally done using a dedicated during the entire duration of data transfer until the teardown phase.
virtual-circuit approach.
Data transferred between the two stations are not packetized (physical layer transfer of the
Switching at Network Layer: At the network layer, we can have packet switching. In this case, either a signal). The data are a continuous flow sent by the source station and received by the destination
virtual-circuit approach or a datagram approach can be used. Currently the Internet uses a datagram station, although there may be periods of silence.
approach, but the tendency is to move to a virtual-circuit approach.
There is no addressing involved during data transfer. The switches route the data based on their
Switching at Application Layer: At the application layer, we can have only message switching. The occupied band (FDM) or time slot (TDM). Of course, there is end-to-end addressing used during
communication at the application layer occurs by exchanging messages. the setup phase.

Conceptually, we can say that communication using e-mail is a kind of message-switched communication, THREE PHASES:
but we do not see any network that actually can be called a message-switched network.
The actual communication in a circuit-switched network requires three phases: connection setup, data
CIRCUIT-SWITCHED NETWORKS: transfer, and connection teardown.

A circuit-switched network consists of a set of switches connected by physical links. A connection between SETUP PHASE: Before the two parties (or multiple parties in a conference call) can communicate, a
two stations is a dedicated path made of one or more links. However, each connection uses only one dedicated circuit (combination of channels in links) needs to be established. The end systems are normally
dedicated channel on each link. Each link is normally divided into n channels by using FDM or TDM. connected through dedicated lines to the switches, so connection setup means creating dedicated channels
between the switches.
Figure 1.52 shows a trivial circuit-switched network with four switches and four links. Each link is divided
into n (n is 3 in the figure) channels by using FDM or TDM. DATA TRANSFER PHASE: After the establishment of the dedicated circuit (channels), the two parties
can transfer data.

TEARDOWN PHASE: When one of the parties needs to disconnect, a signal is sent to each switch to
release the resources.

Efficiency:

It can be argued that circuit-switched networks are not as efficient as the other two types of networks
because resources are allocated during the entire duration of the connection. These resources are unavailable
to other connections. In a telephone network, people normally terminate the communication when they have
finished their conversation.

However, in computer networks, a computer can be connected to another computer even if there is no
activity for a long time. In this case, allowing resources to be dedicated means that other connections are
FIGURE 1.52: A TRIVIAL CIRCUIT-SWITCHED NETWORK deprived.

We have explicitly shown the multiplexing symbols to emphasize the division of the link into channels even Delay:
though multiplexing can be implicitly included in the switch fabric.
Although a circuit-switched network normally has low efficiency, the delay in this type of network is
The end systems, such as computers or telephones, are directly connected to a switch. We have shown only minimal. During data transfer the data are not delayed at each switch; the resources are allocated for the
two end systems for simplicity. When end system A needs to communicate with end system M, system A duration of the connection. Figure 1.53 shows the idea of delay in a circuit-switched network when only two
needs to request a connection to M that must be accepted by all switches as well as by M itself. This is called switches are involved.
the setup phase; a circuit (channel) is reserved on each link, and the combination of circuits or channels
defines the dedicated path. After the dedicated path made of connected circuits (channels) is established, the
data-transfer phase can take place. After all data have been transferred, the circuits are torn down.

We need to emphasize several points here:

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Figure 1.54 shows how the datagram approach is used to deliver four packets from station A to station X.
The switches in a datagram network are traditionally referred to as routers. That is why we use a different
symbol for the switches in the figure.

FIGURE 1.53: DELAY IN A CIRCUIT-SWITCHED NETWORK

As Figure 1.53 shows, there is no waiting time at each switch. The total delay is due to the time needed to
create the connection, transfer data, and disconnect the circuit. The delay caused by the setup is the sum of
four parts: the propagation time of the source computer request (slope of the first gray box), the request FIGURE 1.54: A DATAGRAM NETWORK WITH FOUR SWITCHES (ROUTERS)
signal transfer time (height of the first gray box), the propagation time of the acknowledgment from the
destination computer (slope of the second gray box), and the signal transfer time of the acknowledgment In this example, all four packets (or datagrams) belong to the same message, but may travel different paths to
(height of the second gray box). reach their destination. This is so because the links may be involved in carrying packets from other sources
and do not have the necessary bandwidth available to carry all the packets from A to X.
The delay due to data transfer is the sum of two parts: the propagation time (slope of the colored box) and
data transfer time (height of the colored box), which can be very long. The third box shows the time needed This approach can cause the datagrams of a transmission to arrive at their destination out of order with
to tear down the circuit. We have shown the case in which the receiver requests disconnection, which creates different delays between the packets.
the maximum delay.
Packets may also be lost or dropped because of a lack of resources. In most protocols, it is the responsibility
PACKET SWITCHING: of an upper-layer protocol to reorder the datagrams or ask for lost datagrams before passing them on to the
application.
In data communications, we need to send messages from one end system to another. If the message is going
to pass through a packet-switched network, it needs to be divided into packets of fixed or variable size. The The datagram networks are sometimes referred to as connectionless networks. The term connectionless here
size of the packet is determined by the network and the governing protocol. means that the switch (packet switch) does not keep information about the connection state. There are no
setup or teardown phases. Each packet is treated the same by a switch regardless of its source or destination.
In packet switching, there is no resource allocation for a packet. This means that there is no reserved
bandwidth on the links, and there is no scheduled processing time for each packet. Resources are allocated ROUTING TABLE:
on demand. The allocation is done on a first come, first-served basis. If there are no setup or teardown phases, how are the packets routed to their destinations in a datagram
When a switch receives a packet, no matter what the source or destination is, the packet must wait if there network? In this type of network, each switch (or packet switch) has a routing table which is based on the
are other packets being processed. As with other systems in our daily life, this lack of reservation may create destination address. The routing tables are dynamic and are updated periodically. The destination addresses
delay. For example, if we do not have a reservation at a restaurant, we might have to wait. and the corresponding forwarding output ports are recorded in the tables. Figure 1.55 shows the routing table
for a switch.
In a packet-switched network, there is no resource reservation; resources are allocated on demand.

We can have two types of packet-switched networks: datagram networks and virtual circuit networks.

DATAGRAM NETWORKS:

In a datagram network, each packet is treated independently of all others. Even if a packet is part of a
multipacket transmission, the network treats it as though it existed alone. Packets in this approach are
referred to as datagrams. Datagram switching is normally done at the network layer.

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FIGURE 1.56: DELAY IN A DATAGRAM NETWORK

FIGURE 1.55: ROUTING TABLE IN A DATAGRAM NETWORK
The packet travels through two switches. There are three transmission times (3T), three propagation delays
A switch in a datagram network uses a routing table that is based on the destination address. (slopes 3τ of the lines), and two waiting times (w1 + w2). We ignore the processing time in each switch. The
total delay is
Destination Address: Every packet in a datagram network carries a header that contains, among other
information, the destination address of the packet. When the switch receives the packet, this destination Total delay = 3T + 3τ + w1 + w2
address is examined; the routing table is consulted to find the corresponding port through which the packet
VIRTUAL-CIRCUIT NETWORKS:
should be forwarded. This address, unlike the address in a virtual-circuit network, remains the same during
the entire journey of the packet. A virtual-circuit network is a cross between a circuit-switched network and a datagram network. It has
some characteristics of both.
The destination address in the header of a packet in a datagram network remains the same during the
entire journey of the packet. As in a circuit-switched network, there are setup and teardown phases in addition to the data transfer phase.

Efficiency: Resources can be allocated during the setup phase, as in a circuit-switched network, or on demand, as in a
datagram network.
The efficiency of a datagram network is better than that of a circuit-switched network; resources are
allocated only when there are packets to be transferred. If a source sends a packet and there is a delay of a As in a datagram network, data are packetized and each packet carries an address in the header. However, the
few minutes before another packet can be sent, the resources can be reallocated during these minutes for address in the header has local jurisdiction (it defines what the next switch should be and the channel on
other packets from other sources. which the packet is being carried), not end-to-end jurisdiction.

Delay: As in a circuit-switched network, all packets follow the same path established during the connection. A
virtual-circuit network is normally implemented in the data-link layer, while a circuit-switched network is
There may be greater delay in a datagram network than in a virtual-circuit network. Although there are no
implemented in the physical layer and a datagram network in the network layer. But this may change in the
setup and teardown phases, each packet may experience a wait at a switch before it is forwarded. Figure 1.56
future.
gives an example of delay in a datagram network for one packet.
Figure 1.57 is an example of a virtual-circuit network. The network has switches that allow traffic from
sources to destinations. A source or destination can be a computer, packet switch, bridge, or any other device
that connects other networks.

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Computer Networks Computer Networks

Figure 1.59 shows a frame arriving at port 1 with a VCI of 14. When the frame arrives, the switch looks in its
table to find port 1 and a VCI of 14. When it is found, the switch knows to change the VCI to 22 and send
out the frame from port 3.

Setup Phase:

In the setup phase, a switch creates an entry for a virtual circuit. For example, suppose source A needs to
create a virtual circuit to B. Two steps are required: the setup request and the acknowledgment.

FIGURE 1.57: VIRTUAL-CIRCUIT NETWORK

Addressing: In a virtual-circuit network, two types of addressing are involved: global and local (virtual-
circuit identifier).

Global Addressing: A source or a destination needs to have a global address—an address that can be unique
in the scope of the network or internationally if the network is part of an international network.

However, we will see that a global address in virtual-circuit networks is used only to create a virtual-circuit
identifier.

Virtual-Circuit Identifier:
FIGURE 1.59: SWITCH AND TABLES IN A VIRTUAL-CIRCUIT NETWORK
The identifier that is actually used for data transfer is called the virtual-circuit identifier (VCI) or the label.
Setup Request: A setup request frame is sent from the source to the destination. Figure 1.60 shows the
A VCI, unlike a global address, is a small number that has only switch scope; it is used by a frame between
process.
two switches. When a frame arrives at a switch, it has a VCI; when it leaves, it has a different VCI.

Figure 1.58 shows how the VCI in a data frame changes from one switch to another. Note that a VCI does
not need to be a large number since each switch can use its own unique set of VCIs.

FIGURE 1.58: VIRTUAL-CIRCUIT IDENTIFIER

Three Phases:

As in a circuit-switched network, a source and destination need to go through three phases in a virtual-circuit
network: setup, data transfer, and teardown. In the setup phase, the source and destination use their global
addresses to help switches make table entries for the connection. In the teardown phase, the source and FIGURE 1.60: SETUP REQUEST IN A VIRTUAL-CIRCUIT NETWORK
destination inform the switches to delete the corresponding entry. Data transfer occurs between these two
Acknowledgment:
phases.
A special frame, called the acknowledgment frame, completes the entries in the switching tables. Figure 1.61
Data-Transfer Phase: To transfer a frame from a source to its destination, all switches need to have a table
shows the process.
entry for this virtual circuit. The table, in its simplest form, has four columns. This means that the switch
holds four pieces of information for each virtual circuit that is already set up. Figure 1.59 shows such a Teardown Phase:
switch and its corresponding table.

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Computer Networks Computer Networks

In this phase, source A, after sending all frames to B, sends a special frame called a teardown request.
Destination B responds with a teardown confirmation frame. All switches delete the corresponding entry
from their tables.

FIGURE 1.62: DELAY IN A VIRTUAL-CIRCUIT NETWORK

Circuit-Switched Technology in WANs: virtual-circuit networks are used in switched WANs such as ATM
networks. The data-link layer of these technologies is well suited to the virtual circuit technology.

Switching at the data-link layer in a switched WAN is normally implemented by using virtual-circuit
techniques.

FIGURE 1.61: SETUP ACKNOWLEDGMENT IN A VIRTUAL-CIRCUIT NETWORK ***** USEFUL QUESTIONS with ANSWERS *****

Efficiency: Q: Define the term protocol.

As we said before, resource reservation in a virtual-circuit network can be made during the setup or can be Ans: Set of rules established for users to exchange information.
on demand during the data-transfer phase.
Q: Define the term topology.
In the first case, the delay for each packet is the same; in the second case, each packet may encounter
Ans: Architecture of a network.
different delays.
Q: Define the term deterministic.
There is one big advantage in a virtual-circuit network even if resource allocation is on demand. The source
can check the availability of the resources, without actually reserving it. Consider a family that wants to dine Ans: Access to the network is provided at fixed time intervals
at a restaurant.
Q: What is the difference between a hub and a switch? Ans: Hub – Broadcasts data it receives to all devices
Although the restaurant may not accept reservations (allocation of the tables is on demand), the family can connected to its ports. Switch – Establishes a direct connection from the sender to the destination without
call and find out the waiting time. This can save the family time and effort. passing the data traffic to other networking devices.
Delay in Virtual-Circuit Networks: Q: Cite the three advantages of a wired network. Ans: 1) Faster network data transfer speeds (within the
LAN), 2) Relatively inexpensive to setup & 3) the network is not susceptible to outside interference.
In a virtual-circuit network, there is a one-time delay for setup and a one-time delay for teardown. If
resources are allocated during the setup phase, there is no wait time for individual packets. Figure 1.62 Q: Cite three advantages of a wireless network. Ans: 1) User mobility 2) Simple installations & 3) No
shows the delay for a packet traveling through two switches in a virtual-circuit network. cables
The packet is traveling through two switches (routers). There are three transmission times (3T ), three Q: What does it mean for a wireless networking device to be Wi-Fi compliant? Ans: That the device has
propagation times (3τ), data transfer depicted by the sloping lines, a setup delay (which includes transmission been tested by the Wi-Fi Alliance (Wireless Fidelity) and is certified for compliance with 802.11x wireless
and propagation in two directions), and a teardown delay (which includes transmission and propagation in standards.
one direction).
Q: List five steps that can be used to protect the home network. Ans: 1) Change the default factory
We ignore the processing time in each switch. The total delay time is Total delay + 3T + 3τ + setup delay + passwords. 2) Change the default SSID. “Service Set Identifier” 3) turn on encryption. 4) Turn off the SSID
teardown delay broadcast. & 5) Enable MAC address filtering.

Q: What is Stateful Packet Inspection “SPI”? Ans: A type of firewall protection that inspects incoming data
packets to make sure they correspond to an outgoing request.

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Computer Networks Computer Networks

Above questions link: http://pcr-tech-sc.2302233.n4.nabble.com/Chapter-1-Introduction-to-Computer- 9. A connection provides a dedicated link between two devices.
Networks-Questions-Answers-td3891566.html
A) Point-to-point B) multipoint

C) Primary D) secondary
UNIT –I
10. In a connection, more than two devices can share a single link.
Objective questions
A) Point-to-point B) multipoint
1. The is the physical path over which a message travels.
C) Primary D) secondary
A) Protocol B) Medium
11. In transmission, the channel capacity is shared by both communicating devices at all times.
C) Signal D) All the above
A) Simplex B) half-duplex
2. The information to be communicated in a data communications system is the .
C) full-duplex D) half-simplex
A) Medium B) Protocol
12. In the original ARPANET, were directly connected together.
C) Message D) Transmission
A) IMPs B) host computers
3. Frequency of failure and network recovery time after a failure is measures of the of network.
C) Networks D) routers
A) Performance B) Reliability
13. This was the first network.
C) Security D) Feasibility
A) CSNET B) NSFNET
4. TDM Stands for .
C) ANSNET D) ARPANET
A) Time discrete measures B) Time Division Multiplexing
14. Which organization has authority over interstate and international commerce in the
C) Time division measures D) All the above
Communications field?
5. Which topology requires a central controller or hub?
A) ITU-T B) IEEE
A) Mesh B) Star
C) FCC D) ISOC
C) Bus D) Ring
15. is special-interest groups that quickly test, evaluate, and standardize new technologies.
6. Which topology requires a multipoint connection?
A) Forums B) Regulatory agencies
A) Mesh B) Star
B) C) Standards organizations D) All of the above
C) Bus D) Ring
16. Which agency developed standards for physical connection interfaces and electronic signalling a
7. Communication between a computer and a keyboard involves transmission. specification?

A) Simplex B) half-duplex A) EIA B) ITU-T

C) full-duplex D) automatic C) ANSI D) ISO

8. A television broadcast is an example of transmission. 17. is the protocol suite for the current Internet.

A) Simplex B) half-duplex A) TCP/IP B) NCP

C) full-duplex D) automatic C) UNIX D) ACM

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Computer Networks Computer Networks

18. refers to the structure or format of the data, meaning the order in which they are 27. A single channel is shared by multiple signals by
A) analogy modulation B) digital modulation
Presented.
C) Multiplexing D) none of the mentioned
A) Semantics B) Syntax
28. Wireless transmission can be done via
C) Timing D) All of the above A) radio waves B) microwaves
19. defines how a particular pattern to be interpreted, and what action is to be taken C) Infrared D) all of the mentioned
based on that interpretation. 29. The physical layer translates logical communication requests from the into hardware specific
A) Semantics B) Syntax operations.
A) Data link layer B) network layer
C) Timing D) None of the above
C) Transport layer D) application layer
20. refers to two characteristics: when data should be sent and how fast it can be sent.
30. In asynchronous serial communication the physical layer provides
A) Semantics B) Syntax A) start and stop signalling B) flow control

C) Timing D) none of the above C) Both (a) and (b) D) none of the mentioned

21. The Physical layer of GSM handles of . UNIT –I

A) Radio-specific B) Satellite DESCRIPTIVE QUESTIONS

C) FDM D) DSS 1. List out the advantages and drawbacks of bus topology.

22. Fiber optic cable consists of 2 Explain the Difference between LAN, MAN, WAN.

A) Plastic core B) plastic cladding 3. Explain the OSI Reference model

C) Glass cladding D) ALL 4. Explain about the TCP/IP model

23) Telephone system consists of 5. Explain and draw the ARPANET

A) Local loops B) Trunks &Multiplexing 6 Write any four reasons for using Layer Protocol.

C) Switching D) ALL 7. List out the advantages and disadvantages of OSI Reference model compare with TCP/IP model.

24) In the OSI model, as a data packet moves from the lower to the upper layers, headers are 8. Difference between the datagram packet switching and virtual circuit switching.

A) Added B) removed 9. Write the advantages of optical fiber over twisted pair and coaxial cable.

C) Rearranged D) modified 10. Explain about Public switching telephone network.

25. Which transmission media has the highest transmission speed in a network?
A) Coaxial cable B) twisted pair cable

C) Optical fiber D) electrical cable

26. Physical layer provides

A) mechanical specifications of electrical connectors and cables B) electrical specification of transmission
line signal level

C) Specification for IR over optical fiber D) all of the mentioned

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UNIT 2 layer of the sending node needs to encapsulate the datagram received from the network in a frame, and
INTRODUCTION TO DATALINK LAYER the data-link layer of the receiving node needs to decapsulate the datagram from the frame.

 INTRODUCTION: FRAMING: Definitely, the first service provided by the data-link layer is framing. The data-link layer at
The Internet is a combination of networks glued together by connecting devices (routers or each node needs to encapsulate the datagram (packet received from the network layer) in a frame before
switches). If a packet is to travel from a host to another host, it needs to pass through these networks. sending it to the next node. The node also needs to decapsulate the datagram from the frame received on
Figure shows the same scenario. Communication at the data-link layer is made up of five separate logical the logical channel. Although we have shown only a header for a frame.
connections between the data-link layers in the path.
FLOW CONTROL: The sending data-link layer at the end of a link is a producer of frames; the receiving
data-link layer at the other end of a link is a consumer. If the rate of produced frames is higher than the
rate of consumed frames, frames at the receiving end need to be buffered while waiting to be consumed
(processed). Definitely, we cannot have an unlimited buffer size at the receiving side. We have two
choices. The first choice is to let the receiving data-link layer drop the frames if its buffer is full. The
second choice is to let the receiving data-link layer send a feedback to the sending data-link layer to ask it
to stop or slow down. Different data-link-layer protocols use different strategies for flow control.

ERROR CONTROL: At the sending node, a frame in a data-link layer needs to be changed to bits,
transformed to electromagnetic signals, and transmitted through the transmission media. At the receiving
node, electromagnetic signals are received, transformed to bits, and put together to create a frame. Since
electromagnetic signals are susceptible to error, a frame is susceptible to error. The error needs first to be
detected. After detection, it needs to be either corrected at the receiver node or discarded and
retransmitted by the sending node.

CONGESTION CONTROL: Although a link may be congested with frames, which may result in frame
loss, most data-link-layer protocols do not directly use a congestion control to alleviate congestion,
although some wide-area networks do. In general, congestion control is considered an issue in the
network layer or the transport layer because of its end-to-end nature.

TWO CATEGORIES OF LINKS: Although two nodes are physically connected by a transmission
medium such as cable or air, we need to remember that the data-link layer controls how the medium is
used. We can have a data-link layer that uses the whole capacity of the medium; we can also have a data-
link layer that uses only part of the capacity of the link. In other words, we can have a point-to-point link
or a broadcast link. In a point-to-point link, the link is dedicated to the two devices; in a broadcast link,
the link is shared between several pairs of devices.
COMMUNICATION AT THE DATA-LINK LAYER
Two Sub layers: To better understand the functionality of and the services provided by the link layer, we
The data-link layer at Alice’s computer communicates with the data-link layer at router R2. The data-link
can divide the data-link layer into two sub layers: data link control (DLC) and media access control
layer at router R2 communicates with the data-link layer at router R4, and so on. Finally, the data-link
(MAC). The data link control sub layer deals with all issues common to both point-to-point and broadcast
layer at router R7 communicates with the data-link layer at Bob’s computer. Only one data-link layer is
links; the media access control sub layer deals only with issues specific to broadcast links.
involved at the source or the destination, but two data-link layers are involved at each router.
 LINK-LAYER ADDRESSING:
The reason is that Alice’s and Bob’s computers are each connected to a single network, but each router A link-layer address is sometimes called a link address, sometimes a physical address, and
takes input from one network and sends output to another network. Note that although switches are also sometimes a MAC address.
involved in the data-link-layer communication, for simplicity we have not shown them in the figure. Since a link is controlled at the data-link layer, the addresses need to belong to the data-link layer.
When a datagram passes from the network layer to the data-link layer, the datagram will be encapsulated
SERVICES:
in a frame and two data-link addresses are added to the frame header. These two addresses are changed
The data-link layer is located between the physical and the network layers. The data link layer provides
services to the network layer; it receives services from the physical layer. The duty scope of the data-link every time the frame moves from one link to another. Figure demonstrates the concept in a small internet.
layer is node-to-node. When a packet is travelling in the Internet, the data-link layer of a node (host or
router) is responsible for delivering a datagram to the next node in the path. For this purpose, the data-link

1 2

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In the internet in Figure, we have three links and two routers. We also have shown only two hosts: Alice Anytime a host or a router needs to find the link-layer address of another host or router in its
(source) and Bob (destination). For each host, we have shown two addresses, the IP addresses (N) and the network, it sends an ARP request packet. The packet includes the link-layer and IP addresses of the
link-layer addresses (L). sender and the IP address of the receiver. Because the sender does not know the link-layer address of the
receiver, the query is broadcast over the link using the link-layer broadcast address.
Note that a router has as many pairs of addresses as the number of links the router is connected to.
We have shown three frames, one in each link. Each frame carries the same datagram with the same Every host or router on the network receives and processes the ARP request packet, but only the
source and destination addresses (N1 and N8), but the link-layer addresses of the frame change from link intended recipient recognizes its IP address and sends back an ARP response packet. The response packet
to link. contains the recipient’s IP and link-layer addresses. The packet is unicast directly to the node that sent the
request packet.
In link 1, the link-layer addresses are L1 and L2. In link 2, they are L4 and L5. In link 3, they are
L7 and L8. In Figure (a), the system on the left (A) has a packet that needs to be delivered to another system
(B) with IP address N2. System A needs to pass the packet to its data-link layer for the actual delivery,
but it does not know the physical address of the recipient.

It uses the services of ARP by asking the ARP protocol to send a broadcast ARP request packet to
ask for the physical address of a system with an IP address of N2. This packet is received by every system
on the physical network, but only system B will answer it, as shown in Figure (b).

System B sends an ARP reply packet that includes its physical address. Now system A can send
all the packets it has for this destination using the physical address it received.

FIGURE: IP ADDRESSES AND LINK-LAYER ADDRESSES IN A SMALL INTERNET

Note that the IP addresses and the link-layer addresses are not in the same order. For IP
addresses, the source address comes before the destination address; for link-layer addresses, the
destination address comes before the source.

Address Resolution Protocol (ARP):

Anytime a node has an IP datagram to send to another node in a link, it has the IP address of the
receiving node. The source host knows the IP address of the default router.
Each router except the last one in the path gets the IP address of the next router by using its
forwarding table. The last router knows the IP address of the destination host. However, the IP address of
the next node is not helpful in moving a frame through a link; we need the link-layer address of the next
node. This is the time when the Address Resolution Protocol (ARP) becomes helpful. ARP accepts an
FIGURE: ARP OPERATION
IP address from the IP protocol, maps the address to the corresponding link-layer address, and passes it to
the data-link layer. Packet Format:
Figure shows the format of an ARP packet. The names of the fields are self-explanatory. The
hardware type field defines the type of the link-layer protocol; Ethernet is given the type 1.
The protocol type field defines the network-layer protocol: IPv4 protocol is (0800)16. The source
hardware and source protocol addresses are variable-length fields defining the link-layer and network-
layer addresses of the sender.
The destination hardware address and destination protocol address fields define the receiver link-
layer and network-layer addresses. An ARP packet is encapsulated directly into a data-link frame. The
frame needs to have a field to show that the payload belongs to the ARP and not to the network-layer
FIGURE: POSITION OF ARP IN TCP/IP PROTOCOL SUITE datagram.

3 4

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 Error Detection:
Cyclic Redundancy Check:

FIGURE: ARP PACKET

ERROR DETECTION AND CORRECTION

 Types of Errors:
Whenever bits flow from one point to another, they are subject to unpredictable changes because
of interference. This interference can change the shape of the signal. The term single-bit error means
that only 1 bit of a given data unit (such as a byte, character, or packet) is changed from 1 to 0 or from 0 FIGURE: CRC ENCODER AND DECODER
to 1.
The term burst error means that 2 or more bits in the data unit have changed from 1 to 0 or from Encoder: Let us take a closer look at the encoder. The encoder takes a dataword and augments it with n −
0 to 1. Figure 2.8 shows the effect of a single-bit and a burst error on a data unit. k number of 0s. It then divides the augmented dataword by the divisor, as shown in Figure.

Decoder: The codeword can change during transmission. The decoder does the same division process as
the encoder. The remainder of the division is the syndrome. If the syndrome is all 0s, there is no error
with a high probability; the dataword is separated from the received codeword and accepted. Otherwise,
everything is discarded.

FIGURE: SINGLE-BIT AND BURST ERROR

Redundancy:
The central concept in detecting or correcting errors is redundancy. To be able to detect or
correct errors, we need to send some extra bits with our data. These redundant bits are added by the
sender and removed by the receiver. Their presence allows the receiver to detect or correct corrupted bits.
Detection versus Correction:

The correction of errors is more difficult than the detection. In error detection, we are only
looking to see if any error has occurred. The answer is a simple yes or no. We are not even interested in
the number of corrupted bits. A single-bit error is the same for us as a burst error.

In error correction, we need to know the exact number of bits that are corrupted and, more
importantly, their location in the message. The number of errors and the size of the message are important
factors.

If we need to correct a single error in an 8-bit data unit, we need to consider eight possible error FIGURE: DIVISION IN CRC ENCODER
locations; if we need to correct two errors in a data unit of the same size, we need to consider 28
(permutation of 8 by 2) possibilities. You can imagine the receiver’s difficulty in finding 10 errors in a
data unit of 1000 bits.

5 6

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Suppose the message is a list of five 4-bit numbers that we want to send to a destination. In addition to
sending these numbers, we send the sum of the numbers. For example, if the set of numbers is (7, 11, 12,
0, 6), we send (7, 11, 12, 0, 6, 36), where 36 is the sum of the original numbers.

The receiver adds the five numbers and compares the result with the sum. If the two are the same, the
receiver assumes no error, accepts the five numbers, and discards the sum. Otherwise, there is an error
somewhere and the message is not accepted.

One’s Complement Addition:

The previous example has one major drawback. Each number can be written as a 4-bit word (each is less
than 15) except for the sum. One solution is to use one’s complement arithmetic. In this arithmetic, we
can represent unsigned numbers between 0 and 2m − 1 using only m bits. If the number has more than m
bits, the extra leftmost bits need to be added to the m rightmost bits (wrapping).
In the previous example, the decimal number 36 in binary is (100100)2. To change it to a 4-bit
number we add the extra leftmost bit to the right four bits as shown below.

FIGURE: DIVISION IN THE CRC DECODER FOR TWO CASES (10)2+ (0100)2= (0110)2 → (6)10

The above Figure shows two cases: The left-hand figure shows the value of the syndrome when no error Instead of sending 36 as the sum, we can send 6 as the sum (7, 11, 12, 0, 6, 6). The receiver can
has occurred; the syndrome is 000. The right-hand part of the figure shows the case in which there is a add the first five numbers in one’s complement arithmetic. If the result is 6, the numbers are accepted;
single error. The syndrome is not all 0s (it is 011). otherwise, they are rejected.

ADVANTAGES OF CYCLIC CODES:

We have seen that cyclic codes have a very good performance in detecting single-bit errors,
double errors, an odd number of errors, and burst errors. They can easily be implemented in hardware and
software. They are especially fast when implemented in hardware. This has made cyclic codes a good
candidate for many networks.

CHECKSUM:
Checksum is an error-detecting technique that can be applied to a message of any length. In the
Internet, the checksum technique is mostly used at the network and transport layer rather than the data-
link layer.  FORWARD ERROR CORRECTION:
At the source, the message is first divided into m-bit units. The generator then creates an extra m-
bit unit called the checksum, which is sent with the message. At the destination, the checker creates a new We need to correct the error or reproduce the packet immediately. Several schemes have been designed
checksum from the combination of the message and sent checksum. If the new checksum is all 0s, the and used in this case that is collectively referred to as forward error correction (FEC) techniques.
message is accepted; otherwise, the message is discarded (Figure). Note that in the real implementation,
the checksum unit is not necessarily added at the end of the message; it can be inserted in the middle of HAMMING DISTANCE:
To detect s errors, the minimum Hamming distance should be dmin = s + 1. For error detection, we
the message.
definitely need more distance. It can be shown that to detect t errors, we need to have dmin = 2t + 1. In
other words, if we want to correct 10 bits in a packet, we need to make the minimum hamming distance
21 bits, which means a lot of redundant bits, need to be sent with the data.
To give an example, consider the famous BCH code. In this code, if data is 99 bits, we need to
send 255 bits (extra 156 bits) to correct just 23 possible bit errors. Most of the time we cannot afford such
a redundancy.

CHUNK INTERLEAVING: Another way to achieve FEC in multimedia is to allow some small chunks
to be missing at the receiver. We cannot afford to let all the chunks belonging to the same packet be
missing; however, we can afford to let one chunk be missing in each packet. Figure shows that we can
FIGURE: CHECKSUM divide each packet into 5 chunks (normally the number is much larger).

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DATA LINK CONTROL

 DLC SERVICES:
The data link control (DLC) deals with procedures for communication between two adjacent nodes—
node-to-node communication—no matter whether the link is dedicated or broadcast. Data link control
functions include framing and flow and error control.

FRAMING: The data-link layer, needs to pack bits into frames, so that each frame is distinguishable
from another. Framing in the data-link layer separates a message from one source to a destination by
adding a sender address and a destination address. The destination address defines where the packet is to
go; the sender address helps the recipient acknowledge the receipt.
Although the whole message could be packed in one frame, which is not normally done; one
reason is that a frame can be very large, making flow and error control very inefficient. When a message
is carried in one very large frame, even a single-bit error would require the retransmission of the whole
FIGURE: INTERLEAVING
frame. When a message is divided into smaller frames, a single-bit error affects only that small frame.
We can then create data chunk by chunk (horizontally), but combine the chunks into packets vertically. In
this case, each packet sent carries a chunk from several original packets. If the packet is lost, we miss only Character-Oriented Framing:
one chunk in each packet, which is normally acceptable in multimedia communication. To separate one frame from the next, an 8-bit (1-byte) flag is added at the beginning and the end
of a frame. The flag, composed of protocol-dependent special characters, signals the start or end of a
COMBINING HAMMING DISTANCE AND INTERLEAVING: frame. Figure 2.17 shows the format of a frame in a character-oriented protocol.
Hamming distance and interleaving can be combined. We can first create n-bit packets that can correct t-
bit errors. Then we interleave m rows and send the bits column by column. In this way, we can
automatically correct burst errors up to m × t-bit errors.
FIGURE: A FRAME IN A CHARACTER-ORIENTED PROTOCOL
COMPOUNDING HIGH- AND LOW-RESOLUTION PACKETS:
Still another solution is to create a duplicate of each packet with a low-resolution redundancy and Byte stuffing (or character stuffing), a special byte is added to the data section of the frame when there is
combine the redundant version with the next packet. For example, we can create four low-resolution a character with the same pattern as the flag. The data section is stuffed with an extra byte. This byte is
packets out of five high-resolution packets and send them as shown in Figure. If a packet is lost, we can
use the low-resolution version from the next packet. Note that the low-resolution section in the first usually called the escape character (ESC) and has a predefined bit pattern. Whenever the receiver
packet is empty. encounters the ESC character, it removes it from the data section and treats the next character as data, not
In this method, if the last packet is lost, it cannot be recovered, but we use the low-resolution as a delimiting flag. Figure shows the situation.
version of a packet if the lost packet is not the last one. The audio and video reproduction does not have
Byte stuffing by the escape character allows the presence of the flag in the data section of the
the same quality, but the lack of quality is not recognized most of the time.
frame, but it creates another problem. What happens if the text contains one or more escape characters
followed by a byte with the same pattern as the flag? To solve this problem, the escape characters that are
part of the text must also be marked by another escape character. In other words, if the escape character is
part of the text, an extra one is added to show that the second one is part of the text.

FIGURE: COMPOUNDING HIGH- AND LOW-RESOLUTION PACKETS FIGURE: BYTE STUFFING AND UNSTUFFING

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Bit-Oriented Framing: Error control at the data-link layer is normally very simple and implemented using one of the
Bit stuffing is the process of adding one extra 0 whenever five consecutive 1s follow a 0 in the data, following two methods. In both methods, a CRC is added to the frame header by the sender and checked
so that the receiver does not mistake the pattern 0111110 for a flag. by the receiver.

Figure shows bit stuffing at the sender and bit removal at the receiver. Note that even if we have a  In the first method, if the frame is corrupted, it is silently discarded; if it is not corrupted, the
0 after five 1s, we still stuff a 0. The 0 will be removed by the receiver. This means that if the flag like packet is delivered to the network layer. This method is used mostly in wired LANs such as
pattern 01111110 appears in the data, it will change to 011111010 (stuffed) and is not mistaken for a flag Ethernet.
by the receiver. The real flag 01111110 is not stuffed by the sender and is recognized by the receiver.
 In the second method, if the frame is corrupted, it is silently discarded; if it is not corrupted,
an acknowledgment is sent (for the purpose of both flow and error control) to the sender.

 DATA-LINK LAYER PROTOCOLS:

SIMPLE PROTOCOL:
Our first protocol is a simple protocol with neither flow nor error control. We assume that the
receiver can immediately handle any frame it receives. In other words, the receiver can never be
overwhelmed with incoming frames. Figure shows the layout for this protocol.

FIGURE: SIMPLE PROTOCOL

The data-link layer at the sender gets a packet from its network layer, makes a frame out of it, and
FIGURE: BIT STUFFING AND UNSTUFFING
sends the frame. The data-link layer at the receiver receives a frame from the link, extracts the packet
FLOW AND ERROR CONTROL: from the frame, and delivers the packet to its network layer. The data-link layers of the sender and
If the items are produced faster than they can be consumed, the consumer can be overwhelmed receiver provide transmission services for their network layers.
and may need to discard some items. If the items are produced more slowly than they can be consumed,
the consumer must wait, and the system becomes less efficient. Flow control is related to the first issue.
We need to prevent losing the data items at the consumer site.

FIGURE: FLOW CONTROL AT THE DATA-LINK LAYER

Buffers: Although flow control can be implemented in several ways, one of the solutions is normally to FSM OF SIMPLE PROTOCOL
use two buffers; one at the sending data-link layer and the other at the receiving data-link layer. A buffer
is a set of memory locations that can hold packets at the sender and receiver. The flow control STOP-AND-WAIT PROTOCOL:
Stop-and-Wait protocol uses both flow and error control. In this protocol, the sender sends one frame at
communication can occur by sending signals from the consumer to the producer. When the buffer of the
a time and waits for an acknowledgment before sending the next one. To detect corrupted frames, we
receiving data-link layer is full, it informs the sending data-link layer to stop pushing frames. need to add a CRC to each data frame.
When a frame arrives at the receiver site, it is checked. If its CRC is incorrect, the frame is
Error Control: Since the underlying technology at the physical layer is not fully reliable, we need to corrupted and silently discarded. The silence of the receiver is a signal for the sender that a frame was
implement error control at the data-link layer to prevent the receiving node from delivering corrupted either corrupted or lost.
packets to its network layer. Every time the sender sends a frame, it starts a timer. If an acknowledgment arrives before the
timer expires, the timer is stopped and the sender sends the next frame (if it has one to send). If the timer
expires, the sender resends the previous frame, assuming that the frame was either lost or corrupted.

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Figure shows the outline for the Stop-and-Wait protocol. Note that only one frame and one We have one primary station and multiple secondary stations. A primary station can send
acknowledgment can be in the channels at any time. commands; a secondary station can only respond. The NRM is used for both point-to-point and
multipoint links, as shown in Figure.

FIGURE: STOP-AND-WAIT PROTOCOL

FIGURE: NORMAL RESPONSE MODE

In ABM, the configuration is balanced. The link is point-to-point, and each station can function as
a primary and a secondary (acting as peers), as shown in Figure. This is the common mode today.

FIGURE: ASYNCHRONOUS BALANCED MODE

Framing: To provide the flexibility necessary to support all the options possible in the modes and
configurations just described, HDLC defines three types of frames: information frames (I-frames),
supervisory frames (S-frames), and unnumbered frames (U-frames).

Each type of frame serves as an envelope for the transmission of a different type of message. I-
frames are used to data-link user data and control information relating to user data (piggybacking).

S-frames are used only to transport control information. U-frames are reserved for system
FIG: FSM OF STOP-AND-WAIT PROTOCOL management. Information carried by U-frames is intended for managing the link itself. Each frame in
HDLC may contain up to six fields, as shown in Figure: a beginning flag field, an address field, a control
Piggybacking: The two protocols we discussed in this section are designed for unidirectional
field, an information field, a frame check sequence (FCS) field, and an ending flag field. In multiple-
communication, in which data is flowing only in one direction although the acknowledgment may travel
frame transmissions, the ending flag of one frame can serve as the beginning flag of the next frame.
in the other direction. Protocols have been designed in the past to allow data to flow in both directions.
However, to make the communication more efficient, the data in one direction is piggybacked with the
acknowledgment in the other direction. In other words, when node A is sending data to node B, Node A
also acknowledges the data received from node B. Because piggybacking makes communication at the
data link layer more complicated, it is not a common practice.

HDLC:
High-level Data Link Control (HDLC) is a bit-oriented protocol for communication over point-to-point
and multipoint links. It implements the Stop-and-Wait protocol. FIGURE 2.27: HDLC FRAMES
Configurations and Transfer Modes: HDLC provides two common transfer modes that can be used in
 Flag field. This field contains synchronization pattern 01111110, which identifies both the
different configurations: normal response mode (NRM) and asynchronous balanced mode (ABM). In
beginning and the end of a frame.
normal response mode (NRM), the station configuration is unbalanced.
 Address field. This field contains the address of the secondary station. If a primary station
created the frame, it contains to address. If a secondary station creates the frame, it contains

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from address. The address field can be one byte or several bytes long, depending on the needs expires, that the last frame is lost or damaged. The value of N(R) is the negative
of the network. acknowledgment number.
 Selective reject (SREJ). If the value of the code subfield is 11, it is an SREJ Sframe. This is a
 Control field. The control field is one or two bytes used for flow and error control. NAK frame used in Selective Repeat ARQ. Note that the HDLC Protocol uses the term selective
 Information field. The information field contains the user’s data from the network layer or reject instead of selective repeat. The value of N(R) is the negative acknowledgment number.
management information. Its length can vary from one network to another. Control Field for U-Frames
Unnumbered frames are used to exchange session management and control information between
 FCS field. The frame check sequence (FCS) is the HDLC error detection field. It can contain connected devices. Unlike S-frames, U-frames contain an information field, but one used for system
either a 2- or 4-byte CRC. management information, not user data. As with S-frames, however, much of the information carried by
U-frames is contained in codes included in the control field. U-frame codes are divided into two sections:
The control field determines the type of frame and defines its functionality. The format is specific a 2-bit prefix before the P/F bit and a 3-bit suffix after the P/F bit. Together, these two segments (5 bits)
for the type of frame, as shown in Figure. can be used to create up to 32 different types of U-frames.

POINT-TO-POINT PROTOCOL (PPP):

One of the most common protocols for point-to-point access is the Point-to-Point Protocol
(PPP). Today, millions of Internet users who need to connect their home computers to the server of an
Internet service provider use PPP. The majority of these users have a traditional modem; they are
FIGURE: CONTROL FIELD FORMAT FOR THE DIFFERENT FRAME TYPES connected to the Internet through a telephone line, which provides the services of the physical layer. But
to control and manage the transfer of data, there is a need for a point-to-point protocol at the data-link
Control Field for I-Frames layer. PPP is by far the most common.
I- frames are designed to carry user data from the network layer. In addition, they can include flow- and Services: The designers of PPP have included several services to make it suitable for a point-to-point
error-control information (piggybacking). The subfields in the control field are used to define these protocol, but have ignored some traditional services to make it simple.
functions. The first bit defines the type. If the first bit of the control field is 0, this means the frame is an I-
frame. The next 3 bits, called N(S), define the sequence number of the frame. Note that with 3 bits, we Services Provided by PPP: PPP defines the format of the frame to be exchanged between devices. It also
defines how two devices can negotiate the establishment of the link and the exchange of data. PPP is
can define a sequence number between 0 and 7. The last 3 bits, called N(R), correspond to the designed to accept payloads from several network layers (not only IP).
acknowledgment number when piggybacking is used. The single bit between N(S) and N(R) is called the Authentication is also provided in the protocol, but it is optional. The new version of PPP, called
P/F bit. The P/F field is a single bit with a dual purpose. It has meaning only when it is set (bit = 1) and Multilink PPP, provides connections over multiple links. One interesting feature of PPP is that it provides
can mean poll or final. It means poll when the frame is sent by a primary station to a secondary (when the network address configuration. This is particularly useful when a home user needs a temporary network
address field contains the address of the receiver). It means final when the frame is sent by a secondary to address to connect to the Internet.
a primary (when the address field contains the address of the sender).
Services Not Provided by PPP: PPP does not provide flow control. A sender can send several frames one
Control Field for S-Frames
after another with no concern about overwhelming the receiver. PPP has a very simple mechanism for
Supervisory frames are used for flow and error control whenever piggybacking is either impossible or error control. A CRC field is used to detect errors.
inappropriate. S-frames do not have information fields. If the first 2 bits of the control field are 10, this If the frame is corrupted, it is silently discarded; the upper-layer protocol needs to take care of the
means the frame is an S-frame. The last 3 bits, called N(R), correspond to the acknowledgment number problem. Lack of error control and sequence numbering may cause a packet to be received out of order.
(ACK) or negative acknowledgment number (NAK), depending on the type of S-frame. The 2 bits called PPP does not provide a sophisticated addressing mechanism to handle frames in a multipoint
code are used to define the type of S-frame itself. With 2 bits, we can have four types of S-frames, as configuration.
described below:
 Receive ready (RR). If the value of the code subfield is 00, it is an RR S-frame. This kind of Framing:
frame acknowledges the receipt of a safe and sound frame or group of frames. In this case, the PPP uses a character-oriented (or byte-oriented) frame. Figure shows the format of a PPP frame.
The description of each field follows:
value of the N(R) field defines the acknowledgment number.
 Receive not ready (RNR). If the value of the code subfield is 10, it is an RNR Sframe. This kind
of frame is an RR frame with additional functions. It acknowledges the receipt of a frame or
group of frames, and it announces that the receiver is busy and cannot receive more frames. It
acts as a kind of congestion-control mechanism by asking the sender to slow down. The value of
N(R) is the acknowledgment number. FIGURE: PPP FRAME FORMAT
 Reject (REJ). If the value of the code subfield is 01, it is an REJ S-frame. This is a NAK frame,  Flag. A PPP frame starts and ends with a 1-byte flag with the bit pattern 01111110.
but not like the one used for Selective Repeat ARQ. It is a NAK that can be used in Go-Back-N
ARQ to improve the efficiency of the process by informing the sender, before the sender timer

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 Address. The address field in this protocol is a constant value and set to 11111111 (broadcast c. The system does the same. It applies the same function to the password of the user (known to
address). the system) and the challenge value to create a result. If the result created is the same as the
result sent in the response packet, access is granted; otherwise, it is denied. CHAP is more
 Control. This field is set to the constant value 00000011 (imitating unnumbered frames in
secure than PAP, especially if the system continuously changes the challenge value. Even if
HDLC). As we will discuss later, PPP does not provide any flow control. Error control is also the intruder learns the challenge value and the result, the password is still secret.
limited to error detection.
(iii) Network Control Protocols:
 Protocol. The protocol field defines what is being carried in the data field: either user data or PPP is a multiple-network-layer protocol. It can carry a network-layer data packet from protocols
other information. This field is by default 2 bytes long, but the two parties can agree to use defined by the Internet, OSI, Xerox, DECnet, AppleTalk, Novel, and so on. To do this, PPP has defined a
only 1 byte. specific Network Control Protocol for each network protocol. For example, IPCP (Internet Protocol
Control Protocol) configures the link for carrying IP data packets.
 Payload field. The data field is a sequence of bytes with the default of a maximum of 1500
bytes; but this can be changed during negotiation. IPCP:
One NCP protocol is the Internet Protocol Control Protocol (IPCP). This protocol configures
o The data field is byte-stuffed if the flag byte pattern appears in this field. the link used to carry IP packets in the Internet. IPCP is especially of interest to us. The format of an IPCP
packet is shown in Figure 2.30. IPCP defines seven packets, distinguished by their code values, as shown
o Because there is no field defining the size of the data field, padding is needed if the
in Table.
size is less than the maximum default value or the maximum negotiated value.
Other Protocols: There are other NCP protocols for other network-layer protocols. The OSI
 FCS. The frame check sequence (FCS) is simply a 2-byte or 4-byte standard CRC.
Network Layer Control Protocol has a protocol field value of 8023; the Xerox NS IDP Control Protocol
(i) Link Control Protocol: has a protocol field value of 8025; and so on.
The Link Control Protocol (LCP) is responsible for establishing, maintaining, configuring, and
terminating links. It also provides negotiation mechanisms to set options between the two endpoints. Both
endpoints of the link must reach an agreement about the options before the link can be established.

(ii)Authentication Protocols:
Authentication plays a very important role in PPP because PPP is designed for use over dial-up
links where verification of user identity is necessary. Authentication means validating the identity of a
user who needs to access a set of resources. PPP has created two protocols for authentication: Password
Authentication Protocol and Challenge Handshake Authentication Protocol. Note that these protocols are TABLE 2.4: CODE VALUE FOR IPCP PACKETS
used during the authentication phase.

PAP:
The Password Authentication Protocol (PAP) is a simple authentication procedure with a two-
step process:
a. The user who wants to access a system sends authentication identification (usually the user
name) and a password.
FIGURE 2.30: IPCP PACKET ENCAPSULATED IN PPP FRAME
b. The system checks the validity of the identification and password and either accepts or denies
connection. Multilink PPP:
PPP was originally designed for a single-channel point-to-point physical link. The availability of
CHAP: multiple channels in a single point-to-point link motivated the development of Multilink PPP. In this case,
The Challenge Handshake Authentication Protocol (CHAP) is a three-way handshaking a logical PPP frame is divided into several actual PPP frames. A segment of the logical frame is carried in
authentication protocol that provides greater security than PAP. In this method, the password is kept the payload of an actual PPP frame, as shown in Figure.
secret; it is never sent online.
a. The system sends the user a challenge packet containing a challenge value, usually a few
bytes.

b. The user applies a predefined function that takes the challenge value and the user’s own
password and creates a result. The user sends the result in the response packet to the system.
FIGURE: MULTILINK PPP

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MEDIA ACCESS CONTROL (MAC) A collision involves two or more stations. If all these stations try to resend their frames after the
time-out, the frames will collide again. Pure ALOHA dictates that when the time-out period passes, each
When nodes or stations are connected and use a common link, called a multipoint or broadcast station waits a random amount of time before resending its frame. The randomness will help avoid more
link, we need a multiple-access protocol to coordinate access to the link. The problem of controlling the collisions. We call this time the backoff time TB.
access to the medium is similar to the rules of speaking in an assembly.
Pure ALOHA has a second method to prevent congesting the channel with retransmitted frames.
Many protocols have been devised to handle access to a shared link. All of these protocols belong After a maximum number of retransmission attempts Kmax, a station must give up and try later.
to a sublayer in the data-link layer called media access control (MAC). We categorize them into three
groups, as shown in Figure.

FIGURE: TAXONOMY OF MULTIPLE-ACCESS PROTOCOLS

 RANDOM ACCESS:
In random-access or contention methods, no station is superior to another station and none is
assigned control over another. At each instance, a station that has data to send uses a procedure defined by
the protocol to make a decision on whether or not to send.
This decision depends on the state of the medium (idle or busy). In other words, each station can
Fig: Procedure for pure ALOHA protocol
transmit when it desires on the condition that it follows the predefined procedure, including testing the
state of the medium. Vulnerable time
Let us find the vulnerable time, the length of time in which there is a possibility of collision. We assume
Two features give this method its name. First, there is no scheduled time for a station to transmit. that the stations send fixed-length frames with each frame taking Tfr seconds to send.
Transmission is random among the stations. That is why these methods are called random access.
Second, no rules specify which station should send next. Stations compete with one another to access the
medium. That is why these methods are also called contention methods.

In a random-access method, each station has the right to the medium without being controlled by
any other station. However, if more than one station tries to send, there is an access conflict—collision—
and the frames will be either destroyed or modified.

ALOHA:
Pure ALOHA:
The original ALOHA protocol is called pure ALOHA. This is a simple but elegant protocol. The
idea is that each station sends a frame whenever it has a frame to send (multiple access). However, since
Station B starts to send a frame at time t. Now imagine station A has started to send its frame after t − Tfr.
there is only one channel to share, there is the possibility of collision between frames from different
This leads to a collision between the frames from station B and station A. On the other hand, suppose that
stations.
station C starts to send a frame before time t + Tfr. Here, there is also a collision between frames from
The pure ALOHA protocol relies on acknowledgments from the receiver. When a station sends a station B and station C. Looking at Figure , we see that the vulnerable time during which a collision may
frame, it expects the receiver to send an acknowledgment. If the acknowledgment does not arrive after a occur in pure ALOHA is 2 times the frame transmission time.
time-out period, the station assumes that the frame (or the acknowledgment) has been destroyed and
Pure ALOHA vulnerable time = 2 * Tfr
resends the frame.

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Throughput transmission time. Therefore, if a station generates only one frame in this vulnerable time (and no other
Let us call G the average number of frames generated by the system during one frame transmission time. station generates a frame during this time), the frame will reach its destination successfully.
Then it can be proven that the average number of successfully transmitted frames for pure ALOHA is S =
CSMA:
G × e−2G. The maximum throughput Smax is 0.184, for G = 1/2. In other words, if one-half a frame is
To minimize the chance of collision and, therefore, increase the performance, the CSMA method
generated during one frame transmission time (one frame during two frame transmission times), then 18.4
was developed. The chance of collision can be reduced if a station senses the medium before trying to use
percent of these frames reach their destination successfully. We expect G = 1/2 to produce the maximum
it. Carrier sense multiple access (CSMA) requires that each station first listen to the medium (or check
throughput because the vulnerable time is 2 times the frame transmission time. Therefore, if a station
the state of the medium) before sending. In other words, CSMA is based on the principle “sense before
generates only one frame in this vulnerable time (and no other stations generate a frame during this time),
transmit” or “listen before talk.” CSMA can reduce the possibility of collision, but it cannot eliminate it.
the frame will reach its destination successfully.
Persistence Methods: What should a station do if the channel is busy? What should a station do if
Slotted ALOHA the channel is idle? Three methods have been devised to answer these questions: the 1-persistent
Pure ALOHA has a vulnerable time of 2 × Tfr. This is so because there is no rule that defines when the method, the nonpersistent method, and the p-persistent method
station can send. A station may send soon after another station has started or just before another station
has finished. Slotted ALOHA was invented to improve the efficiency of pure ALOHA. In slotted 1-Persistent: The 1-persistent method is simple and straightforward. In this method, after the
ALOHA we divide the time into slots of Tfr seconds and force the station to send only at the beginning of station finds the line idle, it sends its frame immediately (with probability 1). This method has the highest
the time slot. The following Figure shows an example of frame collisions in slotted ALOHA. chance of collision because two or more stations may find the line idle and send their frames immediately.
We will see later that Ethernet uses this method.
Nonpersistent: In the nonpersistent method, a station that has a frame to send senses the line. If
the line is idle, it sends immediately. If the line is not idle, it waits a random amount of time and then
senses the line again. The nonpersistent approach reduces the chance of collision because it is unlikely
that two or more stations will wait the same amount of time and retry to send simultaneously. However,
this method reduces the efficiency of the network because the medium remains idle when there may be
stations with frames to send.
p-Persistent: The p-persistent method is used if the channel has time slots with a slot duration
equal to or greater than the maximum propagation time. The p-persistent approach combines the
advantages of the other two strategies. It reduces the chance of collision and improves efficiency. In this
Fig:Frames in a slotted ALOHA network method, after the station finds the line idle it follows these steps:
1. With probability p, the station sends its frame.
Because a station is allowed to send only at the beginning of the synchronized time slot, if a station 2. With probability q = 1 − p, the station waits for the beginning of the next
misses this moment, it must wait until the beginning of the next time slot. This means that the station time slot and checks the line again.
which started at the beginning of this slot has already finished sending its frame. Of course, there is still a. If the line is idle, it goes to step 1.
the possibility of collision if two stations try to send at the beginning of the same time slot. However, the b. If the line is busy, it acts as though a collision has occurred and uses the
vulnerable time is now reduced to one-half, equal to Tfr. The following Figure shows the situation. backoff procedure.

CSMA/CD:
The CSMA method does not specify the procedure following a collision. Carrier sense multiple access
with collision detection (CSMA/CD) augments the algorithm to handle the collision.
In this method, a station monitors the medium after it sends a frame to see if the transmission was
successful. If so, the station is finished. If, however, there is a collision, the frame is sent again.

Fig:Vulnerable time for slotted ALOHA protocol

Throughput
It can be proven that the average number of successful transmissions for slotted ALOHA is S = G × e−G.
The maximum throughput Smax is 0.368, when G = 1. In other words, if one frame is generated during
one frame transmission time, then 36.8 percent of these frames reach their destination successfully. We
expect G = 1 to produce maximum throughput because the vulnerable time is equal to the frame
Fig: Collision of bits in CSMA/CD

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To better understand CSMA/CD, let us look at the first bits transmitted by the two stations is sensed, a distant station may have already started transmitting. The distant station’s signal has not
involved in the collision. Although each station continues to send bits in the frame until it detects the yet reached this station. The IFS time allows the front of the transmitted signal by the distant station
collision, we show what happens as the first bits collide. In the Figure , stations A and C are involved in to reach this station. After waiting an IFS time, if the channel is still idle, the station can send, but it
the collision still needs to wait a time equal to the contention window (described next). The IFS variable can also
be used to prioritize stations or frame types. For example, a station that is assigned a shorter IFS has
At time t1, station A has executed its persistence procedure and starts sending the bits of its a higher priority.
frame. At time t2, station C has not yet sensed the first bit sent by A. Station C executes its persistence  Contention Window. The contention window is an amount of time divided into slots. A station that
procedure and starts sending the bits in its frame, which propagate both to the left and to the right. The is ready to send chooses a random number of slots as its wait time. The number of slots in the
collision occurs sometime after time t2. Station C detects a collision at time t3 when it receives the first window changes according to the binary exponential backoff strategy. This means that it is set to one
bit of A’s frame. Station C immediately (or after a short time, but we assume immediately) aborts slot the first time and then doubles each time the station cannot detect an idle channel after the IFS
transmission. Station A detects collision at time t4 when it receives the first bit of C’s frame; it also
time. This is very similar to the p-persistent method except that a random outcome defines the
immediately aborts transmission. Looking at the figure, we see that A transmits for the duration t4 − t1; C number of slots taken by the waiting station. One interesting point about the contention window is
transmits for the duration t3 − t2. that the station needs to sense the channel after each time slot. However, if the station finds the
Minimum Frame Size: For CSMA/CD to work, we need a restriction on the frame size. Before sending channel busy, it does not restart the process; it just stops the timer and restarts it when the channel is
the last bit of the frame, the sending station must detect a collision, if any, and abort the transmission. sensed as idle. This gives priority to the station with the longest waiting time.
This is so because the station, once the entire frame is sent, does not keep a copy of the frame and does  Acknowledgment. With all these precautions, there still may be a collision resulting in destroyed
not monitor the line for collision detection. Therefore, the frame transmission time Tfr must be at least data. In addition, the data may be corrupted during the transmission. The positive acknowledgment
two times the maximum propagation time Tp. To understand the reason, let us think about the worst-case and the time-out timer can help guarantee that the receiver has received the frame.
scenario. If the two stations involved in a collision are the maximum distance apart, the signal from the
first takes time Tp to reach the second, and the effect of the collision takes another time TP to reach the
first. So the requirement is that the first station must still be transmitting after 2Tp.

Fig: Procedure for CSMA/CD

CSMA/CA:
Carrier sense multiple access with collision avoidance (CSMA/CA) was invented for wireless networks.
Collisions are avoided through the use of CSMA/CA’s three strategies: the interframe space, the
contention window, and acknowledgments.
 Interframe Space (IFS). First, collisions are avoided by deferring transmission even if the channel is
found idle. When an idle channel is found, the station does not send immediately. It waits for a
period of time called the interframe space or IFS. Even though the channel may appear idle when it

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 CONNECTING DEVICES Transparent Switches

connecting devices are used to connect hosts together to make a network or to connect networks together A transparent switch is a switch in which the stations are completely unaware of the switch’s existence.
to make an internet. Connecting devices can operate in different layers of the Internet model. Three kinds If a switch is added or deleted from the system, reconfiguration of the stations is unnecessary. According
of connecting devices: hubs, link-layer switches, and routers. Hubs today operate in the first layer of the to the IEEE 802.1d specification, a system equipped with transparent switches must meet three criteria:
Internet model. Link-layer switches operate in the first two layers. Routers operate in the first three layers.  Frames must be forwarded from one station to another.
 The forwarding table is automatically made by learning frame movements in the network.
 Loops in the system must be prevented.
Forwarding
A transparent switch must correctly forward the frames, as discussed in the previous section.
Learning
The earliest switches had switching tables that were static. The system administrator would manually
enter each table entry during switch setup. Although the process was simple, it was not practical. If a
station was added or deleted, the table had to be modified manually. The same was true if a station’s
Hubs MAC address changed, which is not a rare event. For example, putting in a new network card means a
A hub is a device that operates only in the physical layer. A repeater receives a signal and, before it new MAC address. A better solution to the static table is a dynamic table that maps addresses to ports
becomes too weak or corrupted, regenerates and retimes the original bit pattern. The repeater then sends (interfaces) automatically. To make a table dynamic, we need a switch that gradually learns from the
the refreshed signal. In a star topology, a repeater is a multiport device, often called a hub, that can be frames’ movements. To do this, the switch inspects both the destination and the source addresses in each
used to serve as the connecting point and at the same time function as a repeater. Figure shows that when frame that passes through the switch. The destination address is used for the forwarding decision (table
a packet sends from station A to station B arrives at the hub, the signal representing the frame is lookup); the source address is used for adding entries to the table and for updating purposes. Let us
regenerated to remove any possible corrupting noise, but the hub forwards the packet from all outgoing elaborate on this process using Figure.
ports except the one from which the signal was received. In other words, the frame is broadcast. All 1. When station A sends a frame to station D, the switch does not have an entry for either D or A. The
stations in the LAN receive the frame, but only station B keeps it. The rest of the stations discard it. frame goes out from all three ports; the frame floods the network. However, by looking at the source
Figure shows the role of a repeater or a hub in a switched LAN. The figure definitely shows that a hub address, the switch learns that station A must be connected to port 1. This means that frames destined
does not have a filtering capability. It does not have the intelligence to find from which port the frame for A, in the future, must be sent out through port 1. The switch adds this entry to its table. The table
should be sent out. A hub or a repeater is a physical-layer device. They do not have a link-layer address has its first entry now.
and they do not check the link-layer address of the received frame. They just regenerate the corrupted bits 2. When station D sends a frame to station B, the switch has no entry for B, so it floods the network
and send them out from every port. again. However, it adds one more entry to the table related to station D.
3. The learning process continues until the table has information about every port. However, note that the
learning process may take a long time. For example, if a station does not send out a frame (a rare
situation), the station will never have an entry in the table.

Link-Layer Switches
A link-layer switch (or switch) operates in both the physical and the data-link layers. As a link-layer
device, the link-layer switch can check the MAC addresses (source and destination) contained in the
frame.

Filtering
The difference in functionality is between a link-layer switch and a hub is a link-layer switch has filtering
capability. It can check the destination address of a frame and can decide from which outgoing port the
frame should be sent. For example in Figure, we have a LAN with four stations that are connected to a
link-layer switch. If a frame destined for station 71:2B:13:45:61:42 arrives at port 1, the link-layer switch
consults its table to find the departing port. According to its table, frames for 71:2B:13:45:61:42 should
be sent out only through port 2; therefore, there is no need for forwarding the frame through other ports.

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Loop Problem: The Network Layer (Unit-III-1)
Transparent switches work fine as long as there are no redundant switches in the system. Systems
administrators, however, like to have redundant switches (more than one switch between a pair of LANs)  Network Layer Design Issues:
to make the system more reliable. If a switch fails, another switch takes over until the failed one is While designing the network layer we have to consider some of the design issues. These issues include
repaired or replaced. Redundancy can create loops in the system, which is very undesirable. Loops can be the service provided to the transport layer and the internal design of the subnet.
created only when two or more broadcasting LANs (those using hubs, for example) are connected by
more than one switch. Store-and-Forward Packet Switching:
Figure shows a very simple example of a loop created in a system with two LANs connected by two The major components of the system are the carrier's equipment, shown inside the shaded oval, and the
switches. customers' equipment, shown outside the oval. Host H1 is directly connected to one of the carrier's routers,
1. Station A sends a frame to station D. The tables of both switches are empty. Both forward the frame A, by a leased line. In contrast, H2 is on a LAN with a router, F, owned and operated by the customer. This
and update their tables based on the source address A. router also has a leased line to the carrier's equipment. We have shown F as being outside the oval because it
2. Now there are two copies of the frame on LAN 2. The copy sent out by the left switch is received by does not belong to the carrier, but in terms of construction, software, and protocols, it is probably no
the right switch, which does not have any information about the destination address D; it forwards the different from the carrier's routers.
frame. The copy sent out by the right switch is received by the left switch and is sent out for lack of Figure 5-1. The environment of the network layer protocols.
information about D.
3. Now there are two copies of the frame on LAN 1. Step 2 is repeated, and both copies are sent to LAN2.
4. The process continues on and on. Note that switches are also repeaters and regenerate frames. So in
each iteration, there are newly generated fresh copies of the frames.

Spanning Tree Algorithm

To solve the looping problem, the IEEE specification requires that switches use the spanning tree
algorithm to create a loopless topology. In graph theory, a spanning tree is a graph in which there is no
loop.

Here, a host with a packet to send transmits it to the nearest router, either on its own LAN or over a point-to-
point link to the carrier. The packet is stored there until it has fully arrived so the checksum can be verified.
Then it is forwarded to the next router along the path until it reaches the destination host, where it is
delivered. This mechanism is store-and-forward packet switching.
Services Provided to the Transport Layer:
The network layer provides services to the transport layer at the network layer/transport layer interface.
The network layer services have been designed with the following goals in mind.
Routers 1. The services should be independent of the router technology.
A router is a three-layer device; it operates in the physical, data-link, and network layers. As a physical- 2. The transport layer should be shielded from the number, type, and topology of the routers present.
layer device, it regenerates the signal it receives. As a link-layer device, the router checks the physical 3. The network addresses made available to the transport layer should use a uniform numbering plan,
addresses (source and destination) contained in the packet. As a network-layer device, a router checks the even across LANs and WANs.
network-layer addresses. A router can connect networks. In other words, a router is an internetworking
device; it connects independent networks to form an internetwork. According to this definition, two Implementation of Connectionless Service:
networks connected by a router become an internetwork or an internet. There are three major differences Figure 5-2. Routing within a datagram subnet.
between a router and a repeater or a switch.
1. A router has a physical and logical (IP) address for each of its interfaces.
2. A router acts only on those packets in which the link-layer destination address matches the address of
the interface at which the packet arrives.
3. A router changes the link-layer address of the packet (both source and destination) when it forwards the
packet.

1. What is the reminder obtained by dividing x7 +x5 +1 by the generator x3 +1.

2. Draw and explain HDLC frame format.
3. Explain in detail about elementary DLL protocols
4. Given 1101011011 data frame and generator polynomial G(x)= x4 + x+ 1
5. Explain CSMA/CD protocol.
6. Consider the delay of pure ALOHA Vs Slotted ALOHA at low load. Which one is less? Explain
your answer?

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Computer Networks Computer Networks

In connectionless service, packets are injected into the subnet individually and routed independently of each
other. No advance setup is needed. In this context, the packets are frequently called datagrams and the
subnet is called a datagram subnet.
Every router has an internal table telling it where to send packets for each possible destination. Each table
entry is a pair consisting of a destination and the outgoing line to use for that destination. Only directly-
connected lines can be used.
As they arrived at A, packets 1, 2, and 3 were stored briefly (to verify their checksums). Then each was
forwarded to C according to A's table. Packet 1 was then forwarded to E and then to F. When it got to F, it
was encapsulated in a data link layer frame and sent to H2 over the LAN. Packets 2 and 3 follow the same
route However, something different happened to packet 4. When it got to A it was sent to router B, even
though it is also destined for F. The algorithm that manages the tables and makes the routing decisions is
called the routing algorithm.
Implementation of Connection-Oriented Service:
If connection-oriented service is used, a path from the source router to the destination router must be
established before any data packets can be sent. This connection is called a VC (virtual circuit), in analogy
with the physical circuits set up by the telephone system, and the subnet is called a virtual-circuit subnet.
As an example, consider the situation of Fig. 5-3. Here, host H1 has established connection 1 with host
H2. It is remembered as the first entry in each of the routing tables. The first line of A's table says that if a
 Routing Algorithms:
packet bearing connection identifier 1 comes in from H1, it is to be sent to router C and given connection
Characteristics of routing algorithms:
identifier 1. Similarly, the first entry at C routes the packet to E, also with connection identifier 1.
 Correctness: it could able to deliver packets from source to destination without failure or without
Figure 5-3. Routing within a virtual-circuit subnet.
other nodes.
 Simplicity: the function should be simple in operation.
 Robustness: if the network is delivering packets via some route, if any failures or overloads occur, the
function should react to such contingencies without the loss of packets or the breaking of virtual
circuits.
 Stability: The outing function should react to contingencies slowly that are neither fast nor too slow.
Why means, for example, if the network may react to congestion in one area by shifting most of load
to second area. Now the second area is overloaded and the first is under-utilized, causing a second
shift. During these shifts, packets may travel in loops through the network.
 Fairness and Optimality: some performance criteria may give higher priority to the exchange of
packets between neighbor stations compared to an exchange between distant stations. This policy
may maximize average throughput but will appear unfair to the station that primarily needs to
communicate with distant stations.
 Efficiency: The efficiency routing function involves the processing overhead at each node and often a
transmission overhead.
Comparison of Virtual-Circuit and Datagram Subnets:
Classification of routing algorithms:
Routing algorithms can be grouped into two major classes:
 Nonadaptive algorithms do not base their routing decisions on measurements or estimates of the
current traffic and topology. Instead, the choice of the route to use to get from I to J (for all I and J)
is computed in advance, off-line, and downloaded to the routers when the network is booted. This
procedure is sometimes called “static routing”.
 Adaptive algorithms on the contrary are dynamic and online. They collect their information about
the state of the network and make routing decisions based on the latest information, for example,
Distance vector routing and link state routing.

The Optimality Principle:

It states that one can make a general statement about optimal routes without regard to network topology
or traffic. This statement is known as the optimality principle. It states that if router J is on the optimal path
from router I to router K, then the optimal path from J to K also falls along the same route. To see this, call
the part of the route from I to J r1 and the rest of the route r2. If a route better than r2 existed from J to K, it
could be concatenated with r1 to improve the route from I to K, contradicting our statement that r1r2 is
optimal.

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Computer Networks Computer Networks

As the result of optimality principle the set of optimal routes from all sources to a given destination form  Initially, all labels are tentative. When it is discovered that a label represents the shortest possible path
a tree rooted at the destination. Such a tree is called a sink tree and is illustrated in Fig. 5-6, where the from the source to that node, it is made permanent and never changed thereafter.
distance metric is the number of hops. The goal of all routing algorithms is to discover and use the sink trees To illustrate how the labeling algorithm works, look at the weighted, undirected graph of Fig. 5-7(a),
for all routers. Since a sink tree is indeed a tree, it does not contain any loops, so each packet will be where the weights represent, for example, distance. We want to find the shortest path from A to D. We start
delivered within a finite and bounded number of hops. out by marking node A as permanent, indicated by a filled-in circle. Then we examine, in turn, each of the
nodes adjacent to A (the working node), relabeling each one with the distance to A. Whenever a node is
Figure 5-6. (a) A subnet. (b) A sink tree for router B. relabeled, we also label it with the node from which the probe was made so that we can reconstruct the final
path later. Having examined each of the nodes adjacent to A, we examine all the tentatively labeled nodes in
the whole graph and make the one with the smallest label permanent, as shown in Fig. 5-7(b). This one
becomes the new working node.
We now start at B and examine all nodes adjacent to it. If the sum of the label on B and the distance from
B to the node being considered is less than the label on that node, we have a shorter path, so the node is
relabeled.
After all the nodes adjacent to the working node have been inspected and the tentative labels changed if
possible, the entire graph is searched for the tentatively-labeled node with the smallest value. This node is
made permanent and becomes the working node for the next round. Figure 5-7 shows the first five steps of
the algorithm.
Flooding:
Flooding is a static algorithm, in which in which “every incoming packet is sent out on every outgoing
Shortest Path Routing: line except the one it arrived on”. Flooding obviously generates vast numbers of duplicate packets, in fact,
The idea behind routing algorithms is to build a graph of the subnet, with each node of the graph an infinite number unless some measures are taken to damp (Discourage) the process.
representing a router and each arc of the graph representing a communication line. To choose a route  One such measure is to have a hop counter contained in the header of each packet, which is
between a given pair of routers, the algorithm just finds the shortest path between them on the graph. decremented at each hop, with the packet being discarded when the counter reaches zero.
The concept of a shortest path deserves some explanation. One way of measuring path length is the o Ideally, the hop counter should be initialized to the length of the path from source to
number of hops. Using this metric, the paths ABC and ABE in Fig. 5-7 are equally long. Another metric is the destination. If the sender does not know how long the path is, it can initialize the counter to the
geographic distance in kilometers, in which case ABC is clearly much longer than ABE. Another metric is worst case, namely, the full diameter of the subnet.
may be time delay etc., like so many metrics can be used for shortest path routing.  An alternative technique for damming the flood is to keep track of which packets have been flooded,
Figure 5-7. The first five steps used in computing the shortest path to avoid sending them out a second time. Achieve this goal is to have the source router put a sequence
from A to D. The arrows indicate the working node. number in each packet it receives from its hosts. Each router then needs a list per source router telling
which sequence numbers originating at that source have already been seen. If an incoming packet is
on the list, it is not flooded.
Applications and Advantages:
 Flooding is very effective routing approach, when, the information in the routing tables is not
available, such as during system start up.
 Flooding is also effective when the source needs to send a packet to all hosts connected to the network
for example in military applications.
 In distributed data base applications, it is sometimes necessary to update the entire database
concurrently; in such cases flooding is used.
 Flooding always chooses the shortest path, because it chooses every possible path in parallel.
 In wireless networks, all messages transmitted by a station can be received by all other stations within
its radio range, which is, in fact, flooding, and some algorithms utilize this property.

Distance Vector Routing:

Modern computer networks generally use dynamic routing algorithms rather than the static ones because
static algorithms do not take the current network load into account.
Distance vector routing algorithms operate by having each router maintain a table (i.e, a vector) giving
the best known distance to each destination and which line to use to get there. These tables are updated by
exchanging information with the neighbors.
Several algorithms for computing the shortest path between two nodes of a graph are known. This one is The distance vector routing algorithm is also called by other names as, distributed Bellman-Ford routing
due to Dijkstra (1959). Each node is labeled (in parentheses) with its distance from the source node along algorithm and the Ford-Fulkerson algorithm.
the best known path. Initially, no paths are known, so all nodes are labeled with infinity. As the algorithm
This can be explained in Fig. 5-9. Part (a) shows a subnet. The first four columns of part (b) show the
proceeds and paths are found, the labels may change, reflecting better paths. A label may be either tentative
delay vectors received from the neighbors of router J. Suppose that J has measured or estimated its delay to
or permanent.
its neighbors, A, I, H, and K as 8, 10, 12, and 6 msec, respectively.
Figure 5-9. (a) A subnet. (b) Input from A, I, H, K, and the new routing table for J.

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Computer Networks Computer Networks

Link State Routing:

Distance vector routing was replaced by link state routing. Two primary problems caused its demise.
First, since the delay metric was queue length, it did not take line bandwidth into account when
choosing routes.
 Second problem is the count-to-infinity problem.
The idea behind link state routing is simple and can be stated as five parts. Each router must do the
following:
1) Discover its neighbors and learn their network addresses.
2) Measure the delay or cost to each of its neighbors.
3) Construct a packet telling all it has just learned.
4) Send this packet to all other routers.
5) Compute the shortest path to every other router.
Learning about the Neighbors:
When a router is booted, its first task is to learn who its neighbors are. It accomplishes this goal by
sending a special HELLO packet on each point-to-point line.
Measuring Line Cost:
The link state routing algorithm requires each router to know, or at least have a reasonable estimate of,
the delay to each of its neighbors. The most direct way to determine this delay is to send over the line a
Consider how J computes its new route to router G. It knows that it can get to A in 8 msec, and A claims special ECHO packet that the other side is required to send back immediately. By measuring the round-trip
to be able to get to G in 18 msec, so J knows it can count on a delay of 26 msec to G if it forwards packets time and dividing it by two, the sending router can get a reasonable estimate of the delay.
bound for G to A. Similarly, it computes the delay to G via I, H, and K as 41 (31 + 10), 18 (6 + 12), and 37
(31 + 6) msec, respectively. The best of these values is 18, so it makes an entry in its routing table that the Building Link State Packets:
delay to G is 18 msec and that the route to use is via ‘H’. Once the information needed for the exchange has been collected, the next step is for each router to build
a packet containing all the data. The packet starts with the identity of the sender, followed by a sequence
The Count-to-Infinity Problem: number and age, and a list of neighbors. For each neighbor, the delay to that neighbor is given.
consider the five-node (linear) subnet of Fig. 5-10, where the delay metric is the number of hops. An example subnet is given in Fig. 5-13(a) with delays shown as labels on the lines. The corresponding link
Suppose A is down initially and all the other routers know this. In other words, they have all recorded the state packets for all six routers are shown in Fig. 5-13(b).
delay to A as infinity. Figure 5-13. (a) A subnet. (b) The link state packets for this subnet.
Figure 5-10. The count-to-infinity problem.

Distributing the Link State Packets:

The next step after building link state packets is to distribute them across the network. Flooding is used as
the basic algorithm for distributing link state packets. To avoid flooding the same packet, each new packet is
When A comes up, the other routers learn about it via the vector exchanges. At the time of the first given a sequence number. When a packet arrives at a router for flooding then it checks whether this packet is
exchange, B learns that its left neighbor has zero delay to A. B now makes an entry in its routing table that A already seen by using a pair (source router, sequence number) that each router have.
is one hop away to the left. All the other routers still think that A is down. At this point, the routing table When a new link state packet comes in, it is checked against the list of packets already seen. If it is new,
entries for A are as shown in the second row of Fig. 5-10(a). Clearly, the good news is spreading at the rate it is forwarded on all lines except the one it arrived on. If it is a duplicate, it is discarded. If a packet with a
of one hop per exchange, then, C,D,E routers are updated as 2,3,4 respectively. sequence number lower than the highest one seen so far ever arrives, it is rejected as being obsolete ( outdated)
Now let us consider the situation of Fig. 5-10(b), in which all the lines and routers are initially up. since the router has more recent data. The age of each packet decrement it once per second. When the age
Routers B, C, D, and E have distances to A of 1, 2, 3, and 4, respectively. Suddenly A goes down, or hits zero, the information from that router is discarded.
alternatively, the line between A and B is cut, which is effectively the same thing from B's point of view. The data structure used by router ‘B’ for the subnet is depicted in Fig. 5- 14. Each row here corresponds
 When A goes down, line between A and B is out. to a recently-arrived, but as yet not fully-processed, link state packet. The table records where the packet
 B does not hear anything from A. originated, its sequence number and age, and the data. In addition, there are send and acknowledgement
 C informs B, I Have Path to A, of length 2. flags for each of B's three lines (to A, C, and F, respectively). The send flags mean that the packet must be
 If metric used is Time Delay, there is no well-defined upper bound. So, high value is needed to prevent sent on the indicated line. The acknowledgement flags mean that it must be acknowledged there.
a path with a long delay from being treated as down. This problem is known as count-to-infinity.

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Figure 5-14. The packet buffer for router B in Fig. 5-13.

In above table, the link state packet from A arrives directly, so it must be sent to C and F and acknowledged
to A, as indicated by the flag bits. Similarly, the packet from F has to be forwarded to A and C and
acknowledged to F.

Computing the New Routes:

Once a router has accumulated a full set of link state packets, router builds the entire subnet graph
because every link is represented. Every link is, in fact, represented twice, once for each direction. The two
values can be averaged or used separately.
Figure 5-15 gives a quantitative example of routing in a two-level hierarchy with five regions. The full
Now Dijkstra's algorithm can be run locally to construct the shortest path to all possible destinations.
routing table for router 1A has 17 entries, as shown in Fig. 5-15(b). When routing is done hierarchically, as in
The results of this algorithm can be installed in the routing tables, and normal operation resumed
Fig. 5-15(c), there are entries for all the local routers as before, but all other regions have been condensed
(continued).
into a single router, so all traffic for region 2 goes via the “1B -2A” line, but the rest of the remote traffic
goes via the “1C -3B” line. Hierarchical routing has reduced the table from 17 to 7 entries. “As the ratio of
Hierarchical Routing:
the number of regions to the number of routers per region grows, the savings in table space increase”.
Hierarchical routing is an algorithm for routing packets hierarchically. It is used due to the following
reasons.
Broadcast Routing:
 As networks grow in size, the router routing tables grow proportionally.
Sending a packet to all destinations simultaneously is called broadcasting; the algorithms used for
 Router memory consumed by ever-increasing tables. broadcasting are called broadcast routing. Various methods have been proposed for doing it. They are,
 More CPU time is needed to scan them and more bandwidth is needed to send status reports about 1) Distinct point-to-point routing
them. 2) Flooding
 At a certain point the network may grow to the point where it is no longer feasible for every router to 3) Multi-destination routing
have an entry for every other router. 4) Use of spanning tree
When hierarchical routing is used, the routers are divided into regions, with each router knowing all the 5) Reverse path forwarding
details about how to route packets to destinations within its own region, but knowing nothing about the
internal structure of other regions. When different networks are interconnected, then each one can be treated Distinct point-to-point routing:
as a separate region in order to free the routers in one network from having to know the topological structure This is the simplest method for broadcasting.in this method ‘sender simply sends a distinct packet to each
of the other ones. destination or to all the nodes in the network’. Thus it takes no special features of the subnet. This method is
Figure 5-15. Hierarchical routing. not desirable due to two reasons. First, Wasteful of bandwidth, Second it requires the source to have a
complete list of all destinations.

Flooding:
Flooding is another obvious candidate. This algorithm sends a packet on every outgoing line except the
line on which it arrived. The problem with flooding as a broadcast technique is the same problem it has as a
point-to-point routing algorithm: “it generates too many packets and consumes too much bandwidth”.
Multi-destination routing:
A third algorithm is multi-destination routing. If this method is used, each packet contains either a list
of destinations or a bit map indicating the desired destinations. When a packet arrives at a router, the router
checks all the destinations to determine the set of output lines that will be needed. The router generates a new
copy of the packet for each output line to be used and includes in each packet only those destinations that are
to use the line. In effect, the destination set is partitioned among the output lines. After a sufficient number of
hops, each packet will carry only one destination and can be treated as a normal packet.
Use of spanning tree:
A spanning tree is a subset of the subnet that includes all the routers but contains no loops.

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If each router knows which of its lines belong to the spanning tree, it can copy an incoming broadcast Figure 5-17. (a) A network. (b) A spanning tree for the leftmost
packet onto all the spanning tree lines except the one it arrived on. This method makes excellent use of router. (c) A multicast tree for group 1. (d) A multicast tree for group 2
bandwidth, generating the absolute minimum number of packets necessary to do the job.

Reverse path forwarding:

An example of reverse path forwarding is shown in Fig. 5-16. Part (a) shows a subnet, part (b) shows a
sink tree for router I of that subnet, and part (c) shows how the reverse path algorithm works. On the first
hop, I send packets to F, H, J, and N, as indicated by the second row of the tree. Each of these packets arrives
on the preferred path to I (assuming that the preferred path falls along the sink tree) and is so indicated by a
circle around the letter. On the second hop, eight packets are generated, two by each of the routers that
received a packet on the first hop. As it turns out, all eight of these arrive at previously unvisited routers, and
five of these arrive along the preferred line. Of the six packets generated on the third hop, only three arrive
on the preferred path (at C, E, and K); the others are duplicates. After five hops and 24 packets, the
broadcasting terminates, compared with four hops and 14 packets had the sink tree been followed exactly.
Figure 5-16. Reverse path forwarding. (a) A subnet. (b) A sink tree.
(c) The tree built by reverse path forwarding.

When a process sends a multicast packet to a group, the first router examines its spanning tree and prunes
it, removing all lines that do not lead to hosts that are members of the group. In our example, Fig. 5-17(c)
shows the pruned spanning tree for group 1. Similarly, Fig. 5- 17(d) shows the pruned spanning tree for
group 2. Multicast packets are forwarded only along the appropriate spanning tree.
Advantages: Various ways of pruning the spanning tree are possible. The simplest one can be used if link state routing
 The reverse path forwarding is that it is both reasonably efficient and easy to implement. is used and each router is aware of the complete topology, including which hosts belong to which groups.
 It does not require routers to know about spanning trees. Then the spanning tree can be pruned, starting at the end of each path, working toward the root, and
 It does not have the overhead of a destination list or bit map in each broadcast packet as does multi- removing all routers that do not belong to the group in question.
destination addressing.
 It does not require any special mechanism to stop the process. Routing for Mobile Hosts:
Figure 5-18. A WAN to which LANs, MANs, and wireless cells are attached.
Multicast Routing:
For some applications such as tele conferencing, a source may want to send packets to multiple
destinations simultaneously or a group of processes implementing a distributed database systems. It is
frequently necessary for one process to send a message to all the other members of the group.
 If the group is small, it can just send each other member a point-to-point message.
 If the group is large, this strategy is expensive.
Thus, we need a way to send messages to well defined groups that are numerically large in size but small
compared to the network as a whole.
Sending a message to such a group is called multicasting, and its routing algorithm is called multicast
routing. In the model of Fig. 5-18, the world is divided up (geographically) into small units. Let us call them
Multicasting requires group management. areas, where an area is typically a LAN or wireless cell. Each area has one or more foreign agents, which
 To create and destroy groups and are processes that keep track of all mobile hosts visiting the area. In addition, each area has a home agent,
 To allow processes to join and leave groups. which keeps track of hosts whose home is in the area, but who are currently visiting another area.
The routing algorithm does not know how these tasks are accomplished but when a process joins a group; When a new host enters an area, either by connecting to it (e.g., plugging into the LAN) or just wandering
it informs its host of this fact. It is important that routers know which of their hosts belong to which groups. (travelling) into the cell, his computer must register itself with the foreign agent there. The registration
Either host must inform their routers about changes in group membership, or routers must query their hosts procedure typically works like this:
periodically. Either way, routers learn about which of their hosts are in which groups. Routers tell their 1) Periodically, each foreign agent broadcasts a packet announcing its existence and address. A newly-
neighbors, so the information propagates through the subnet. arrived mobile host may wait for one of these messages, but if none arrives quickly enough, the
To do multicast routing, each router computes a spanning tree covering all other routers. For example, in mobile host can broadcast a packet saying: Are there any foreign agents around?
Fig. 5-17(a) we have two groups, 1 and 2. Some routers are attached to hosts that belong to one or both of 2) The mobile host registers with the foreign agent, giving its home address, current data link layer
these groups, as indicated in the figure. A spanning tree for the leftmost router is shown in Fig. 5-17(b). address, and some security information.

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3) The foreign agent contacts the mobile host's home agent and says: One of your hosts is over here. The (c) After C, F, and G have received A's broadcast.
message from the foreign agent to the home agent contains the foreign agent's network address. It (d) After E, H, and I have received A's broadcast.
also includes the security information to convince the home agent that the mobile host is really there. The shaded nodes are new recipients. The arrows show the possible reverse routes.
4) The home agent examines the security information, which contains a timestamp, to prove that it was
generated within the past few seconds. If it is happy, it tells the foreign agent to proceed.
5) When the foreign agent gets the acknowledgement from the home agent, it makes an entry in its tables
and informs the mobile host that it is now registered.
Figure 5-19. Packet routing for mobile hosts.

To locate I, A constructs a special ROUTE REQUEST packet and broadcasts it. The packet reaches B and
D, as illustrated in Fig. 5-20(a). In fact, the reason B and D are connected to A in the graph is that they can
receive communication from A. F, for example, is not shown with an arc to A because it cannot receive A's
radio signal. Thus, F is not connected to A.
Route Maintenance:
Ideally, when a host leaves an area, that, too, should be announced to allow deregistration, but many users Because nodes can move or be switched off, the topology can change spontaneously. Periodically, each
abruptly turn off their computers when done. node broadcasts a Hello message. Each of its neighbors is expected to respond to it. If no response is
When a packet is sent to a mobile host, it is routed to the host's home LAN because that is what the forthcoming, the broadcaster knows that that neighbor has moved out of range and is no longer connected to
address says should be done, as illustrated in step 1 of Fig. 5-19. it. Similarly, if it tries to send a packet to a neighbor that does not respond, it learns that the neighbor is no
The home agent then does two things. longer available.
 First, it encapsulates the packet in the payload field of an outer packet and sends the latter to the This information is used to purge (remove) routes that no longer work. For each possible destination, each
foreign agent (step 2 in Fig. 5-19). This mechanism is called tunneling; node, N, keeps track of its neighbors that have fed it a packet for that destination during the last ΔT seconds.
 Second, the home agent tells the sender to henceforth send packets to the mobile host by These are called N's active neighbors for that destination.
encapsulating them in the payload of packets explicitly addressed to the foreign agent instead of just
sending them to the mobile host's home address (step 3). Subsequent packets can now be routed  Congestion Control Algorithms:
directly to the host via the foreign agent (step 4), bypassing the home location entirely. When too many packets are present in (a part of) the subnet, performance degrades. This situation is
called congestion.
Routing in Ad Hoc Networks:
Here routing can be done when the hosts are mobile and also routers themselves are mobile. Among the Figure depicts the onset of congestion. When the number of packets hosts send into the network is well
possibilities are: within its carrying capacity, the number delivered is proportional to the number sent. If twice as many are
1. Military vehicles on a battlefield with no existing infrastructure. sent, twice as many are delivered. However, as the offered load approaches the carrying capacity, bursts of
2. A fleet of ships at sea. traffic occasionally fill up the buffers inside routers and some packets are lost. These lost packets consume
3. Emergency workers at an earthquake that destroyed the infrastructure. some of the capacity, so the number of delivered packets falls below the ideal curve. The network is now
4. A gathering of people with notebook computers in an area lacking 802.11. congested.
In all these cases, and others, each node consists of a router and a host, usually on the same computer.
Networks of nodes that just happen to be near each other are called ad hoc networks or MANETs (Mobile
Ad hoc NETworks).
A variety of routing algorithms for ad hoc networks have been proposed. One of the more interesting ones
is the AODV (Ad hoc On-demand Distance Vector) routing algorithm. It is a distant relative of the
Bellman-Ford distance vector algorithm but adapted to work in a mobile environment and takes into account
the limited bandwidth and low battery life found in this environment. If all of a sudden, streams of packets begin arriving on three or four input lines and all need the same output
It is an on-demand algorithm, that is, it determines a route to some destination only when somebody line, a queue will build up. If there is insufficient memory to hold all of them, packets will be lost. Adding
wants to send a packet to that destination. more memory may help up to a point that if routers have an infinite amount of memory, congestion gets
worse, not better. This is because by the time packets get to the front of the queue, they have already timed
Route Discovery: out (repeatedly) and duplicates have been sent. This makes matters worse, not better—it leads to congestion
Consider the ad hoc network of Fig. 5-20, in which a process at node A wants to send a packet to node I. collapse.
The AODV algorithm maintains a table at each node, keyed (enetered) by destination, giving information
about that destination, including which neighbor to send packets to in order to reach the destination. Suppose Low-bandwidth links or routers that process packets more slowly than the line rate can also become
that A looks in its table and does not find an entry for I. It now has to discover a route to I. This property of congested. In this case, the situation can be improved by directing some of the traffic away from the
discovering routes only when they are needed is what makes this algorithm ''on demand.'' bottleneck to other parts of the network. Eventually, however, all regions of the network will be congested.
Figure 5-20. (a) Range of A's broadcast. (b) After B and D have received A's broadcast

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Approaches to Congestion Control By analogy, in the telephone system, when a switch gets overloaded it practices admission control by not
giving dial tones. The task is straightforward in the telephone network because of the fixed bandwidth of
calls (64 kbps for uncompressed audio). However, virtual circuits in computer networks come in all shapes
and sizes. Thus, the circuit must come with some characterization of its traffic if we are to apply admission
control.

The most basic way to avoid congestion is to build a network that is well matched to the traffic that it carries.
If there is a low-bandwidth link on the path along which most traffic is directed, congestion is likely.
Sometimes resources can be added dynamically when there is serious congestion, for example, turning on
spare routers or enabling lines that are normally used only as backups (to make the system fault tolerant) or
purchasing bandwidth on the open market. This is called provisioning and happens on a time scale of
months, driven by long-term traffic trends.
Some local radio stations have helicopters flying around their cities to report on road congestion to make it
possible for their mobile listeners to route their packets (cars) around hotspots. This is called traffic-aware
routing. Splitting traffic across multiple paths is also helpful.
For example, consider the network illustrated in Fig(a). in which two routers are congested, as indicated.
Sometimes it is not possible to increase capacity. The only way then to beat back the congestion is to Suppose that a host attached to router A wants to set up a connection to a host attached to router B. Normally,
decrease the load. In a virtual-circuit network, new connections can be refused if they would cause the this connection would pass through one of the congested routers. To avoid this situation, we can redraw the
network to become congested. This is called admission control. network as shown in Fig(b). omitting the congested routers and all of their lines. The dashed line shows a
possible route for the virtual circuit that avoids the congested routers.
At a finer granularity, when congestion is imminent the network can deliver feedback to the sources whose
traffic flows are responsible for the problem. The network can request these sources to throttle their traffic, Traffic Throttling
or it can slow down the traffic itself. Two difficulties with this approach are how to identify the onset of When congestion is imminent, it must tell the senders to throttle back their transmissions and slow down.
congestion, and how to inform the source that needs to slow down. The term congestion avoidance is sometimes used to contrast this operating point with the one in which the
To tackle the first issue, routers can monitor the average load, queueing delay, or packet loss. In all cases, network has become (overly) congested.
rising numbers indicate growing congestion. To tackle the second issue, routers must participate in a
feedback loop with the sources. Choke Packets
The most direct way to notify a sender of congestion is to tell it directly. In this approach, the router selects a
Finally, when all else fails, the network is forced to discard packets that it cannot deliver. The general name congested packet and sends a choke packet back to the source host, giving it the destination found in the
for this is load shedding. A good policy for choosing which packets to discard can help to prevent packet. The original packet may be tagged (a header bit is turned on) so that it will not generate any
congestion collapse. more choke packets farther along the path and then forwarded in the usual way. To avoid increasing load on
the network during a time of congestion, the router may only send choke packets at a low rate.
Traffic-Aware Routing
The goal in taking load into account when computing routes is to shift traffic away from hotspots that will be When the source host gets the choke packet, it is required to reduce the traffic sent to the specified
the first places in the network to experience congestion. The most direct way to do this is to set the link destination, for example, by 50%. In a datagram network, choke packets to be sent to fast senders, because
weight to be a function of the (fixed) link bandwidth and propagation delay plus the (variable) measured load they will have the most packets in the queue. The host should ignore these additional chokes for the fixed
or average queuing delay. Least-weight paths will then favour paths that are more lightly loaded, all else time interval until its reduction in traffic takes effect. After that period, further choke packets indicate
being equal. that the network is still congested.

Explicit Congestion Notification

Instead of generating additional packets to warn of congestion, a router can tag any packet it forwards (by
setting a bit in the packet’s header) to signal that it is experiencing congestion. When the network delivers
the packet, the destination can note that there is congestion and inform the sender when it sends a reply
packet. The sender can then throttle its transmissions as before. This design is called ECN (Explicit
Congestion Notification). Packets are unmarked when they are sent, as illustrated in Fig. If any of the
routers they pass through is congested, that router will then mark the packet as having experienced
Consider the network of Fig. which is divided into two parts, East and West, connected by two links, CF and congestion as it is forwarded. The destination will then echo any marks back to the sender as an explicit
EI. Suppose that most of the traffic between East and West is using link CF, and, as a result, this link is congestion signal in its next reply packet.
heavily loaded with long delays. Including queuing delay in the weight used for the shortest path calculation
will make EI more attractive. After the new routing tables have been installed, most of the East-West traffic
will now go over EI, loading this link. Consequently, in the next update, CF will appear to be the shortest
path. As a result, the routing tables may oscillate wildly, leading to erratic routing and many potential
problems.

Admission Control
Hop-by-Hop Backpressure

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At high speeds or over long distances, many new packets may be transmitted after congestion has been contrast, for real-time media, a new packet is worth more than an old one. This is because packets become
signalled because of the delay before the signal takes effect. Consider, for example, a host in San Francisco useless if they are delayed and miss the time at which they must be played out to the user.
(router A in Fig.) that is sending traffic to a host in New York (router D in Fig.) at the speed of 155 Mbps. If
the New York host begins to run out of buffers, it will take about 40 msec for a choke packet to get back to Random Early Detection
San Francisco to tell it to slow down. An ECN indication will take even longer because it is delivered via the A popular algorithm for doing this is called RED (Random Early Detection). To determine when to start
destination. discarding, routers maintain a running average of their queue lengths. When the average queue length on
some link exceeds a threshold, the link is said to be congested and a small fraction of the packets are dropped
Choke packet propagation is illustrated as at random. Picking packets at random makes it more likely that the fastest senders will see a packet drop; this
the second, third, and fourth steps in is the best option since the router cannot tell which source is causing the most trouble in a datagram network.
Fig(a). In those 40 msec, another 6.2 The affected sender will notice the loss when there is no acknowledgement, and then the transport protocol
megabits will have been sent. Even if the will slow down. The lost packet is thus delivering the same message as a choke packet, but implicitly,
host in San Francisco completely shuts without the router sending any explicit signal. RED routers improve performance compared to routers that
down immediately, the 6.2 megabits in the drop packets only when their buffers are full, though they may require tuning to work well. For example, the
pipe will continue to pour in and have to ideal number of packets to drop depends on how many senders need to be notified of congestion. However,
be dealt with. Only in the seventh diagram ECN is the preferred option if it is available. It works in exactly the same manner, but delivers a congestion
in Fig(a) will the New York router notice a signal explicitly rather than as a loss; RED is used when hosts cannot receive explicit signals.
slower flow. An alternative approach is to
have the choke packet take effect at every QUALITY OF SERVICE:
hop it passes through, as shown in the An easy solution to provide good quality of service is to build a network with enough capacity for whatever
sequence of Fig.(b). Here, as soon as the traffic will be thrown at it. The name for this solution is over provisioning. The resulting network will carry
choke packet reaches F, F is required to application traffic without significant loss and, assuming a decent routing scheme, will deliver packets with
reduce the flow to D. Doing so will low latency. Performance doesn’t get any better than this.
require F to devote more buffers to the To some extent, the telephone system is over provisioned because it is rare to pick up a telephone and not get
connection, since the source is still a dial tone instantly. There is simply so much capacity available that demand can almost always be met. The
sending away at full blast, but it gives D trouble with this solution is that it is expensive.
immediate relief. In the next step, the Four issues must be addressed to ensure quality of service:
choke packet reaches E, which tells E to 1. What applications need from the network?
reduce the flow to F. This action puts a 2. How to regulate the traffic that enters the network.
greater demand on E’s buffers but gives F 3. How to reserve resources at routers to guarantee performance.
immediate relief. Finally, the choke packet 4. Whether the network can safely accept more traffic.
reaches A and the flow genuinely slows No single technique deals efficiently with all these issues. Instead, a variety of techniques have been
down. The net effect of this hop-by-hop developed for use at the network (and transport) layer. Practical quality-of-service solutions combine
scheme is to provide quick relief at the multiple techniques. To this end, we will describe two versions of quality of service for the Internet called
point of congestion, at the price of using Integrated Services and Differentiated Services.
up more buffers upstream. In this way, APPLICATION REQUIREMENTS:
congestion can be nipped in the bud A stream of packets from a source to a destination is called a flow. A flow might be all the packets of a
without losing any packets. connection in a connection-oriented network, or all the packets sent from one process to another process in a
connectionless network. The needs of each flow can be characterized by four primary parameters:
bandwidth, delay, jitter, and loss. Together, these determine the QoS (Quality of Service) the flow requires.
Figure 5-25 depicts the symptom. Several common applications and the stringency (meaning toughness/flexibility) of their network
requirements are listed in Fig. 3.11. The applications differ in their bandwidth needs, with email, audio in all
forms, and remote login not needing much, but file sharing and video in all forms needing a great deal.
More interesting are the delay requirements. File transfer applications, including email and video, are not
delay sensitive. If all packets are delayed uniformly by a few seconds, no harm is done.
Interactive applications, such as Web surfing and remote login, are more delay sensitive. Real-time
applications, such as telephony and videoconferencing, have strict delay requirements. If all the words in a
telephone call are each delayed by too long, the users will find the connection unacceptable. On the other
hand, playing audio or video files from a server does not require low delay.
The variation (i.e., standard deviation) in the delay or packet arrival times is called jitter. The first three
Load Shedding applications in Fig. 3.11 are not sensitive to the packets arriving with irregular time intervals between them.
Load shedding is a fancy way of saying that when routers are being inundated by packets that they cannot Remote login is somewhat sensitive to that, since updates on the screen will appear in little bursts if the
handle, they just throw them away. The key question for a router drowning in packets is which packets to connection suffers much jitter.
drop. The preferred choice may depend on the type of applications that use the network. For a file transfer, Video and especially audio are extremely sensitive to jitter. If a user is watching a video over the network
an old packet is worth more than a new one. This is because dropping packet 6 and keeping packets 7 and the frames are all delayed by exactly 2.000 seconds, no harm is done. But if the transmission time varies
through 10, for example, will only force the receiver to do more work to buffer data that it cannot yet use. In randomly between 1 and 2 seconds, the result will be terrible unless the application hides the jitter. For
audio, a jitter of even a few milliseconds is clearly audible.

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Computer Networks Computer Networks

A second resource that is often in short supply is buffer space. When a packet arrives, it is buffered inside the
router until it can be transmitted on the chosen outgoing line. The purpose of the buffer is to absorb small
bursts of traffic as the flows contend with each other.
If no buffer is available, the packet has to be discarded since there is no place to put it. For good quality of
service, some buffers might be reserved for a specific flow so that flow does not have to compete for buffers
with other flows. Up to some maximum value, there will always be a buffer available when the flow needs
one.
Finally, CPU cycles may also be a scarce resource. It takes router CPU time to process a packet, so a router
can process only a certain number of packets per second. While modern routers are able to process most
packets quickly, some kinds of packets require greater CPU processing, such as the ICMP packets. Making
sure that the CPU is not overloaded is needed to ensure timely processing of these packets.
FIGURE 3.11: STRINGENCY OF APPLICATIONS’ QUALITY-OF-SERVICE REQUIREMENTS
To accommodate a variety of applications, networks may support different categories of QoS. An influential How Networks Can Be Connected
example comes from ATM networks. They support: Networks can be interconnected by different devices.
1. Constant bit rate (e.g., telephony). In the physical layer, networks can be connected by repeaters or hubs, which just
2. Real-time variable bit rate (e.g., compressed videoconferencing).
move the bits from one network to an identical network. These are mostly analog
3. Non-real-time variable bit rate (e.g., watching a movie on demand).
4. Available bit rate (e.g., file transfer).
devices and do not understand anything about digital protocols.
These categories are also useful for other purposes and other networks. In the Data Link Layer, networks are connected by bridges and switches. They can
TRAFFIC SHAPING: Before the network can make QoS guarantees, it must know what traffic is being accept frames, examine the MAC addresses, and forward the frames to a different
guaranteed. In the telephone network, this characterization is simple. For example, a voice call (in network while doing minor protocol translation in the process.
uncompressed format) needs 64 kbps and consists of one 8-bit sample every 125 μsec. In the network layer, we have routers that can connect two networks. If two
However, traffic in data networks is bursty. It typically arrives at nonuniform rates as the traffic rate varies networks have dissimilar network layers, the router may be able to translate between
(e.g., videoconferencing with compression), users interact with applications (e.g., browsing a new Web the packet formats, although packet translation is now increasingly rare. A router that
page), and computers switch between tasks. Bursts of traffic are more difficult to handle than constant-rate can handle multiple protocols is called a multiprotocol router.
traffic because they can fill buffers and cause packets to be lost.
Traffic shaping is a technique for regulating the average rate and burstiness of a flow of data that enters the
In the transport layer we find transport gateways, which can interface between two
network. The goal is to allow applications to transmit a wide variety of traffic that suits their needs, including transport connections. Transport gateway has a different transport protocol, by
some bursts, yet have a simple and useful way to describe the possible traffic patterns to the network. essentially gluing one connection to another connection.
When a flow is set up, the user and the network (i.e., the customer and the provider) agree on a certain traffic Finally, in the application layer, application gateways translate message semantics.
pattern (i.e., shape) for that flow. In effect, the customer says to the provider ‘‘my transmission pattern will
look like this; can you handle it?’’ Here, we will focus on internetworking in the network layer. To see how that differs from switching
Sometimes this agreement is called an SLA (Service Level Agreement), especially when it is made over in the data link layer, examine Fig. 5-44. In Fig. 5-44(a), the source machine, S, wants to send a
aggregate flows and long periods of time, such as all of the traffic for a given customer. As long as the packet to the destination machine, D. These machines are on different Ethernets, connected by a
customer fulfills her part of the bargain and only sends packets according to the agreed-on contract, the switch. S encapsulates the packet in a frame and sends it on its way. The frame arrives at the switch,
provider promises to deliver them all in a timely fashion. which then determines that the frame has to go to LAN 2 by looking at its MAC address. The
Traffic shaping reduces congestion and thus helps the network live up to its promise. However, to make it switch just removes the frame from LAN 1 and deposits it on LAN 2.
work, there is also the issue of how the provider can tell if the customer is following the agreement and what
to do if the customer is not. Packets in excess of the agreed pattern might be dropped by the network, or they
might be marked as having lower priority. Monitoring a traffic flow is called traffic policing.
PACKET SCHEDULING:
Being able to regulate the shape of the offered traffic is a good start. However, to provide a performance
guarantee, we must reserve sufficient resources along the route that the packets take through the network. To
do this, we are assuming that the packets of a flow follow the same route. Spraying them over routers at
random makes it hard to guarantee anything. As a consequence, something similar to a virtual circuit has to
be set up from the source to the destination, and all the packets that belong to the flow must follow this route.
Algorithms that allocate router resources among the packets of a flow and between competing flows are
called packet scheduling algorithms. Three different kinds of resources can potentially be reserved for
different flows:
1. Bandwidth.
2. Buffer space.
3. CPU cycles.
The first one, bandwidth, is the most obvious. If a flow requires 1 Mbps and the outgoing line has a capacity
of 2 Mbps, trying to direct three flows through that line is not going to work. Thus, reserving bandwidth
means not oversubscribing any output line.

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INTERNETWORKING: FIGURE 3.12: SOME OF THE MANY WAYS NETWORKS CAN DIFFER.

HOW NETWORKS DIFFER: How Networks Can Be Connected

Networks can differ in many ways. Some of the differences, such as different There are two basic choices for connecting different networks: we can build
modulation techniques or frame formats, are internal to the physical and data link devices that translate or convert packets from each kind of network into packets for
layers. These differences will not concern us here. Instead, in Fig. 3.12 we list some each other network, or, like good computer scientists, we can try to solve the
of the differences that can be exposed to the network layer. It is papering over problem by adding a layer of indirection and building a common layer on top of the
these differences that makes internetworking more difficult than operating within a different networks. In either case, the devices are placed at the boundaries
single network. between networks.

When packets sent by a source on one network must transit one or more Internetworking has been very successful at building large networks, but it
foreign networks before reaching the destination network, many problems can only works when there is a common network layer. There have, in fact, been many
occur at the interfaces between networks. To start with, the source needs to be network protocols over time. Getting everybody to agree on a single format is
able to address the destination. difficult when companies perceive it to their commercial advantage to have a
proprietary format that they control.
What do we do if the source is on an Ethernet network and the destination is
on a WiMAX network? Assuming we can even specify a WiMAX destination from an A router that can handle multiple network protocols is called a
Ethernet network, packets would cross from a connectionless network to a multiprotocol router. It must either translate the protocols, or leave connection
connection-oriented one. for a higher protocol layer. Neither approach is entirely satisfactory. Connection at a
higher layer, say, by using TCP, requires that all the networks implement TCP
This may require that a new connection be set up on short notice, which (which may not be the case). Then, it limits usage across the networks to
injects a delay, and much overhead if the connection is not used for many more applications that use TCP (which does not include many real-time applications).
packets. Many specific differences may have to be accommodated as well. How do
we multicast a packet to a group with some members on a network that does not TUNNELING:
support multicast?
Handling the general case of making two different networks interwork is
The differing max packet sizes used by different networks can be a major exceedingly difficult. However, there is a common special case that is manageable
nuisance, too. How do you pass an 8000-byte packet through a network whose even for different network protocols. This case is where the source and destination
maximum size is 1500 bytes? If packets on a connection-oriented network transit a hosts are on the same type of network, but there is a different network in between.
connectionless network, they may arrive in a different order than they were sent. As an example, think of an international bank with an IPv6 network in Paris, an
That is something the sender likely did not expect, and it might come as an IPv6 network in London and connectivity between the offices via the IPv4 Internet.
(unpleasant) surprise to the receiver as well. This situation is shown in Fig. 3.13.

FIGURE 3.13: TUNNELING A PACKET FROM PARIS TO LONDON

The solution to this problem is a technique called tunneling. To send an IP

packet to a host in the London office, a host in the Paris office constructs the packet

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containing an IPv6 address in London, and sends it to the multiprotocol router that Across the networks that make up the internet, an interdomain or exterior
connects the Paris IPv6 network to the IPv4 Internet. gateway protocol is used. The networks may all use different intradomain
protocols, but they must use the same interdomain protocol.
When this router gets the IPv6 packet, it encapsulates the packet with an
IPv4 header addressed to the IPv4 side of the multiprotocol router that connects to In the Internet, the interdomain routing protocol is called BGP (Border
the London IPv6 network. Gateway Protocol).

That is, the router puts a (IPv6) packet inside a (IPv4) packet. When this There is one more important term to introduce. Since each network is
wrapped packet arrives, the London router removes the original IPv6 packet and operated independently of all the others, it is often referred to as an AS
sends it onward to the destination host. The path through the IPv4 Internet can be (Autonomous System). A good mental model for an AS is an ISP network. In fact,
seen as a big tunnel extending from one multiprotocol router to the other. an ISP network may be comprised of more than one AS, if it is managed, or, has
been acquired, as multiple networks. But the difference is usually not significant.
The IPv6 packet just travels from one end of the tunnel to the other, snug in
its nice box. It does not have to worry about dealing with IPv4 at all. Neither do the PACKET FRAGMENTATION: Each network or link imposes some maximum
hosts in Paris or London. Only the multiprotocol routers have to understand both size on its packets. These limits have various causes, among them:
IPv4 and IPv6 packets.
1. Hardware (e.g., the size of an Ethernet frame).
In effect, the entire trip from one multiprotocol router to the other is like a
2. Operating system (e.g., all buffers are 512 bytes).
hop over a single link. Tunneling is widely used to connect isolated hosts and
networks using other networks. 3. Protocols (e.g., the number of bits in the packet length field).

4. Compliance with some (inter)national standard.

INTERNETWORK ROUTING: 5. Desire to reduce error-induced retransmissions to some level.
Routing through an internet poses the same basic problem as routing within 6. Desire to prevent one packet from occupying the channel too long.
a single network, but with some added complications. To start, the networks may
internally use different routing algorithms. For example, one network may use link The result of all these factors is that the network designers are not free to
state routing and another distance vector routing. Since link state algorithms need choose any old maximum packet size they wish. Maximum payloads for some
to know the topology but distance vector algorithms do not, this difference alone common technologies are 1500 bytes for Ethernet and 2272 bytes for 802.11. IP is
would make it unclear how to find the shortest paths across the internet. more generous, allows for packets as big as 65,515 bytes.

Networks run by different operators lead to bigger problems. First, the Hosts usually prefer to transmit large packets because this reduces packet
operators may have different ideas about what is a good path through the network. overheads such as bandwidth wasted on header bytes. An obvious internetworking
One operator may want the route with the least delay, while another may want the problem appears when a large packet wants to travel through a network whose
most inexpensive route. This will lead the operators to use different quantities to maximum packet size is too small. This nuisance has been a persistent issue, and
set the shortest-path costs. solutions to it have evolved along with much experience gained on the Internet.

Finally, the internet may be much larger than any of the networks that One solution is to make sure the problem does not occur in the first place.
comprise it. It may therefore require routing algorithms that scale well by using a However, this is easier said than done. A source does not usually know the path a
hierarchy, even if none of the individual networks need to use a hierarchy. packet will take through the network to a destination, so it certainly does not know
how small packets must be to get there. This packet size is called the Path MTU
All of these considerations lead to a two-level routing algorithm. Within each (Path Maximum Transmission Unit).
network, an intradomain or interior gateway protocol is used for routing.
(‘‘Gateway’’ is an older term for ‘‘router.’’) It might be a link state protocol of the The alternative solution to the problem is to allow routers to break up
Kind. packets into fragments, sending each fragment as a separate network layer

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packet. However, as every parent of a small child knows, converting a large object For file transfer, error-free transmission is more important than fast
into small fragments is considerably easier than the reverse process. transmission. The Type of service field provided 3 bits to signal priority and 3 bits
to signal whether a host cared more about delay, throughput, or reliability.
THE NETWORK LAYER IN THE INTERNET
The Total length includes everything in the datagram—both header and data.
THE IP VERSION 4 PROTOCOL: The maximum length is 65,535 bytes. At present, this upper limit is tolerable, but
with future networks, larger datagrams may be needed.
An appropriate place to start our study of the network layer in the Internet is
with the format of the IP datagrams themselves. An IPv4 datagram consists of a The Identification field is needed to allow the destination host to determine
header part and a body or payload part. The header has a 20-byte fixed part and a which packet a newly arrived fragment belongs to. All the fragments of a packet
variable-length optional part. The header format is shown in Fig. 3.14. The bits are contain the same Identification value.
transmitted from left to right and top to bottom, with the high-order bit of the
Version field going first. (This is a ‘‘big-endian’’ network byte order. DF stands for Don’t Fragment. It is an order to the routers not to fragment
the packet. Originally, it was intended to support hosts incapable of putting the
On little-endian machines, such as Intel x86 computers, a software pieces back together again.
conversion is required on both transmission and reception.) In retrospect, little-
endian would have been a better choice, but at the time IP was designed, no one MF stands for More Fragments. All fragments except the last one have this
knew it would come to dominate computing. bit set. It is needed to know when all fragments of a datagram have arrived.

The Fragment offset tells where in the current packet this fragment belongs.
All fragments except the last one in a datagram must be a multiple of 8 bytes, the
elementary fragment unit. Since 13 bits are provided, there is a maximum of 8192
fragments per datagram, supporting a maximum packet length up to the limit of
the Total length field. Working together, the Identification, MF, and Fragment offset
fields are used to implement fragmentation.

The TtL (Time to live) field is a counter used to limit packet lifetimes. It was
originally supposed to count time in seconds, allowing a maximum lifetime of 255
sec.

When the network layer has assembled a complete packet, it needs to know
what to do with it. The Protocol field tells it which transport process to give the
FIGURE 3.14: THE IPV4 (INTERNET PROTOCOL) HEADER
packet to. TCP is one possibility, but so are UDP and some others.
The Version field keeps track of which version of the protocol the datagram
Since the header carries vital information such as addresses, it rates its own
belongs to.
checksum for protection, the Header checksum. The algorithm is to add up all the
Since the header length is not constant, a field in the header, IHL, is 16-bit halfwords of the header as they arrive, using one’s complement arithmetic,
provided to tell how long the header is, in 32-bit words. The minimum value is 5, and then take the one’s complement of the result. For purposes of this algorithm,
which applies when no options are present. The maximum value of this 4-bit field is the Header checksum is assumed to be zero upon arrival. Such a checksum is
15, which limits the header to 60 bytes, and thus the Options field to 40 bytes. useful for detecting errors while the packet travels through the network.

The Differentiated services field is one of the few fields that have changed its The Source address and Destination address indicate the IP address of the
meaning (slightly) over the years. Originally, it was called the Type of service field. source and destination network interfaces.
Various combinations of reliability and speed are possible. For digitized voice, fast
The Options field was designed to provide an escape to allow subsequent
delivery beats accurate delivery.
versions of the protocol to include information not present in the original design, to
permit experimenters to try out new ideas, and to avoid allocating header bits to

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information that is rarely needed. The options are of variable length. The Options This format is referred to as dotted-decimal notation. Note that because each
field is padded out to a multiple of 4 bytes. Originally, the five options listed in Fig. byte (octet) is only 8 bits, each number in the dotted-decimal notation is between 0
3.15. and 255. We sometimes see an IPv4 address in hexadecimal notation. Each
hexadecimal digit is equivalent to four bits. This means that a 32-bit address has 8
hexadecimal digits. This notation is often used in network programming. Figure
3.16 shows an IP address in the three discussed notations.

HIERARCHY IN ADDRESSING: A 32-bit IPv4 address is also hierarchical,

but divided only into two parts. The first part of the address, called the prefix,
defines the network; the second part of the address, called the suffix, defines the
FIGURE 3.15: SOME OF THE IP OPTIONS node (connection of a device to the Internet).

IPV4 ADDRESSES: Figure 3.17 shows the prefix and suffix of a 32-bit IPv4 address. The prefix
length is n bits and the suffix length is (32 − n) bits.
The identifier used in the IP layer of the TCP/IP protocol suite to identify the
connection of each device to the Internet is called the Internet address or IP
address. An IPv4 address is a 32-bit address that uniquely and universally defines
the connection of a host or a router to the Internet. The IP address is the address
of the connection, not the host or the router, because if the device is moved to
another network, the IP address may be changed.

IPv4 addresses are unique in the sense that each address defines one, and
only one, connection to the Internet. If a device has two connections to the
FIGURE 3.16: THREE DIFFERENT NOTATIONS IN IPV4 ADDRESSING
Internet, via two networks, it has two IPv4 addresses. IPv4 addresses are universal
in the sense that the addressing system must be accepted by any host that wants
to be connected to the Internet.

Address Space

A protocol like IPv4 that defines addresses has an address space. An

address space is the total number of addresses used by the protocol. If a protocol
uses b bits to define an address, the address space is 2b because each bit can have
two different values (0 or 1). IPv4 uses 32-bit addresses, which means that the
address space is 232 or 4,294,967,296 (more than four billion). If there were no
restrictions, more than 4 billion devices could be connected to the Internet.
FIGURE 3.17: HIERARCHY IN ADDRESSING
Notation
A prefix can be fixed length or variable length. The network identifier in the
There are three common notations to show an IPv4 address: binary notation IPv4 was first designed as a fixed-length prefix. This scheme, which is now
(base 2), dotted-decimal notation (base 256), and hexadecimal notation (base 16). obsolete, is referred to as classful addressing. The new scheme, which is referred to
In binary notation, an IPv4 address is displayed as 32 bits. To make the address as classless addressing, uses a variable-length network prefix. First, we briefly
more readable, one or more spaces are usually inserted between each octet (8 discuss Classful addressing; then we concentrate on classless addressing.
bits). Each octet is often referred to as a byte. To make the IPv4 address more
Classful Addressing:
compact and easier to read, it is usually written in decimal form with a decimal
point (dot) separating the bytes. When the Internet started, an IPv4 address was designed with a fixed-length
prefix, but to accommodate both small and large networks, three fixed-length

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prefixes were designed instead of one (n = 8, n = 16, and n = 24). The whole the Internet was faced with the problem of the addresses being rapidly used up,
address space was divided into five classes (class A, B, C, D, and E), as shown in resulting in no more addresses available for organizations and individuals that
Figure 3.18. This scheme is referred to as classful addressing. needed to be connected to the Internet.

In class A, the network length is 8 bits, but since the first bit, which is 0, Subnetting and Supernetting: To alleviate address depletion, two
defines the class, we can have only seven bits as the network identifier. This means strategies were proposed and, to some extent, implemented: subnetting and
there are only 27 = 128 networks in the world that can have a class A address. Supernetting. In subnetting, a class A or class B block is divided into several
subnets.
In class B, the network length is 16 bits, but since the first two bits, which
are (10)2, define the class, we can have only 14 bits as the network identifier. This Each subnet has a larger prefix length than the original network. While
means there are only 214 = 16,384 networks in the world that can have a class B subnetting was devised to divide a large block into smaller ones, Supernetting was
address. devised to combine several class C blocks into a larger block to be attractive to
organizations that need more than the 256 addresses available in a class C block.
All addresses that start with (110)2 belong to class C. In class C, the network This idea did not work either because it makes the routing of packets more difficult.
length is 24 bits, but since three bits define the class, we can have only 21 bits as
the network identifier. This means there are 221 = 2,097,152 networks in the world Classless Addressing:
that can have a class C address.
Subnetting and Supernetting in classful addressing did not really solve the
address depletion problem. With the growth of the Internet, it was clear that a
larger address space was needed as a long-term solution. The larger address space,
however, requires that the length of IP addresses also be increased, which means
the format of the IP packets needs to be changed.

Although the long-range solution has already been devised and is called
IPv6, a short-term solution was also devised to use the same address space but to
change the distribution of addresses to provide a fair share to each organization.
The short-term solution still uses IPv4 addresses, but it is called classless
addressing. In other words, the class privilege was removed from the distribution to
compensate for the address depletion.

FIGURE 3.18: OCCUPATION OF THE ADDRESS SPACE IN CLASSFUL In classless addressing, the whole address space is divided into variable
ADDRESSING length blocks. The prefix in an address defines the block (network); the suffix
defines the node (device). Theoretically, we can have a block of 20, 21, 22, . . . ,
Class D is not divided into prefix and suffix. It is used for multicast 232 addresses. One of the restrictions, as we discuss later, is that the number of
addresses. All addresses that start with 1111 in binary belong to class E. As in Class addresses in a block needs to be a power of 2. An organization can be granted one
D, Class E is not divided into prefix and suffix and is used as reserve. block of addresses. Figure 3.19 shows the division of the whole address space into
nonoverlapping blocks.
Advantage of Classful Addressing:

Although classful addressing had several problems and became obsolete, it

had one advantage: Given an address, we can easily find the class of the address
and, since the prefix length for each class is fixed, we can find the prefix length
immediately. In other words, the prefix length in classful addressing is inherent in FIGURE 3.19: VARIABLE-LENGTH BLOCKS IN CLASSLESS
the address; no extra information is needed to extract the prefix and the suffix. ADDRESSING

Address Depletion: The reason that classful addressing has become Unlike classful addressing, the prefix length in classless addressing is
obsolete is address depletion. Since the addresses were not distributed properly, variable. We can have a prefix length that ranges from 0 to 32. The size of the

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network is inversely proportional to the length of the prefix. A small prefix means a
larger network; a large prefix means a smaller network.

We need to emphasize that the idea of classless addressing can be easily

applied to classful addressing. An address in class A can be thought of as a
classless address in which the prefix length is 8. An address in class B can be
thought of as a classless address in which the prefix is 16, and so on. In other
words, classful addressing is a special case of classless addressing.

Prefix Length: Slash Notation:

FIGURE 3.21: INFORMATION EXTRACTION IN CLASSLESS
The first question that we need to answer in classless addressing is how to ADDRESSING
find the prefix length if an address is given. Since the prefix length is not inherent
in the address, we need to separately give the length of the prefix. In this case, the Example:
prefix length, n, is added to the address, separated by a slash. The notation is
A classless address is given as 167.199.170.82/27. We can find the above
informally referred to as slash notation and formally as classless interdomain
three pieces of information as follows. The number of addresses in the network is
routing or CIDR (pronounced cider) strategy. An address in classless addressing
232 − n = 25 = 32 addresses.
can then be represented as shown in Figure 3.20.
The first address can be found by keeping the first 27 bits and changing the
rest of the bits to 0s.

Address: 167.199.170.82/27 10100111 11000111 10101010

01010010

First address: 167.199.170.64/27 10100111 11000111 10101010

FIGURE 3.20: SLASH NOTATION (CIDR)
01000000
Extracting Information from an Address:
The last address can be found by keeping the first 27 bits and changing the
Given any address in the block, we normally like to know three pieces of rest of the bits to 1s.
information about the block to which the address belongs: the number of
Address: 167.199.170.82/27 10100111 11000111 10101010
addresses, the first address in the block, and the last address. Since the value of
01011111
prefix length, n, is given, we can easily find these three pieces of information, as
shown in Figure 3.21. Last address: 167.199.170.95/27 10100111 11000111 10101010
01011111
1. The number of addresses in the block is found as N = 2 32−n
.
IP VERSION 6:
2. To find the first address, we keep the n leftmost bits and set the (32 − n)
rightmost bits all to 0s. IP has been in heavy use for decades. It has worked extremely well, as
demonstrated by the exponential growth of the Internet. Unfortunately, IP has
3. To find the last address, we keep the n leftmost bits and set the (32 − n)
become a victim of its own popularity: it is close to running out of addresses. Even
rightmost bits all to 1s.
with CIDR and NAT using addresses more sparingly, the last IPv4 addresses are
expected to be assigned by ICANN before the end of 2012.

IPv6 (IP version 6) is a replacement design that does just that. It uses
128-bit addresses; a shortage of these addresses is not likely any time in the
foreseeable future. However, IPv6 has proved very difficult to deploy. It is a
different network layer protocol that does not really interwork with IPv4, despite
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many similarities. Also, companies and users are not really sure why they should previously were required are now optional (because they are not used
want IPv6 in any case. so often).

In 1990 IETF started work on a new version of IP, one that would never run o In addition, the way options are represented is different, making
out of addresses, would solve a variety of other problems, and be more flexible and it simple for routers to skip over options not intended for them.
efficient as well. Its major goals were: This feature speeds up packet processing time.

1. Support billions of hosts, even with inefficient address allocation. A fourth area in which IPv6 represents a big advance is in security.

2. Reduce the size of the routing tables. Finally, more attention has been paid to quality of service.

3. Simplify the protocol, to allow routers to process packets faster.

4. Provide better security (authentication and privacy). The Main IPv6 Header:

5. Pay more attention to the type of service, particularly for real-time data. The IPv6 header is shown in Fig. 3.22. The Version field is always 6 for IPv6
(and 4 for IPv4). During the transition period from IPv4, which has already taken
6. Aid multicasting by allowing scopes to be specified. more than a decade, routers will be able to examine this field to tell what kind of
7. Make it possible for a host to roam without changing its address. packet they have.

8. Allow the protocol to evolve in the future. As an aside, making this test wastes a few instructions in the critical path,
given that the data link header usually indicates the network protocol for
9. Permit the old and new protocols to coexist for years. demultiplexing, so some routers may skip the check.

The design of IPv6 presented a major opportunity to improve all of the The Differentiated services field (originally called Traffic class) is used to
features in IPv4 that fall short of what is now wanted. One proposal was to run TCP distinguish the class of service for packets with different real-time delivery
over CLNP, the network layer protocol designed for OSI. With its 160-bit addresses, requirements.
CLNP would have provided enough address space forever.
The Flow label field provides a way for a source and destination to mark
IPv6 meets IETF’s goals fairly well. It maintains the good features of IP, groups of packets that have the same requirements and should be treated in the
discards or deemphasizes the bad ones, and adds new ones where needed. In same way by the network, forming a pseudo connection.
general, IPv6 is not compatible with IPv4, but it is compatible with the other
auxiliary Internet protocols, including TCP, UDP, ICMP, IGMP, OSPF, BGP, and DNS, The Payload length field tells how many bytes follow the 40-byte header of
with small modifications being required to deal with longer addresses. Fig. 3.22. The name was changed from the IPv4 Total length field because the
meaning was changed slightly: the 40 header bytes are no longer counted as part
The main features of IPv6 are discussed below. of the length (as they used to be). This change means the payload can now be
65,535 bytes instead of a mere 65,515 bytes.
First and foremost, IPv6 has longer addresses than IPv4. They are 128
bits long, which solves the problem that IPv6 set out to solve:
providing an effectively unlimited supply of Internet addresses.

The second major improvement of IPv6 is the simplification of the

header. It contains only seven fields (versus 13 in IPv4). This change
allows routers to process packets faster and thus improves throughput
and delay.

The third major improvement is better support for options. This

change was essential with the new header because fields that

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defined. Each ICMP message type is carried encapsulated in an IP packet. The most
important ones are listed in Fig. 3.23.

FIGURE 3.23: THE PRINCIPAL ICMP MESSAGE TYPES

The DESTINATION UNREACHABLE message is used when the router cannot

FIGURE 3.22: THE IPV6 FIXED HEADER (REQUIRED) locate the destination or when a packet with the DF bit cannot be delivered because
The Next header field tells which transport protocol handler (e.g., TCP, UDP) a ‘‘small-packet’’ network stands in the way.
to pass the packet to. The TIME EXCEEDED message is sent when a packet is dropped because its
The Hop limit field is used to keep packets from living forever. It is, in TtL (Time to live) counter has reached zero. This event is a symptom that packets
practice, the same as the Time to live field in IPv4, namely, a field that is are looping, or that the counter values are being set too low.
decremented on each hop. In The PARAMETER PROBLEM message indicates that an illegal value has been
Next come the Source address and Destination address fields. A new notation detected in a header field. This problem indicates a bug in the sending host’s IP
has been devised for writing 16-byte addresses. They are written as eight groups of software or possibly in the software of a router transited.
four hexadecimal digits with colons between the groups, like this: The SOURCE QUENCH message was long ago used to throttle hosts that were
8000:0000:0000:0000:0123:4567:89AB:CDEF sending too many packets. When a host received this message, it was expected to
slow down.
Since many addresses will have many zeros inside them, three optimizations
have been authorized. First, leading zeros within a group can be omitted, so 0123 The REDIRECT message is used when a router notices that a packet seems to
can be written as 123. Second, one or more groups of 16 zero bits can be replaced be routed incorrectly. It is used by the router to tell the sending host to update to a
by a pair of colons. Thus, the above address now becomes better route.

8000::123:4567:89AB:CDEF The TIMESTAMP REQUEST and TIMESTAMP REPLY messages are similar,
except that the arrival time of the message and the departure time of the reply are
INTERNET CONTROL PROTOCOLS: recorded in the reply. This facility can be used to measure network performance.

In addition to IP, which is used for data transfer, the Internet has several OSPF—AN INTERIOR GATEWAY ROUTING PROTOCOL:
companion control protocols that are used in the network layer. They include ICMP,
ARP, and DHCP. The Internet is made up of a large number of independent networks or ASes
(Autonomous Systems) that are operated by different organizations, usually a
ICMP—The Internet Control Message Protocol: company, university, or ISP. Inside of its own network, an organization can use its
own algorithm for internal routing, or intradomain routing, as it is more
The operation of the Internet is monitored closely by the routers. When
commonly known. Nevertheless, there are only a handful of standard protocols that
something unexpected occurs during packet processing at a router, the event is
are popular.
reported to the sender by the ICMP (Internet Control Message Protocol). ICMP
is also used to test the Internet. About a dozen types of ICMP messages are
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An intradomain routing protocol is also called an interior gateway OSPF works by exchanging information between adjacent routers, which is
protocol. We will study the problem of routing between independently operated not the same as between neighboring routers. In particular, it is inefficient to have
networks, or interdomain routing. For that case, all networks must use the same every router on a LAN talk to every other router on the LAN. To avoid this situation,
interdomain routing protocol or exterior gateway protocol. The protocol that is one router is elected as the designated router. It is said to be adjacent to all the
used in the Internet is BGP (Border Gateway Protocol). other routers on its LAN, and exchanges information with them.

Early intradomain routing protocols used a distance vector design, based on In effect, it is acting as the single node that represents the LAN. Neighboring
the distributed Bellman-Ford algorithm inherited from the ARPANET. It works well in routers that are not adjacent do not exchange information with each other. A
small systems, but less well as networks get larger. It also suffers from the count- backup designated router is always kept up to date to ease the transition should
to-infinity problem and generally slow convergence. the primary designated router crash and need to be replaced immediately.

The ARPANET switched over to a link state protocol in May 1979 because of During normal operation, each router periodically floods LINK STATE UPDATE
these problems, and in 1988 IETF began work on a link state protocol for messages to each of its adjacent routers. These messages gives its state and
intradomain routing. That protocol, called OSPF (Open Shortest Path First), provide the costs used in the topological database. The flooding messages are
became a standard in 1990. It drew on a protocol called IS-IS (Intermediate- acknowledged, to make them reliable.
System to Intermediate-System), which became an ISO standard.
Each message has a sequence number, so a router can see whether an
Given the long experience with other routing protocols, the group designing incoming LINK STATE UPDATE is older or newer than what it currently has. Routers
OSPF had a long list of requirements that had to be met. First, the algorithm had to also send these messages when a link goes up or down or its cost changes.
be published in the open literature, hence the ‘‘O’’ in OSPF.
DATABASE DESCRIPTION messages give the sequence numbers of all the
Second, the new protocol had to support a variety of distance metrics, link state entries currently held by the sender. By comparing its own values with
including physical distance, delay, and so on. Third, it had to be a dynamic those of the sender, the receiver can determine who has the most recent values.
algorithm, one that adapted to changes in the topology automatically and quickly. These messages are used when a link is brought up.

Fourth, and new for OSPF, it had to support routing based on type of service. All these messages are sent directly in IP packets. The five kinds of
The new protocol had to be able to route real-time traffic one way and other traffic messages are summarized in Fig. 3.24.
a different way. At the time, IP had a Type of service field, but no existing routing
protocol used it. This field was included in OSPF but still nobody used it, and it was
eventually removed.

Fifth, and related to the above, OSPF had to do load balancing, splitting the
load over multiple lines. Most previous protocols sent all packets over a single best
route, even if there were two routes that were equally good. The other route was
not used at all. In many cases, splitting the load over multiple routes gives better
performance.
FIGURE 3.24: THE FIVE TYPES OF OSPF MESSAGES
Sixth, support for hierarchical systems was needed. By 1988, some networks
BGP—THE EXTERIOR GATEWAY ROUTING PROTOCOL:
had grown so large that no router could be expected to know the entire topology.
OSPF had to be designed so that no router would have to. Within a single AS, OSPF and IS-IS are the protocols that are commonly
used. Between ASes, a different protocol, called BGP (Border Gateway Protocol),
OSPF supports both point-to-point links (e.g., SONET) and broadcast
is used. A different protocol is needed because the goals of an intradomain protocol
networks (e.g., most LANs). Actually, it is able to support networks with multiple
and an interdomain protocol are not the same. All an intradomain protocol has to
routers, each of which can communicate directly with the others (called multi-
do is move packets as efficiently as possible from the source to the destination.
access networks) even if they do not have broadcast capability. Earlier protocols
did not handle this case well.

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BGP is a form of distance vector protocol, but it is quite unlike intradomain

distance vector protocols such as RIP. We have already seen that policy, instead of
minimum distance, is used to pick which routes to use. Another large difference is INTERNET PROTOCOL (IP):
that instead of maintaining just the cost of the route to each destination, each BGP The network layer in version 4 can be thought of as one main protocol and
router keeps track of the path used. This approach is called a path vector three auxiliary ones. The main protocol, Internet Protocol version 4 (IPv4), is
protocol. responsible for packetizing, forwarding, and delivery of a packet at the network
The path consists of the next hop router (which may be on the other side of layer.
the ISP, not adjacent) and the sequence of ASes, or AS path, that the route has The Internet Control Message Protocol version 4 (ICMPv4) helps IPv4 to
followed (given in reverse order). Finally, pairs of BGP routers communicate with handle some errors that may occur in the network-layer delivery. The Internet
each other by establishing TCP connections. Operating this way provides reliable Group Management Protocol (IGMP) is used to help IPv4 in multicasting. The
communication and also hides all the details of the network being passed through. Address Resolution Protocol (ARP) is used to glue the network and data-link layers
An example of how BGP routes are advertised is shown in Fig. 3.25. There in mapping network-layer addresses to link-layer addresses. Figure 3.26 shows the
are three ASes and the middle one is providing transit to the left and right ISPs. A positions of these four protocols in the TCP/IP protocol suite.
route advertisement to prefix C starts in AS3. When it is propagated across the link
to R2c at the top of the figure, it has the AS path of simply AS3 and the next hop
router of R3a.

At the bottom, it has the same AS path but a different next hop because it
came across a different link. This advertisement continues to propagate and crosses
the boundary into AS1. At router R1a, at the top of the figure, the AS path is AS2,
AS3 and the next hop is R2a.

Carrying the complete path with the route makes it easy for the receiving
router to detect and break routing loops. The rule is that each router that sends a
route outside of the AS prepends its own AS number to the route. (This is why the
list is in reverse order.)
FIGURE 3.26: POSITION OF IP & OTHER NETWORK-LAYER
PROTOCOLS IN TCP/IP PROTOCOL SUITE

IPv4 is an unreliable datagram protocol—a best-effort delivery service. The

term best-effort means that IPv4 packets can be corrupted, be lost, arrive out of
order, or be delayed, and may create congestion for the network. If reliability is
important, IPv4 must be paired with a reliable transport-layer protocol such as TCP.

IPv4 is also a connectionless protocol that uses the datagram approach. This
means that each datagram is handled independently, and each datagram can follow
a different route to the destination. This implies that datagrams sent by the same
source to the same destination could arrive out of order. Again, IPv4 relies on a
FIGURE 3.25: PROPAGATION OF BGP ROUTE ADVERTISEMENTS
higher-level protocol to take care of all these problems.
When a router receives a route, it checks to see if its own AS number is
DATAGRAM FORMAT:
already in the AS path. If it is, a loop has been detected and the advertisement is
discarded. Packets used by the IP are called datagrams. Figure 3.27 shows the IPv4
datagram format. A datagram is a variable-length packet consisting of two parts:

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header and payload (data). The header is 20 to 60 bytes in length and contains Protocol. In TCP/IP, the data section of a packet, called the payload, carries the
information essential to routing and delivery. It is customary in TCP/IP to show the whole packet from another protocol. A datagram, for example, can carry a
header in 4-byte sections. packet belonging to any transport-layer protocol such as UDP or TCP. A
datagram can also carry a packet from other protocols that directly use the
Version Number. The 4-bit version number (VER) field defines the version of service of the IP, such as some routing protocols or some auxiliary protocols.
the IPv4 protocol, which, obviously, has the value of 4.
Header checksum. IP is not a reliable protocol; it does not check whether the
Header Length. The 4-bit header length (HLEN) field defines the total length of payload carried by a datagram is corrupted during the transmission. IP puts the
the datagram header in 4-byte words. The IPv4 datagram has a variable-length
burden of error checking of the payload on the protocol that owns the payload,
header. such as UDP or TCP. The datagram header, however, is added by IP, and its
Service Type. In the original design of the IP header, this field was referred to error-checking is the responsibility of IP.
as type of service (TOS), which defined how the datagram should be handled. Source and Destination Addresses. These 32-bit source and destination
address fields define the IP address of the source and destination respectively.
The source host should know its IP address. The destination IP address is either
known by the protocol that uses the service of IP or is provided by the DNS.

Options. A datagram header can have up to 40 bytes of options. Options can be

used for network testing and debugging. Although options are not a required
part of the IP header, option processing is required of the IP software.

Payload. Payload, or data, is the main reason for creating a datagram. Payload
is the packet coming from other protocols that use the service of IP. Comparing
a datagram to a postal package, payload is the content of the package; the
header is only the information written on the package.

ICMPv4:

The IPv4 has no error-reporting or error-correcting mechanism. The IP

protocol also lacks a mechanism for host and management queries. A host
sometimes needs to determine if a router or another host is alive. And sometimes a
FIGURE 3.27: IP DATAGRAM network manager needs information from another host or router.

Total Length. This 16-bit field defines the total length (header plus data) of the The Internet Control Message Protocol version 4 (ICMPv4) has been
IP datagram in bytes. A 16-bit number can define a total length of up to 65,535 designed to compensate for the above two deficiencies. It is a companion to the IP
(when all bits are 1s). protocol. ICMP itself is a network-layer protocol.

Identification, Flags, and Fragmentation Offset. These three fields are However, its messages are not passed directly to the data-link layer as would
related to the fragmentation of the IP datagram when the size of the datagram be expected. Instead, the messages are first encapsulated inside IP datagrams
is larger than the underlying network can carry. before going to the lower layer. When an IP datagram encapsulates an ICMP
message, the value of the protocol field in the IP datagram is set to 1 to indicate
Time-to-live. The time-to-live (TTL) field is used to control the maximum that the IP payroll is an ICMP message.
number of hops (routers) visited by the datagram. When a source host sends
the datagram, it stores a number in this field. This value is approximately two MESSAGES:
times the maximum number of routers between any two hosts. Each router that
ICMP messages are divided into two broad categories: error-reporting
processes the datagram decrements this number by one. If this value, after
messages and query messages.
being decremented, is zero, the router discards the datagram.
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The error-reporting messages report problems that a router or a host Source Quench: Another error message is called the source quench (type 4)
(destination) may encounter when it processes an IP packet. message, which informs the sender that the network has encountered congestion
and the datagram has been dropped; the source needs to slow down sending more
The query messages, which occur in pairs, help a host or a network datagrams.
manager get specific information from a router or another host. For example,
nodes can discover their neighbors. Also, hosts can discover and learn about
routers on their network and routers can help a node redirect its messages.

An ICMP message has an 8-byte header and a variable-size data section.

Although the general format of the header is different for each message type, the
first 4 bytes are common to all.

As Figure 3.28 shows, the first field, ICMP type, defines the type of the
message. The code field specifies the reason for the particular message type. The
last common field is the checksum field (to be discussed later in the chapter). The
rest of the header is specific for each message type.

The data section in error messages carries information for finding the original
packet that had the error. In query messages, the data section carries extra
information based on the type of query.

Error Reporting Messages: Since IP is an unreliable protocol, one of the FIGURE 3.28: GENERAL FORMAT OF ICMP MESSAGES
main responsibilities of ICMP is to report some errors that may occur during the
Redirection Message: The redirection message (type 5) is used when the
processing of the IP datagram. ICMP does not correct errors, it simply reports
source uses a wrong router to send out its message. The router redirects the
them.
message to the appropriate router, but informs the source that it needs to change
Error correction is left to the higher-level protocols. Error messages are its default router in the future. The IP address of the default router is sent in the
always sent to the original source because the only information available in the message.
datagram about the route is the source and destination IP addresses.
Parameter Problem: A parameter problem message (type 12) can be sent
ICMP uses the source IP address to send the error message to the source when either there is a problem in the header of a datagram (code 0) or some
(originator) of the datagram. To make the error-reporting process simple, ICMP options are missing or cannot be interpreted (code 1).
follows some rules in reporting messages:
Query Messages: Query messages in ICMP can be used independently
First, no error message will be generated for a datagram having a without relation to an IP datagram. Of course, a query message needs to be
multicast address or special address (such as this host or loopback). encapsulated in a datagram, as a carrier.

Second, no ICMP error message will be generated in response to a Query messages are used to probe or test the liveliness of hosts or routers in
datagram carrying an ICMP error message. the Internet, find the one-way or the round-trip time for an IP datagram between
two devices, or even find out whether the clocks in two devices are synchronized.
Third, no ICMP error message will be generated for a fragmented Naturally, query messages come in pairs: request and reply.
datagram that is not the first fragment.

Destination Unreachable: The most widely used error message is the

destination unreachable (type 3). This message uses different codes (0 to 15) to IGMP:
define the type of error message and the reason why a datagram has not reached
The protocol that is used today for collecting information about group
its final destination.
membership is the Internet Group Management Protocol (IGMP). IGMP is a

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protocol defined at the network layer; it is one of the auxiliary protocols, like ICMP, or hosts. The message is encapsulated in a datagram with the destination
which is considered part of the IP. IGMP messages, like ICMP messages, are address set to the corresponding multicast address. Although all hosts
encapsulated in an IP datagram. receive this message, those not interested drop it.

Messages: Report Message

There are only two types of messages in IGMP version 3, query and report A report message is sent by a host as a response to a query message. The
messages, as shown in Figure 3.29. A query message is periodically sent by a message contains a list of records in which each record gives the identifier of the
router to all hosts attached to it to ask them to report their interests about corresponding group (multicast address) and the addresses of all sources that the
membership in groups. A report message is sent by a host as a response to a query host is interested in receiving messages from (inclusion).
message.
The record can also mention the source addresses from which the host does
not desire to receive a group message (exclusion). The message is encapsulated in
a datagram with the multicast address 224.0.0.22 (multicast address assigned to
IGMPv3).

In IGMPv3, if a host needs to join a group, it waits until it receives a query

message and then sends a report message. If a host needs to leave a group, it
does not respond to a query message. If no other host responds to the
corresponding message, the group is purged from the router database.
FIGURE 3.29: IGMP OPERATION
Propagation of Membership Information:
Query Message:
After a router has collected membership information from the hosts and
The query message is sent by a router to all hosts in each interface to collect other routers at its own level in the tree, it can propagate it to the router located in
information about their membership. There are three versions of query messages, a higher level of the tree. Finally, the router at the tree root can get the
as described below: membership information to build the multicast tree. The process, however, is more
complex than what we can explain in one paragraph. Interested readers can check
a. A general query message is sent about membership in any group. It is the book website for the complete description of this protocol.
encapsulated in a datagram with the destination address 224.0.0.1 (all
hosts and routers). Note that all routers attached to the same network Encapsulation: The IGMP message is encapsulated in an IP datagram with
receive this message to inform them that this message is already sent the value of the protocol field set to 2 and the TTL field set to 1. The destination IP
and that they should refrain from resending it. address of the datagram, however, depends on the type of message, as shown in
figure 3.30.
b. A group-specific query message is sent from a router to ask about the
membership related to a specific group. This is sent when a router does
not receive a response about a specific group and wants to be sure that
there is no active member of that group in the network. The group
identifier (multicast address) is mentioned in the message. The message
is encapsulated in a datagram with the destination address set to the
FIGURE 3.30: DESTINATION IP ADDRESSES
corresponding multicast address. Although all hosts receive this message,
those not interested drop it.

c. A source-and-group-specific query message is sent from a router to

ask about the membership related to a specific group when the message
comes from a specific source or sources. Again the message is sent when
the router does not hear about a specific group related to a specific host
25 26
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9. The network layer protocol of internet is

a) Ethernet b) internet protocol
c) hypertext transfer protocol d) none of the mentioned

10. ICMP is primarily used for

a) error and diagnostic functions b) addressing
c) forwarding
UNIT-III d) none of the mentioned
NETWORK LAYER
OBJECTIVE QUESTIONS 11. The operation of subnet is controlled by .
a. Network Layer.
1. The network layer concerns with b. Data Link Layer
a) bits b) frames c. Data Layer
c) packets d) none of the mentioned d. Transport Layer

2. Which one of the following is not a function of network layer?

a) routing b) inter-networking 12. are two popular examples of distance vector routing protocols.
c) congestion control d) none of the mentioned
a. OSPF and RIP b. RIP and BGP
3. The 4 byte IP address consists of c. BGP and OSPF d. BGP and SPF
a) network address b) host address
c) both (a) and (b) d) none 13. deals with the issues of creating and maintaining routing tables.
4. In virtual circuit network each packet contains a. Forwarding b. Routing
a) full source and destination address b) a short VC number
c. Directing d. None directing
c) both (a) and (b) d) none
14. During an adverse condition, the length of time for every device in the network
5. Which one of the following routing algorithm can be used for network layer to produce an accurate routing table is called the .
design?
a. Accurate time b. integrated time
a) shortest path algorithm b) distance vector routing
c) link state routing d) all the above c. Convergence time d. Average time

6. 6. Multidestination routing
a) is same as broadcast routing b) contains the list of all destinations 15. A routing table contains information entered manually.
c) data is not sent by packets d) none
a. Static b. Dynamic
c. Hierarchical d. Non static
7. A subset of a network that includes all the routers but contains no loops is
called
16. Which of the following is/are the uses of static routing methods?
a) spanning tree b) spider structure
c) spider tree d) none a. To manually define a default route. b. To provide more secure network
environment.
8. Which one of the following algorithm is not used for congestion control? c. To provide more efficient resource utilization. d. All of the above
a) traffic aware routing b) admission control
c) load shedding d) none of the mentioned 17. A routing table is updated periodically using one of the dynamic routing
protocols.
a. static b. dynamic
c. hierarchical d. non static

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27. A one-to-all communication between one source and all hosts on a network is
18.Which of the following is not the category of dynamic routing algorithm? classified as a .
a. Distance vector protocols b. Link state protocols a. unicast b. multicast
c. Hybrid protocols d. Automatic state protocols c. broadcast d. point to point

19. In forwarding, the full IP address of a destination is given in the 28. allow the exchange of summary information between autonomous
routing table. systems.
a. next-hop b. network-specific a. Interior Gateway Protocol (IGP) b. Exterior Gateway Protocol
c. host-specific d. default (EGP)
c. Border Gateway Protocol (BGP) d. Dynamic Gateway Protocol (DGP)
20. To build the routing table, algorithms allow routers to automatically
discover and maintain awareness or the paths through the network. 29).A robust routing protocol provides the ability to .......... build and manage the
a. Static routing b. dynamic routing information in the IP routing table.
c. Hybrid routing d. automatic routing a. Dynamically b. Statically
c. Hierarchically d. All of the above
21. In forwarding, the mask and destination addresses are both 0.0.0.0 in
the routing table. 30. State True of False for definition of an autonomous system(AS).
a. next-hop b. network-specific i) An AS is defined as a physical portion of a larger IP network.
c. host-specific d. default ii) An AS is normally comprised of an internetwork within an organization.
a. i-True, ii-True b. i-True, ii-False
22).To build the routing table, method use preprogrammed definitions c. i-False, ii-True d. i-False, ii-False
representing paths through the network.
a. Static routing b. dynamic routing 31. What are the parameters on which two networks differ.
c. Hybrid routing d. automatic routing a) Packet sized used b) use flow and error control technique
c) Connectionless control and security mechanism d) all
23).In forwarding, the destination addresses is a network address in the
32. are the limitations that cause different networks have different
routing table. packet size.
a. next-hop b. network-specific a) hardware b) operating system
c. host-specific d. default c) protocols d) all

24). allows routers to exchange information within an AS. 33. Fragmentation means
a. Interior Gateway Protocol (IGP) b. Exterior Gateway Protocol a) adding of sma;ll packets to form large packets
b) breaking the large packet into small packets
(EGP)
c) forwarding packet through different networks
c. Border Gateway Protocol (BGP) d. Static Gateway Protocol
d) None
(SGP)
34. The header part of a fragment contains number of fields
25. In forwarding, the routing table holds the address of just the next hop a) 2 b)3 c)1 d)4
instead of complete route information.
a. next-hop b. network-specific 35. The header checksum is the IP header is used to verify .
c. host-specific d. default a) only header b) only data c) both d) None

36. The higest IPV4 address in digital notation is

26. Which of the following is an example of Exterior Gateway Protocol?
a) 255.0.0.0 b)255.255.0.0 c)255.255.255.255
a. Open Short Path First (OSPF) b. Border Gateway Protocol (BGP) d)255.255.255.255
c. Routing Information Protocol (RIP) d. All of the above
37. Which class of IP address is used for used for more networks and less hosts.
29 30
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a) class-A b) class-B c)class-C d) class-E

38. In IPV4 addressing, the digital notation address 222.255.255.255 belongs to

a) class-A b) class-B c)class-C d) class-E

39. classless inter domain routing contains .

a)32 bit IP address b)32 bit mask address c)both d)None
UNIT-III
40. To indicate a sender that,it has no data,to the receiver bit is set Descriptive Questions
a)PSH b)RST c)FIN d) ACK
1. What are the Services provided by Network layer to the Transport layer ?
41. connecting different networks is called
a) intranet working b)internetworking c)multiple networking d)ALL 2. Discuss the functions of the communication subnet to provide datagram service.

42. Bridges are used in layer. 3. What is meant by connection state information in a virtual circuit network?
a) physical b)MAC c)network d)application
4. Compare Virtual-Circuit and Datagram Subnets.
43. Which is a intranet working device
a) router b)gateway c)bridge d)ALL 5. What is routing algorithm? What are the classifications of it?

44. Gateways are used at layer. 6. What is the Optimality Principle?

a) datalink layer b)network layer c)application d)ALL
7. With an example explain shortest path routing algorithm.
45. The max length of the option load field in IP datagram is .
a)40 bytes b)80 bytes c)16 bytes d)any number of bytes multipleof 4 8. Explain flooding

46. The length of the subnet mask is bits . 9. Explain distance vector routing algorithm.
a)16 bits b)32bits c)64bits d)any
10. Explain count-to-infinity problem.
47. Address resolution protocol is used to MAP the IP address on to the . 11. Write short notes on the following
a) data link layer b)internet address c)network address d)port (a) IPV4 (b) IPV6
address 12. Write about Internet Control Protocols.

48. RARP is used to map the data link layer address onto
address.
a)network b)port c)IP d)None

49. which are the following option are used in IPV4.

a) security b)timestamp c)source routing d)ALL

50. Which class of IP addressing provide more number of hosts in each network
a)class-A b)class-B c)class-c d)class-D

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Computer Networks Computer Networks

UNIT-IV
Furthermore, the connectionless transport service is also very similar to
TRANSPORT LAYER: the connectionless network service. However, note that it can be difficult to provide
a connectionless transport service on top of a connection-oriented network service,
The transport layer in the TCP/IP suite is located between the application since it is inefficient to set up a connection to send a single packet and then tear
layer and the network layer. It provides services to the application layer and (meaning run/rip/rush) it down immediately afterwards.
receives services from the network layer.

The transport layer acts as a liaison between a client program and a server
program, a process-to-process connection. The transport layer is the heart of the
TCP/IP protocol suite; it is the end-to-end logical vehicle for transferring data from
one point to another in the Internet.

Introduction:

The transport layer is located between the application layer and the network
layer. It provides a process-to-process communication between two application
layers, one at the local host and the other at the remote host.

Communication is provided using a logical connection, which means that the

two application layers, which can be located in different parts of the globe, assume Figure 4.1: The network, transport, and application layers
that there is an imaginary direct connection through which they can send and
receive messages. Transport service primitives:

THE TRANSPORT SERVICE: To allow users to access the transport service, the transport layer must
provide some operations to application programs, that is, a transport service
Services provided to the upper layers: interface. Each transport service has its own interface.

The ultimate goal of the transport layer is to provide efficient, reliable, and The transport service is similar to the network service, but there are also
cost-effective data transmission service to its users, normally processes in the some important differences. The main difference is that the network service is
application layer. To achieve this, the transport layer makes use of the services intended to model the service offered by real networks and all. Real networks can
provided by the network layer. The software and/or hardware within the transport lose packets, so the network service is generally unreliable.
layer that does the work is called the transport entity.
The connection-oriented transport service, in contrast, is reliable. Of course,
The transport entity can be located in the operating system kernel, in a real networks are not error-free, but that is precisely the purpose of the transport
library package bound into network applications, in a separate user process, or layer—to provide a reliable service on top of an unreliable network.
even on the network interface card. The first two options are most common on the
Internet. The (logical) relationship of the network, transport, and application layers A second difference between the network service and transport service is
is illustrated in Fig. 4.1. whom the services are intended for. The network service is used only by the
transport entities. Few users write their own transport entities, and thus few users
Just as there are two types of network service, connection-oriented and or programs ever (meaning always/forever/still) see the bare network service.
connectionless, there are also two types of transport service. The connection-
oriented transport service is similar to the connection-oriented network service Berkeley sockets: Let us now briefly inspect another set of transport
in many ways. In both cases, connections have three phases: establishment, data primitives, the socket primitives as they are used for TCP. Sockets were first
transfer, and release. Addressing and flow control are also similar in both layers. released as part of the Berkeley UNIX 4.2BSD software distribution in 1983. They
quickly became popular.

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Computer Networks Computer Networks

The primitives are now widely used for Internet programming on many For another thing, the process of establishing a connection over the wire of
operating systems, especially UNIX-based systems, and there is a socket-style API Fig. 4.3(a) is simple: the other end is always there (unless it has crashed, in which
for Windows called ‘‘winsock.’’ The primitives are listed in Fig. 4.2. case it is not there). Either way, there is not much to do.

Even on wireless links, the process is not much different. Just sending a
message is sufficient to have it reach all other destinations. If the message is not
acknowledged due to an error, it can be resent. In the transport layer, initial
connection establishment is complicated.

Addressing:

When an application (e.g., a user) process wishes to set up a connection to a

remote application process, it must specify which one to connect to.
(Connectionless transport has the same problem: to whom should each message be
sent?) The method normally used is to define transport addresses to which
Figure 4.2: The socket primitives for TCP
processes can listen for connection requests.
Note: An Example of Socket Programming: An Internet File Server
In the Internet, these endpoints are called ports. We will use the generic
ELEMENTS OF TRANSPORT PROTOCOLS: term TSAP (Transport Service Access Point) to mean a specific endpoint in the
transport layer. The analogous endpoints in the network layer (i.e., network layer
The transport service is implemented by a transport protocol used between addresses) are naturally called NSAPs (Network Service Access Points). IP
the two transport entities. In some ways, transport protocols resemble the data link addresses are examples of NSAPs.
protocols. Both have to deal with error control, sequencing, and flow control,
among other issues. Figure 4.4 illustrates the relationship between the NSAPs, the TSAPs, and a
transport connection.
However, significant differences between the two also exist. These
differences are due to major dissimilarities between the environments in which the
two protocols operate, as shown in Fig. 4.3.

Figure 4.3: Environment of the (a) data link layer (b) transport layer

At the data link layer, two routers communicate directly via a physical
channel, whether wired or wireless, whereas at the transport layer, this physical
channel is replaced by the entire network.

For one thing, over point-to-point links such as wires or optical fiber, it is
usually not necessary for a router to specify which router it wants to talk to—each
outgoing line leads directly to a particular router. In the transport layer, explicit Figure 4.4: TSAPs, NSAPs, and Transport connections
addressing of destinations is required.

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Computer Networks Computer Networks

Application processes, both clients and servers, can attach themselves to a The sender then times out and sends them all again. This time the packets
local TSAP to establish a connection to a remote TSAP. These connections run take the shortest route and are delivered quickly so the sender releases the
through NSAPs on each host, as shown in figure 4.4. connection.

A possible scenario for a transport connection is as follows: Unfortunately, eventually the initial batch of packets finally come out of
hiding and arrive at the destination in order, asking the bank to establish a new
1. A mail server process attaches itself to TSAP 1522 on host 2 to wait connection and transfer money (again). The bank has no way of telling that these
for an incoming call. A call such as our LISTEN might be used, for are duplicates. It must assume that this is a second, independent transaction, and
example. transfers the money again.

2. An application process on host 1 wants to send an email message, so

The crux (meaning root) of the problem is that the delayed duplicates are
it attaches itself to TSAP 1208 and issues a CONNECT request.
thought to be new packets. We cannot prevent packets from being duplicated and
delayed. But if and when this happens, the packets must be rejected as duplicates
o The request specifies TSAP 1208 on host 1 as the source and
and not processed as fresh packets.
TSAP 1522 on host 2 as the destination. This action ultimately
results in a transport connection being established between the
The problem can be attacked in various ways, none of them very
application process and the server.
satisfactory. One way is to use throwaway transport addresses. In this approach,
3. The application process sends over the mail message. each time a transport address is needed, a new one is generated. When a
connection is released, the address is discarded and never used again. Delayed
4. The mail server responds to say that it will deliver the message. duplicate packets then never find their way to a transport process and can do no
damage.
5. The transport connection is released.
Note: However, this approach makes it more difficult to connect with a process in
Connection Establishment:
the first place.
Establishing a connection sounds easy, but it is actually surprisingly tricky. At
Another possibility is to give each connection a unique identifier (i.e., a
first glance, it would seem sufficient for one transport entity to just send a
sequence number incremented for each connection established) chosen by the
CONNECTION REQUEST segment to the destination and wait for a CONNECTION
initiating party and put in each segment, including the one requesting the
ACCEPTED reply. The problem occurs when the network can lose, delay, corrupt,
connection.
and duplicate packets. This behavior causes serious complications.
After each connection is released, each transport entity can update a table
Imagine a network that is so congested that acknowledgements hardly ever
listing obsolete connections as (peer transport entity, connection identifier) pairs.
get back in time and each packet times out and is retransmitted two or three times.
Whenever a connection request comes in, it can be checked against the table to see
Suppose that the network uses datagrams inside and that every packet follows a
if it belongs to a previously released connection.
different route.
Unfortunately, this scheme has a basic flaw: it requires each transport entity
Some of the packets might get stuck in a traffic jam inside the network and
to maintain a certain amount of history information indefinitely. This history must
take a long time to arrive. That is, they may be delayed in the network and pop out
persist at both the source and destination machines. Otherwise, if a machine
much later, when the sender thought that they had been lost.
crashes and loses its memory, it will no longer know which connection identifiers
have already been used by its peers.
The worst possible nightmare is as follows. A user establishes a connection
with a bank, sends messages telling the bank to transfer a large amount of money
Instead, we need to take a different tack to simplify the problem. Rather
to the account of a not-entirely-trustworthy person. Unfortunately, the packets
than allowing packets to live forever within the network, we devise a mechanism to
decide to take the scenic route to the destination and go off exploring a remote
kill off aged packets that are still hobbling about.
corner of the network.

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Computer Networks Computer Networks

Packet lifetime can be restricted to a known maximum using one (or more) When it arrives, the recipient sends back a DR segment and starts a timer,
of the following techniques: just in case its DR is lost. When this DR arrives, the original sender sends back an
ACK segment and releases the connection.
1. Restricted network design.

2. Putting a hop counter in each packet.

3. Timestamping each packet.

TCP uses three-way handshake to establish connections in the presence of

delayed duplicate control segments as shown in figure 4.5.

Connection Release:

Releasing a connection is easier than establishing one. There are two styles
of terminating a connection: asymmetric release and symmetric release.

Asymmetric release is the way the telephone system works: when one
party hangs up, the connection is broken.

Symmetric release treats the connection as two separate unidirectional

connections and requires each one to be released separately.

Asymmetric release is abrupt and may result in data loss. Consider the
scenario of Fig. 4.6. After the connection is established, host 1 sends a segment
that arrives properly at host 2. Then host 1 sends another segment.

Unfortunately, host 2 issues a DISCONNECT before the second segment

arrives. The result is that the connection is released and data are lost.

Symmetric release does the job when each process has a fixed amount of Figure 4.5: Three protocol scenarios for establishing a connection
data to send and clearly knows when it has sent it. In other situations, determining using a three-way handshake. CR denotes Connection Request. (a) normal
that all the work has been done and the connection should be terminated is not so operation. (b) old duplicate connection request appearing out of nowhere.
obvious. (c) duplicate connection request and duplicate ack.

One can envision a protocol in which host 1 says ‘‘I am done. Are you done Finally, when the ACK segment arrives, the receiver also releases the
too?’’ If host 2 responds: ‘‘I am done too. Goodbye, the connection can be safely connection. Releasing a connection means that the transport entity removes the
released.’’ information about the connection from its table of currently open connections and
signals the connection’s owner (the transport user) somehow.
In practice, we can avoid this quandary (meaning dilemma/difficulty) by
foregoing the need for agreement and pushing the problem up to the transport If the final ACK segment is lost, as shown in Fig. 4.7(b), the situation is
user, letting each side independently decide when it is done. This is an easier saved by the timer. When the timer expires, the connection is released anyway.
problem to solve. Now consider the case of the second DR being lost.

Figure 4.7 illustrates four scenarios of releasing using a three-way The user initiating the disconnection will not receive the expected response,
handshake. While this protocol is not infallible, it is usually adequate. In Fig. 4.7(a), will time out, and will start all over again. In Fig. 4.7(c), we see how this works,
we see the normal case in which one of the users sends a DR (DISCONNECTION assuming that the second time no segments are lost and all segments are delivered
REQUEST) segment to initiate the connection release. correctly and on time.

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Computer Networks Computer Networks

Figure 4.6: Abrupt disconnection with loss of data

Our last scenario, Fig. 4.7(d), is the same as Fig. 4.7(c) except that now we
assume all the repeated attempts to retransmit the DR also fail due to lost
segments. After N retries, the sender just gives up and releases the connection.
Meanwhile, the receiver times out and also exits.

Error control and Flow control:

Figure 4.7: Four protocol scenarios for releasing a connection. (a) normal
case of three-way handshake. (b) final ACK lost. (c) response lost. (d) response
Error control is ensuring that the data is delivered with the desired level of lost and subsequent DRs lost.
reliability, usually that all of the data is delivered without any errors. Flow control is
keeping a fast transmitter from overrunning a slow receiver.

MULTIPLEXING:

Multiplexing, or sharing several conversations over connections, virtual

circuits, and physical links plays a role in several layers of the network architecture.
In the transport layer, the need for multiplexing can arise in a number of ways. For
example, if only one network address is available on a host, all transport
connections on that machine have to use it.

When a segment comes in, some way is needed to tell which process to give
it to. This situation, called multiplexing, is shown in Fig. 4.8(a). In this figure, four
distinct transport connections all use the same network connection (e.g., IP
address) to the remote host.
Multiplexing can also be useful in the transport layer for another reason.
Figure 4.8: (A) Multiplexing (B) Inverse Multiplexing
Suppose, for example, that a host has multiple network paths that it can use. If a
user needs more bandwidth or more reliability than one of the network paths can
provide, a way out is to have a connection that distributes the traffic among
multiple network paths on a round-robin basis, as indicated in Fig. 4.8(b).

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Computer Networks Computer Networks

CRASH RECOVERY: Max-Min Fairness:

If hosts and routers are subject to crashes or connections are long-lived In the preceding discussion, we did not talk about how to divide bandwidth
(e.g., large software or media downloads), recovery from these crashes becomes between different transport senders. This sounds like a simple question to answer—
an issue. give all the senders an equal fraction of the bandwidth—but it involves several
considerations.
If the transport entity is entirely within the hosts, recovery from network and
router crashes is straightforward. The transport entities expect lost segments all Perhaps the first consideration is to ask what this problem has to do with
the time and know how to cope with them by using retransmissions. congestion control.

A more troublesome problem is how to recover from host crashes. In A second consideration is what a fair portion means for flows in a network. It
particular, it may be desirable for clients to be able to continue working when is simple enough if N flows use a single link, in which case they can all have 1/N of
servers crash and quickly reboot. the bandwidth (although efficiency will dictate that they use slightly less if the
traffic is bursty).
CONGESTION CONTROL:
But what happens if the flows have different, but overlapping, network
If the transport entities on many machines send too many packets into the paths? For example, one flow may cross three links, and the other flows may cross
network too quickly, the network will become congested, with performance one link. The three-link flow consumes more network resources. It might be fairer
degraded as packets are delayed and lost. in some sense to give it less bandwidth than the one-link flows.

Controlling congestion to avoid this problem is the combined responsibility of The form of fairness that is often desired for network usage is max-min
the network and transport layers. Congestion occurs at routers, so it is detected at fairness. An allocation is max-min fair if the bandwidth given to one flow cannot be
the network layer. increased without decreasing the bandwidth given to another flow with an allocation
that is no larger.
However, congestion is ultimately caused by traffic sent into the network by
the transport layer. The only effective way to control congestion is for the transport Convergence:
protocols to send packets into the network more slowly.
A final criterion is that the congestion control algorithm converge quickly to a
DESIRABLE BANDWIDTH ALLOCATION: fair and efficient allocation of bandwidth. The discussion of the desirable operating
point above assumes a static network environment.
Before we describe how to regulate traffic, we must understand what we are
trying to achieve by running a congestion control algorithm. That is, we must However, connections are always coming and going in a network, and the
specify the state in which a good congestion control algorithm will operate the bandwidth needed by a given connection will vary over time too. Because of the
network. variation in demand, the ideal operating point for the network varies over time.

The goal is more than to simply avoid congestion. It is to find a good A good congestion control algorithm should rapidly converge to the ideal
allocation of bandwidth to the transport entities that are using the network. A good operating point, and it should track that point as it changes over time. If the
allocation will deliver good performance because it uses all the available bandwidth convergence is too slow, the algorithm will never be close to the changing operating
but avoids congestion, it will be fair across competing transport entities, and it will point. If the algorithm is not stable, it may fail to converge to the right point in
quickly track changes in traffic demands. some cases, or even oscillate around the right point.

Efficiency and Power: Regulating the sending rate:

An efficient allocation of bandwidth across transport entities will use all of the Now it is time to regulate the sending rates to obtain a desirable bandwidth
network capacity that is available. However, it is not quite right to think that if allocation. The sending rate may be limited by two factors.
there is a 100-Mbps link, five transport entities should get 20 Mbps each. They
should usually get less than 20 Mbps for good performance.

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The first is flow control, in the case that there is insufficient buffering at Wireless networks lose packets all the time due to transmission errors. To
the receiver. function well, the only packet losses that the congestion control algorithm should
observe are losses due to insufficient bandwidth, not losses due to transmission
The second is congestion, in the case that there is insufficient capacity in errors. One solution to this problem is to mask the wireless losses by using
the network. retransmissions over the wireless link.

In Fig. 4.9, we see this problem illustrated hydraulically. In Fig. 4.9(a), we

THE INTERNET TRANSPORT PROTOCOLS:
see a thick pipe leading to a small-capacity receiver. This is a flow-control limited
situation. As long as the sender does not send more water than the bucket can
UDP:
contain, no water will be lost.
The Internet has two main protocols in the transport layer, a connectionless
In Fig. 4.9(b), the limiting factor is not the bucket capacity, but the internal
protocol and a connection-oriented one. The protocols complement each other.
carrying capacity of the network. If too much water comes in too fast, it will back
up and some will be lost (in this case, by overflowing the funnel).
The connectionless protocol is UDP. It does almost nothing beyond sending
packets between applications, letting applications build their own protocols on top
The way that a transport protocol should regulate the sending rate depends
as needed.
on the form of the feedback returned by the network. Different network layers may
return different kinds of feedback. The feedback may be explicit or implicit, and it The connection-oriented protocol is TCP. It does almost everything. It makes
may be precise or imprecise. connections and adds reliability with retransmissions, along with flow control and
congestion control, all on behalf of the applications that use it.

INTRODUCTION TO UDP:

The Internet protocol suite supports a connectionless transport protocol

called UDP (User Datagram Protocol).

UDP provides a way for applications to send encapsulated IP datagrams

without having to establish a connection. UDP is described in RFC 768.

UDP transmits segments consisting of an 8-byte header followed by the

payload. The header is shown in Fig. 4.10. The two ports serve to identify the
endpoints within the source and destination machines.

When a UDP packet arrives, its payload is handed to the process attached to
the destination port. This attachment occurs when the BIND primitive or something
similar is used.

Figure 4.9: (a) a fast network feeding a low-capacity receiver. (b) a

slow network feeding a high-capacity receiver.

Wireless issues:
Figure 4.10: the UDP header
Transport protocols such as TCP that implement congestion control should be
independent of the underlying network and link layer technologies. That is a good Think of ports as mailboxes that applications can rent to receive packets. In
theory, but in practice there are issues with wireless networks. The main issue is fact, the main value of UDP over just using raw IP is the addition of the source and
that packet loss is often used as a congestion signal, including by TCP. destination ports.

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Without the port fields, the transport layer would not know what to do with
each incoming packet. With them, it delivers the embedded segment to the correct
application.

The source port is primarily needed when a reply must be sent back to the
source. By copying the Source port field from the incoming segment into the
Destination port field of the outgoing segment, the process sending the reply can
specify which process on the sending machine is to get it.

The UDP length field includes the 8-byte header and the data. The minimum
length is 8 bytes, to cover the header. The maximum length is 65,515 bytes, which
is lower than the largest number that will fit in 16 bits because of the size limit on
IP packets.

An optional Checksum is also provided for extra reliability. It checksums the Figure 4.12: Steps in making a remote procedure call, the stubs are
shaded
header, the data, and a conceptual IP pseudoheader. When performing this
computation, the Checksum field is set to zero and the data field is padded out with

an additional zero byte if its length is an odd number. Step 4 is the operating system passing the incoming packet to the server stub.

The checksum algorithm is simply to add up all the 16-bit words in one’s 
complement and to take the one’s complement of the sum. Finally, step 5 is the server stub calling the server procedure with the
unmarshaled parameters.
Remote procedure call: The reply traces the same path in the other direction.

In a certain sense, sending a message to a remote host and getting a reply The key item to note here is that the client procedure, written by the user,
back is a lot like making a function call in a programming language. The idea just makes a normal (i.e., local) procedure call to the client stub, which has the
behind RPC is to make a remote procedure call look as much as possible like a local same name as the server procedure. Since the client procedure and client stub are
one. in the same address space, the parameters are passed in the usual way.

In the simplest form, to call a remote procedure, the client program must be Similarly, the server procedure is called by a procedure in its address space
bound with a small library procedure, called the client stub, that represents the with the parameters it expects. To the server procedure, nothing is unusual.
server procedure in the client’s address space.
Real-Time Transport Protocols
Similarly, the server is bound with a procedure called the server stub.
These procedures hide the fact that the procedure call from the client to the server Client-server RPC is one area in which UDP is widely used. Another one is for
is not local. The actual steps in making an RPC are shown in Fig. 4.12. real-time multimedia applications.

 In particular, as Internet radio, Internet telephony, music-on-demand,

Step 1 is the client calling the client stub. This call is a local procedure call, with
the parameters pushed onto the stack in the normal way. videoconferencing, video-on-demand, and other multimedia applications became
more commonplace, people have discovered that each application was reinventing
 more or less the same real-time transport protocol. Thus was RTP (Real-time
Step 2 is the client stub packing the parameters into a message and
making a system call to send the message. Packing the parameters is Transport Protocol) born.
called marshaling.
It is described in RFC 3550 and is now in widespread use for multimedia

Step 3 is the operating system sending the message from the client machine to applications. There are two aspects of real-time transport . The first is the RTP
the server machine.
protocol for transporting audio and video data in packets. The second is the
processing that takes place, mostly at the receiver, to play out the audio and video
at the right time.

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Computer Networks Computer Networks

RTP—The Real-Time Transport Protocol: to. It is the method used to multiplex and demultiplex multiple data streams onto a
single stream of UDP packets.
The basic function of RTP is to multiplex several real-time data streams onto
a single stream of UDP packets. The UDP stream can be sent to a single destination Finally, the Contributing source identifiers, if any, are used when mixers are
(unicasting) or to multiple destinations (multicasting). present.

Because RTP just uses normal UDP, its packets are not treated specially by RTCP—The Real-time Transport Control Protocol
the routers unless some normal IP quality-of-service features are enabled. In
particular, there are no special guarantees about delivery, and packets may be lost, RTP has a little sister protocol (little sibling protocol?) called RTCP
delayed, corrupted, etc. (Realtime Transport Control Protocol). It is defined along with RTP in RFC 3550
and handles feedback, synchronization, and the user interface. It does not transport
The RTP format contains several features to help receivers work with any media samples.
multimedia information. The RTP header is illustrated in Fig. 4.13. It consists of
three 32-bit words and potentially some extensions. THE INTERNET TRANSPORT PROTOCOLS:

TCP

UDP is a simple protocol and it has some very important uses, such as
clientserver interactions and multimedia, but for most Internet applications,
reliable, sequenced delivery is needed. UDP cannot provide this, so another protocol
is required. It is called TCP and is the main workhorse of the Internet.

Introduction to TCP:

TCP (Transmission Control Protocol) was specifically designed to provide

a reliable end-to-end byte stream over an unreliable internetwork. An internetwork
differs from a single network because different parts may have wildly different
Figure 4.13: The RTP header topologies, bandwidths, delays, packet sizes, and other parameters.
The first word contains the Version field, which is already at 2.
TCP was designed to dynamically adapt to properties of the internetwork and
The P bit indicates that the packet has been padded to a multiple of 4 bytes. to be robust in the face of many kinds of failures. TCP was formally defined in RFC
793 in September 1981.
The X bit indicates that an extension header is present.
As time went on, many improvements have been made, and various errors
The CC field tells how many contributing sources are present, from 0 to 15. and inconsistencies have been fixed. To give you a sense of the extent of TCP, the
important RFCs are now RFC 793 plus: clarifications and bug fixes in RFC 1122;
The M bit is an application-specific marker bit. It can be used to mark the extensions for high-performance in RFC 1323.
start of a video frame, the start of a word in an audio channel, or something else
that the application understands. Selective acknowledgements in RFC 2018; congestion control in RFC 2581;
repurposing of header fields for quality of service in RFC 2873; improved
The Payload type field tells which encoding algorithm has been used (e.g., retransmission timers in RFC 2988; and explicit congestion notification in RFC 3168.
uncompressed 8-bit audio, MP3, etc.).
The IP layer gives no guarantee that datagrams will be delivered properly, nor any
indication of how fast datagrams may be sent.
The Sequence number is just a counter that is incremented on each RTP
packet sent. It is used to detect lost packets.

The Timestamp is produced by the stream’s source to note when the first
sample in the packet was made.

The Synchronization source identifier tells which stream the packet belongs

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The TCP Protocol:

It is up to TCP to send datagrams fast enough to make use of the capacity A key feature of TCP, and one that dominates the protocol design, is that
but not cause congestion, and to time out and retransmit any datagrams that are every byte on a TCP connection has its own 32-bit sequence number. When the
not delivered. Datagrams that do arrive may well do so in the wrong order; it is
Internet began, the lines between routers were mostly 56-kbps leased lines, so a
also up to TCP to reassemble them into messages in the proper sequence.
host blasting away at full speed took over 1 week to cycle through the sequence
The TCP Service Model: numbers.

TCP service is obtained by both the sender and the receiver creating end The sending and receiving TCP entities exchange data in the form of
points, called sockets. Each socket has a socket number (address) consisting of segments. A TCP segment consists of a fixed 20-byte header (plus an optional
the IP address of the host and a 16-bit number local to that host, called a port. A part) followed by zero or more data bytes. The TCP software decides how big
port is the TCP name for a TSAP. segments should be.

For TCP service to be obtained, a connection must be explicitly established It can accumulate data from several writes into one segment or can split data
between a socket on one machine and a socket on another machine. A socket may from one write over multiple segments. Two limits restrict the segment size. First,
be used for multiple connections at the same time. In other words, two or more each segment, including the TCP header, must fit in the 65,515- byte IP payload.
connections may terminate at the same socket. Second, each link has an MTU (Maximum Transfer Unit).

Port numbers below 1024 are reserved for standard services that can usually Each segment must fit in the MTU at the sender and receiver so that it can
only be started by privileged users (e.g., root in UNIX systems). They are called be sent and received in a single, unfragmented packet. However, it is still possible
well-known ports. for IP packets carrying TCP segments to be fragmented when passing over a
network path for which some link has a small MTU.
For example, any process wishing to remotely retrieve mail from a host can
connect to the destination host’s port 143 to contact its IMAP daemon. The list of If this happens, it degrades performance and causes other problems.
well-known ports is given at www.iana.org. Over 700 have been assigned. A few of Instead, modern TCP implementations perform path MTU discovery by using the
the better-known ones are listed in Fig. 4.14. technique outlined in RFC 1191. This technique uses ICMP error messages to find
the smallest MTU for any link on the path. TCP then adjusts the segment size
downwards to avoid fragmentation.

The basic protocol used by TCP entities is the sliding window protocol with a
dynamic window size. When a sender transmits a segment, it also starts a timer.
When the segment arrives at the destination, the receiving TCP entity sends back a
segment (with data if any exist, and otherwise without) bearing an
acknowledgement number equal to the next sequence number it expects to receive
and the remaining window size.

If the sender’s timer goes off before the acknowledgement is received, the
sender transmits the segment again.

Figure 4.14: Some assigned ports The TCP Segment Header:

All TCP connections are full duplex and point-to-point. Full duplex means that Figure 4.15 shows the layout of a TCP segment. Every segment begins with a
traffic can go in both directions at the same time. Point-to-point means that each fixed-format, 20-byte header. The fixed header may be followed by header options.
connection has exactly two end points. TCP does not support multicasting or After the options, if any, up to 65,535 − 20 − 20 = 65,495 data bytes may follow,
broadcasting. where the first 20 refer to the IP header and the second to the TCP header.
A TCP connection is a byte stream, not a message stream. Message
oundaries are not preserved end to end.

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Segments without any data are legal and are commonly used for The ACK bit is set to 1 to indicate that the Acknowledgement number is valid.
acknowledgements and control messages. This is the case for nearly all packets. If ACK is 0, the segment does not contain an
acknowledgement, so the Acknowledgement number field is ignored.

The PSH bit indicates PUSHed data. The receiver is hereby kindly requested
to deliver the data to the application upon arrival and not buffer it until a full buffer
has been received (which it might otherwise do for efficiency).

The RST bit is used to abruptly reset a connection that has become confused
due to a host crash or some other reason.

The SYN bit is used to establish connections. The FIN bit is used to release a
connection.

The Window size field tells how many bytes may be sent starting at the byte
acknowledged.

A Checksum is also provided for extra reliability. The Options field provides a
way to add extra facilities not covered by the regular header.

Figure 4.15: The TCP Header TCP Connection Establishment:

The Source port and Destination port fields identify the local end points of Connections are established in TCP by means of the three-way handshake.
the connection. The source and destination end points together identify the To establish a connection, one side, say, the server, passively waits for an incoming
connection. This connection identifier is called a 5 tuple because it consists of five connection by executing the LISTEN and ACCEPT primitives in that order, either
specifying a specific source or nobody in particular.
pieces of information: the protocol (TCP), source IP and source port, and
destination IP and destination port.
The other side, say, the client, executes a CONNECT primitive, specifying the
The Sequence number and Acknowledgement number fields perform their IP address and port to which it wants to connect, the maximum TCP segment size it
usual functions. is willing to accept, and optionally some user data (e.g., a password). The
CONNECT primitive sends a TCP segment with the SYN bit on and ACK bit off and
The Sequence number and Acknowledgement number fields perform their waits for a response.
usual functions.
When this segment arrives at the destination, the TCP entity there checks to
The TCP header length tells how many 32-bit words are contained in the TCP see if there is a process that has done a LISTEN on the port given in the Destination
header. This information is needed because the Options field is of variable length, port field. If not, it sends a reply with the RST bit on to reject the connection.
so the header is, too.
TCP Connection Release
Now come eight 1-bit flags. CWR and ECE are used to signal congestion
when ECN (Explicit Congestion Notification) is used. CWR is set to signal Congestion Although TCP connections are full duplex, to understand how connections are
Window Reduced from the TCP sender to the TCP receiver so that it knows the released it is best to think of them as a pair of simplex connections. Each simplex
sender has slowed down and can stop sending the ECN-Echo. connection is released independently of its sibling.

URG is set to 1 if the Urgent pointer is in use. The Urgent pointer is used to To release a connection, either party can send a TCP segment with the FIN
indicate a byte offset from the current sequence number at which urgent data are bit set, which means that it has no more data to transmit. When the FIN is
to be found. acknowledged, that direction is shut down for new data.

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Measuring network performance and parameters has many potential pitfalls.

We list a few of them here. Any systematic attempt to measure network
performance should be careful to avoid these.
Data may continue to flow indefinitely in the other direction, however. When
both directions have been shut down, the connection is released. 1) Make Sure That the Sample Size Is Large Enough
TCP Congestion Control:
Do not measure the time to send one segment, but repeat the measurement,
say, one million times and take the average.
The network layer detects congestion when queues grow large at routers and
tries to manage it, if only by dropping packets. It is up to the transport layer to 2) Make Sure That the Samples Are Representative
receive congestion feedback from the network layer and slow down the rate of
traffic that it is sending into the network. Ideally, the whole sequence of one million measurements should be repeated
at different times of the day and the week to see the effect of different network
In the Internet, TCP plays the main role in controlling congestion, as well as conditions on the measured quantity.
the main role in reliable transport. That is why it is such a special protocol.
3) Caching Can Wreak Havoc with Measurements
PERFORMANCE PROBLEMS IN COMPUTER NETWORKS
Repeating a measurement many times will return an unexpectedly fast
Some performance problems, such as congestion, are caused by temporary answer if the protocols use caching mechanisms.
resource overloads. If more traffic suddenly arrives at a router than the router can
handle, congestion will build up and performance will suffer. 4) Be Sure That Nothing Unexpected Is Going On during Your Tests

Performance also degrades when there is a structural resource imbalance. Making measurements at the same time that some user has decided to run a
For example, if a gigabit communication line is attached to a low-end PC, the poor video conference over your network will often give different results than if there is
host will not be able to process the incoming packets fast enough and some will be no video conference.
lost. These packets will eventually be retransmitted, adding delay, wasting
5) Be Careful When Using a Coarse-Grained Clock
bandwidth, and generally reducing performance.
Computer clocks function by incrementing some counter at regular intervals.
Overloads can also be synchronously triggered. As an example, if a segment
contains a bad parameter , in many cases the receiver will thoughtfully send back 6) Be Careful about Extrapolating the Results
an error notification.
Suppose that you make measurements with simulated network loads running
Another tuning issue is setting timeouts. When a segment is sent, a timer is from 0 (idle) to 0.4 (40% of capacity).
set to guard against loss of the segment. If the timeout is set too short,
unnecessary retransmissions will occur, clogging the wires. If the timeout is set too
long, unnecessary delays will occur after a segment is lost.

NETWORK PERFORMANCE MEASUREMENT:

When a network performs poorly, its users often complain to the folks
running it, demanding improvements. To improve the performance, the operators
must first determine exactly what is going on. To find out what is really happening,
the operators must make measurements.

Measurements can be made in different ways and at many locations (both in

the protocol stack and physically). The most basic kind of measurement is to start a
timer when beginning some activity and see how long that activity takes.

Other measurements are made with counters that record how often some
event has happened (e.g., number of lost segments).

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UNIT-V
Standard Application-Layer Protocols:
INTRODUCTION TO APPLICATION LAYER:
There are several application-layer protocols that have been standardized
INTRODUCTION: and documented by the Internet authority, and we are using them in our daily
interaction with the Internet.
The application layer provides services to the user. Communication is
provided using a logical connection, which means that the two application layers Each standard protocol is a pair of computer programs that interact with the
assume that there is an imaginary direct connection through which they can send user and the transport layer to provide a specific service to the user.
and receive messages.
Nonstandard Application-Layer Protocols:
Providing Services:
A programmer can create a nonstandard application-layer program if she can
All communication networks that started before the Internet were designed write two programs that provide service to the user by interacting with the
to provide services to network users. Most of these networks, however, were transport layer.
originally designed to provide one specific service. For example, the telephone
Application-Layer Paradigms
network was originally designed to provide voice service: to allow people all over
the world to talk to each other. This network, however, was later used for some It should be clear that to use the Internet we need two application programs
other services, such as facsimile (fax), enabled by users adding some extra to interact with each other: one running on a computer somewhere in the world,
hardware at both ends. the other running on another computer somewhere else in the world. The two
programs need to send messages to each other through the Internet infrastructure.
The Internet was originally designed for the same purpose: to provide service
to users around the world. The layered architecture of the TCP/IP protocol suite, However, we have not discussed what the relationship should be between
however, makes the Internet more flexible than other communication networks these programs.
such as postal or telephone networks.
Should both application programs be able to request services and provide
Each layer in the suite was originally made up of one or more protocols, but services, or should the application programs just do one or the other?
new protocols can be added or some protocols can be removed or replaced by the
Internet authorities. However, if a protocol is added to each layer, it should be Two paradigms have been developed during the lifetime of the Internet to
designed in such a way that it uses the services provided by one of the protocols at answer this question: the client-server paradigm and the peer-to-peer paradigm.
the lower layer.
Traditional Paradigm: Client-Server:
If a protocol is removed from a layer, care should be taken to change the
The traditional paradigm is called the client-server paradigm. It was the
protocol at the next higher layer that supposedly uses the services of the removed
most popular paradigm until a few years ago. In this paradigm, the service provider
protocol. The application layer, however, is somewhat different from other layers in
is an application program, called the server process; it runs continuously, waiting
that it is the highest layer in the suite.
for another application program, called the client process, to make a connection
The protocols in this layer do not provide services to any other protocol in through the Internet and ask for service.
the suite; they only receive services from the protocols in the transport layer. This
means that protocols can be removed from this layer easily. New protocols can be There are normally some server processes that can provide a specific type of
service, but there are many clients that request service from any of these server
also added to this layer as long as the new protocols can use the services provided
processes. The server process must be running all the time; the client process is
by one of the transport-layer protocols.
started when the client needs to receive service.

Standard and Nonstandard Protocols:

New Paradigm: Peer-to-Peer:
To provide smooth operation of the Internet, the protocols used in the first
four layers of the TCP/IP suite need to be standardized and documented.
A new paradigm, called the peer-to-peer paradigm (often abbreviated P2P

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paradigm) has emerged to respond to the needs of some new applications. suite.

In this paradigm, there is no need for a server process to be running all the Several APIs have been designed for communication. One of the most
time and waiting for the client processes to connect. The responsibility is shared common one is: socket interface. The socket interface is a set of instructions that
between peers. provide communication between the application layer and the operating system, as
shown in Figure 5.1.
A computer connected to the Internet can provide service at one time and
receive service at another time. A computer can even provide and receive services
at the same time.

CLIENT-SERVER PROGRAMMING:

In a client-server paradigm, communication at the application layer is

between two running application programs called processes: a client and a server.

A client is a running program that initializes the communication by sending a

request; a server is another application program that waits for a request from a
client. FIGURE 5.1: Position Of The Socket Interface

The server handles the request received from a client, prepares a result, and It is a set of instructions that can be used by a process to communicate with
sends the result back to the client. This definition of a server implies that a server another process. The idea of sockets allows us to use the set of all instructions
must be running when a request from a client arrives, but the client needs to be already designed in a programming language for other sources and sinks.
run only when it is needed.
For example, in most computer languages, like C, C++, or Java, we have
This means that if we have two computers connected to each other several instructions that can read and write data to other sources and sinks such as
somewhere, we can run a client process on one of them and the server on the a keyboard (a source), a monitor (a sink), or a file (source and sink). We can use
other. However, we need to be careful that the server program is started before we the same instructions to read from or write to sockets.
start running the client program.
Sockets:
Application Programming Interface:
Although a socket is supposed to behave like a terminal or a file, it is not a
A client process communicate with a server process with the help of a physical entity like them; it is an abstraction. It is an object that is created and
computer program which is normally written in a computer language with a used by the application program.
predefined set of instructions that tells the computer what to do.
Socket Addresses:
A computer language has a set of instructions for mathematical operations, a
The interaction between a client and a server is two-way communication. In
set of instructions for string manipulation, a set of instructions for input/output
a two-way communication, we need a pair of addresses: local (sender) and remote
access, and so on.
(receiver). The local address in one direction is the remote address in the other
If we need a process to be able to communicate with another process, we direction and vice versa.
need a new set of instructions to tell the lowest four layers of the TCP/IP suite to
Since communication in the client-server paradigm is between two sockets,
open the connection, send and receive data from the other end, and close the
we need a pair of socket addresses for communication: a local socket address
connection. A set of instructions of this kind is normally referred to as an
and a remote socket address. However, we need to define a socket address in
application programming interface (API).
terms of identifiers used in the TCP/IP protocol suite.
An interface in programming is a set of instructions between two entities. In
A socket address should first define the computer on which a client or a
this case, one of the entities is the process at the application layer and the other is
server is running. Socket address should be a combination of an IP address (32 bit)
the operating system that encapsulates the first four layers of the TCP/IP protocol

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and a port number (16 bit). A pair of processes provide services to the users of the Internet, human or
Since a socket defines the end-point of the communication, we can say that a programs. A pair of processes, however, need to use the services provided by the
socket is identified by a pair of socket addresses, a local and a remote. transport layer for communication because there is no physical communication at
the application layer.
Finding Socket Addresses: How can a client or a server find a pair of
socket addresses for communication? The situation is different for each site. WORLD WIDE WEB AND HTTP:

Server Site: The server needs a local (server) and a remote (client) socket World Wide Web:
address for communication.
The idea of the Web was first proposed by Tim Berners-Lee in 1989. The Web
Local Socket Address The local (server) socket address is provided by the today is a repository of information in which the documents, called web pages, are
operating system. The operating system knows the IP address of the computer on distributed all over the world and related documents are linked together.
which the server process is running. The port number of a server process, however,
needs to be assigned. The popularity and growth of the Web can be related to two terms in the
above statement: distributed and linked. Distribution allows the growth of the Web.
If the server process is a standard one defined by the Internet authority, a Each web server in the world can add a new web page to the repository and
port number is already assigned to it. For example, the assigned port number for a announce it to all Internet users without overloading a few servers.
Hypertext Transfer Protocol (HTTP) is the integer 80, which cannot be used by any
other process. Linking allows one web page to refer to another web page stored in another
server somewhere else in the world. The linking of web pages was achieved using a
Remote Socket Address The remote socket address for a server is the concept called hypertext, which was introduced many years before the advent of
socket address of the client that makes the connection. Since the server can serve the Internet.
many clients, it does not know beforehand the remote socket address for
communication. The idea was to use a machine that automatically retrieved another
document stored in the system when a link to it appeared in the document. The
The server can find this socket address when a client tries to connect to the Web implemented this idea electronically to allow the linked document to be
server. The client socket address, which is contained in the request packet sent to retrieved when the link was clicked by the user.
the server, becomes the remote socket address that is used for responding to the
client. Today, the term hypertext, coined to mean linked text documents, has been
changed to hypermedia, to show that a web pagecan be a text document, an
Client Site: The client also needs a local (client) and a remote (server) image, an audio file, or a video file.
socket address for communication.
Architecture:
Local Socket Address The local (client) socket address is also provided by
the operating system. The operating system knows the IP address of the computer The WWW today is a distributed client-server service, in which a client using
on which the client is running. The port number, however, is a 16-bit temporary a browser can access a service using a server. However, the service provided is
integer that is assigned to a client process each time the process needs to start the distributed over many locations called sites. Each site holds one or more web
communication. pages.

The port number, however, needs to be assigned from a set of integers Each web page, however, can contain some links to other web pages in the
defined by the Internet authority and called the ephemeral (temporary) port same or other sites. In other words, a web page can be simple or composite. A
numbers. The operating system, however, needs to guarantee that the new port simple web page has no links to other web pages; a composite web page has one
number is not used by any other running client process. or more links to other web pages. Each web page is a file with a name and address.

Web Client (Browser): A variety of vendors offer commercial browsers

Remote Socket Address Finding the remote (server) socket address for a
client, however, needs more work. When a client process starts, it should know the that interpret and display a web page, and all of them use nearly the same
socket address of the server it wants to connect to. architecture. Each browser usually consists of three parts: a controller, client
Using Services of the Transport Layer: protocols, and interpreters (figure 5.2).

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Port. The port, a 16-bit integer, is normally predefined for the client-server
application.

Path. The path identifies the location and the name of the file in the
underlying operating system. The format of this identifier normally depends on the
operating system.
Figure 5.2: Browser
The controller receives input from the keyboard or the mouse and uses the To combine these four pieces together, the uniform resource locator
client programs to access the document. After the document has been accessed, (URL) has been designed; it uses three different separators between the four
the controller uses one of the interpreters to display the document on the screen. pieces as shown below:

The client protocol can be one of the protocols described later, such as HTTP
or FTP. The interpreter can be HTML, Java, or JavaScript, depending on the type of
document. Some commercial browsers include Internet Explorer, Netscape
Navigator, and Firefox. Web Documents:

Web Server: The web page is stored at the server. Each time a request The documents in the WWW can be grouped into three broad categories:
static, dynamic, and active.
arrives, the corresponding document is sent to the client. To improve efficiency,
servers normally store requested files in a cache in memory; memory is faster to Static Documents:
access than a disk.
Static documents are fixed-content documents that are created and stored
A server can also become more efficient through multithreading or in a server. The client can get a copy of the document only. In other words, the
multiprocessing. In this case, a server can answer more than one request at a time. contents of the file are determined when the file is created, not when it is used.
Some popular web servers include Apache and Microsoft Internet Information
Server. Static documents are prepared using one of several languages: HyperText
Markup Language (HTML), Extensible Markup Language (XML), Extensible Style
Language (XSL), and Extensible Hypertext Markup Language (XHTML).
Uniform Resource Locator (URL):
Dynamic Documents:
A web page, as a file, needs to have a unique identifier to distinguish it from
other web pages. To define a web page, we need three identifiers: host, port, and A dynamic document is created by a web server whenever a browser
path. requests the document. When a request arrives, the web server runs an application
program or a script that creates the dynamic document.
However, before defining the web page, we need to tell the browser what
clientserver application we want to use, which is called the protocol. This means we The server returns the result of the program or script as a response to the
need four identifiers to define the web page. browser that requested the document. Because a fresh document is created for
each request, the contents of a dynamic document may vary from one request to
The first is the type of vehicle to be used to fetch the web page; the last
another. A very simple example of a dynamic document is the retrieval of the time
three make up the combination that defines the destination object (web page).
and date from a server.
Protocol. The first identifier is the abbreviation for the client-server program
that we need in order to access the web page.
Active Documents:

Although most of the time the protocol is HTTP (HyperText Transfer For many applications, we need a program or a script to be run at the client
Protocol), we can also use other protocols such as FTP (File Transfer Protocol). site. These are called active documents. For example, suppose we want to run a
program that creates animated graphics on the screen or a program that interacts
Host. The host identifier can be the IP address of the server or the unique with the user.
name given to the server. IP addresses can be defined in dotted decimal notation.
HyperText Transfer Protocol (HTTP):

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Message Formats:
The HyperText Transfer Protocol (HTTP) is used to define how the client-
server programs can be written to retrieve web pages from the Web. An HTTP client The HTTP protocol defines the format of the request and response messages.
sends a request; an HTTP server returns a response. The server uses the port Each message is made of four sections. The first section in the request message is
number 80; the client uses a temporary port number. HTTP uses the services of called the request line; the first section in the response message is called the status
TCP, which, as discussed before, is a connection-oriented and reliable protocol. line.

Nonpersistent versus Persistent Connections: The other three sections have the same names in the request and response
messages. However, the similarities between these sections are only in the names;
If the web pages, objects to be retrieved, are located on different servers, we they may have different contents. We discuss each message type separately.
do not have any other choice than to create a new TCP connection for retrieving
each object. However, if some of the objects are located on the same server, we Request Message:
have two choices: to retrieve each object using a new TCP connection or to make a
TCP connection and retrieve them all. The first method is referred to as a There are three fields in this line separated by one space and terminated by
two characters (carriage return and line feed). The fields are called method, URL,
nonpersistent connection, the second as a persistent connection.
and version.
Nonpersistent Connections
The method field defines the request types. Several methods are defined like
In a nonpersistent connection, one TCP connection is made for each GET, PUT, HEAD, POST, TRACE, DELETE, etc. The URL defines the address and
request/response. name of the corresponding web page. The version field gives the version of the
protocol; the most current version of HTTP is 1.1.
The following lists the steps in this strategy:
Response Message:
1. The client opens a TCP connection and sends a request.
A response message consists of a status line, header lines, a blank line, and
2. The server sends the response and closes the connection. sometimes a body. The first line in a response message is called the status line.
There are three fields in this line separated by spaces and terminated by a carriage
3. The client reads the data until it encounters an end-of-file marker; it then
return and line feed.
closes the connection.

Persistent Connections The first field defines the version of HTTP protocol, currently 1.1. The status
code field defines the status of the request. It consists of three digits. Whereas the
HTTP version 1.1 specifies a persistent connection by default. In a codes in the 100 range are only informational, the codes in the 200 range indicate a
persistent connection, the server leaves the connection open for more requests successful request.
after sending a response.
The codes in the 300 range redirect the client to another URL, and the codes
The server can close the connection at the request of a client or if a time-out in the 400 range indicate an error at the client site. Finally, the codes in the 500
has been reached. The sender usually sends the length of the data with each range indicate an error at the server site.
response. However, there are some occasions when the sender does not know the
The status phrase explains the status code in text form. After the status line,
length of the data.
we can have zero or more response header lines. Each header line sends additional
This is the case when a document is created dynamically or actively. In these information from the server to the client.
cases, the server informs the client that the length is not known and closes the
connection after sending the data so the client knows that the end of the data has Web Caching: Proxy Servers:
been reached. Time and resources are saved using persistent connections.
HTTP supports proxy servers. A proxy server is a computer that keeps
copies of responses to recent requests. The HTTP client sends a request to the
Only one set of buffers and variables needs to be set for the connection at
proxy server. The proxy server checks its cache.
each site. The round trip time for connection establishment and connection
termination is saved.
If the response is not stored in the cache, the proxy server sends the request

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to the corresponding server. Incoming responses are sent to the proxy server and
stored for future requests from other clients. 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 proxy server reduces the load on the original server, decreases traffic,
and improves latency. However, to use the proxy server, the client must be Control Connection:
configured to access the proxy instead of the target server.
During this control connection, commands are sent from the client to the
HTTP Security: server and responses are sent from the server to the client. Commands, which are
sent from the FTP client control process, are in the form of ASCII uppercase, which
HTTP per se does not provide security. HTTP can be run over the Secure may or may not be followed by an argument. Some of the most common
Socket Layer (SSL). In this case, HTTP is referred to as HTTPS. HTTPS provides commands are shown in table below:
confidentiality, client and server authentication, and data integrity.
Command Argument(s) Description
FTP: ABOR Abort the previous command
CDUP Change to parent directory
File Transfer Protocol (FTP) is the standard protocol provided by TCP/IP CWD Directory name Change to another directory
for copying a file from one host to another. Although transferring files from one DELE File name Delete a file
system to another seems simple and straightforward, some problems must be dealt LIST Directory name List subdirectories or files
with first. MKD Directory name Create a new directory
PASS User password Password
PASV Server chooses a port
Although we can transfer files using HTTP, FTP is a better choice to transfer PORT Port identifier Client chooses a port
large files or to transfer files using different formats. Figure 5.3 shows the basic PWD Display name of current directory
model of FTP. The client has three components: the user interface, the client QUIT Log out of the system
control process, and the client data transfer process. The server has two Every FTP command generates at least one response. A response has two
components: the server control process and the server data transfer process. parts: a three-digit number followed by text. The numeric part defines the code;
the text part defines needed parameters or further explanations. The first digit
defines the status of the command. The second digit defines the area in which the
status applies. The third digit provides additional information.

Code Description Code Description

125 Data Connection Open 250 Request file action OK
150 File Status OK 331 User name OK; password is needed
Figure 5.3: FTP 200 Command OK 425 Cannot open data connection

ELECTRONIC MAIL:
The control connection is made between the control processes. The data Electronic mail (or e-mail) allows users to exchange messages. The nature of
connection is made between the data transfer processes. Separation of commands this application, however, is different from other applications discussed so far. In an
and data transfer makes FTP more efficient. The control connection uses very application such as HTTP or FTP, the server program is running all the time, waiting
simple rules of communication. We need to transfer only a line of command or a for a request from a client. When the request arrives, the server provides the
line of response at a time. The data connection, on the other hand, needs more service. There is a request and there is a response.
complex rules due to the variety of data types transferred.
In the case of electronic mail, the situation is different. First, e-mail is
Two Connections
considered a one-way transaction. When Alice sends an email to Bob, she may
expect a response, but this is not a mandate. Bob may or may not respond. If he
The two connections in FTP have different lifetimes. The control connection
does respond, it is another one-way transaction.
remains connected during the entire interactive FTP session. The data connection is
opened and then closed for each file transfer activity. Second, it is neither feasible nor logical for Bob to run a server program and

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wait until someone sends an e-mail to him. Bob may turn off his computer when he in the queue to be sent.
is not using it.
This means that the idea of client/server programming should be The user agent at the Bob site allows Bob to read the received message. Bob
implemented in another way: using some intermediate computers (servers). The later uses an MAA client to retrieve the message from an MAA server running on
users run only client programs when they want and the intermediate servers apply the second server.
the client/server paradigm
User Agent: The first component of an electronic mail system is the user
Architecture: agent (UA). It provides service to the user to make the process of sending and
receiving a message easier.
To explain the architecture of e-mail, we give a common scenario as shown
in Figure 5.4. A user agent is a software package (program) that composes, reads, replies
to, and forwards messages. It also handles local mailboxes on the user computers.

Message Transfer Agent: SMTP: Based on the common scenario, we can

say that the e-mail is one of those applications that needs three uses of client-
server paradigms to accomplish its task. It is important that we distinguish these
three when we are dealing with e-mail.

The formal protocol that defines the MTA client and server in the Internet is
called Simple Mail Transfer Protocol (SMTP). SMTP is used two times, between
the sender and the sender’s mail server and between the two mail servers. SMTP
simply defines how commands and responses must be sent back and forth.

Message Access Agent: POP and IMAP: The first and second stages of
mail delivery use SMTP. However, SMTP is not involved in the third stage because
In the common scenario, the sender and the receiver of the e-mail, Alice and SMTP is a push protocol; it pushes the message from the client to the server.On the
Bob respectively, are connected via a LAN or a WAN to two mail servers. The other hand, the third stage needs a pull protocol; the client must pull messages
administrator has created one mailbox for each user where the received messages from the server. The direction of the bulk data is from the server to the client. The
are stored. third stage uses a message access agent.

A mailbox is part of a server hard drive, a special file with permission Currently two message access protocols are available: Post Office Protocol,
version 3 (POP3) and Internet Mail Access Protocol, version 4 (IMAP4).
restrictions. Only the owner of the mailbox has access to it. The administrator has
also created a queue (spool) to store messages waiting to be sent.
POP3:

A simple e-mail from Alice to Bob takes nine different steps. Alice and Bob
Post Office Protocol, version 3 (POP3) is simple but limited in
use three different agents: a user agent (UA), a message transfer agent functionality. The client POP3 software is installed on the recipient computer; the
(MTA), and a message access agent (MAA). When Alice needs to send a server POP3 software is installed on the mail server.
message to Bob, she runs a UA program to prepare the message and send it to her
mail server. Mail access starts with the client when the user needs to download its e-mail
from the mailbox on the mail server. The client opens a connection to the server on
The mail server at her site uses a queue (spool) to store messages waiting to TCP port 110. It then sends its user name and password to access the mailbox. The
be sent. The message, however, needs to be sent through the Internet from Alice’s user can then list and retrieve the mail messages, one by one.
site to Bob’s site using an MTA. Here two message transfer agents are needed: one
client and one server. POP3 has two modes: the delete mode and the keep mode. In the delete
mode, the mail is deleted from the mailbox after each retrieval. In the keep mode,
Like most client-server programs on the Internet, the server needs to run all the mail remains in the mailbox after retrieval.
the time because it does not know when a client will ask for a connection. The
client, on the other hand, can be triggered by the system when there is a message IMAP4:

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TELNET:
Another mail access protocol is Internet Mail Access Protocol, version 4
(IMAP4). IMAP4 is similar to POP3, but it has more features; IMAP4 is more A server program can provide a specific service to its corresponding client
powerful and more complex. program. For example, the FTP server is designed to let the FTP client store or
retrieve files on the server site. However, it is impossible to have a client/server
POP3 is deficient in several ways. It does not allow the user to organize her pair for each type of service we need; the number of servers soon becomes
mail on the server; the user cannot have different folders on the server. In intractable which is not scalable.
addition, POP3 does not allow the user to partially check the contents of the mail
before downloading. Another solution is to have a specific client/server program for a set of
common scenarios, but to have some generic client/server programs that allow a
IMAP4 provides the following extra functions: user on the client site to log into the computer at the server site and use the
 services available there.
A user can check the e-mail header prior to downloading.

 For example, if a student needs to use the Java compiler program at her
A user can search the contents of the e-mail for a specific string of characters
prior to downloading. university lab, there is no need for a Java compiler client and a Java compiler
server. The student can use a client logging program to log into the university
 server and use the compiler program at the university. We refer to these generic
A user can partially download e-mail. This is especially useful if bandwidth
is limited and the e-mail contains multimedia with high bandwidth client/server pairs as remote logging applications.
requirements.
 One of the original remote logging protocols is TELNET, which is an
A user can create, delete, or rename mailboxes on the mail server.
abbreviation for TErminaL NETwork. Although TELNET requires a logging name and
password, it is vulnerable to hacking because it sends all data including the
password in plaintext (not encrypted).

A hacker can eavesdrop and obtain the logging name and password. Because
of this security issue, the use of TELNET has diminished in favor of another
protocol, Secure Shell (SSH).

Although TELNET is almost replaced by SSH, we briefly discuss TELNET here

for two reasons:

1. The simple plaintext architecture of TELNET allows us to explain the issues and
challenges related to the concept of remote logging, which is also used in SSH
when it serves as a remote logging protocol.

2. Network administrators often use TELNET for diagnostic and debugging

purposes.

Local versus Remote Logging:

When a user logs into a local system, it is called local logging. As a user
types at a terminal or at a workstation running a terminal emulator, the keystrokes
are accepted by the terminal driver.

The terminal driver passes the characters to the operating system. The
operating system, in turn, interprets the combination of characters and invokes the
desired application program or utility.

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Computer Networks Computer Networks

However, when a user wants to access an application program or utility

located on a remote machine, she performs remote logging. Here the TELNET client SECURE SHELL (SSH):
and server programs come into use. The user sends the keystrokes to the terminal
driver where the local operating system accepts the characters but does not Although Secure Shell (SSH) is a secure application program that can be
interpret them. used today for several purposes such as remote logging and file transfer, it was
originally designed to replace TELNET.
The characters are sent to the TELNET client, which transforms the
characters into a universal character set called Network Virtual Terminal (NVT) There are two versions of SSH: SSH-1 and SSH-2, which are totally
characters and delivers them to the local TCP/IP stack. incompatible. The first version, SSH-1, is now deprecated because of security flaws
in it. In this section, we discuss only SSH-2.
The commands or text, in NVT form, travel through the Internet and arrive at
the TCP/IP stack at the remote machine. Here the characters are delivered to the Components: SSH is an application-layer protocol with three components.
operating system and passed to the TELNET server, which changes the characters
to the corresponding characters understandable by the remote computer. SSH Transport-Layer Protocol (SSH-TRANS):

However, the characters cannot be passed directly to the operating system Since TCP is not a secured transport-layer protocol, SSH first uses a protocol
because the remote operating system is not designed to receive characters from a that creates a secured channel on top of the TCP. This new layer is an independent
TELNET server; it is designed to receive characters from a terminal driver. protocol referred to as SSH-TRANS.

The solution is to add a piece of software called a pseudoterminal driver, When the procedure implementing this protocol is called, the client and
which pretends that the characters are coming from a terminal. The operating server first use the TCP protocol to establish an insecure connection. Then they
system then passes the characters to the appropriate application program. exchange several security parameters to establish a secure channel on top of the
TCP. The services provided by this protocol are:
NVT uses two sets of characters, one for data and one for control. Both are
8-bit bytes. For data, NVT normally uses what is called NVT ASCII. This is an 8-bit 1. Privacy or confidentiality of the message exchanged.
character set in which the seven lowest order bits are the same as US ASCII and
2. Data integrity, which means that it is guaranteed that the messages exchanged
the highest order bit is 0.
between the client and server are not changed by an intruder.
To send control characters between computers (from client to server or vice
3. Server authentication, which means that the client is now sure that the server is
versa), NVT uses an 8-bit character set in which the highest order bit is set to 1.
the one that it claims to be.
Options: TELNET lets the client and server negotiate options before or
4. Compression of the messages, which improves the efficiency of the system and
during the use of the service.
makes attack more difficult.
User Interface:
SSH Authentication Protocol (SSH-AUTH):
The operating system (UNIX, for example) defines an interface with user-
After a secure channel is established between the client and the server and
friendly commands. An example of such a set of commands can be found in Table
the server is authenticated for the client, SSH can call another procedure that can
beow:
authenticate the client for the server. The client authentication process in SSH is
Command Name Meaning very similar to what is done in Secure Socket Layer (SSL).
open Connect to a remote computer
close Close the connections This layer defines a number of authentication tools similar to the ones used
display Show the operating parameters in SSL. Authentication starts with the client, which sends a request message to the
mode Change to line or character mode server. The request includes the user name, server name, the method of
Quit Exit TELNET authentication, and the required data. The server responds with either a success
send Send special characters message, which confirms that the client is authenticated, or a failed message,
which means that the process needs to be repeated with a new request message.

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Computer Networks Computer Networks

SSH Connection Protocol (SSH-CONN):

After the secured channel is established and both server and client are
authenticated for each other, SSH can call a piece of software that implements the
third protocol, SSHCONN.

One of the services provided by the SSH-CONN protocol is multiplexing. SSH-

CONN takes the secure channel established by the two previous protocols and lets
the client create multiple logical channels over it. Each channel can be used for a
Figure 5.5: Purpose of DNS
different purpose, such as remote logging, file transfer, and so on.
Name Space:
Applications:

A name space that maps each address to a unique name can be organized
Although SSH is often thought of as a replacement for TELNET, SSH is, in
in two ways: flat or hierarchical. In a flat name space, a name is assigned to an
fact, a general-purpose protocol that provides a secure connection between a client
address.
and server.

SSH for Remote Logging: A name in this space is a sequence of characters without structure. The
names may or may not have a common section; if they do, it has no meaning. The
Several free and commercial applications use SSH for remote logging. Among main disadvantage of a flat name space is that it cannot be used in a large system
them, we can mention PuTTy, by Simon Tatham, which is a client SSH program that such as the Internet because it must be centrally controlled to avoid ambiguity and
can be used for remote logging. Another application program is Tectia, which can duplication.
be used on several platforms.
In a hierarchical name space, each name is made of several parts. The first
SSH for File Transfer: part can define the nature of the organization, the second part can define the name
of an organization, the third part can define departments in the organization, and
One of the application programs that is built on top of SSH for file transfer is so on. In this case, the authority to assign and control the name spaces can be
the Secure File Transfer Program (sftp). The sftp application program uses one of decentralized.
the channels provided by the SSH to transfer files. Another common application is
called Secure Copy (scp). This application uses the same format as the UNIX copy A central authority can assign the part of the name that defines the nature of
command, cp, to copy files. the organization and the name of the organization. The responsibility for the rest of
the name can be given to the organization itself.
DOMAIN NAME SYSTEM (DNS):
The organization can add suffixes (or prefixes) to the name to define its host
Since the Internet is so huge today, a central directory system cannot hold or resources. The management of the organization need not worry that the prefix
all the mapping. In addition, if the central computer fails, the whole communication chosen for a host is taken by another organization because, even if part of an
network will collapse. address is the same, the whole address is different.

A better solution is to distribute the information among many computers in Domain Name Space:
the world. In this method, the host that needs mapping can contact the closest
computer holding the needed information. This method is used by the Domain To have a hierarchical name space, a domain name space was designed. In
Name System (DNS). this design the names are defined in an inverted-tree structure with the root at the
top. The tree can have only 128 levels: level 0 (root) to level 127 (see Figure 5.6).
Figure 5.5 shows how TCP/IP uses a DNS client and a DNS server to map a
name to an address. A user wants to use a file transfer client to access the
corresponding file transfer server running on a remote host. The user knows only
the file transfer server name, such as afilesource.com.

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Computer Networks Computer Networks

Figure 5.8: Domains

SNMP:

Several network management standards have been devised during the last
Figure 5.6: Domain name space few decades. The most important one is Simple Network Management Protocol
(SNMP), used by the Internet.
Label:
SNMP is a framework for managing devices in an internet using the TCP/IP
Each node in the tree has a label, which is a string with a maximum of 63 protocol suite. It provides a set of fundamental operations for monitoring and
characters. The root label is a null string (empty string). DNS requires that children maintaining an internet. SNMP uses the concept of manager and agent. That is, a
of a node (nodes that branch from the same node) have different labels, which manager, usually a host, controls and monitors a set of agents, usually routers or
guarantees the uniqueness of the domain names. servers (see Figure 5.9).
Domain Name:

Each node in the tree has a domain name. A full domain name is a
sequence of labels separated by dots (.). The domain names are always read from
the node up to the root.

The last label is the label of the root (null). This means that a full domain
name always ends in a null label, which means the last character is a dot because
the null string is nothing. Figure 5.7 shows some domain names. Figure 5.9: SNMP concept

Domain: SNMP is an application-level protocol in which a few manager stations control

a set of agents. The protocol is designed at the application level so that it can
A domain is a subtree of the domain name space. The name of the domain monitor devices made by different manufacturers and installed on different physical
is the name of the node at the top of the subtree. Figure 5.8 shows some domains. networks.
Note that a domain may itself be divided into domains.
In other words, SNMP frees management tasks from both the physical
characteristics of the managed devices and the underlying networking technology.
It can be used in a heterogeneous internet made of different LANs and WANs
connected by routers made by different manufacturers.

Managers and Agents: A management station, called a manager, is a host

that runs the SNMP client program. A managed station, called an agent, is a router
(or a host) that runs the SNMP server program.

Figure 5.7: Domain names and labels

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Computer Networks

Management is achieved through simple interaction between a manager and

an agent. The agent keeps performance information in a database. The manager
has access to the values in the database.

For example, a router can store in appropriate variables the number of

packets received and forwarded. The manager can fetch and compare the values of
these two variables to see if the router is congested or not.

The manager can also make the router perform certain actions. For example,
a router periodically checks the value of a reboot counter to see when it should
reboot itself. It reboots itself, for example, if the value of the counter is 0. The
manager can use this feature to reboot the agent remotely at any time. It simply
sends a packet to force a 0 value in the counter.

Agents can also contribute to the management process. The server program
running on the agent can check the environment and, if it notices something
unusual, it can send a warning message (called a Trap) to the manager. In other
words, management with SNMP is based on three basic ideas:

1. A manager checks an agent by requesting information that reflects the

behavior of the agent.

2. A manager forces an agent to perform a task by resetting values in the agent

database.

3. An agent contributes to the management process by warning the manager of

an unusual situation.

Management Components: To do management tasks, SNMP uses two

other protocols: Structure of Management Information (SMI) and
Management Information Base (MIB).

Role of SNMP: SNMP has some very specific roles in network management.
It defines the format of the packet to be sent from a manager to an agent and vice
versa. It also interprets the result and creates statistics (often with the help of
other management software).

Role of SMI: To use SNMP, we need rules for naming objects. This is
particularly important because the objects in SNMP form a hierarchical structure.
Part of a name can be inherited from the parent. We also need rules to define the
types of objects.

Role of MIB: MIB creates a set of objects defined for each entity in a
manner similar to that of a database (mostly metadata in a database, names &
types without values).

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