10ec71 - CCN - Notes PDF

COMPUTER COMMUNICATION NETWORKS
UNIT 1 : INTRODUCTION TO NETWORKS
Networking engineering is a complicated task, which involves software, firmware, chip level
engineering, hardware, and electric pulses. To ease network engineering, the whole networking
concept is divided into multiple layers. Each layer is involved in some particular task and is
independent of all other layers. But as a whole, almost all networking tasks depend on all of
these layers. Layers share data between them and they depend on each other only to take input
and send output.
Layered Tasks
In layered architecture of Network Model, one whole network process is divided into small
tasks. Each small task is then assigned to a particular layer which works dedicatedly to process
the task only. Every layer does only specific work.
In layered communication system, one layer of a host deals with the task done by or to be done
by its peer layer at the same level on the remote host. The task is either initiated by layer at the
lowest level or at the top most level. If the task is initiated by the-top most layer, it is passed on
to the layer below it for further processing. The lower layer does the same thing, it processes the
task and passes on to lower layer. If the task is initiated by lower most layer, then the reverse
path is taken.
Every layer clubs together all procedures, protocols, and methods which it requires to execute
its piece of task. All layers identify their counterparts by means of encapsulation header and tail.
OSI reference model (Open Systems Interconnection)
OSI (Open Systems Interconnection) is reference model for how applications can communicate
over a network. A reference model is a conceptual framework for understanding relationships.
The purpose of the OSI reference model is to guide vendors and developers so the digital
communication products and software programs they create will interoperate, and to facilitate
clear comparisons among communications tools. Most vendors involved in telecommunications
make an attempt to describe their products and services in relation to the OSI model. And
although useful for guiding discussion and evaluation, OSI is rarely actually implemented, as
few network products or standard tools keep all related functions together in well-defined layers
as related to the model. The TCP/IP protocols, which define the Internet, do not map cleanly to
the OSI model.
Developed by representatives of major computer and telecommunication companies beginning

in 1983, OSI was originally intended to be a detailed specification of actual interfaces. Instead,
the committee decided to establish a common reference model for which others could then
develop detailed interfaces, which in turn could become standards. OSI was officially adopted
as an international standard by the International Organization of Standards (ISO).
OSI layers
The main concept of OSI is that the process of communication between two endpoints in a
telecommunication network can be divided into seven distinct groups of related functions, or
layers. Each communicating user or program is at a computer that can provide those seven
layers of function. So in a given message between users, there will be a flow of data down
through the layers in the source computer, across the network and then up through the layers in
the receiving computer. The seven layers of function are provided by a combination of
applications, operating systems, network card device drivers and networking hardware that
enable a system to put a signal on a network cable or out over Wi-Fi or other wireless protocol).
The seven Open Systems Interconnection layers are:
Layer 7: The application layer. This is the layer at which communication partners are identified
(Is there someone to talk to?), network capacity is assessed (Will the network let me talk to
them right now?), and that creates a thing to send or opens the thing received. (This layer
is not the application itself, it is the set of services an application should be able to make use of
directly, although some applications may perform application layer functions.)
Layer 6: The presentation layer. This layer is usually part of an operating system (OS) and
converts incoming and outgoing data from one presentation format to another (for example,
from clear text to encrypted text at one end and back to clear text at the other).
Layer 5: The session layer. This layer sets up, coordinates and terminates conversations.
Services include authentication and reconnection after an interruption. On the Internet,
Transmission Control Protocol (TCP) and User Datagram Protocol (UDP) provide these
services for most applications.
Layer 4: The transport layer. This layer manages packetization of data, then the delivery of the
packets, including checking for errors in the data once it arrives. On the Internet, TCP and UDP
provide these services for most applications as well.
Layer 3: The network layer. This layer handles the addressing and routing of the data (sending it
in the right direction to the right destination on outgoing transmissions and receiving incoming
transmissions at the packet level). IP is the network layer for the Internet.
Layer 2: The data-link layer. This layer sets up links across the physical network, putting
packets into network frames. This layer has two sub-layers, the Logical Link Control Layer and
the Media Access Control Layer. Ethernet is the main data link layer in use.
Layer 1: The physical layer. This layer conveys the bit stream through the network at the
electrical, optical or radio level. It provides the hardware means of sending and receiving data
on a carrier network.
The TCP/IP Protocol Suite
The TCP/IP protocol suite maps to a four-layer conceptual model known as the DARPA model,
which was named after the U.S. government agency that initially developed TCP/IP. The four
layers of the DARPA model are: Application, Transport, Internet, and Network Interface. Each
layer in the DARPA model corresponds to one or more layers of the seven-layer OSI model.
Figure 2-1 shows the architecture of the TCP/IP protocol suite.

Figure 2-1 The architecture of the TCP/IP protocol suite
The TCP/IP protocol suite has two sets of protocols at the Internet layer:
• IPv4, also known as IP, is the Internet layer in common use today on private intranets and
the Internet.
• IPv6 is the new Internet layer that will eventually replace the existing IPv4 Internet layer.
Network Interface Layer

The Network Interface layer (also called the Network Access layer) sends TCP/IP packets on
the network medium and receives TCP/IP packets off the network medium. TCP/IP was
designed to be independent of the network access method, frame format, and medium.
Therefore, you can use TCP/IP to communicate across differing network types that use LAN
technologies—such as Ethernet and 802.11 wireless LAN—and WAN technologies—such as
Frame Relay and Asynchronous Transfer Mode (ATM). By being independent of any specific
network technology, TCP/IP can be adapted to new technologies.
The Network Interface layer of the DARPA model encompasses the Data Link and Physical
layers of the OSI model. The Internet layer of the DARPA model does not take advantage of
sequencing and acknowledgment services that might be present in the Data Link layer of the
OSI model. The Internet layer assumes an unreliable Network Interface layer and that reliable
communications through session establishment and the sequencing and acknowledgment of
packets is the responsibility of either the Transport layer or the Application layer.
Internet Layer
The Internet layer responsibilities include addressing, packaging, and routing functions. The
Internet layer is analogous to the Network layer of the OSI model.
The core protocols for the IPv4 Internet layer consist of the following:
• The Address Resolution Protocol (ARP) resolves the Internet layer address to a Network
Interface layer address such as a hardware address.
• The Internet Protocol (IP) is a routable protocol that addresses, routes, fragments, and
reassembles packets.
• The Internet Control Message Protocol (ICMP) reports errors and other information to
help you diagnose unsuccessful packet delivery.
• The Internet Group Management Protocol (IGMP) manages IP multicast groups.
For more information about the core protocols for the IPv4 Internet layer, see "IPv4 Internet
Layer" later in this chapter.
The core protocols for the IPv6 Internet layer consist of the following:
• IPv6 is a routable protocol that addresses and routes packets.
• The Internet Control Message Protocol for IPv6 (ICMPv6) reports errors and other
information to help you diagnose unsuccessful packet delivery.
• The Neighbor Discovery (ND) protocol manages the interactions between neighboring
IPv6 nodes.
• The Multicast Listener Discovery (MLD) protocol manages IPv6 multicast groups.
Transport Layer
The Transport layer (also known as the Host-to-Host Transport layer) provides the Application
layer with session and datagram communication services. The Transport layer encompasses the
responsibilities of the OSI Transport layer. The core protocols of the Transport layer are TCP
and UDP.
TCP provides a one-to-one, connection-oriented, reliable communications service. TCP
establishes connections, sequences and acknowledges packets sent, and recovers packets lost
during transmission.
In contrast to TCP, UDP provides a one-to-one or one-to-many, connectionless, unreliable
communications service. UDP is used when the amount of data to be transferred is small (such as
the data that would fit into a single packet), when an application developer does not want the
overhead associated with TCP connections, or when the applications or upper-layer protocols
provide reliable delivery.
TCP and UDP operate over both IPv4 and IPv6 Internet layers.
Note The Internet Protocol (TCP/IP) component of Windows contains separate versions of the
TCP and UDP protocols than the Microsoft TCP/IP Version 6 component does. The versions in
the Microsoft TCP/IP Version 6 component are functionally equivalent to those provided with
the Microsoft Windows NT® 4.0 operating systems and contain all the most recent security
updates. The existence of separate protocol components with their own versions of TCP and
UDP is known as a dual stack architecture. The ideal architecture is known as a dual IP layer, in
which the same versions of TCP and UDP operate over both IPv4 and IPv6 (as Figure 2-1
shows). Windows Vista has a dual IP layer architecture for the TCP/IP protocol components.
Application Layer
The Application layer allows applications to access the services of the other layers, and it defines
the protocols that applications use to exchange data. The Application layer contains many
protocols, and more are always being developed.
The most widely known Application layer protocols help users exchange information:
• The Hypertext Transfer Protocol (HTTP) transfers files that make up pages on the World
Wide Web.
• The File Transfer Protocol (FTP) transfers individual files, typically for an interactive
user session.
• The Simple Mail Transfer Protocol (SMTP) transfers mail messages and attachments.
Additionally, the following Application layer protocols help you use and manage TCP/IP
networks:
• The Domain Name System (DNS) protocol resolves a host name, such as
www.microsoft.com, to an IP address and copies name information between DNS
servers.
• The Routing Information Protocol (RIP) is a protocol that routers use to exchange routing
information on an IP network.
• The Simple Network Management Protocol (SNMP) collects and exchanges network
management information between a network management console and network devices
such as routers, bridges, and servers.
Windows Sockets and NetBIOS are examples of Application layer interfaces for TCP/IP
applications.
IPv4 Internet Layer

The IPv4 Internet layer consists of the following protocols:
• ARP
• IP (IPv4)
• ICMP
• IGMP
The following sections describe each of these protocols in more detail.
ARP
When IP sends packets over a shared access, broadcast-based networking technology such as
Ethernet or 802.11 wireless LAN, the protocol must resolve the media access control (MAC)
addresses corresponding to the IPv4 addresses of the nodes to which the packets are being
forwarded, also known as the next-hop IPv4 addresses. As RFC 826 defines, ARP uses MAC-
level broadcasts to resolve next-hop IPv4 addresses to their corresponding MAC addresses.
Based on the destination IPv4 address and the route determination process, IPv4 determines the
next-hop IPv4 address and interface for forwarding the packet. IPv4 then hands the IPv4 packet,
the next-hop IPv4 address, and the next-hop interface to ARP.
If the IPv4 address of the packet’s next hop is the same as the IPv4 address of the packet’s
destination, ARP performs a direct delivery to the destination. In a direct delivery, ARP must
resolve the IPv4 address of the packet’s destination to its MAC address.
If the IPv4 address of the packet’s next hop is not the same as the IPv4 address of the packet’s
destination, ARP performs an indirect delivery to a router. In an indirect delivery, ARP must
resolve the IPv4 address of the router to its MAC address
To resolve the IPv4 address of a packet’s next hop to its MAC address, ARP uses the
broadcasting facility on shared access networking technologies (such as Ethernet or 802.11) to
send out a broadcast ARP Request frame. In response, the sender receives an ARP Reply frame,
which contains the MAC address that corresponds to the IPv4 address of the packet’s next hop.
ARP Cache
To minimize the number of broadcast ARP Request frames, many TCP/IP protocol
implementations incorporate an ARP cache, which is a table of recently resolved IPv4 addresses
and their corresponding MAC addresses. ARP checks this cache before sending an ARP Request
frame. Each interface has its own ARP cache.
Depending on the vendor implementation, the ARP cache can have the following qualities:
ARP cache entries can be dynamic (based on ARP replies) or static. Static ARP cache entries are
permanent, and you add them manually using a TCP/IP tool, such as the Arp tool provided with
Windows. Static ARP cache entries prevent nodes from sending ARP requests for commonly
used local IPv4 addresses, such as those for routers and servers. The problem with static ARP
cache entries is that you must manually update them when network adapter equipment changes.
Dynamic ARP cache entries have time-out values associated with them so that they are removed
from the cache after a specified period of time. For example, dynamic ARP cache entries for
Windows are removed after no more than 10 minutes.
To view the ARP cache on a Windows–based computer, type arp -a at a command prompt. You
can also use the Arp tool to add or delete static ARP cache entries.
ARP Process
When sending the initial packet as the sending host or forwarding the packet as a router, IPv4
sends the IPv4 packet, the next-hop IPv4 address, and the next-hop interface to ARP. Whether
performing a direct or indirect delivery, ARP performs the following process:
Based on the next-hop IPv4 address and interface, ARP checks the appropriate ARP cache for an
entry that matches the next-hop IPv4 address. If ARP finds an entry, ARP skips to step 6.
If ARP does not find an entry, ARP builds an ARP Request frame. This frame contains the MAC
and IPv4 addresses of the interface from which the ARP request is being sent and the IPv4
packet's next-hop IPv4 address. ARP then broadcasts the ARP Request frame from the
appropriate interface.
All nodes on the subnet receive the broadcasted frame and process the ARP request. If the next-
hop address in the ARP request corresponds to the IPv4 address assigned to an interface on the
subnet, the receiving node updates its ARP cache with the IPv4 and MAC addresses of the ARP
requestor. All other nodes silently discard the ARP request.
The receiving node that is assigned the IPv4 packet’s next-hop address formulates an ARP reply
that contains the requested MAC address and sends the reply directly to the ARP requestor.
When the ARP requestor receives the ARP reply, the requestor updates its ARP cache with the
address mapping. With the exchange of the ARP request and the ARP reply, both the ARP
requestor and ARP responder have each other's address mappings in their ARP caches.
The ARP requestor sends the IPv4 packet to the next-hop node by addressing it to the resolved
MAC address.
Comparison of OSI Reference Model and TCP/IP Reference Model
Following are some major differences between OSI Reference Model and TCP/IP Reference
Model, with diagrammatic comparison below.
OSI(Open System Interconnection) TCP/IP(Transmission Control Protocol /

Internet Protocol)
1. OSI is a generic, protocol 1. TCP/IP model is based on standard

independent standard, acting as a protocols around which the Internet has
communication gateway between developed. It is a communication protocol,
the network and end user. which allows connection of hosts over a
network.
2. In OSI model the transport layer 2. In TCP/IP model the transport layer does
guarantees the delivery of packets. not guarantees delivery of packets. Still the
TCP/IP model is more reliable.
3. Follows vertical approach. 3. Follows horizontal approach.
4. OSI model has a separate 4. TCP/IP does not have a separate

Presentation layer and Session Presentation layer or Session layer.
layer.
5. OSI is a reference model around 5. TCP/IP model is, in a way implementation

which the networks are built. of the OSI model.
Generally it is used as a guidance
tool.
6. Network layer of OSI model 6. The Network layer in TCP/IP model

provides both connection oriented provides connectionless service.
and connectionless service.
7. OSI model has a problem of 7. TCP/IP model does not fit any protocol
fitting the protocols into the model.
8. Protocols are hidden in OSI 8. In TCP/IP replacing protocol is not easy.

model and are easily replaced as the
technology changes.
9. OSI model defines services, 9. In TCP/IP, services, interfaces and

interfaces and protocols very protocols are not clearly separated. It is also
clearly and makes clear distinction protocol dependent.
between them. It is protocol
independent.
10. It has 7 layers 10. It has 4 layers

Using Telephone and Cable Networks for Data Transmission
The telephone, which is referred to as the plain old telephone system (POTS), was originally an
analog system. During the last decade, the telephone network has undergone many technical
changes. The network is now digital as well as analog.
A home computer can access the Internet through the existing telephone system or through a
cable TV system.
The telephone network is made of three major components: local loops, trunks, and switching
offices. It has several levels of switching offices such as end offices, tandem offices, and regional
offices.
Telephone companies provide two types of services: analog and digital.

The United States is divided into many local access transport areas (LATAs). The services
offered inside a LATA are called intra-LATA services. The carrier that handles these services is
called a local exchange carrier (LEC). The services between LATAs are handled
by interexchange carriers (lXCs).
A LATA is a small or large metropolitan area that according to the divestiture of 1984 was under
the control of a single telephone-service provider.
In in-band signaling, the same circuit is used for both signaling and data. In out-of band
signaling, a portion of the bandwidth is used for signaling and another portion for data. The
protocol that is used for signaling in the telephone network is called Signaling System Seven
(SS7).
• Telephone companies provide two types of services: analog and digital. We can
categorize analog services as either analog switched services or analog leased services.
The two most common digital services are switched/56 service and digital data service
(DDS).
• Data transfer using the telephone local loop was traditionally done using a dial-up
modem. The term modem is a composite word that refers to the two functional entities
that make up the device: a signal modulator and a signal demodulator.
• Most popular modems available are based on the V-series standards. The V.32 modem
has a data rate of 9600 bps. The V32bis modem supports 14,400-bps transmission. V90
modems, called 56K modems, with a downloading rate of 56 kbps and uploading rate
of 33.6 kbps are very common. The standard above V90 is called V92. These modems
can adjust their speed, and if the noise allows, they can upload data at the rate of 48 kbps.
• Telephone companies developed another technology, digital subscriber line (DSL), to

provide higher-speed access to the Internet. DSL technology is a set of technologies, each
differing in the first letter (ADSL, VDSL, HDSL, and SDSL. ADSL provides higher
speed in the downstream direction than in the upstream direction. The high-bitrate digital
subscriber line (HDSL) was designed as an alternative to the T-l line (1.544 Mbps). The
symmetric digital subscriber line (SDSL) is a one twisted-pair version of HDSL. The
very high-bit-rate digital subscriber line (VDSL) is an alternative approach that is similar
to ADSL.
• DSL supports high-speed digital communications over the existing telephone local loops.
• ADSL technology allows customers a bit rate of up to 1 Mbps in the upstream direction
and up to 8 Mbps in the downstream direction.
• ADSL uses a modulation technique called DMT which combines QAM and FDM.
• ADSL is an asymmetric communication technology designed for residential users; it is

not suitable for businesses.
• ADSL is an adaptive technology. The system uses a data rate based on the condition of
the local loop line.
• SDSL, HDSL, and VDSL are other DSL technologies.
• Theoretically, the coaxial cable used for cable TV allows Internet access with a bit rate of
up to 12 Mbps in the upstream direction and up to 30 Mbps in the downstream direction.
• An HFC network allows Internet access through a combination of fiber-optic and coaxial
cables.
• The coaxial cable bandwidth is divided into a video band, a downstream data band, and
an upstream data band. Both upstream and downstream bands are shared among
subscribers.
• DOCSIS defines all protocols needed for data transmission on an HFC network.
• Synchronous Optical Network (SONET) is a synchronous high-data-rate TDM network

for fiber-optic networks.
• SONET has defined a hierarchy of signals (similar to the DS hierarchy) called

synchronous transport signals (STSs).
• Optical carrier (OC) levels are the implementation of STSs.
• A SONET frame can be viewed as a matrix of nine rows of 90 octets each.
• SONET is backward compatible with the current DS hierarchy through the virtual
tributary (VT) concept. VT's are a partial payload consisting of an m-by-n block of
octets. An STS payload can be a combination of several VT's.
• STSs can be multiplexed to get a new STS with a higher data range.
• Community antenna TV (CATV) was originally designed to provide video services for
the community. The traditional cable TV system used coaxial cable end to end. The
second generation of cable networks is called a hybrid fiber-coaxial (HFC) network. The
network uses a combination of fiber-optic and coaxial cable.
• Communication in the traditional cable TV network is unidirectional.
• Communication in an HFC cable TV network can be bidirectional.
• To provide Internet access, the cable company has divided the available bandwidth of the
coaxial cable into three bands: video, downstream data, and upstream data. The
downstream-only video band occupies frequencies from 54 to 550 MHz. The downstream
data occupies the upper band, from 550 to 750 MHz. The upstream data occupies the
lower band, from 5 to 42 MHz.
In a telephone network, the telephone numbers of the caller and callee are serving as source and
destination addresses. These are used only during the setup (dialing) and teardown (hanging-
up)phases.
Three Major Components of Telephone System
• Local loops - a twisted-pair cable that connects the subscriber telephone to the nearest
end office or local central office. The local loop, when used for voice, has a bandwidth
of 4000 Hz (4 kHz). The existing local loops can handle bandwidths up to 1.1 MHz.
• Trunks - transmission media that handle the communication between offices. A trunk
normally handles hundreds or thousands of connections through multiplexing.
Transmission is usually through optical fibers or satellite links.
• Switching offices - A switch connects several local loops or trunks and allows a
connection between different subscribers.
Telephone Line Bandwidth
Bandwidth division in ADSL

Summary of DSL technologies
Division of coaxial cable band by CATV
A SONET system can use the following equipment:
STS multiplexer - combines several optical signals to make an STS signal.

Regenerator - removes noise from an optical signal.
Add/drop multiplexer - adds STSs from different paths and removes STSs from a path.
Modem (also called a 'dial-up modem')
This is a piece of equipment used for sending and receiving data from one computer to
another computer using the existing phone network.
A modem works by taking the packets of data (which are digital signals) from a computer
and converting them into analogue signals, which the phone network uses. The analogue
signals then pass along the phone network from computer to computer, from network to
network, until the final destination is reached. At the destination, another modem converts
back the analogue signals into the original digital ones and then passes these to the
destination computer.
Advantages and disadvantages of modems
In this way, the existing, widespread phone network, which only uses analogue signals, can
be used by computers, which are digital devices. On the other hand, modems cannot send
large volumes of data at once. They are therefore of little use when streaming files, and the
time taken to download files such as music or films may make them frustrating. In addition,
the phone line, just like making a phone call, is engaged whilst the modem is in use. This is
why it is also known as a 'dial-up' modem; you have to dial up the phone number provided by
your Internet Service Provider (ISP) to make a connection to the Internet and all the time
you are on the phone using the Internet, your phone cannot be used for other things, such as
making and receiving phone calls or sending faxes.
DIGITAL SUBSCRIBER LINE
After traditional modems reached their peak data rate, telephone companies developed
another technology, DSL, to provide higher-speed access to the Internet. Digital subscriber
line (DSL) technology is one of the most promising for supporting high-speed digital
communication over the existing local loops.
The original DSL system was designed to produce data rates of 1.5 Mbits/s to 8 Mbits/s
downstream from the telephone company to the subscriber and a lower rate upstream. Most
Internet access involves more downloading and less uploading of data. The resulting design
is referred to as asymmetrical DSL or ADSL. Most DSL formats are asymmetrical, although
there are DSL variations that deliver the same rates in both directions.
The great attenuation, noise, and crosstalk problems of bundling multiple twisted-pair lines
are the primary limitations of the POTS. These lines are effectively long low-pass filters with
upper frequency limits that reduce the bandwidth of the line and limit the data rate that can
be achieved. Line bandwidth is a function of the length of the UTP. Shorter cable runs have
wider bandwidths, so they are clearly more capable of high data rates than longer runs. But
despite this limitation, developments in digital signal processing have made this once limited
communications medium capable of high-speed data delivery.
Cable TV Systems
Cable TV systems were developed to provide reliable TV service to local communities.

Along with the hundreds of TV channels available, cable companies offer services such as
high-speed Internet access. Some even offer voice over IP (VoIP) telephone service. Cable
companies usually offer a “triple-play” package that bundles TV, phone, and Internet
services.
Systems have been upgraded from pure analog transmission to digital. Early systems were
based on coax cable, but today the most common configuration is fiber-optic cable and coax.
Hybrid fiber coax is one of the most common configurations (Fig. 1).
1. The typical hybrid fiber coax (HFC) cable TV distribution system used throughout the
consists of fiber-optic cable to neighborhood nodes that then distribute the signals to
homes with RG-6/U coax.
All of the services originate from the cable company’s facilities, known as the headend,
where the company collects the video from local TV stations and cable TV programming
suppliers via satellite. The company then packages multiple channels into bundles for basic
cable as well as two or three other options of premium movie and/or sports channels. The
headend also has an interconnection to the Internet, where it can supply Internet services or
connect to a separate Internet service provider.
The headend connects to the end user via a network of fiber-optic and coax cables. The TV
channels and Internet channels are frequency multiplexed and modulated on to the main
fiber-optic cable for transport out to distribution hubs that rejuvenate the signals over longer
cable runs. From the one or more distribution hubs, the signal travels to multiple optical
nodes located in various city or suburban neighborhoods. In a typical configuration, a single
fiber is split to serve four fiber optical nodes. Most fiber nodes serve up to 500 homes. With
this arrangement, each fiber serves up to 2000 homes, although not all homes passed have a
cable modem or service.
The optical nodes convert the optical signals into electrical signals for the final distribution
via coax cable. The most common cable is RG-6/U 75-ohm coax using F-type connectors.
All of the homes receive the same signal, just like a bus network topology. In some areas
with longer distances, amplifiers are added along the way to mitigate the large cable losses
that are common.
All of the TV signals and Internet data are transmitted in a spectrum of 6-MHz wide
channels. Since a coax cable has a bandwidth as wide as 850 MHz to 1 GHz, the system can
accommodate from 140 to 170 downstream channels of 6 MHz each. The TV signals or
Internet data are modulated on to carriers in each channel. There are also upstream channels
that allow the consumer to transmit data back to the headend. This communication takes
places in 6-MHz channels as well that occupy the cable spectrum from 5 MHz to 40 MHz or
in some systems up to 65 MHz.
The composite video signal is developed in equipment called the cable modem termination
system (CMTS). In older systems, the video information is modulated on to the 6-MHz
channel carriers and then all channels are combined or linearly mixed to form the composited
cable signal (Fig. 2a). However, today it’s possible to synthesize a full block of modulated
channels digitally. The digitized video is sent to an ASIC or FPGA programmed to produce
the desired quadrature amplitude modulation (QAM) for each channel (Fig. 2b). The signals
are then digitally upconverted to the final frequency and sent to a wideband digital-to-analog
converter (DAC) that produces the composite multi-channel signal to be sent to the cable.
The original DSL system was designed to produce data rates of 1.5 Mbits/s to 8 Mbits/s
downstream from the telephone company to the subscriber and a lower rate upstream. Most
Internet access involves more downloading and less uploading of data. The resulting design
is referred to as asymmetrical DSL or ADSL. Most DSL formats are asymmetrical, although
there are DSL variations that deliver the same rates in both directions.
The great attenuation, noise, and crosstalk problems of bundling multiple twisted-pair lines
are the primary limitations of the POTS. These lines are effectively long low-pass filters with
upper frequency limits that reduce the bandwidth of the line and limit the data rate that can
be achieved. Line bandwidth is a function of the length of the UTP. Shorter cable runs have
wider bandwidths, so they are clearly more capable of high data rates than longer runs. But
despite this limitation, developments in digital signal processing have made this once limited
communications medium capable of high-speed data delivery.
UNIT 2 :
DATA LINK CONTROL - OSI Model
Data link layer is most reliable node to node delivery of data. It forms frames from the
packets that are received from network layer and gives it to physical layer. It also
synchronizes the information which is to be transmitted over the data. Error controlling is
easily done. The encoded data are then passed to physical.
Error detection bits are used by the data link layer. It also corrects the errors. Outgoing
messages are assembled into frames. Then the system waits for the acknowledgements to be
received after the transmission. It is reliable to send message.
FUNCTIONS OF DATA LINK LAYER:
• Framing: Frames are the streams of bits received from the network layer into
manageable data units. This division of stream of bits is done by Data Link Layer.
• Physical Addressing: The Data Link layer adds a header to the frame in order to
define physical address of the sender or receiver of the frame, if the frames are to be
distributed to different systems on the network.
• Flow Control: A flow control mechanism to avoid a fast transmitter from running a
slow receiver by buffering the extra bit is provided by flow control. This prevents
traffic jam at the receiver side.
• Error Control: Error control is achieved by adding a trailer at the end of the frame.
Duplication of frames are also prevented by using this mechanism. Data Link Layers
adds mechanism to prevent duplication of frames.
• Access Control: Protocols of this layer determine which of the devices has control
over the link at any given time, when two or more devices are connected to the same
link.
Data Link Layer - Creating a Frame
The description of a frame is a key element of each Data Link layer protocol. Data Link layer
protocols require control information to enable the protocols to function. Control information
may tell:
• Which nodes are in communication with each other

• When communication between individual nodes begins and when it ends
• Which errors occurred while the nodes communicated
• Which nodes will communicate next
The Data Link layer prepares a packet for transport across the local media by encapsulating it
with a header and a trailer to create a frame. Unlike the other PDUs that have been discussed
in this course, the Data Link layer frame includes:
Data - The packet from the Network layer
Header - Contains control information, such as addressing, and is located at the beginning of
the PDU
Trailer - Contains control information added to the end of the PDU

Formatting Data for Transmission
When data travels on the media, it is converted into a stream of bits, or 1s and 0s. If a node is
receiving long streams of bits, how does it determine where a frame starts and stops or which
bits represent the address?
Framing breaks the stream into decipherable groupings, with control information inserted in
the header and trailer as values in different fields. This format gives the physical signals a
structure that can be received by nodes and decoded into packets at the destination.
Typical field types include:
Start and stop indicator fields - The beginning and end limits of the frame
Naming or addressing fields
Type field - The type of PDU contained in the frame

Quality - control fields
A data field -The frame payload (Network layer packet)
Fields at the end of the frame form the trailer. These fields are used for error detection and
mark the end of the frame. Not all protocols include all of these fields. The standards for a
specific Data Link protocol define the actual frame format. Examples of frame formats will
be discussed at the end of this chapter.
Framing technique does not work satisfactorily, because networks generally do not make any
guarantees about the timing. So some other methods are derived.
Framing methods :
Following methods are used for carrying out framing.

1. Character count
2. Starting and ending characters, with character stuffing.
3. Starting and ending flags with bit stuffing.
4. Physical layer coding violations.
Character count :
In this method, a field in the header is used to specify the number of characters in the frame. This
number helps the receiver to know the number of characters in the frame following this count.
The character count method is illustrated in Fig. 5.
Character Count Method
The two frames shown in Fig. 5 are of 3 and 8 characters respectively. The disadvantage of this
method is that, an error can change the character count. If the wrong character count number is
received then the receiver will get out of synchronization and will be unable to locate the start of
next frame. The character count method is rarely used in practice.
Starting and ending character with character stuffing :
The problem of character count method is solved here by using a starting character before the
starting of each frame and an ending character at the end of each frame. Each frame is preceded
by the transmission of ASCII character sequence DLE STX. (DLE stands for data link escape
and STX is start of Text). After each frame the ASCII character sequence DLE ETX is
transmitted. Here DLE stands for Data Link Escape and ETX stands for End of Text. So if the
receiver loses the synchronization, it just to search for the DLE STX or DLE ETX characters to
return back on track. This is shown in Fig. 6.
Starting and ending character with character stuffing
Character stuffing :
The problem with this system is that the characters DLE STX or DLE ETX can be a part of data
as well. If so, they will be misinterpreted by the receiver as start or end of frame. This problem is
solved by using a technique called character stuffing. Which is as follows :
The data link layer at the sending end inserts an ASCII DLE character just before each accidental
DLE character in the data. The data link layer at the receiving end will remove these DLE
characters before handing over the data to the network layer.
Character Stuffing
Thus the framing DLE STX or DLE ETX can be distinguished from the one in data because
DLEs in the data are always doubled. This is called character stuffing and it is shown in Fig. 7.
At the receiving end the destuffing is essential.
Starting and ending flags, with bit stuffing :
This technique allows the frames to contain an arbitrary number of bits and codes different from
ASCII code. At the beginning and end of each frame, a specific bit pattern 0111 1110 called flag
byte is Since there are six consecutive Is in this byte a technique called bit stuffing which is
similar to character stuffing is used.
Bit stuffing :
Whenever the sender data link layer detects the presence of five consecutive ones in the data, it
automatically stuffs a O bit into the outgoing bit stream. This is called bit stuffing and it is as
shown in fig. 8.
Bit Stuffing and Destuffing
When a receiver detects presence of five consecutive ones in the received bit stream, it
automatically deletes the O bit following the five ones. This is called de-stuffing. It is shown in
Fig. 8. Due to bit stuffing, the possible problem if the data contains the flag byte pattern (0111
1110) is eliminated.
Data-link layer is responsible for implementation of point-to-point flow and error control
mechanism.
Flow Control
When a data frame (Layer-2 data) is sent from one host to another over a single medium, it is
required that the sender and receiver should work at the same speed. That is, sender sends at
a speed on which the receiver can process and accept the data. What if the speed
(hardware/software) of the sender or receiver differs? If sender is sending too fast the
receiver may be overloaded, (swamped) and data may be lost.
Two types of mechanisms can be deployed to control the flow:
Stop and Wait
This flow control mechanism forces the sender after transmitting a data frame to stop and
wait until the acknowledgement of the data-frame sent is received.
Sliding Window
In this flow control mechanism, both sender and receiver agree on the number of data-frames
after which the acknowledgement should be sent. As we learnt, stop and wait flow control
mechanism wastes resources, this protocol tries to make use of underlying resources as much
as possible.
Error Control
When data-frame is transmitted, there is a probability that data-frame may be lost in the
transit or it is received corrupted. In both cases, the receiver does not receive the correct data-
frame and sender does not know anything about any loss.In such case, both sender and
receiver are equipped with some protocols which helps them to detect transit errors such as
loss of data-frame. Hence, either the sender retransmits the data-frame or the receiver may
request to resend the previous data-frame.
Requirements for error control mechanism:
Error detection - The sender and receiver, either both or any, must ascertain that there is
some error in the transit.
Positive ACK - When the receiver receives a correct frame, it should acknowledge it.
Negative ACK - When the receiver receives a damaged frame or a duplicate frame, it sends
a NACK back to the sender and the sender must retransmit the correct frame.
Retransmission: The sender maintains a clock and sets a timeout period. If an

acknowledgement of a data-frame previously transmitted does not arrive before the timeout
the sender retransmits the frame, thinking that the frame or it’s acknowledgement is lost in
transit.
NOISELESS CHANNELS
Let us first assume we have an ideal channel in which no frames are lost, duplicated, or
corrupted. We introduce two protocols for this type of channel. The first is a protocol that
does not use flow control; the second is the one that does. Of course, neither has error control
because we have assumed that the channel is a perfect noiseless channel.
In order to appreciate the step by step development of efficient and complex protocols
such as SDLC, HDLC etc., we will begin with a simple but unrealistic protocol. In
this protocol:
• Data are transmitted in one direction only

• The transmitting (Tx) and receiving (Rx) hosts are always ready
• Processing time can be ignored
• Infinite buffer space is available
• No errors occur; i.e. no damaged frames and no lost frames (perfect channel)
The protocol consists of two procedures, a sender and receiver as depicted below:
/* protocol 1 */
Sender()
{
forever
{
from_host(buffer);
S.info = buffer;
sendf(S);
}
}
Receiver()
{
forever
{
wait(event);
getf(R);
to_host(R.info);
}
}
A simplex stop-and-wait protocol

In this protocol we assume that
• Data are transmitted in one direction only

• No errors occur (perfect channel)
• The receiver can only process the received information at a finite rate
These assumptions imply that the transmitter cannot send frames at a rate faster than the receiver
can process them.
The problem here is how to prevent the sender from flooding the receiver.
A general solution to this problem is to have the receiver provide some sort of
feedback to the sender. The process could be as follows: The receiver send an
acknowledge frame back to the sender telling the sender that the last received frame
has been processed and passed to the host; permission to send the next frame is
granted. The sender, after having sent a frame, must wait for the acknowledge frame
from the receiver before sending another frame. This protocol is known as stop-and-
wait.
The protocol is as follows:
/* protocol 2 */
Sender()
{
forever
{
from_host(buffer);
S.info = buffer;
sendf(S);
wait(event);
}
}
Receiver()
{
forever
{
wait(event);
getf(R);
to_host(R.info);
sendf(S);
}}
A simplex protocol for a noisy channel
In this protocol the unreal "error free" assumption in protocol 2 is dropped. Frames
may be either damaged or lost completely. We assume that transmission errors in the
frame are detected by the hardware checksum.
One suggestion is that the sender would send a frame, the receiver would send an
ACK frame only if the frame is received correctly. If the frame is in error the receiver
simply ignores it; the transmitter would time out and would retransmit it.
One fatal flaw with the above scheme is that if the ACK frame is lost or damaged,
duplicate frames are accepted at the receiver without the receiver knowing it.
Imagine a situation where the receiver has just sent an ACK frame back to the sender
saying that it correctly received and already passed a frame to its host. However, the
ACK frame gets lost completely, the sender times out and retransmits the frame.
There is no way for the receiver to tell whether this frame is a retransmitted frame or a
new frame, so the receiver accepts this duplicate happily and transfers it to the host.
The protocol thus fails in this aspect.
To overcome this problem it is required that the receiver be able to distinguish a frame
that it is seeing for the first time from a retransmission. One way to achieve this is to
have the sender put a sequence number in the header of each frame it sends. The
receiver then can check the sequence number of each arriving frame to see if it is a
new frame or a duplicate to be discarded.
The receiver needs to distinguish only 2 possibilities: a new frame or a duplicate; a 1-

bit sequence number is sufficient. At any instant the receiver expects a particular
sequence number. Any wrong sequence numbered frame arriving at the receiver is
rejected as a duplicate. A correctly numbered frame arriving at the receiver is
accepted, passed to the host, and the expected sequence number is incremented by 1
(modulo 2).
The protocol is depicted below:
/* protocol 3 */
Sender()
{
NFTS = 0; /* NFTS = Next Frame To Send */
from_host(buffer);
forever
{
S.seq = NFTS;
S.info = buffer;
sendf(S);
start_timer(S.seq);
wait(event);
if(event == frame_arrival)
{
from_host(buffer);
++NFTS; /* modulo 2 operation */
}
}
}
Receiver()
{
FE = 0; /* FE = Frame Expected */
forever
{
wait(event);
if(event == frame_arrival)
{
getf(R);
if(R.seq == FE)
{
to_host(R.info);
++FE; /* modulo 2 operation */
}
sendf(S); /* ACK */
}
}
}
This protocol can handle lost frames by timing out. The timeout interval has to be
long enough to prevent premature timeouts which could cause a "deadlock" situation.
Stop and Wait ARQ
Stop and Wait transmission is the simplest reliability technique and is adequate for a
very simple communications protocol. A stop and wait protocol transmits a Protocol
Data Unit (PDU) of information and then waits for a response. The receiver receives
each PDU and sends an Acknowledgement (ACK) PDU if a data PDU is received
correctly, and a Negative Acknowledgement (NACK) PDU if the data was not
received. In practice, the receiver may not be able to reliably identify whether a PDU
has been received, and the transmitter will usually also need to implement a timer to
recover from the condition where the receiver does not respond.
Under normal transmission the sender will receive an ACK for the data and then
commence transmission of the next data block. For a long delay link, the sender may
have to wait an appreciable time for this response. While it is waiting the sender is
said to be in the "idle" state and is unable to send further data.
Stop and Wait ARQ - Waiting for Acknowledgment (ACK) from the remote node.
The blue arrows show the sequence of data PDUs being sent across the link from the
sender (top to the receiver (bottom). A Stop and Wait protocol relies on two way
transmission (full duplex or half duplex) to allow the receiver at the remote node to
return PDUs acknowledging the successful transmission. The acknowledgements are
shown in green in the diagram, and flow back to the original sender. A small
processing delay may be introduced between reception of the last byte of a Data PDU
and generation of the corresponding ACK.
When PDUs are lost, the receiver will not normally be able to identify the loss (most
receivers will not receive anything, not even an indication that something has been
corrupted). The transmitter must then rely upon a timer to detect the lack of a
response.
Stop and Wait ARQ - Retransmission due to timer expiry
In the diagram, the second PDU of Data is corrupted during transmission. The
receiver discards the corrupted data (by noting that it is followed by an invalid
data checksum). The sender is unaware of this loss, but starts a timer after sending
each PDU. Normally an ACK PDU is received before this the timer expires. In this
case no ACK is received, and the timer counts down to zero and triggers
retransmission of the same PDU by the sender. The sender always starts a timer
following transmission, but in the second transmission receives an ACK PDU before
the timer expires, finally indicating that the data has now been received by the remote
node.
The state diagram (also showing the operation of NACK) is shown below:
State Diagram for a simple stop and wait protocol

Go Back n Protocol
Go Back n is a connection oriented protocol in which the transmitter has a window of

sequence numbers that may be transmitted without acknowledgment. The receiver
will only accept the next sequence number it is expecting - other sequence nubmers
are silently ignored.
The protocol simulation shows a time-sequence diagram with users A and B, protocol
entities A and B that support them, and a communications medium that carries
messages. Users request data transmissions with DatReq(DATAn), and receive data
transmissions as DatInd(DATAn). Data messages are simply
numbered DATA0, DATA1, etc. without explicit content. The transmitting protocol
sends the protocol message DT(n) that gives only the sequence number, not the data.
Once sequence numbers reach a maximum number (like 7), they wrap back round to
0. An acknowledgement AK(n) means that the DT message numbered n is the next
one expected (i.e. all messages up to but not including this number have been
received). Since sequence numbers wrap round, an acknowledgement with sequence
number 1 refers to messages 0, 1, 7, 6, etc. Note that if aDT message is received again
due to re-transmission, it is acknowledged but discarded.
The protocol has a maximum number of messages that can be sent without
acknowledgement. If this window becomes full, the protocol is blocked until an
acknowledgement is received for the earliest outstanding message. At this point the
transmitter is clear to send more messages.
The receiver delivers the protocol messages DT(n) to the user in order. Any received
out of order are ignored.
Selective Repeat Error Recovery
Selective Repeat error recovery is a procedure which is implemented in

some communications protocols to provide reliability. It is the most complex of a set
of procedures which may provide error recovery, it is however the most efficient
scheme. Selective repeat is employed by the TCPtransport protocol.
Features required for Selective Repeat ARQ
• To support Go-Back-N ARQ, a protocol must number each PDU which is sent. (PDUs
are normally numbered using modulo arithmetic, which allows the same number to be re-
used after a suitably long period of time. The time period is selected to ensure the same
PDU number is never used again for a different PDU, until the first PDU has "left the
network" (e.g. it may have been acknowledged)).
• The local node must also keep a buffer of all PDUs which have been sent, but have not
yet been acknowledged.
• The receiver at the remote node keeps a record of the highest numbered PDU which has
been correctly received. This number corresponds to the last acknowledgement PDU
which it may have sent.
The above features are also required for Go-Back-N, however for selective repeat, the
receiver must also maintain a buffer of frames which have been received, but not
acknowledged.
Recovery of lost PDUs using Selective Repeat ARQ
The recovery of a corrupted PDU proceeds in four stages:
• First, the corrupted PDU is discarded at the remote node's receiver.

• Second, the remote node requests retransmission of the missing PDU using a
control PDU (sometimes called a Selective Reject). The receiver then stores all
out-of-sequence PDUs in the receive buffer until the requested PDU has been
retransmitted.
• The sender receives the retransmission request and then transmits the lost
PDU(s).
• The receiver forwards the retransmitted PDU, and all subsequent in-sequence
PDUs which are held in the receive buffer.
A remote node may request retransmission of corrupted PDUs by initiating Selective

Repeat error recovery by sending a control PDU indicating the missing PDU. This
allows the remote node to instruct the sending node where to retransmit the PDU
which has not been received. The remotestores any out-of-sequence PDUs (i.e. which
do not have the expected sequence number) until the retransmission is complete.
Upon receipt of a Selective Repeat control PDU (by the local node), the
transmitter sends a single PDU from its buffer of unacknowledged PDUs. The
transmitter then continues normal transmission of new PDUs until the PDUs are
acknowledged or another selective repeat request is received.
HDLC Encapsulation
ISO Standard - current ISO 13239
▪ bit-oriented protocol
▪ Developed from the Synchronous Data Link Control
▪ Provides connection-oriented and connectionless service
▪ Synchronous serial transmission to provide error-free communication between devices
▪ HDLC uses a frame delimiter to mark the beginning and end of each frame, something that a
layer 1 protocol cannot do over synchronous nor asynchronous transfer.
▪ Cisco has a proprietary, but vendor-friendly, implementation of HDLC that solves
multiprotocol support problems.
Notable Frame Fields
▪ Flag - The flag field initiates and terminates error checking. The frame always starts and ends
with an 8-bit flag field. The bit pattern is 01111110. Because there is a likelihood that this
pattern occurs in the actual data, the sending HDLC system always inserts a 0 bit after every
five 1s in the data field, so in practice the flag sequence can only occur at the frame ends.
The receiving system strips out the inserted bits. When frames are transmitted consecutively,
the end flag of the first frame is used as the start flag of the next frame.
▪ Address - The address field contains the HDLC address of the secondary station. This
address can contain a specific address, a group address, or a broadcast address. A primary
address is either a communication source or a destination, which eliminates the need to
include the address of the primary.
▪ Control
▪ Information (I) frame: I-frames carry upper layer information and some control
information. This frame sends and receives sequence numbers, and the poll final (P/F) bit
performs flow and error control. The send sequence number refers to the number of the
frame to be sent next. The receive sequence number provides the number of the frame to
be received next. Both sender and receiver maintain send and receive sequence numbers.
A primary station uses the P/F bit to tell the secondary whether it requires an immediate
response. A secondary station uses the P/F bit to tell the primary whether the current
frame is the last in its current response.
▪ Supervisory (S) frame: S-frames provide control information. An S-frame can request
and suspend transmission, report on status, and acknowledge receipt of I-frames. S-
frames do not have an information field.
▪ Unnumbered (U) frame: U-frames support control purposes and are not sequenced. A U-
frame can be used to initialize secondaries. Depending on the function of the U-frame, its
control field is 1 or 2 bytes. Some U-frames have an information field.
▪ Protocol - (only used in Cisco HDLC) This field specifies the protocol type encapsulated
within the frame (e.g. 0x0800 for IP).
▪ Data - The data field contains a path information unit (PIU) or exchange identification (XID)
information.
▪ Frame check sequence (FCS) - The FCS precedes the ending flag delimiter and is usually a
cyclic redundancy check (CRC) calculation remainder. The CRC calculation is redone in the
receiver. If the result differs from the value in the original frame, an error is assumed.
Point-to-Point Protocol (PPP)
• PPP is most commonly used data link protocol. It is used to connect the Home PC to
the server of ISP via a modem.
• This protocol offers several facilities that were not present in SLIP. Some of these
facilities are:
1. PPP defines the format of the frame to be exchanged between the devices.
2. It defines link control protocol (LCP) for:-
(a) Establishing the link between two devices.

(b) Maintaining this established link.
(c) Configuring this link.
(d) Terminating this link after the transfer.
3. It defines how network layer data are encapsulated in data link frame.
4. PPP provides error detection.
5. Unlike SLIP that supports only IP, PPP supports multiple protocols.
6. PPP allows the IP address to be assigned at the connection time i.e. dynamically. Thus
a temporary IP address can be assigned to each host.
7. PPP provides multiple network layer services supporting a variety of network layer
protocol. For this PPP uses a protocol called NCP (Network Control Protocol).
8. It also defines how two devices can authenticate each other.
PPP Frame Format
The frame format of PPP resembles HDLC frame. Its various fields are:
PPP Frame Format
The frame format of PPP resembles HDLC frame. Its various fields are:
1. Flag field: Flag field marks the beginning and end of the PPP frame. Flag byte is
01111110. (1 byte).
2. Address field: This field is of 1 byte and is always 11111111. This address is the
broadcast address i.e. all the stations accept this frame.
3. Control field: This field is also of 1 byte. This field uses the format of the U-frame
(unnumbered) in HDLC. The value is always 00000011 to show that the frame does not
contain any sequence numbers and there is no flow control or error control.
4. Protocol field: This field specifies the kind of packet in the data field i.e. what is
being carried in data field.
5. Data field: Its length is variable. If the length is not negotiated using LCP during line
set up, a default length of 1500 bytes is used. It carries user data or other information.
6. FCS field: The frame checks sequence. It is either of 2 bytes or 4 bytes. It contains
the checksum.
Transition Phases in PPP
The PPP connection goes through different states as shown in fig.
1. Dead: In dead phase the link is not used. There is no active carrier and the line is
quiet.
2. Establish: Connection goes into this phase when one of the nodes start
communication. In this phase, two parties negotiate the options. If negotiation is
successful, the system goes into authentication phase or directly to networking phase.
LCP packets are used for this purpose.
3. Authenticate: This phase is optional. The two nodes may decide during the
establishment phase, not to skip this phase. However if they decide to proceed with
authentication, they send several authentication packets. If the result is successful, the
connection goes to the networking phase; otherwise, it goes to the termination phase.
4. Network: In network phase, negotiation for the network layer protocols takes place.
PPP specifies that two nodes establish a network layer agreement before data at the
network layer can be exchanged. This is because PPP supports several protocols at
network layer. If a node is running multiple protocols simultaneously at the network
layer, the receiving node needs to know which protocol will receive the data.
5. Open: In this phase, data transfer takes place. The connection remains in this phase
until one of the endpoints wants to end the connection.
6. Terminate: In this phase connection is terminated.
Point-to-point protocol Stack
PPP uses several other protocols to establish link, authenticate users and to carry the
network layer data.
The various protocols used are:
1. Link Control Protocol
2. Authentication Protocol
3. Network Control Protocol
1. Link Control Protocol
• It is responsible for establishing, maintaining, configuring and terminating the link.
• It provides negotiation mechanism to set options between two endpoints.
• All LCP packets are carried in the data field of the PPP frame.
• The presence of a value C02116 in the protocol field of PPP frame indicates that LCP
packet is present in the data field.
• The various fields present in LCP packet are:
1. Code: 1 byte-specifies the type of LCP packet.
2. ID: 1 byte-holds a value used to match a request with the reply.
3. Length: 2 byte-specifies the length of entire LCP packet.

4. Information: Contains extra information required for some LCP packet.
• There are eleven different type of LCP packets. These are categorized in three groups:
1. Configuration packet: These are used to negotiate options between the two ends.
For example: configure-request, configure-ack, configure-nak, configure-reject are some
configuration packets.
2. Link termination packets: These are used to disconnect the link between two end
points. For example: terminate-request, terminate-ack, are some link termination
packets.
3. Link monitoring and debugging packets: These are used to monitor and debug
the links. For example: code-reject, protocol-reject, echo-request, echo-reply and
discard-request are some link monitoring and debugging packets.
2. Authentication Protocol
Authentication protocols help to validate the identity of a user who needs to access the
resources.
There are two authentication protocols:
1. Password Authentication Protocols (PAP)
2. Challenge Handshake Authentication Protocol (CHAP)
1. PAP (Password Authentication Protocol)
This protocol provides two step authentication procedures:
Step 1: User name and password is provided by the user who wants to access a system.
Step 2: The system checks the validity of user name and password and either accepts or
denies the connection.
• PAP packets are also carried in the data field of PPP frames.
• The presence of PAP packet is identified by the value C02316 in the protocol field of
PPP frame.
• There are three PAP packets.
1. Authenticate-request: used to send user name & password.
2. Authenticate-ack: used by system to allow the access.
3. Authenticate-nak: used by system to deny the access.
2. CHAP (Challenge Handshake Authentication Protocol)
• It provides more security than PAP.
• In this method, password is kept secret, it is never sent on-line.

• It is a three-way handshaking authentication protocol:
1. System sends. a challenge packet to the user. This packet contains a value, usually a
few bytes.
2. Using a predefined function, a user combines this challenge value with the user
password and sends the resultant packet back to the system.
3. System then applies the same function to the password of the user and challenge
value and creates a result. If result is same as the result sent in the response packet,
access is granted, otherwise, it is denied.
• There are 4 types of CHAP packets:
1. Challenge-used by system to send challenge value.
2. Response-used by the user to return the result of the calculation.
3. Success-used by system to allow access to the system.
4. Failure-used by the system to deny access to the system.
3. Network Control Protocol (NCP)
• After establishing the link and authenticating the user, PPP connects to the network
layer. This connection is established by NCP.
• Therefore NCP is a set of control protocols that allow the encapsulation of the data
coming from network layer.
• After the network layer configuration is done by one of the NCP protocols, the users
can exchange data from the network layer.
• PPP can carry a network layer data packet from protocols defined by the Internet,
DECNET, Apple Talk, Novell, OSI, Xerox and so on.
• None of the NCP packets carry networks layer data. They just configure the link at the
network layer for the incoming data.
Piggy Backing
In all practical situations, the transmission of data needs to be bi-directional. This is called as
full-duplex transmission.
We can achieve this full duplex transmission i.e. by having two separate channels-one for
forward data transfer and the other for separate transfer i.e. for acknowledgements.
• A better solution would be to use each channel (forward & reverse) to transmit frames both
ways, with both channels having the same capacity. If A and B are two users. Then the data
frames from A to Bare intermixed with the acknowledgements from A to B.
• One more improvement that can be made is piggybacking. The concept is explained as follows:
In two way communication, Whenever a data frame is received, the received waits and
does not send the control frame (acknowledgement) back to the sender immediately.
The receiver waits until its network layer passes in the next data packet. The delayed
acknowledgement is then attached to this outgoing data frame.
This technique of temporarily delaying the acknowledgement so that it can be hooked
with next outgoing data frame is known as piggybacking.
• The major advantage of piggybacking is better use of available channel bandwidth.
• The disadvantages of piggybacking are:
1. Additional complexity.
2. If the data link layer waits too long before transmitting the acknowledgement, then
retransmission of frame would take place.
Objectives:
The student shall be able to:
Describe how CSMA/CD and CSMA/CA work, and the differences between the two.
Be able to determine the number of collisions and successful transmissions given an
example situation for the following technologies: Aloha, Slotted Aloha, CSMA/CD,
CSMA/CA.
Define FDMA, TDMA, CDMA, Channelization, Scheduling, Reservation, Polling, and
briefly define how they work.
Define switch, bridge, hub.
UNIT 3,4,5 : Medium Access Control (MAC)

Local Area Networks (LAN)
Introduction
LAN: used to interconnect distributed terminals located within a single building or
localized group of buildings.
Network
Switche Broadcast
Channelization Dynamic
Scheduling Random
Reservatio Polling
Medium Access Control: Coordinate access to a channel so that information can be

transmitted from a source to a destination in a broadcast network.
Channelization Scheme: Partitions medium into separate channels that are dedicated
to particular users.
• Base Station or Controller  Station: Forward or Downstream transmission
• Station  Base Station or Controller: Reverse or Upstream transmission
Examples: Cellular FDMA, TDMA, CDMA; Multidrop
Dynamic MAC Scheme: Direct communication between all M stations

Dynamic sharing of the medium on a per packet basis.
No central control
Must minimize collision of packet transmission by multiple sources simultaneously.
Example: Ethernet, Ad Hoc Wireless LANs, Multitapped bus.
Dynamic Medium Access Control: Random Access
Random Access: Transmit immediately and then randomly if collision occurred

Coordination improves throughput.
Aloha Protocol
Aloha protocol: From University of Hawaii

Everyone transmits when they want to transmit.
Result: a lot of collisions.
Maximum throughput: 18.4% (when something to send 50% of time).
Throughput calculations:
S = Throughput
G = Average number of frames attempting transmission
Assumes frames are of equal length
P(0 frames) = e-2G
S = G(e-2G)
Highest throughput achieved when: G=0.5 S=18.4%
Slotted Aloha protocol:

Terminal can only transmit during fixed time intervals called timeslots.
Max throughput achieved at: 37% successes, 37% slots empty, 26% collisions
(assumes Poisson arrival rate).
Current Use: Cellular Random Access Channel
Throughput calculations:
S = Throughput
G = Average number of frames attempting transmission
Assumes Poisson Distribution
P(0 frames) = e-G
S = G(e-G)
E = Expected # of transmissions to achieve successful transmission:

E = eG
Highest throughput achieved when G = 1
S = 36.79%
E = 2.718
CSMA/CD
Carrier Sense Multiple Access with Collision Detection (CSMA/CD): Ethernet: 802.3
Carrier Sense: Before transmitting a frame the source first listens to see if someone
is using the medium, and waits until the transmission is complete.
• Multiple Access: Multiple stations can access media concurrently – broadcast
mode
o 1-Persistent CSMA: As soon as channel becomes idle, transmit packet
o P-Persistent CSMA: Transmit packet with probability P as channel becomes
idle – else wait one propagation delay and try again.
Collision Detect:
1. When transmitting, listen to see if own signal is received back.
2. If collision has occurred jam and then stop the transmission.
3. Retransmit after a random period of time.
Collision Determination: How long will it take to determine if a collision has occurred?
Assume a 1km long coax cable.
t = propagation delay between 2 furthest end points = 5 usec
Assume Node 1 is at one end of cable and transmits first.
Assume Node 2 is at other end of cable and begins to transmit just as first bit is
arriving at Node 2.
Collision detect will take twice the propagation delay of sending bits from one end to
other end of cable.
Example: Ethernet
Uses optical fiber, twisted pair or coax from 10 Mbps to 1 Gbps.
Ethernet Frame format:

1. Preamble: 10101010: synchronizes sender and receiver.
2. Start of frame delimiter: 10101011: indicates start of frame.
3. Destination address 16 or 48 bits.
4. Source address
5. Type of above layer: IP, ARP, or RARP
o IEEE 802.3: Replaces type with length.
6. Data field: 0 to 1500 bytes
7. Pad: fills out to minimum 64 bytes if necessary.
8. FCS: CRC 4 octets.
Hub: Forwards transmissions in all directions

Has ports to multiple terminals in star configuration
Received frames broadcast to all other lines

If collision detected, hub jams to cause backoff algorithm
Used with 10BaseT: Twisted pair LAN at 10 Mbps.
Dynamic Medium Access Control: Scheduling

Scheduling: Dynamic form of Time Division Multiplexing
Assumes centrally scheduled system
Reservation
Two Periods:
Reservation Interval: Stations bid for time
M minislots: Each minislot allocated to 1 terminal OR
Random Access bidding
Data Transmission Interval: Packets transmitted in order (optionally: and duration and
number) as specified during the Reservation Interval
Frame transmission = 1 + v where v = time for minislot
Maximum throughput = 1 / (1 + v) for fixed-size 1-frame transmissions
Example: Cellular Packet Data (General Packet Radio Services)
Polling
Central Controller polls stations
CC: Do you have anything to send?
Station: Yes, here it is .. I am done OR No, I have nothing to send
Walk time: time to poll a station: overhead
Total Walk time: Sum of Walk times for 1 polling round.
Example:
IBM SNRM,
Token: Traverses the ring, giving owner of token permission to transmit
o Packet can be removed at destination or origination
Channelization
Frequency Division Multiple Access (FDMA): A station is allocated a frequency all

the time
Time Division Multiple Access (TDMA): Stations have assigned slot to transmit and
thus take turns transmitting.
Code Division Multiple Access (CDMA): Station transmits over all the frequency all
the time.
Each bit is transmitted multiple times according to the chip rate, using a unique
binary pseudorandom sequence, called the chip sequence.
The received signal is correlated with the terminals chip sequence and summed
since:
o C12 + C2 2 + C3 2+ … + CG 2= G
o Where each C is either –1 or +1, and –12 = +1 and 12 = +1
Frequency Division Duplex (FDD): Base station and terminal transmit to each other
on different frequencies.
Time Division Duplex (TDD): Base station and terminals transmit on same frequency
at different times.
Switched Networks
Switch: A form of lower-layer router

Has ports to multiple terminals in star configuration
Incoming frames are transferred to single appropriate outgoing port
May buffer data (packet switch)
Uses routing tables to forward packets to correct destination
Multiplexes packets toward destination.
Forwards according to layer 2 address.
Two Modes:
1) Dedicated lines in both directions to single terminal: No collisions
2) Dedicated lines in both directions to hub and multiple terminals:
o Collisions possible on uplink.
Fast Ethernet: 100 Mbps

Uses frame structure of IEEE 802.3 standard
Physical Layer:
o 100BaseT4: 4 x UTP 3: single direction.
o 100BaseTX: 2 x UTP 5: can operate in full-duplex
o 100BaseFX: 2 x fiber: full duplex
Uses hubs or switches
Gigabit Ethernet: 1 Gbps speed

Uses frame structure of IEEE 802.3 standard
Star configuration
Physical layer:
o 1000BaseSX: 2 x Optical fiber @ 550 m
o 1000BaseLX: 2 x Optical fiber @ 5 km
o 1000BaseCX: Shielded copper @ 25 m
o 1000BaseT: Cat. 5 UTP @ 100 m
Problem: TX of frame can occur without recognizing collision occurred.
Solution: Minimum slot size of 512 bytes
o Packet Bursting: TX a burst of small packets.
IEEE 802.11: Wireless LAN
IEEE 802.11 Wireless LAN protocol.

Supports rates of 11, 5.5, 2, 1 Mbps.
Cannot hear any interference since transmitted signal is always the loudest signal.
Multiple stations may send simultaneously as long as intended receiver is closest to
the desired transmitting station.
Hidden Terminal: A terminal which can hear transmissions from few terminals
Time divided into two periods:

Point Coordination Function (PCF): Contention Free Period
Access Point (AP) uses polling to ensure low-delay applications get serviced
regularly
Distributed Coordination Function (DCF): Contention Period
Uses CSMA/Collision Avoidance to coordinate Random Access
CSMA/CA: Carrier Sense Multiple Access / Collision Avoidance

Only allowed to send if no other station is sending.
Collision Avoidance: Random backoff procedure
Certain types of packets may be transmitted if no transmission has occurred in a
period of T time, called an Interframe Space (IFS)
o SIFS or Short IFS: Can transmit Acks, poll responses, remaining packet
segments
o PIFS: PCF IFS: Access Point can request control, then Contention Free
period starts
o DIFS: DCF IFS: A random access packet can be transmitted
Reservation algorithm includes:
1. Send a Request To Send (RTS) packet notifying all how long the message
will be.
2. Receiver replies with Clear To Send (CTS) packet acknowledging RTS and
repeating length of message.
3. RTS/CTS notifies all neighbors within earshot of both transmitter and receiver
to not transmit during the message transmission.
Exercise on Multiple Access:
Assume transmissions are to be transmitted from the indicated terminals at the following
times. Assume each frame (for simplicity sake) is 0.99 unit long. Assume any
propagation delay is at most 0.1. All numbers are in time units.
Terminals are configured as follows:
+--------+-----------------------------------+----------------------------------+--------+
A B E C D
Terminal Time Frame

Generated
A->B 1.5
D->C 1.7
B->E 3.0
C->D 4.2
D->E 5.0
E->A 5.1
How many collisions will occur for:

Aloha?
Slotted Aloha?
CSMA/CD?
• IEEE 802.11?
More LAN Vocabulary
LLC: Logical Link Control

Addressing: Identifies the layer 3 protocol entity (IP, Novell IPX) in addition to:
Error control:
o Type 1: Unacknowledged Connectionless (i.e. UI or Unnumbered Info)
o Type 2: Reliable Connection Oriented (i.e. SABM and I-frames)
o Type 3: Acknowledged Connectionless: Individual frames with acks
Layer 2 consists of:
LLC
MAC
NIC:
NIC: Network Interface Card or LAN Adapter Card

Implements MAC protocol
Transfers data in serial mode over network: performs parallel to serial conversion
Own unique physical address is burned into ROM.
Interface with computer with PCMCIA card or other type card into expansion slot.
Routing / Repeaters:
Repeater: Repeats at the Physical Layer:
No intelligence: No checking of CRC
Bridge: Relay occurs at MAC layer.
Interfaces two LANs.
Builds routing tables from source addresses.
Router: Routing occurs at Network layer.
Application Layer:
Application Gateway
Transport Layer:
Transport Gateway
Network Layer:
Router
Data Link Layer:
Bridge, Switch
Physical Layer:
Repeater, Hub
UNIT 4
Wired Networks
Wired networks, also called Ethernet networks, are the most common type of local area
network (LAN) technology. A wired network is simply a collection
of two or more computers, printers, and other devices linked by
Ethernet cables. Ethernet is the fastest wired network protocol,
with connection speeds of 10 megabits per second (Mbps) to 100
Mbps or higher. Wired networks can also be used as part of other
wired and wireless networks. To connect a computer to a network
with an Ethernet cable, the computer must have an Ethernet
adapter (sometimes called a network interface card, or NIC). Ethernet adapters can be internal
(installed in a computer) or external (housed in a separate case). Some computers include a built-
in Ethernet adapter port, which eliminates the need for a separate adapter (Microsoft). There are
three basic network topologies that are most commonly used today. (Homenthelp.com)
The star network, a general more simplistic type of topology, has one central hub that
connects to three or more computers and the ability to network printers. This type can be used for
small businesses and even home networks. The star network is very useful for applications where
some processing must be centralized and some must be performed locally. The major
disadvantage is the star network is its vulnerability. All data must pass through one central host
computer and if that host fails the entire network will fail.
On the other hand the bus network has no central computer and all computers are linked
on a single circuit. This type broadcasts signals in all directions and it uses special software to
identify which computer gets what signal. One disadvantage
with this type of network is that only one signal can be sent at
one time, if two signals are sent at the same time they will
collide and the signal will fail to reach its destination. One
advantage is that there is no central computer so if one
computer goes down others will not be affected and will be
able to send messages to one another. (Laudon)
The third type of network is the ring network. Similar to the

bus network, the ring network does not rely on a central host
computer either. Each computer in the network can communicate
directly with any other computer, and each processes its own
applications independently. A ring network forms a closed loop and
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data is sent in one direction only and if a computer in the network fails the data is still able to be
transmitted.
Typically the range of a wired network is within a 2,000-foot-radius. The disadvantage

of this is that data transmission over this distance may be slow or nonexistent. The benefit of a
wired network is that bandwidth is very high and that interference is very limited through direct
connections. Wired networks are more secure and can be used in many situations; corporate
LANs, school networks and hospitals. The biggest drawback to this type of network is that it
must be rewired every time it is moved.
Wireless Networks
A wireless network, which uses high-frequency radio waves rather than wires to
communicate between nodes, is another option for home or business networking. Individuals
and organizations can use this option to expand their existing wired network or to go completely
wireless. Wireless allows for devices to be shared without networking cable which increases
mobility but decreases range. There are two main types of wireless networking; peer to peer or
ad-hoc and infrastructure. (Wi-fi.com)
An ad-hoc or peer-to-peer wireless network consists of a number of computers each

equipped with a wireless networking interface card. Each computer can communicate
directly with all of the other wireless enabled computers. They can share files and
printers this way, but may not be able to access wired LAN resources, unless one of the
computers acts as a bridge to the wired LAN using special software.
An infrastructure wireless network consists of an access point or a base station. In this type
of network the access point acts like a hub, providing connectivity for the wireless computers. It
can connect or bridge the wireless LAN to a wired LAN, allowing wireless computer access to
LAN resources, such as file servers or existing Internet Connectivity. (compnetworking.about.com)
There are four basic types of transmissions standards for wireless networking. These
types are produced by the Institute of Electrical and Electronic Engineers (IEEE). These
standards define all aspects of radio frequency wireless networking. They have established four
transmission standards; 802.11, 802.11a, 802.11b, 802.11g.
The basic differences between these four types are connection speed and radio frequency.
802.11 and 802.11b are the slowest at 1 or 2 Mbps and 5.5 and 11Mbps respectively. They both
operate off of the 2.4 GHz radio frequency. 802.11a operates off of a 5 GHz frequency and can
transmit up to 54 Mbps and the 802.11g operates off of the 2.4 GHz frequency and can transmit
up to 54 Mbps. Actual transmission speeds vary depending on such factors as the number and
size of the physical barriers within the network and any interference in the radio transmissions.
(Wi-fi.com)
Wireless networks are reliable, but when interfered with it can reduce the range and the
quality of the signal. Interference can be caused by other devices operating on the same radio
frequency and it is very hard to control the addition of new devices on the same frequency.
Usually if your wireless range is compromised considerably, more than likely, interference is to
blame. (Laudon)
A major cause of interference with any radio signals are the materials in your
surroundings, especially metallic substances, which have a tendency to reflect radio signals.
Needless to say, the potential sources of metal around a home are numerous--things like metal
studs, nails, building insulation with a foil backing and even lead paint can all possibly reduce
the quality of the wireless radio signal. Materials with a high density, like concrete, tend to be
harder for radio signals to penetrate, absorbing more of the energy. Other devices utilizing the
same frequency can also result in interference with your wireless. For example, the 2.4GHz
frequency used by 802.11b-based wireless products to communicate with each other. Wireless
devices don't have this frequency all to themselves. In a business environment, other devices that
use the 2.4GHz band include microwave ovens and certain cordless phones. (Laundon)
On the other hand, many wireless networks can increase the range of the signal by using
many different types of hardware devices. A wireless extender can be used to relay the radio
frequency from one point to another without losing signal strength. Even though this device
extends the range of a wireless signal it has some drawbacks. One drawback is that it extends
the signal, but the transmission speed will be slowed.
There are many benefits to a wireless network. The most important one is the option to
expand your current wired network to other areas of your organization where it would otherwise
not be cost effective or practical to do so. An organization can also install a wireless network
without physically disrupting the current workplace or wired network. (Wi-Fi.org) Wireless
networks are far easier to move than a wired network and adding users to an existing wireless
network is easy. Organizations opt for a wireless network in conference rooms, lobbies and
offices where adding to the existing wired network may be too expensive to do so.
Wired vs. Wireless Networking
The biggest difference between these two types of networks is one uses network cables
and one uses radio frequencies. A wired network allows for a faster and more secure connection
and can only be used for distances shorter than 2,000 feet. A wireless network is a lot less secure
and transmission speeds can suffer from outside interference. Although wireless networking is a
lot more mobile than wired networking the range of the network is usually 150-300 indoors and
up to 1000 feet outdoors depending on the terrain. (Homelanextream.com)
The cost for wired networking has become rather inexpensive. Ethernet cables, hubs and
switches are very inexpensive. Some connection sharing software packages, like ICS, are free;
some cost a nominal fee. Broadband routers cost more, but these are optional components of a
wired network, and their higher cost is offset by the benefit of easier installation and built-in
security features.
Wireless gear costs somewhat more than the equivalent wired Ethernet products. At full
retail prices, wireless adapters and access points may cost three or four times as much as
Ethernet cable adapters and hubs/switches, respectively. 802.11b products have dropped in price
considerably with the release of 802.11g. (Homelanextream.com)
Wired LANs offer superior performance. A traditional Ethernet connection offers only 10
Mbps bandwidth, but 100 Mbps Fast Ethernet technology costs a little more and is readily
available. Fast Ethernet should be sufficient for file sharing, gaming, and high-speed Internet
access for many years into the future. (Wi-Fi.org) Wired LANs utilizing hubs can suffer
performance slowdown if computers heavily utilize the network simultaneously. Use Ethernet
switches instead of hubs to avoid this problem; a switch costs little more than a hub.
Wireless networks using 802.11b support a maximum bandwidth of 11 Mbps, roughly the
same as that of old, traditional Ethernet. 802.11a and 802.11g LANs support 54 Mbps, that is
approximately one-half the bandwidth of Fast Ethernet. Furthermore, wireless networking
performance is distance sensitive, meaning that maximum performance will degrade on
computers farther away from the access point or other communication endpoint. As more
wireless devices utilize the 802.11 LAN more heavily, performance degrades even further. (Wi-
Fi.org)
The greater mobility of wireless LANs helps offset the performance disadvantage.
Mobile computers do not need to be tied to an Ethernet cable and can roam freely within the
wireless network range. However, many computers are larger desktop models, and even mobile
computers must sometimes be tied to an electrical cord and outlet for power. This undermines
the mobility advantage of wireless networks in many organizations and homes.
For any wired network connected to the Internet, firewalls are the primary security
consideration. Wired Ethernet hubs and switches do not support firewalls. However, firewall
software products like Zone Alarm can be installed on the computers themselves. Broadband
routers offer equivalent firewall capability built into the device, configurable through its own
software.
In theory, wireless LANs are less secure than wired LANs, because wireless
communication signals travel through the air and can easily be intercepted. The weaknesses of
wireless security are more theoretical than practical. (Wi-Fi.org) Wireless networks protect their
data through the Wired Equivalent Privacy (WEP) encryption standard that makes wireless
communications reasonably as safe as wired ones.
No computer network is completely secure. Important security considerations for

organizations tend to not be related to whether the network is wired or wireless but rather
ensuring that the firewall is properly configured, employees are aware of the dangers of spoof
emails, they are away of spy ware and how to avoid and that anyone outside the organization
does not have unauthorized access to the network.
Ethernet Network Topologies and Structures
LANs take on many topological configurations, but regardless of their size or complexity, all
will be a combination of only three basic interconnection structures or network building blocks.
The simplest structure is the point-to-point interconnection, shown in Figure: Example Point-to-
Point Interconnection. Only two network units are involved, and the connection may be DTE-to-
DTE, DTE-to-DCE, or DCE-to-DCE. The cable in point-to-point interconnections is known as a
network link. The maximum allowable length of the link depends on the type of cable and the
transmission method that is used.
Figure: Example Point-to-Point Interconnection
The original Ethernet networks were implemented with a coaxial bus structure, as shown
in Figure: Example Coaxial Bus Topology. Segment lengths were limited to 500 meters, and up
to 100 stations could be connected to a single segment. Individual segments could be
interconnected with repeaters, as long as multiple paths did not exist between any two stations on
the network and the number of DTEs did not exceed 1024. The total path distance between the
most-distant pair of stations was also not allowed to exceed a maximum prescribed value.
Figure: Example Coaxial Bus Topology
Although new networks are no longer connected in a bus configuration, some older bus-
connected networks do still exist and are still useful.
Since the early 1990s, the network configuration of choice has been the star-connected topology,
shown in Figure: Example Star-Connected Topology. The central network unit is either a
multiport repeater (also known as a hub) or a network switch. All connections in a star network
are point-to-point links implemented with either twisted-pair or optical fiber cable.
Figure: Example Star-Connected Topology
The IEEE 802.3 Logical Relationship to the OSI Reference Model
Figure: Ethernet’s Logical Relationship to the OSI Reference Model shows the IEEE 802.3
logical layers and their relationship to the OSI reference model. As with all IEEE 802 protocols,
the OSI data link layer is divided into two IEEE 802 sublayers, the Media Access Control
(MAC) sublayer and the MAC-client sublayer. The IEEE 802.3 physical layer corresponds to the
OSI physical layer.
Figure: Ethernet’s Logical Relationship to the OSI Reference Model
The MAC-client sublayer may be one of the following:
▪ Logical Link Control (LLC), if the unit is a DTE. This sublayer provides the interface
between the Ethernet MAC and the upper layers in the protocol stack of the end station. The
LLC sublayer is defined by IEEE 802.2 standards.
▪ Bridge entity, if the unit is a DCE. Bridge entities provide LAN-to-LAN interfaces between
LANs that use the same protocol (for example, Ethernet to Ethernet) and also between
different protocols (for example, Ethernet to Token Ring). Bridge entities are defined by
IEEE 802.1 standards.
Because specifications for LLC and bridge entities are common for all IEEE 802 LAN protocols,
network compatibility becomes the primary responsibility of the particular network protocol.
Figure: MAC and Physical Layer Compatibility Requirements for Basic Data
Communication shows different compatibility requirements imposed by the MAC and physical
levels for basic data communication over an Ethernet link.
Figure: MAC and Physical Layer Compatibility Requirements for Basic Data Communication
The MAC layer controls the node's access to the network media and is specific to the individual
protocol. All IEEE 802.3 MACs must meet the same basic set of logical requirements, regardless
of whether they include one or more of the defined optional protocol extensions. The only
requirement for basic communication (communication that does not require optional protocol
extensions) between two network nodes is that both MACs must support the same transmission
rate.
The 802.3 physical layer is specific to the transmission data rate, the signal encoding, and the
type of media interconnecting the two nodes. Gigabit Ethernet, for example, is defined to operate
over either twisted-pair or optical fiber cable, but each specific type of cable or signal-encoding
procedure requires a different physical layer implementation.
The Ethernet MAC Sublayer
The MAC sublayer has two primary responsibilities:
▪ Data encapsulation, including frame assembly before transmission, and frame parsing/error
detection during and after reception
▪ Media access control, including initiation of frame transmission and recovery from
transmission failure
The Basic Ethernet Frame Format

The IEEE 802.3 standard defines a basic data frame format that is required for all MAC
implementations, plus several additional optional formats that are used to extend the protocol's
basic capability. The basic data frame format contains the seven fields shown in Figure: The
Basic IEEE 802.3 MAC Data Frame Format.
▪ Preamble (PRE) - Consists of 7 bytes. The PRE is an alternating pattern of ones and zeros
that tells receiving stations that a frame is coming, and that provides a means to synchronize
the frame-reception portions of receiving physical layers with the incoming bit stream.
▪ Start-of-frame delimiter (SOF) - Consists of 1 byte. The SOF is an alternating pattern of
ones and zeros, ending with two consecutive 1-bits indicating that the next bit is the left-most
bit in the left-most byte of the destination address.
▪ Destination address (DA) - Consists of 6 bytes. The DA field identifies which station(s)
should receive the frame. The left-most bit in the DA field indicates whether the address is an
individual address (indicated by a 0) or a group address (indicated by a 1). The second bit
from the left indicates whether the DA is globally administered (indicated by a 0) or locally
administered (indicated by a 1). The remaining 46 bits are a uniquely assigned value that
identifies a single station, a defined group of stations, or all stations on the network.
▪ Source addresses (SA) - Consists of 6 bytes. The SA field identifies the sending station. The
SA is always an individual address and the left-most bit in the SA field is always 0.
▪ Length/Type - Consists of 2 bytes. This field indicates either the number of MAC-client data
bytes that are contained in the data field of the frame, or the frame type ID if the frame is
assembled using an optional format. If the Length/Type field value is less than or equal to
1500, the number of LLC bytes in the Data field is equal to the Length/Type field value. If
the Length/Type field value is greater than 1536, the frame is an optional type frame, and the
Length/Type field value identifies the particular type of frame being sent or received.
▪ Data - Is a sequence of n bytes of any value, where n is less than or equal to 1500. If the
length of the Data field is less than 46, the Data field must be extended by adding a filler (a
pad) sufficient to bring the Data field length to 46 bytes.
▪ Frame check sequence (FCS) - Consists of 4 bytes. This sequence contains a 32-bit cyclic
redundancy check (CRC) value, which is created by the sending MAC and is recalculated by
the receiving MAC to check for damaged frames. The FCS is generated over the DA, SA,
Length/Type, and Data fields.
Figure: The Basic IEEE 802.3 MAC Data Frame Format
{{note:Individual addresses are also known as unicast addresses because they refer to a single
MAC and are assigned by the NIC manufacturer from a block of addresses allocated by the
IEEE. Group addresses (a.k.a. multicast addresses) identify the end stations in a workgroup and
are assigned by the network manager. A special group address (all 1s-the broadcast address)
indicates all stations on the network.}}
Frame Transmission
Whenever an end station MAC receives a transmit-frame request with the accompanying address
and data information from the LLC sublayer, the MAC begins the transmission sequence by
transferring the LLC information into the MAC frame buffer.
▪ The preamble and start-of-frame delimiter are inserted in the PRE and SOF fields.
▪ The destination and source addresses are inserted into the address fields.
▪ The LLC data bytes are counted, and the number of bytes is inserted into the Length/Type
field.
▪ The LLC data bytes are inserted into the Data field. If the number of LLC data bytes is less
than 46, a pad is added to bring the Data field length up to 46.
▪ An FCS value is generated over the DA, SA, Length/Type, and Data fields and is appended
to the end of the Data field.
After the frame is assembled, actual frame transmission will depend on whether the MAC is
operating in half-duplex or full-duplex mode.
The IEEE 802.3 standard currently requires that all Ethernet MACs support half-duplex
operation, in which the MAC can be either transmitting or receiving a frame, but it cannot be
doing both simultaneously. Full-duplex operation is an optional MAC capability that allows the
MAC to transmit and receive frames simultaneously.
Half-Duplex Transmission-The CSMA/CD Access Method

The CSMA/CD protocol was originally developed as a means by which two or more stations
could share a common media in a switch-less environment when the protocol does not require
central arbitration, access tokens, or assigned time slots to indicate when a station will be
allowed to transmit. Each Ethernet MAC determines for itself when it will be allowed to send a
frame.
The CSMA/CD access rules are summarized by the protocol's acronym:
▪ Carrier sense - Each station continuously listens for traffic on the medium to determine
when gaps between frame transmissions occur.
▪ Multiple access - Stations may begin transmitting any time they detect that the network is
quiet (there is no traffic).
▪ Collision detect - If two or more stations in the same CSMA/CD network (collision domain)
begin transmitting at approximately the same time, the bit streams from the transmitting
stations will interfere (collide) with each other, and both transmissions will be unreadable. If
that happens, each transmitting station must be capable of detecting that a collision has
occurred before it has finished sending its frame. Each must stop transmitting as soon as it
has detected the collision and then must wait a quasirandom length of time (determined by a
back-off algorithm) before attempting to retransmit the frame.
The worst-case situation occurs when the two most-distant stations on the network both need to
send a frame and when the second station does not begin transmitting until just before the frame
from the first station arrives. The collision will be detected almost immediately by the second
station, but it will not be detected by the first station until the corrupted signal has propagated all
the way back to that station. The maximum time that is required to detect a collision (the
collision window, or "slot time") is approximately equal to twice the signal propagation time
between the two most-distant stations on the network.
This means that both the minimum frame length and the maximum collision diameter are directly
related to the slot time. Longer minimum frame lengths translate to longer slot times and larger
collision diameters; shorter minimum frame lengths correspond to shorter slot times and smaller
collision diameters.
The trade-off was between the need to reduce the impact of collision recovery and the need for
network diameters to be large enough to accommodate reasonable network sizes. The
compromise was to choose a maximum network diameter (about 2500 meters) and then to set the
minimum frame length long enough to ensure detection of all worst-case collisions.
The compromise worked well for 10 Mbps, but it was a problem for higher data-rate Ethernet
developers. Fast Ethernet was required to provide backward compatibility with earlier Ethernet
networks, including the existing IEEE 802.3 frame format and error-detection procedures, plus
all applications and networking software running on the 10-Mbps networks.
Although signal propagation velocity is essentially constant for all transmission rates, the time
required to transmit a frame is inversely related to the transmission rate. At 100 Mbps, a
minimum-length frame can be transmitted in approximately one-tenth of the defined slot time,
and any collision that occurred during the transmission would not likely be detected by the
transmitting stations. This, in turn, meant that the maximum network diameters specified for 10-
Mbps networks could not be used for 100-Mbps networks. The solution for Fast Ethernet was to
reduce the maximum network diameter by approximately a factor of 10 (to a little more than 200
meters).
The same problem also arose during specification development for Gigabit Ethernet, but
decreasing network diameters by another factor of 10 (to approximately 20 meters) for 1000-
Mbps operation was simply not practical. This time, the developers elected to maintain
approximately the same maximum collision domain diameters as 100-Mbps networks and to
increase the apparent minimum frame size by adding a variable-length nondata extension field to
frames that are shorter than the minimum length (the extension field is removed during frame
reception).
Figure: MAC Frame with Gigabit Carrier Extension shows the MAC frame format with the
gigabit extension field, and the following table shows the effect of the trade-off between the
transmission data rate and the minimum frame size for 10-Mbps, 100-Mbps, and 1000-Mbps
Ethernet.
Figure: MAC Frame with Gigabit Carrier Extension
Table: Limits for Half-Duplex Operation

Parameter 10 Mbps 100 Mbps 1000 Mbps
520 bytes (with extension

Minimum frame size 64 bytes 64 bytes
field added)
Maximum collision diameter, DTE 100 meters 100 meters 100 meters UTP
to DTE UTP UTP
316 meters fiber

412 meters
fiber
Maximum collision diameter with

2500 meters 205 meters 200 meters
repeaters
Maximum number of repeaters in

5 2 1
network path
Note: 520 bytes applies to 1000Base-T implementations. The minimum frame size with
extension field for 1000Base-X is reduced to 416 bytes because 1000Base-X encodes
and transmits 10 bits for each byte.
Another change to the Ethernet CSMA/CD transmit specification was the addition of frame
bursting for gigabit operation. Burst mode is a feature that allows a MAC to send a short
sequence (a burst) of frames equal to approximately 5.4 maximum-length frames without having
to relinquish control of the medium. The transmitting MAC fills each interframe interval with
extension bits, as shown in Figure: A Gigabit Frame-Burst Sequence, so that other stations on the
network will see that the network is busy and will not attempt transmission until after the burst is
complete.
Figure: A Gigabit Frame-Burst Sequence
If the length of the first frame is less than the minimum frame length, an extension field is added
to extend the frame length to the value indicated in Table: Limits for Half-Duplex Operation.
Subsequent frames in a frame-burst sequence do not need extension fields, and a frame burst
may continue as long as the burst limit has not been reached. If the burst limit is reached after a
frame transmission has begun, transmission is allowed to continue until that entire frame has
been sent.
Frame extension fields are not defined, and burst mode is not allowed for 10 Mbps and 100
Mbps transmission rates.
Full-Duplex Transmission-An Optional Approach to Higher Network Efficiency

Full-duplex operation is an optional MAC capability that allows simultaneous two-way
transmission over point-to-point links. Full duplex transmission is functionally much simpler
than half-duplex transmission because it involves no media contention, no collisions, no need to
schedule retransmissions, and no need for extension bits on the end of short frames. The result is
not only more time available for transmission, but also an effective doubling of the link
bandwidth because each link can now support full-rate, simultaneous, two-way transmission.
Transmission can usually begin as soon as frames are ready to send. The only restriction is that
there must be a minimum-length interframe gap between successive frames, as shown in Figure:
Full Duplex Operation Allows Simultaneous Two-Way Transmission on the Same Link, and
each frame must conform to Ethernet frame format standards.
Figure: Full Duplex Operation Allows Simultaneous Two-Way Transmission on the Same Link
Flow Control
Full-duplex operation requires concurrent implementation of the optional flow-control capability
that allows a receiving node (such as a network switch port) that is becoming congested to
request the sending node (such as a file server) to stop sending frames for a selected short period
of time. Control is MAC-to-MAC through the use of a pause frame that is automatically
generated by the receiving MAC. If the congestion is relieved before the requested wait has
expired, a second pause frame with a zero time-to-wait value can be sent to request resumption
of transmission. An overview of the flow control operation is shown in Figure: An Overview of
the IEEE 802.3 Flow Control Sequence.
Figure: An Overview of the IEEE 802.3 Flow Control Sequence
The full-duplex operation and its companion flow control capability are both options for all
Ethernet MACs and all transmission rates. Both options are enabled on a link-by-link basis,
assuming that the associated physical layers are also capable of supporting full-duplex operation.
Pause frames are identified as MAC control frames by an exclusive assigned (reserved)
length/type value. They are also assigned a reserved destination address value to ensure that an
incoming pause frame is never forwarded to upper protocol layers or to other ports in a switch.
Frame Reception
Frame reception is essentially the same for both half-duplex and full-duplex operations, except
that full-duplex MACs must have separate frame buffers and data paths to allow for
simultaneous frame transmission and reception.
Frame reception is the reverse of frame transmission. The destination address of the received
frame is checked and matched against the station's address list (its MAC address, its group
addresses, and the broadcast address) to determine whether the frame is destined for that station.
If an address match is found, the frame length is checked and the received FCS is compared to
the FCS that was generated during frame reception. If the frame length is okay and there is an
FCS match, the frame type is determined by the contents of the Length/Type field. The frame is
then parsed and forwarded to the appropriate upper layer.
The VLAN Tagging Option
VLAN tagging is a MAC option that provides three important capabilities not previously
available to Ethernet network users and network managers:
▪ Provides a means to expedite time-critical network traffic by setting transmission priorities

for outgoing frames.
▪ Allows stations to be assigned to logical groups, to communicate across multiple LANs as

though they were on a single LAN. Bridges and switches filter destination addresses and
forward VLAN frames only to ports that serve the VLAN to which the traffic belongs.
▪ Simplifies network management and makes adds, moves, and changes easier to administer.
A VLAN-tagged frame is simply a basic MAC data frame that has had a 4-byte VLAN header
inserted between the SA and Length/Type fields, as shown in Figure: VLAN-Tagged Frames Are
Identified When the MAC Finds the LAN Type Value in the Normal Length/Type Field
Location.
Figure: VLAN-Tagged Frames Are Identified When the MAC Finds the LAN Type Value in the
Normal Length/Type Field Location
The VLAN header consists of two fields:
▪ A reserved 2-byte type value, indicating that the frame is a VLAN frame
▪ A two-byte Tag-Control field that contains both the transmission priority (0 to 7, where 7 is
the highest) and a VLAN ID that identifies the particular VLAN over which the frame is to
be sent
The receiving MAC reads the reserved type value, which is located in the normal Length/Type
field position, and interprets the received frame as a VLAN frame. Then the following occurs:
▪ If the MAC is installed in a switch port, the frame is forwarded according to its priority level
to all ports that are associated with the indicated VLAN identifier.
▪ If the MAC is installed in an end station, it removes the 4-byte VLAN header and processes
the frame in the same manner as a basic data frame.
VLAN tagging requires that all network nodes involved with a VLAN group be equipped with
the VLAN option.
The Ethernet Physical Layers
Because Ethernet devices implement only the bottom two layers of the OSI protocol stack, they
are typically implemented as network interface cards (NICs) that plug into the host device's
motherboard. The different NICs are identified by a three-part product name that is based on the
physical layer attributes.
The naming convention is a concatenation of three terms indicating the transmission rate, the
transmission method, and the media type/signal encoding. For example, consider this:
▪ 10Base-T = 10 Mbps, baseband, over two twisted-pair cables

▪ 100Base-T2 = 100 Mbps, baseband, over two twisted-pair cables
▪ 100Base-T4 = 100 Mbps, baseband, over four-twisted pair cables
▪ 1000Base-LX = 1000 Mbps, baseband, long wavelength over optical fiber cable
A question sometimes arises as to why the middle term always seems to be "Base." Early
versions of the protocol also allowed for broadband transmission (for example, 10Broad), but
broadband implementations were not successful in the marketplace. All current Ethernet
implementations use baseband transmission.
Encoding for Signal Transmission
In baseband transmission, the frame information is directly impressed upon the link as a
sequence of pulses or data symbols that are typically attenuated (reduced in size) and distorted
(changed in shape) before they reach the other end of the link. The receiver's task is to detect
each pulse as it arrives and then to extract its correct value before transferring the reconstructed
information to the receiving MAC.
Filters and pulse-shaping circuits can help restore the size and shape of the received waveforms,
but additional measures must be taken to ensure that the received signals are sampled at the
correct time in the pulse period and at same rate as the transmit clock:
▪ The receive clock must be recovered from the incoming data stream to allow the receiving
physical layer to synchronize with the incoming pulses.
▪ Compensating measures must be taken for a transmission effect known as baseline wander.
Clock recovery requires level transitions in the incoming signal to identify and synchronize on
pulse boundaries. The alternating 1s and 0s of the frame preamble were designed both to indicate
that a frame was arriving and to aid in clock recovery. However, recovered clocks can drift and
possibly lose synchronization if pulse levels remain constant and there are no transitions to detect
(for example, during long strings of 0s).
Baseline wander results because Ethernet links are AC-coupled to the transceivers and because
AC coupling is incapable of maintaining voltage levels for more than a short time. As a result,
transmitted pulses are distorted by a droop effect similar to the exaggerated example shown
in Figure: A Concept Example of Baseline Wander. In long strings of either 1s or 0s, the droop
can become so severe that the voltage level passes through the decision threshold, resulting in
erroneous sampled values for the affected pulses.
Figure: A Concept Example of Baseline Wander
Fortunately, encoding the outgoing signal before transmission can significantly reduce the effect
of both these problems, as well as reduce the possibility of transmission errors. Early Ethernet
implementations, up to and including 10Base-T, all used the Manchester encoding method,
shown in Figure: Transition-Based Manchester Binary Encoding. Each pulse is clearly identified
by the direction of the midpulse transition rather than by its sampled level value.
Figure: Transition-Based Manchester Binary Encoding
Unfortunately, Manchester encoding introduces some difficult frequency-related problems that

make it unsuitable for use at higher data rates. Ethernet versions subsequent to 10Base-T all use
different encoding procedures that include some or all of the following techniques:
▪ Using data scrambling - A procedure that scrambles the bits in each byte in an orderly (and
recoverable) manner. Some 0s are changed to 1s, some 1s are changed to 0s, and some bits
are left the same. The result is reduced run-length of same-value bits, increased transition
density, and easier clock recovery.
▪ Expanding the code space - A technique that allows assignment of separate codes for data
and control symbols (such as start-of-stream and end-of-stream delimiters, extension bits,
and so on) and that assists in transmission error detection.
▪ Using forward error-correcting codes - An encoding in which redundant information is
added to the transmitted data stream so that some types of transmission errors can be
corrected during frame reception.
Note: Forward error-correcting codes are used in 1000Base-T to achieve an effective

reduction in the bit error rate. Ethernet protocol limits error handling to detection of
bit errors in the received frame. Recovery of frames received with uncorrectable
errors or missing frames is the responsibility of higher layers in the protocol stack.
The 802.3 Physical Layer Relationship to the ISO Reference Model

Although the specific logical model of the physical layer may vary from version to version, all
Ethernet NICs generally conform to the generic model shown in Figure: The Generic Ethernet
Physical Layer Reference Model:
Figure: The Generic Ethernet Physical Layer Reference Model
The physical layer for each transmission rate is divided into sublayers that are independent of the
particular media type and sublayers that are specific to the media type or signal encoding.
▪ The reconciliation sublayer and the optional media-independent interface (MII in 10-Mbps
and 100-Mbps Ethernet, GMII in Gigabit Ethernet) provide the logical connection between
the MAC and the different sets of media-dependent layers. The MII and GMII are defined
with separate transmit and receive data paths that are bit-serial for 10-Mbps implementations,
nibble-serial (4 bits wide) for 100-Mbps implementations, and byte-serial (8 bits wide) for
1000-Mbps implementations. The media-independent interfaces and the reconciliation
sublayer are common for their respective transmission rates and are configured for full-
duplex operation in 10Base-T and all subsequent Ethernet versions.
▪ The media-dependent physical coding sublayer (PCS) provides the logic for encoding,
multiplexing, and synchronization of the outgoing symbol streams as well symbol code
alignment, demultiplexing, and decoding of the incoming data.
▪ The physical medium attachment (PMA) sublayer contains the signal transmitters and
receivers (transceivers), as well as the clock recovery logic for the received data streams.
▪ The medium-dependent interface (MDI) is the cable connector between the signal
transceivers and the link.
▪ The Auto-negotiation sublayer allows the NICs at each end of the link to exchange
information about their individual capabilities, and then to negotiate and select the most
favorable operational mode that they both are capable of supporting. Auto-negotiation is
optional in early Ethernet implementations and is mandatory in later versions.
Depending on which type of signal encoding is used and how the links are configured, the PCS
and PMA may or may not be capable of supporting full-duplex operation.
10-Mbps Ethernet-10Base-T
10Base-T provides Manchester-encoded 10-Mbps bit-serial communication over two unshielded
twisted-pair cables. Although the standard was designed to support transmission over common
telephone cable, the more typical link configuration is to use two pair of a four-pair Category 3
or 5 cable, terminated at each NIC with an 8-pin RJ-45 connector (the MDI), as shown in Figure:
The Typical 10Base-T Link Is a Four-Pair UTP Cable in Which Two Pairs Are Not Used pair is
configured as a simplex link where transmission is in one direction only, the 10Base-T physical
layers can support either half-duplex or full-duplex operation.
Figure: The Typical 10Base-T Link Is a Four-Pair UTP Cable in Which Two Pairs Are Not Used
Although 10Base-T may be considered essentially obsolete in some circles, it is included here
because there are still many 10Base-T Ethernet networks, and because full-duplex operation has
given 10BaseT an extended life.
10Base-T was also the first Ethernet version to include a link integrity test to determine the
health of the link. Immediately after powerup, the PMA transmits a normal link pulse (NLP) to
tell the NIC at the other end of the link that this NIC wants to establish an active link connection:
▪ If the NIC at the other end of the link is also powered up, it responds with its own NLP.
▪ If the NIC at the other end of the link is not powered up, this NIC continues sending an NLP
about once every 16 ms until it receives a response.
The link is activated only after both NICs are capable of exchanging valid NLPs.
100 Mbps-Fast Ethernet
Increasing the Ethernet transmission rate by a factor of ten over 10Base-T was not a simple task,
and the effort resulted in the development of three separate physical layer standards for 100
Mbps over UTP cable: 100Base-TX and 100Base-T4 in 1995, and 100Base-T2 in 1997. Each
was defined with different encoding requirements and a different set of media-dependent
sublayers, even though there is some overlap in the link cabling.
Table: Summary of 100Base-T Physical Layer Characteristics compares the physical layer
characteristics of 10Base-T to the various 100Base versions.
Table: Summary of 100Base-T Physical Layer Characteristics

Ethernet Transmit Full-Duplex
Encoding Cabling
Version Symbol Rate Operation
Two pairs of UTP Category -

10Base-T 10 MBd Manchester Supported
3 or better

100Base-TX 125 MBd 4B/5B Supported
5 or Type 1 STP
Four pairs of UTP Category -

100Base-T4 33 MBd 8B/6T Not supported
3 or better

100Base-T2 25 MBd PAM5x5 Supported
3 or better
Note: One baud is equal to one transmitted symbol per second, where the transmitted
symbol may contain the equivalent value of 1 or more binary bits.
Although not all three 100-Mbps versions were successful in the marketplace, all three have been
discussed in the literature, and all three did impact future designs. As such, all three are
important to consider here.
100Base-X
100Base-X was designed to support transmission over either two pairs of Category 5 UTP
copper wire or two strands of optical fiber. Although the encoding, decoding, and clock recovery
procedures are the same for both media, the signal transmission is different-electrical pulses in
copper and light pulses in optical fiber. The signal transceivers that were included as part of the
PMA function in the generic logical model of the following figure were redefined as the separate
physical media-dependent (PMD) sublayers shown in Figure: The 100Base-X Logical Model.
Figure: The 100Base-X Logical Model
The 100Base-X encoding procedure is based on the earlier FDDI optical fiber physical media-
dependent and FDDI/CDDI copper twisted-pair physical media-dependent signaling standards
developed by ISO and ANSI. The 100Base-TX physical media-dependent sublayer (TP-PMD)
was implemented with CDDI semiconductor transceivers and RJ-45 connectors; the fiber PMD
was implemented with FDDI optical transceivers and the Low Cost Fibre Interface Connector
(commonly called the duplex SC connector).
The 4B/5B encoding procedure is the same as the encoding procedure used by FDDI, with only
minor adaptations to accommodate Ethernet frame control. Each 4-bit data nibble (representing
half of a data byte) is mapped into a 5-bit binary code-group that is transmitted bit-serial over the
link. The expanded code space provided by the 32 5-bit code-groups allow separate assignment
for the following:
▪ The 16 possible values in a 4-bit data nibble (16 code-groups).

▪ Four control code-groups that are transmitted as code-group pairs to indicate the start-of-
stream delimiter (SSD) and the end-of-stream delimiter (ESD). Each MAC frame is
"encapsulated" to mark both the beginning and end of the frame. The first byte of preamble is
replaced with SSD code-group pair that precisely identifies the frame's code-group
boundaries. The ESD code-group pair is appended after the frame's FCS field.
▪ A special IDLE code-group that is continuously sent during interframe gaps to maintain
continuous synchronization between the NICs at each end of the link. The receipt of IDLE is
interpreted to mean that the link is quiet.
▪ Eleven invalid code-groups that are not intentionally transmitted by a NIC (although one is
used by a repeater to propagate receive errors). Receipt of any invalid code-group will cause
the incoming frame to be treated as an invalid frame.
Figure: The 100Base-X Code-Group Stream with Frame Encapsulation shows how a MAC
frame is encapsulated before being transmitted as a 100Base-X code-group stream.
Figure: The 100Base-X Code-Group Stream with Frame Encapsulation
100Base-TX transmits and receives on the same link pairs and uses the same pin assignments on
the MDI as 10Base-T. 100Base-TX and 100Base-FX both support half-duplex and full-duplex
transmission.
100Base-T4
100Base-T4 was developed to allow 10BaseT networks to be upgraded to 100-Mbps operation
without requiring existing four-pair Category 3 UTP cables to be replaced with the newer
Category 5 cables. Two of the four pairs are configured for half-duplex operation and can
support transmission in either direction, but only in one direction at a time. The other two pairs
are configured as simplex pairs dedicated to transmission in one direction only. Frame
transmission uses both half-duplex pairs, plus the simplex pair that is appropriate for the
transmission direction, as shown in Figure: The 100Base-T4 Wire-Pair Usage During Frame
Transmission. The simplex pair for the opposite direction provides carrier sense and collision
detection. Full-duplex operation cannot be supported on 100Base-T4.
Figure: The 100Base-T4 Wire-Pair Usage During Frame Transmission
100Base-T4 uses an 8B6T encoding scheme in which each 8-bit binary byte is mapped into a
pattern of six ternary (three-level: +1, 0, -1) symbols known as 6T code-groups. Separate 6T
code-groups are used for IDLE and for the control code-groups that are necessary for frame
transmission. IDLE received on the dedicated receive pair indicates that the link is quiet.
During frame transmission, 6T data code-groups are transmitted in a delayed round-robin

sequence over the three transmit wire-pairs, as shown in Figure: The 100Base-T4 Frame
Transmission Sequence. Each frame is encapsulated with start-of-stream and end-of-packet 6T
code-groups that mark both the beginning and end of the frame, and the beginning and end of the
6T code-group stream on each wire pair. Receipt of a non-IDLE code-group over the dedicated
receive-pair any time before the collision window expires indicates that a collision has occurred.
Figure: The 100Base-T4 Frame Transmission Sequence
100Base-T2
The 100Base-T2 specification was developed as a better alternative for upgrading networks with
installed Category 3 cabling than was being provided by 100Base-T4. Two important new goals
were defined:
▪ To provide communication over two pairs of Category 3 or better cable

▪ To support both half-duplex and full-duplex operation
100Base-T2 uses a different signal transmission procedure than any previous twisted-pair
Ethernet implementations. Instead of using two simplex links to form one full-duplex link, the
100Base-T2 dual-duplex baseband transmission method sends encoded symbols simultaneously
in both directions on both wire pairs, as shown in Figure: The 100Base-T2 Link Topology. The
term "TDX<3:2>" indicates the 2 most significant bits in the nibble before encoding and
transmission. "RDX<3:2>" indicates the same 2 bits after receipt and decoding.
Figure: The 100Base-T2 Link Topology
Dual-duplex baseband transmission requires the NICs at each end of the link to be operated in a
master/slave loop-timing mode. Which NIC will be master and which will be slave is determined
by autonegotiation during link initiation. When the link is operational, synchronization is based
on the master NIC's internal transmit clock. The slave NIC uses the recovered clock for both
transmit and receive operations, as shown inFigure: The 100Base-T2 Loop Timing
Configuration. Each transmitted frame is encapsulated, and link synchronization is maintained
with a continuous stream of IDLE symbols during interframe gaps.
Figure: The 100Base-T2 Loop Timing Configuration
The 100Base-T2 encoding process first scrambles the data frame nibbles to randomize the bit
sequence. It then maps the two upper bits and the two lower bits of each nibble into two five-
level (+2, +1, 0, -1, -2) pulse amplitude-modulated (PAM5) symbols that are simultaneously
transmitted over the two wire pairs (PAM5x5). Different scrambling procedures for master and
slave transmissions ensure that the data streams traveling in opposite directions on the same wire
pair are uncoordinated.
Signal reception is essentially the reverse of signal transmission. Because the signal on each wire
pair at the MDI is the sum of the transmitted signal and the received signal, each receiver
subtracts the transmitted symbols from the signal received at the MDI to recover the symbols in
the incoming data stream. The incoming symbol pair is then decoded, unscrambled, and
reconstituted as a data nibble for transfer to the MAC.
1000 Mbps-Gigabit Ethernet
The Gigabit Ethernet standards development resulted in two primary specifications: 1000Base-T
for UTP copper cable and 1000Base-X STP copper cable, as well as single and multimode
optical fiber (see Figure: Gigabit Ethernet Variations).
Figure 7-22 Gigabit Ethernet Variations
1000Base-T
1000Base-T Ethernet provides full-duplex transmission over four-pair Category 5 or better UTP
cable. 1000Base-T is based largely on the findings and design approaches that led to the
development of the Fast Ethernet physical layer implementations:
▪ 100Base-TX proved that binary symbol streams could be successfully transmitted over
Category 5 UTP cable at 125 MBd.
▪ 100Base-T4 provided a basic understanding of the problems related to sending multilevel
signals over four wire pairs.
▪ 100Base-T2 proved that PAM5 encoding, coupled with digital signal processing, could
handle both simultaneous two-way data streams and potential crosstalk problems resulting
from alien signals on adjacent wire pairs.
1000Base-T scrambles each byte in the MAC frame to randomize the bit sequence before it is
encoded using a 4-D, 8-State Trellis Forward Error Correction (FEC) coding in which four
PAM5 symbols are sent at the same time over four wire pairs. Four of the five levels in each
PAM5 symbol represent 2 bits in the data byte. The fifth level is used for FEC coding, which
enhances symbol recovery in the presence of noise and crosstalk. Separate scramblers for the
master and slave PHYs create essentially uncorrelated data streams between the two opposite-
travelling symbol streams on each wire pair.
The1000Base-T link topology is shown in Figure: The 1000Base-T Link Topology. The term
"TDX<7:6>" indicates the 2 most significant bits in the data byte before encoding and
transmission. "RDX<7:6>" indicates the same 2 bits after receipt and decoding.
Figure: The 1000Base-T Link Topology
The clock recovery and master/slave loop timing procedures are essentially the same as those
used in 100Base-T2 (see Figure: 1000Base-T Master/Slave Loop Timing Configuration). Which
NIC will be master (typically the NIC in a multiport intermediate network node) and which will
be slave is determined during auto negotiation.
Figure: 1000Base-T Master/Slave Loop Timing Configuration
Each transmitted frame is encapsulated with start-of-stream and end-of-stream delimiters, and
loop timing is maintained by continuous streams of IDLE symbols sent on each wire pair during
interframe gaps. 1000Base-T supports both half-duplex and full-duplex operation.
1000Base-X
All three 1000Base-X versions support full-duplex binary transmission at 1250 Mbps over two
strands of optical fiber or two STP copper wire-pairs, as shown in Figure: 1000Base-X Link
Configuration. Transmission coding is based on the ANSI Fibre Channel 8B/10B encoding
scheme. Each 8-bit data byte is mapped into a 10-bit code-group for bit-serial transmission. Like
earlier Ethernet versions, each data frame is encapsulated at the physical layer before
transmission, and link synchronization is maintained by sending a continuous stream of IDLE
code-groups during interframe gaps. All 1000Base-X physical layers support both half-duplex
and full-duplex operation.
Figure: 1000Base-X Link Configuration
The principal differences among the 1000Base-X versions are the link media and connectors that
the particular versions will support and, in the case of optical media, the wavelength of the
optical signal (see Table: 1000Base-X Link Configuration Support).
Table: 1000Base-X Link Configuration Support
1000Base-SX (850 nm 1000Base-LX (1300
Link Configuration 1000Base-CX
Wavelength) nm Wavelength)
150 Ω STP copper Supported Not supported Not supported
125/62.5 µm multimode
Not supported Supported Supported
optical fiber
125/50 µm multimode
Not supported Supported Supported
optical fiber
125/10 µm single mode

Not supported Not supported Supported
optical fiber
IEC style 1 or Fibre SFF MT-RJ or Duplex SFF MT-RJ or Duplex

Allowed connectors
Channel style 2 SC SC
The 125/62.5 µm specification refers to the cladding and core diameters of the optical fiber.
Network Cabling-Link Crossover Requirements
Link compatibility requires that the transmitters at each end of the link be connected to the
receivers at the other end of the link. However, because cable connectors at both ends of the link
are keyed the same, the conductors must cross over at some point to ensure that transmitter
outputs are always connected to receiver inputs.
Unfortunately, when this requirement first came up in the development of 10Base-T, IEEE 802.3
chose not to make a hard rule as to whether the crossover should be implemented in the cable as
shown in Figure: Alternative Ways for Implementing the Link Crossover Requirement (a) or
whether it should be implemented internally as shown in Figure: Alternative Ways for
Implementing the Link Crossover Requirement (b).
Figure: Alternative Ways for Implementing the Link Crossover Requirement
Instead, IEEE 802.3 defined two rules and made two recommendations:
▪ There must be an odd number of crossovers in all multi-conductor links.

▪ If a PMD is equipped with an internal crossover, its MDI must be clearly labeled with the
graphical X symbol.
▪ Implementation of an internal crossover function is optional.
▪ When a DTE is connected to a repeater or switch (DCE) port, it is recommended that the
crossover be implemented within the DCE port.
The eventual result was that ports in most DCEs were equipped with PMDs that contained
internal crossover circuitry and that DTEs had PMDs without internal crossovers. This led to the
following oft-quoted de facto installation rule:
▪ Use a straight-through cable when connecting DTE to DCE. Use a crossover cable when
connecting DTE to DTE or DCE to DCE.
Unfortunately, the de facto rule does not apply to all Ethernet versions that have been developed
subsequent to 10Base-T. As things now stand, the following is true:
▪ All fiber-based systems use cables that have the crossover implemented within the cable.
▪ All 100Base systems using twisted-pair links use the same rules and recommendations as
10Base-T.
▪ 1000Base-T NICs may implement a selectable internal crossover option that can be
negotiated and enabled during auto-negotiation. When the selectable crossover option is not
implemented, 10Base-T rules and recommendations apply.
UNIT 6: NETWORK LAYER
Protocol Design Concepts, IP and Routing
Introduction
A central idea in the design of protocols is that of layering; and a guiding principle of Internet
protocols is the “end-to-end” principle. In this chapter, we review these ideas and describe the
transport and network layers in the Internet stack.
Protocols and Layering
Protocols are complex, distributed pieces of software. Abstraction and modular design are
standard techniques used by software engineers to deal with complexity. By abstraction, we
mean that a subset of functions is carefully chosen and setup as a “black-box” or module (see
Figure 1). The module has an interface describing its input/output behavior. The interface
outlives the implementation the module in the sense that the technology used to implement the
interface may change often, but the interface tends to remain constant. Modules may be built and
maintained by different entities. The software modules are then used as building blocks in a
larger design. Placement of functions to design the right building blocks and interfaces is a core
activity in software engineering.
Input Interface Software Output Interface
Figure 1: Abstraction of Functionality into Modules
Protocols have an additional constraint of being distributed. Therefore software modules have to
communicate with one or more software modules at a distance. Such interfaces across a distance
are termed as “peer-to-peer” interfaces; and the local interfaces are termed as “service”
interfaces (Figure 2). Since protocol function naturally tend to be a sequence of functions, the
modules on each end are organized as a (vertical) sequence called “layers”. The set of modules
organized as layers is also commonly called a “protocol stack”. The concept of layering is
illustrated in Figure 3.
Service Interface Service Interface
Input Peer-to-Peer Interface Output

Software
Software
Output Input
Figure 2: Communicating Software Modules
Service Interfaces
Layer i+1 Layer i+1

Peer-to-Peer Interface
Service Interfaces
Layer i Peer-to-Peer Interface Layer i
Figure 3: Layered Communicating Software Modules (Protocols)

Over the years, some layered models have been standardized. The ISO Open Systems
Interconnection (ISO/OSI) layered model has seven layers and was developed by a set of
committees under the auspices of International Standards Organization (ISO). The TCP/IP has a
4-layered stack that has become a de facto standard of the Internet. The TCP/IP and ISO/OSI
stacks and their rough correspondences are illustrated in Figure 4. The particular protocols of the
TCP/IP stack are also shown.
Application
Application FTP Telnet HTTP
Presentation
Session
Transport TCP UDP
Transport
Internetwork IP Network
Host to Ether Packet Point-to- Datalink

Network net Radio Point
Physical
Figure 4: TCP/IP vs ISO/OSI Protocol Stack
The physical layer deals with getting viable bit-transmission out of the underlying medium
(fiber, copper, coax cable, air etc). The data-link layer then converts this bit-transmission into a
framed-transmission on the link. “Frames” are used to allow multiplexing of several streams; and
defines the unit of transmission used in error detection and flow control. In the event that the link
is actually shared (i.e. multiple access), the data link layer defines the medium access control
(MAC) protocol as well. The network layer deals with packet transmission across multiple links
from the source node to the destination node. The functions here include routing, signaling and
mechanisms to deal with the heterogeneous link layers at each link.
The transport layer protocol provides “end-to-end” communication services, i.e., it allows
applications to multiplex the network service, and may add other capabilities like connection
setup, reliability and flow/congestion control. Examples of communication abstractions provided
by the transport layer include a reliable byte-stream service (TCP) and an unreliable datagram
service (UDP). These abstractions are made available through application-level programming
interfaces (APIs) such as the BSD socket interface. The application layers (session, presentation,
application) then use the communication abstractions provided by the transport layer to create
the basis for interesting applications like email, web, file transfer, multimedia conference, peer-
to-peer applications etc. Examples of such protocols include SMTP, HTTP, DNS, H.323 and
SIP.
The End-to-End Principle in Internet Protocol Design
A key principle used in the design of the TCP/IP protocols is the so-called “end-to-end” principle
that guides the placement of functionality in a complex distributed system. The principle
suggests that “…functions placed at the lower levels may be redundant or of little value when
compared to the cost of providing them at the lower level…” In other words, a system (or
subsystem level) should consider only functions that can be completely and correctly
implemented within it. All other functions are best moved to the system level where it can be
completely and correctly implemented.
In the context of the Internet, it implies that several functions like reliability, congestion control,
session/connection management are best moved to the end-systems (i.e. performed on an “end-
to-end” basis), and the network layer focuses on functions which it can fully implement, i.e.
routing and datagram delivery. As a result, the end-systems are intelligent and in control of the
communication while the forwarding aspects of the network is kept simple. This leads to a
philosophy diametrically opposite to the telephone world which sports dumb end-systems (the
telephone) and intelligent networks. Indeed the misunderstanding of the end-to-end principle has
been a primary cause for friction between the “telephony” and “internet” camps. Arguably the
telephone world developed as such due to technological and economic reasons because
intelligent and affordable end-systems were not possible until 1970s. Also, as an aside, note that
there is a misconception that the end-to-end principle implies a “dumb” network. Routing is a
good example of a very complex function that is consistent with the end-to-end principle, but is
non-trivial in terms of complexity. Routing is kept at the network level because it can be
completely implemented at that level, and the costs of involving the end-systems in routing are
formidable.
The end-to-end principle further argues that even if the network layer did provide connection
management and reliability, transport levels would have to add reliability to account for the
interaction at the transport-network boundary; or if the transport needs more reliability than what
the network provides. Removing these concerns from the lower layer packet-forwarding devices
streamlines the forwarding process, contributing to system-wide efficiency and lower costs. In
other words, the costs of providing the “incomplete” function at the network layers would
arguably outweigh the benefits.
It should be noted that the end-to-end principle emphasizes function placement vis-a-vis
correctness, completeness and overall system costs. The argument does say that, “…sometimes
an incomplete version of the function provided by the communication system may be useful as a
performance enhancement…” In other words, the principle does allow a cost-performance

tradeoff, and incorporation of economic concerns. However, it cautions that the choice of such
“incomplete versions of functions” to be placed inside the network should be made very
prudently. Lets try to understand some implications of this aspect.
One issue regarding the “incomplete network-level function” is the degree of “state” maintained
inside the network. Lack of state removes any requirement for the network nodes to notify each
other as endpoint connections are formed or dropped. Furthermore, the endpoints are not, and
need not be, aware of any network components other than the destination, first hop router(s), and
an optional name resolution service. Packet integrity is preserved through the network, and
transport checksums and any address-dependent security functions are valid end-to-end. If state
is maintained only in the endpoints, in such a way that the state can only be destroyed when the
endpoint itself breaks (also termed “fate-sharing”), then as networks grow in size, likelihood of
component failures affecting a connection becomes increasingly frequent. If failures lead to loss
of communication, because key state is lost, then the network becomes increasingly brittle, and
its utility degrades. However, if an endpoint itself fails, then there is no hope of subsequent
communication anyway. Therefore one quick interpretation of the end-to-end model is that it
suggests that only the endpoints should hold critical state. But this is flawed.
Let us consider the economic issues of Internet Service Provider (ISPs) into this mix. ISPs need
to go beyond the commoditised mix of access and connectivity services to provide differentiated
network services. Providing Quality of Service (QoS) and charging for it implies that some part
of the network has to participate in decisions of resource sharing, and billing, which cannot be
entrusted to end-systems. A correct application of the end-to-end principle in this scenario is as
follows: due to the economic and trust model issues, these functions belong to the network.
Applications may be allowed to participate in the decision process, but the control belongs to the
network, not the end-system in this matter. The differentiated services architecture discussed
later in this chapter has the notion of the “network edge” which is the repository of these
functions.
In summary, the end-to-end principle has guided a vast majority of function placement decisions
in the Internet and it remains relevant today even as the design decisions are intertwined with
complex economic concerns of multiple ISPs and vendors.
Network Layer
The network layer in the TCP/IP stack deals with internetworking and routing. The core
problems of internetworking are heterogeneity and scale. Heterogeneity is the problem of
dealing with disparate layer 2 networks to create a viable forwarding and addressing paradigm;
and the problem of providing meaningful service to a range of disparate applications. Scale is the
problem of allowing the Internet to grow without bounds to meet its intended user demands. The
Internet design applies the end-to-end principle to deal with these problems.
Network Service Models
One way of dealing with heterogeneity is to provide translation services between the
heterogeneous entities when forwarding across them is desired. Examples of such design include
multi-protocol bridges and multi-protocol routers. But this gets too complicated and does not
allow scaling because every new entity that wishes to join the Internet will require changes in all
existing infrastructure. A more preferable requirement is to be able to “incrementally upgrade”
the network. The alternative strategy is called an “overlay” model where a new protocol (IP)
with its own packet format and address space is developed and the mapping is done between all
protocols and this intermediate protocol.
IP has to be simple by necessity so that the mapping between IP and lower layer protocols is
simplified. As a result, IP opts for a best-effort, unreliable datagram service model where it
forwards datagrams between sources and destinations situated on, and separated by a set of
disparate networks. IP expects a minimal link-level frame forwarding service from lower layers.
The mapping between IP and lower layers involve address mapping issues (eg: address
resolution) and packet format mapping issues (eg: fragmentation/reassembly). Experience has
shown that this mapping is straightforward in many subnetworks, especially those that are not
too large, and those which support broadcast at the LAN level. The address resolution can be a
complex problem on non-broadcast multiple access (NBMA) sub-networks; and the control
protocols associated with IP (esp BGP routing) can place other requirements on large sub-
networks (eg: ATM networks) which make the mapping problems hard. Hybrid technologies like
MPLS are used to address these mapping concerns, and to enable new traffic engineering
capabilities in core networks.
For several applications, it turns out that the simple best-effort service provided by IP can be
augmented with end-to-end transport protocols like TCP, UDP and RTP to be sufficient. Other
applications having stringent performance expectations (eg: telephony) need to either adapt
and/or use augmented QoS capabilities from the network. While several mechanisms and
protocols for this have been developed in the last decade, a fully QoS-capable Internet is still a
holy grail for the Internet community. The hard problems surround routing, inter-domain/multi-
provider issues, and the implications of QoS on a range of functions (routing, forwarding,
scheduling, signaling, application adaptation etc).
In summary, the best-effort, overlay model of IP has proved to be enormously successful, it has
faced problems in being mapped to large NBMA sub-networks and continues to faces challenges
in the inter-domain/multi-provider and QoS areas.
The Internet Protocol (IP): Forwarding Paradigm
The core service provided by IP is datagram forwarding over disparate networks. This itself is a
non-trivial problem. The end-result of this forwarding service is to provide connectivity. The two
broad approaches to getting connectivity are: direct connectivity and indirect connectivity. Direct
connectivity refers to the case where the destination is only a single link away (this includes
shared and unshared media). Indirect connectivity refers to connectivity achieved by going
through intermediate components or intermediate networks. The intermediate components
(bridges, switches, routers, NAT boxes etc) are dedicated to functions to deal with the problem
of scale and/or heterogeneity. Indeed the function of providing indirect connectivity through
intermediate networks can be thought of as a design of a large virtual intermediate component,
the Internet. These different forms of connectivity are shown in Figures 5-7.
…….
Bus
Full mesh
Figure 5: Direct Connectivity Architectures

Star Ring
Tree
Figure 6: Indirect Connectivity though Intermediate Components
… …
=
Internet
… …
Figure 7: Indirect Connectivity though Intermediate Neworks & Components
The problem of scaling with respect to a parameter (eg: number of nodes) is inversely related to
the efficiency characteristics of the architecture with respect to the same parameter. For example,
direct connectivity architectures do not scale because of finite capacity of shared medium, or
finite interface slots; or high costs of provisioning a full mesh of links. A way to deal with this is
to build a switched network, where the intermediate components (“switches”) provide filtering
and forwarding capabilities to isolate multiple networks to keep them within their scaling limits,
and yet providing scalable interconnection. In general, the more efficient the filtering and
forwarding of these components, the more scalable is the architecture. Layer 1 hubs do pure
broadcast, and hence do no filtering, but can forward signals. Layer 2 bridges and switches can
filter to an extent using forwarding tables learnt by snooping; but their default to flooding on a
spanning tree when the forwarding table does not contain the address of the receiver. This default
behavior of flooding or broadcast is inefficient, and hence limits scalability. This behavior if also
partially a result of the flat addressing structure used by L2 networks.
In contrast, layer 3 (IP) switches (aka routers) never broadcast across sub-networks; and rely on a
set of routing protocols and a concatenated set of local forwarding decisions to deliver packets
across the Internet. IP addressing is designed hierarchically, and address assignment is
coordinated with routing design. This enables intermediate node (or hosts) to do a simple
determination: whether the destination is directly or indirectly connected. In the former case,
simple layer 2 forwarding is invoked; and in the latter case, a layer 3 forwarding decision is
made to determine the next-hop that is an intermediate node on the same sub-network, and then
the layer 2 forwarding is invoked.
Heterogeneity is supported by IP because it invokes only a minimal forwarding service of the

underlying L2 protocol. Before invoking this L2 forwarding service, the router has to a)
determine the L2 address of the destination (or next-hop) -- an address resolution problem; and
b) map the datagram to the underlying L2 frame format. If the datagram is too large, it has to do
something -- fragmentation/reassembly. IP does not expect any other special feature in lower
layers and hence can work over a range of L2 protocols.
In summary, the IP forwarding paradigm naturally comes out of the notions of direct and indirect
connectivity. The “secret sauce” is in the way addressing is designed to enable the
directly/indirectly reachable query; and the scalable design of routing protocols to aid the
determination of the appropriate next-hop if the destination is indirectly connected.
Heterogeneity leads to mapping issues, which are simplified because of the minimalist
expectations of IP from its lower layers (only an forwarding capability expected). All other
details of lower layers are abstracted out.
The Internet Protocol: Packet Format, Addressing, Fragmentation/Reassembly
IP Packet Format
The IP packet format is shown in Figure 8. The biggest fields in the header are the source and
destination 32-bit IP address fields. The second 32-bit line (ID, flags, frag offset) are related to
fragmentation/reassembly and will be explained later. The length field indicates the length of the
entire datagram, and is required because IP accepts variable length payloads. The checksum field
covers only the header and not the payload and is used to catch any header errors to avoid mis-
routing garbled packets. Error detection in the payload is the responsibility of the transport layer
(both UDP and TCP provide error detection). The protocol field allows IP to demultiplex the
datagram and deliver it to a higher-level protocol. Since it has only 8-bits, IP does not support
application multiplexing. Providing port number fields to enable application multiplexing is
another required function in transport protocols on IP.
The time-to-live (TTL) field is decremented at every hop and the packet is discarded if the field is
0; this prevents packets from looping forever in the Internet. The TTL field is also used a simple
way to scope the reach of the packets, and can be used in conjunction with ICMP, multicast etc
to support administrative functions. The type-of-service (TOS) field was designed to allow
optional support for differential forwarding, but has not been extensively used. Recently, the
differentiated services (diff-serv) WG in IETF renamed this field to the DS byte to be used to
support diff-serv. The version field indicates the version of IP and allows extensibility. The
current version of IP is version 4. IPv6 is the next generation of IP that may be deployed over the
next decade to support a larger 128-bit IP address space. Header length is a field used because
options can be variable length. But options are rarely used in modern IP deployments, so we
don’t discuss them any further.
32 bits
head. type of
length
ver
fragment
16-bit identifier flgs
time to Internet
protocol
32 bit source IP address
32 bit destination IP address
Options (if any)
data
(variable length,
Figure 8: IP Packet Format

3.3.2 IP Addressing and Address Allocation
An address is a unique “computer-understandable” identifier. Uniqueness is defined in a

domain. Outside that domain one needs to have either a larger address space, or do translation.
An address should ideally be valid regardless of the location of the source, but may change if the
destination moves. Fixed size addresses can be processed faster.
The concept of addresses is fundamental to networking. There is no (non-trivial) network

without addresses. Address space size also limits the scalability of networks. A large address
space allows a large network, i.e. it is fundamentally required for network scalability. Large
address space also makes it easier to assign addresses and minimize configuration. In
connectionless networks, the most interesting differences revolve around addresses. After all, a
connectionless net basically involves putting an address in a packet and sending it hoping it will
get to the destination.
IPv4 uses 32-bit addresses whereas IPv6 uses 128-bit addresses. For convenience of writing, a
dotted decimal notation became popular. Each byte is summarized as a base-10 integer, and dots
placed between these numbers (eg: 128.113.40.50).
IP addresses have two parts -- a network part (prefix), and a host part (suffix). This is illustrated
in Figure 9. Recall that the intermediate nodes (or hosts) have to make a determination whether
the destination is directly or indirectly connected. Examining the network part of the IP address
allows us to make this determination. If the destination is directly connected, the network part
matches the network part of an outgoing interface of the intermediate node. This hierarchical
structure of addressing which is fundamental to IP scaling is not seen in layer 2 (IEEE 802)
addresses. The structure has implications on address allocation because all interfaces on a single
sub-network have to be assigned the same network part of the address (to enable the forwarding
test mentioned above).
Network ID Host ID
Demarcator
Figure 9: Hierarchical Structure of an IP Address
Unfortunately address allocation was not well thought out during the early days of IP, and hence
it has followed a number of steps of evolution. Part of the evolution was forced because of the
then unforeseen sustained exponential growth of the Internet. The evolution largely centered
around the placement of the conceptual demarcator between the network ID and Host ID as
shown in Figure 9.
Initially, the addressing followed a “classful” scheme where the address space was divided into a
few blocks and static demarcators assigned to each block. Class A has a 8-bit demarcator; Class
B has a 16-bit demarcator; Class C has a 24-bit demarcator. Class D was reserved for multicast
and Class E for future use. This scheme is shown in Figure 10. This scheme ran into trouble in
early 1980s because of two reasons: a) class B’s were popular (class Cs largely unallocated) and
b) the host space in class As and class Bs were largely unused because no single sub-network
(eg: Ethernets) was large enough to utilize the space fully. The solution to these problems is
simple -- allow the host space to be further subdivided; and allow demarcators to be placed more
flexibly rather than statically.
These realizations led to the development of “subnet” and “supernet” masking respectively. A
mask is a 32-bit pattern, the ones of which indicate the bits belonging to the network ID and the
zeros indicate the host ID bits. For simplicity, the ones in the masks are contiguous. For example,
a subnet mask 255.255.255.0 applied to IP address 128.113.40.50 indicates that the network ID
has been extended from 16-bits (since this is a class B address) to 24-bits. Supernet masks are
used between autonomous systems to indicate address allocations or to advertise networks for
routing. For example the notation 198.28.29.0/18 indicates an 18-bit address space. The supernet
mask written as /18 is actually 255.255.192.0. Observe that the 198.28.29.0 belonged to the class
C space according to the earlier classful scheme and class C admits only of /24 networks (i.e.
with host space of 8 bits).
class
1.0.0.0 to
A 0 network host
128.0.0.0 to
B 10 network host
192.0.0.0 to
C 110 network host
multicast address 224.0.0.0 to

D 1110
32 bits
Figure 10: Initial Classful Addressing for IPv4

Since these class boundaries are no longer valid with the supernet masks, this allocation scheme
is also called “classless” allocation; and the routing scheme which accompanied this
development is called “Classless Inter-Domain Routing” (CIDR). One effect of CIDR and
supernet masking is that it is possible for a destination address to match multiple prefixes of
different lengths. To resolve this, CIDR prescribes that the longest-prefix match be chosen for
the L3 forwarding decision. As a result, all routers in the mid 1980s had to replace their
forwarding algorithms. Similarly when subnet masking was introduced, hosts and routers had to
be configured with subnet masks; and had to apply the mask in the forwarding process to
determine the true network ID. Recall that the network ID is used to determine if the destination
is directly or indirectly connected. These evolutionary changes are examples of how control-
plane changes (CIDR and address allocation) could also affect the data-plane (IP forwarding)
operation.
In modern networks, two other schemes are also used to further conserve public address space:
DHCP and NAT. The Dynamic Host Configuration Protocol (DHCP) was originally a network
“booting” protocol that configured essential parameters to hosts and routers. Now, it is primarily
used to lease a pool of scarce public addresses among hosts who need it for connecting to the
Internet. Observe that the leasing model means that host interfaces no longer “own” IP addresses.
The Network Address Translator (NAT) system enables the use of private address spaces within
large enterprises. The Internet Assigned Numbers Authority (IANA) has reserved the following
three blocks of the IP address space for private internets:
10.0.0.0 - 10.255.255.255 (10/8 prefix)
172.16.0.0 - 172.31.255.255 (172.16/12 prefix)
192.168.0.0 - 192.168.255.255 (192.168/16 prefix)
The NAT boxes at the edge of these private networks then translate public addresses to private
addresses for all active sessions. Since early applications (eg: FTP) overloaded the semantics of
IP addresses and included them in application-level fields, NAT has to transform these addresses
as well. NAT breaks certain security protocols, notably IPSEC, which in part tries to ensure
integrity of the IP addresses during transmission.
The combination of these techniques has delayed the deployment of IPv6 that proposes a more
long-lived solution to address space shortage. IETF and the IPv6 Forum have been planning the
deployment of IPv6 for over a decade now, and it remains to be seen what will be the major
catalyst for IPv6 adoption. The potential growth of 3G wireless networks and/or the strain on
inter-domain routing due to multi-homing have been recently cited as possible catalysts. ISPs
project that the IPv4 address space can be prolonged for another decade with the above
techniques.
ARP, Fragmentation and Reassembly
Recall that the overlay model used by IP results in two mapping problems: address mapping and
packet format mapping. The address mapping is resolved by a sub-IP protocol called ARP, while
the packet mapping is done within IP by the fragmentation/reassembly procedures. The mapping
problems in IP are far simpler than other internetworking protocols in the early 80s because IP
has minimal expectations from the lower layers.
The address mapping problem occurs once the destination or next-hop is determined at the IP
level (i.e. using the L3 forwarding table). The problem is as follows: the node knows the IP
address of the next hop (which by definition is directly connected (i.e. accessible through layer 2
forwarding). But now to be able to use L2 forwarding, it needs to find out the next-hop’s L2
address. Since the address spaces of L2 and L3 are independently assigned, the mapping is not a
simple functional relationship, i.e., it has to be discovered dynamically. The protocol used to
discover the L3 address to L2 address mapping is called the Address Resolution Protocol (ARP).
The ARP at the node sends out a link-level broadcast message requesting the mapping. Since the
next hop is on the same layer-2 “wire,” it will respond with a unicast ARP reply to the node
giving its L2 address. Then the node uses this L2 address, and encloses the IP datagram in the L2
frame payload and “drops” the frame on the L2 “wire.” ARP then uses caching (i.e. an ARP
mapping table) to avoid the broadcast request-response for future packets. In fact, other nodes on
the same L2 wire also snoop and update their ARP tables, thus reducing the need for redundant
ARP broadcasts. Since the mapping between L3 and L2 addresses could change (because both
L2 and L3 address can be dynamically assigned), the ARP table entries are aged and expunged
after a timeout period.
The packet-mapping problem occurs when the IP datagram to be forwarded is larger than the
maximum transmission unit (MTU) possible in the link layer. Every link typically has a MTU for
reasons such as fairness in multiplexing, error detection efficiency etc. For example, Ethernet has
an MTU of 1518 bytes. The solution is for the IP datagram to be fragmented such that each
fragment fits the L2 payload. Each fragment now becomes an independent IP datagram; hence
the IP header is copied over. However, it also needs to indicate the original datagram, the
position (or offset) of the fragment in the original datagram and whether it is the last datagram.
These pieces of information are filled into the fragmentation fields in the IP header (ID, flags,
frag offset) respectively. The reassembly is then done at the IP layer in the ultimate destination.
Fragments may come out-of-order or be delayed. A reassembly table data structure and a time-
out per datagram is maintained at the receiver to implement this function. Reassembly is not
attempted at intermediate routers because all fragments may not be routed through the same path.
In general, though fragmentation is a necessary function for correctness, it has severe

performance penalties. This is because any one of the fragments lost leads to the entire datagram
being discarded at the receiver. Moreover, the remaining fragments that have reached the
receiver (and are discarded) have consumed and effectively wasted scare resources at
intermediate nodes. Therefore, modern transport protocols try to avoid fragmentation as much as
possible by first discovering the minimum MTU of the path. This procedure is also known as
“path-MTU discovery.” Periodically (every 6 seconds or so), an active session will invoke the
path-MTU procedure. The procedure starts by sending a maximum sized datagram with the “do
not fragment” bit set in the flags field. When a router is forced to consider fragmentation due to a
smaller MTU than the datagram, it drops the datagram and sends an ICMP message indicating
the MTU of the link. The host then retries the procedure with the new MTU. This process is
repeated till an appropriately sized packet reaches the receiver, the size of which is used as the
maximum datagram size for future transmissions.
In summary, the mapping problems in IP are solved by ARP (a separate protocol) and
fragmentation/reassembly procedures. Fragmentation avoidance is a performance imperative and
is carried out through path MTU discovery. This completes the discussion of the key data-plane
concepts in IP. The missing pieces now are the routing protocols used to populate forwarding
tables such that a concatenation of local decisions (forwarding) leads to efficient global
connectivity.
Routing in the Internet
Routing is the magic enabling connectivity. It is the control-plane function, which sets up the
local forwarding tables at the intermediate nodes, such that a concatenation of local forwarding
decisions leads to global connectivity. The global connectivity is also “efficient” in the sense that
loops are avoided in the steady state.
Internet routing is scalable because it is hierarchical. There are two categories of routing in the
Internet: inter-domain routing and intra-domain routing. Inter-domain routing is performed
between autonomous systems (AS’s). An autonomous system defines the locus of single
administrative control and is internally connected, i.e., employs appropriate routing so that two
internal nodes need not use an external route to reach each other. The internal connectivity in an
AS is achieved through intra-domain routing protocols.
Once the nodes and links of a network are defined and the boundary of the routing architecture is
defined, then the routing protocol is responsible for capturing and condensing the appropriate
global state into local state (i.e. the forwarding table). Two issues in routing are completeness
and consistency.
In the steady state, the routing information at nodes must be consistent, i.e., a series of
independent local forwarding decisions must lead to connectivity between any (source,
destination) pair in the network. If this condition is not true, then the routing algorithm is said to
not have “converged” to steady state, i.e., it is in a transient state. In certain routing protocols,
convergence may take a long time. In general a part of the routing information may be consistent
while the rest may be inconsistent. If packets are forwarded during the period of convergence,
they may end up in loops or arbitrarily traverse the network without reaching the destination.
This is why the TTL field in the IP header is used. In general, a faster convergence algorithm is
preferred, and is considered more stable; but this may come at the expense of complexity.
Longer convergence times also limit the scalability of the algorithm, because with more nodes,
there are more routes, and each could have convergence issues independently.
Completeness means that every node has sufficient information to be able to compute all paths in
the entire network locally. In general, with more complete information, routing algorithms tend
to converge faster, because the chances of inconsistency reduce. But this means that more
distributed state must be collected at each node and processed. The demand for more
completeness also limits the scalability of the algorithm. Since both consistency and
completeness pose scalability problems, large networks have to be structured hierarchically (eg:
as areas in OSPF) where each area operates independently and views the other areas as a single
border node.
Distance Vector and Link-State Algorithms and Protocols
In packet switched networks, the two main types of routing are link-state and distance vector.
Distance vector protocols maintain information on a per-node basis (i.e. a vector of elements),
where each element of the vector represents a distance or a path to that node. Link state protocols
maintain information on a per-link basis where each element represents a weight or a set of
attributes of a link. If a graph is considered as a set of nodes and links, it is easy to see that the
link-state approach has complete information (information about links also implicitly indicates
the nodes which are the end-points of the links) whereas the distance vector approach has
incomplete information.
The basic algorithms of the distance vector (Bellman-Ford) and the link-state (Dijkstra) attempt
to find the shortest paths in a graph, in a fully distributed manner, assuming that distance vector
or link-state information can only be exchanged between immediate neighbors. Both algorithms
rely on a simple recursive equation. Assume that the shortest distance path from node i to node j
has distance D(i,j), and it passes through neighbor k to which the cost from i is c(i,k), then we
have the equation:
D(i, j) = c(i,k) + D(k,j) (1)
In other words, the subset of a shortest path is also the shortest path between the two
intermediate nodes.
The distance vector (Bellman-Ford) algorithm evaluates this recursion iteratively by starting
with initial distance values:
D(i,i) = 0 ;
D(i,k) = c(i,k) if k is a neighbor (i.e. k is one-hop away); and
D(i,k) = INFINITY for all other non-neighbors k.
Observe that the set of values D(i,*) is a distance vector at node i. The algorithm also maintains
a nexthop value for every destination j, initialized as:
next-hop(i) = i;
next-hop(k) = k if k is a neighbor, and
next-hop(k) = UNKNOWN if k is a non-neighbor.
Note that the next-hop values at the end of every iteration go into the forwarding table used at
node i.
In every iteration each node i exchanges its distance vectors D(i,*) with its immediate neighbors.
Now each node i has the values used in equation (1), i.e. D(i,j) for any destination and D(k,j) and
c(i,k) for each of its neighbors k. Now if c(i,k) + D(k,j) is smaller than the current value of D(i,j),
then D(i,j) is replaced with c(i,k) + D(k,j), as per equation (1). The next-hop value for destination
j is set now to k. Thus after m iterations, each node knows the shortest path possible to any other
node which takes m hops or less. Therefore the algorithm converges in O(d) iterations where d is
the maximum diameter of the network. Observe that each iteration requires information
exchange between neighbors. At the end of each iteration, the next-hop values for every
destination j are output into the forwarding table used by IP.
The link state (Dijkstra) algorithm pivots around the link cost c(i,k) and the destinations j, rather
than the distance D(i,j) and the source i in the distance-vector approach. It follows a greedy
iterative approach to evaluating (1), but it collects all the link states in the graph before running
the Dijkstra algorithm locally. The Dijkstra algorithm at node i maintains two sets: set N that
contains nodes to which the shortest paths have been found so far, and set M that contains all
other nodes. Initially, the set N contains node i only, and the next hop (i) = i. For all other nodes
k a value D(i,k) is maintained which indicates the current value of the path cost (distance) from i
to k. Also a value p(k) indicates what is the predecessor node to k on the shortest known path
from i (i.e. p(k) is a neighbor of k). Initially,
D(i,i) = 0 and p(i) = i;
D(i,k) = c(i,k) and p(k) = i if k is a neighbor of i

D(i,k) = INFINITY and p(k) = UNKNOWN if k is not a neighbor of i
Set N contains node i only, and the next hop (i) = i.
Set M contains all other nodes j.
In each iteration, a new node j is moved from set M into the set N. Such a node j has the
minimum distance among all current nodes in M, i.e. D(i,j) = min {l  M} D(i,l). If multiple nodes
have the same minimum distance, any one of them is chosen as j. Node j is moved from set M to
set N, and the next-hop(j) is set to the neighbor of i on the shortest path to j. Now, in addition,
the distance values of any neighbor k of j in set M is reset as:
If D(i,k) < c(j,k) + D(i,j), then D(i,k) = c(j,k) + D(i,j), and p(k) = j.
This operation called “relaxing” the edges of j is essentially the application of equation (1). This
defines the end of the iteration. Observe that at the end of iteration p the algorithm has
effectively explored paths, which are p hops or smaller from node i. At the end of the algorithm,
the set N contains all the nodes, and knows all the next-hop(j) values which are entered into the
IP forwarding table. The set M is empty upon termination. The algorithm requires n iterations
where n is the number of nodes in the graph. But since the Dijkstra algorithm is a local
computation, they are performed much quicker than in the distance vector approach. The
complexity in the link-state approach is largely due to the need to wait to get all the link states
c(j,k) from the entire network.
The protocols corresponding to the distance-vector and link-state approaches for intra-domain
routing are RIP and OSPF respectively. In both these algorithms if a link or node goes down, the
link costs or distance values have to be updated. Hence information needs to be distributed and
the algorithms need to be rerun. RIP is used for fairly small networks mainly due to a
convergence problem called “count-to-infinity.” The advantage of RIP is simplicity (25 lines of
code!). OSPF is a more complex standard that allows hierarchy and is more stable than RIP.
Therefore it is used in larger networks (esp enterprise and ISP internal networks). Another
popular link-state protocol commonly used in ISP networks is IS-IS, which came from the
ISO/OSI world, but was adapted to IP networks.
Routers and Routing Algorithms
Router: a network device working in the network layer; it receives packets, puts them in a queue
and dispatches the packets to the links toward their destinations. To do this, it uses the IP header
of packets together with its precalculated forwarding table.
Routing
 Intra-AS:
▪ Routing within a single autonomous system: a network with a single

administrator.
▪ Example: OSPF and RIP routing protocols
 Inter-AS:
▪ sometimes called policy routing;
▪ competition and security issues may arise because of communicating among

networks with different administrators.
▪ Example: BGP routing protocol.
OSPF
OSPF1 is a routing protocol based on the Dijkstra’s algorithm. It constructs a shortest path tree
for each source. It is robust against link failures and converges quite rapidly to new solutions in
response to dynamic changes in the network.
RIP
RIP acts based on the Bellman-Ford algorithm. Each node computes and stores a local routing
table containing information about which nodes can be reached by how much cost (distance) and
through which neighbor (next-hop). To propagate this information throughout the network, each
node sends its (node, distance) information to all its neighbors. This happens on a regular basis
after each t time units to ensure that the node is live, or when an update occurs.
When a link cost goes down:
Consider the following example, where cost of the link zw changes from ∞ to 1.
Initially, routing tables for w and z are:
After the link appears, z and w know that they are now connected with a weight 1 link. They
send their routing tables to each other, and update the routing tables based on the new
information:
Apparently, in this case the algorithm converges rapidly.
When a link cost goes up:
Now consider the graph below, where cost of the edge ab increases from 1 to 20.
Initially, routing tables of c and b are as follows:
When a and b discover that the link cost has been increased, they update their tables and send
them to their neighbors. As the link cost is high, b and a no more route through each other. But c
doesn’t realize the fact, and feeds a and b with obsolete information. This causes a long sequence
of alternating changes in routing tables of b and c, until c realizes that to reach a, it shouldn’t
rout through b.
As you see, when link cost increases, convergence time is very high. This problem is known as
“The Counting to Infinity Problem”. It is caused by the fact that b and c are engaged in a pattern
of mutual deception. Each claims to be able to get to a via the other. To solve the problem, we
note that it is never useful to claim reachability for a destination node to the neighbor from which
the route passes. “Split horizon with poisoned reverse” solves this problem based on this fact:
If node b is next hop on your path to a, advertise “infinity to a” to b.
But the problem still remains for longer loops:
Initially, the routing tables are:
Then, b finds out that it is no more linked to a. Also, b receives “∞ to a” from d. Thus, it can
route toward a using c. What happens is in an infinite loop, b,c, and d update their route to a: b
routes through c, c through d and d through a. Although the “Split horizon with poisoned
reverse” scheme is applied, it turns out to fail for loops of length three or more.
To come up with the problem: BGP has put an upperbound on the number of hops. Thus infinity
is no more unachievable; a specific number e.g. 15 is considered as infinity.
Firewalls
 Firewalls are sets of Rules: a list of criteria a packet should satisfy or it will be blocked
and won’t be allowed to enter the network.
 TCP flags SYN and FIN: SYN and FIN flags in TCP header used to manage TCP
connections. A SYN/FIN packet is the most well known illegal combination 2.
 Example rules:
▪ If a packet comes from a trusted source, accept.
▪ If SYN and FIN bits are both set, reject.
 If there is more than one match, the first match is important, i.e. we look through the
rules and we stop once we find the first match.
In a sense, this works similar to the forwarding table. The forwarding table in a router determines
to which link a packet should be dispatched, considering a prefix of its destination IP address. In
case of a forwarding table, if multiple matches exist, the longest should be taken into account.
Forwarding tables
Tables used in network layer to route packets based on the prefix of their destination IP address.
Remark: IP classes
IP address is a 32 bit number. These addresses admit a hierarchical structure. All hosts within a
single network have similar parts of IP used as the network address. To route a packet to its
destination, we first route it toward the destination network which in turn directs the packet to its
destination. Thus at the first step, the host address doesn’t matter. This is analogous to
hierarchical structure of mailing addresses: as long as a package hasn’t reached your house, the
apartment number is not important.
Previously, IP addresses belonged to one of the following classes3:
 Class A: A small number (126) of networks with huge number (224) of hosts.
2
Since a SYN packet is used to initiate a connection, it should never have the FIN flag set in conjunction. It
is always a malicious attempt at getting past your firewall. Thus most firewalls are now aware of SYN/FIN packets.
3
There are also two other classes D and E.
 Class B: For medium sized networks.
 Class C: For small networks of at most 256 hosts.
This way of classification caused waste of IP space for two reasons: Few people liked class C
addresses because they were too restrictive; also no network with a class A IP could fill up a
reasonable fraction of 224 IPs. This happens because the gap between type B and C is large,
making the options too discrete. To prevent waste of IP addresses, we have CIDR4. In CIDR, we
dedicate some bits of the network address to the host address. Network addresses are numbers
like 112.56.0/14 where /14 means we should only look at the first 14 bits for subnet mask (IP
range).
Routing tables do “Longest Prefix Matching”: they route packets based on the longest match in
the routing table that is a prefix of destination address.
Example.
prefix next-hop
1) 01101* A
2) 011* B
3) 001111
* A
4) 001110
* B
5) 110111
* C
6) 1* A
7) * C
4
Classless Inter-Domain Routing
a) 01100001…….. : Matches 2 and 7, hence goes through B.
b) 01101111…….. : Matches 1, 2 and 7, thus goes through A based on (1).
c) 00110000…….. : Matches default column 7: goes through C.
To better represent routing tables, they are stored in data structures called Trie. A trie is a tree in
which edges represent characters, and each of the nodes represents the string obtained by
traversing all edges from the root to that node. To store the routing table using trie, we insert
each of the rows as a string in the trie, and for the nodes at the end of each string, we determine
the next-hop. To achieve a more compressed representation, we may combine the nodes which
have only one child with their descendents. Here is a trie for the above example:
C C
0 1 0 1
A A
1 10111
0 1 0111
0 11 C
1 B 01
1 1
B B 0 1 A A
1 0 1
1
A 1 C
1
B 0 1 A

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COMPUTER COMMUNICATION NETWORKS

COMPUTER COMMUNICATION NETWORKS

UNIT 1 : INTRODUCTION TO NETWORKS

OSI reference model (Open Systems Interconnection)

Developed by representatives of major computer and telecommunication companies beginning

The seven Open Systems Interconnection layers are:

The TCP/IP Protocol Suite

Figure 2-1 shows the architecture of the TCP/IP protocol suite.

Figure 2-1 The architecture of the TCP/IP protocol suite

Network Interface Layer

IPv4 Internet Layer

Comparison of OSI Reference Model and TCP/IP Reference Model

OSI(Open System Interconnection) TCP/IP(Transmission Control Protocol /

1. OSI is a generic, protocol 1. TCP/IP model is based on standard

3. Follows vertical approach. 3. Follows horizontal approach.

4. OSI model has a separate 4. TCP/IP does not have a separate

5. OSI is a reference model around 5. TCP/IP model is, in a way implementation

6. Network layer of OSI model 6. The Network layer in TCP/IP model

8. Protocols are hidden in OSI 8. In TCP/IP replacing protocol is not easy.

9. OSI model defines services, 9. In TCP/IP, services, interfaces and

10. It has 7 layers 10. It has 4 layers

Using Telephone and Cable Networks for Data Transmission

Telephone companies provide two types of services: analog and digital.

• Telephone companies developed another technology, digital subscriber line (DSL), to

• ADSL is an asymmetric communication technology designed for residential users; it is

• SDSL, HDSL, and VDSL are other DSL technologies.

• Synchronous Optical Network (SONET) is a synchronous high-data-rate TDM network

• SONET has defined a hierarchy of signals (similar to the DS hierarchy) called

• Optical carrier (OC) levels are the implementation of STSs.

• A SONET frame can be viewed as a matrix of nine rows of 90 octets each.

• Communication in the traditional cable TV network is unidirectional.

• Communication in an HFC cable TV network can be bidirectional.

Three Major Components of Telephone System

Telephone Line Bandwidth

Bandwidth division in ADSL

Summary of DSL technologies

Division of coaxial cable band by CATV

A SONET system can use the following equipment:

STS multiplexer - combines several optical signals to make an STS signal.

Modem (also called a 'dial-up modem')

Advantages and disadvantages of modems

DIGITAL SUBSCRIBER LINE

Cable TV systems were developed to provide reliable TV service to local communities.

FUNCTIONS OF DATA LINK LAYER:

Data Link Layer - Creating a Frame

• Which nodes are in communication with each other

Data - The packet from the Network layer

Trailer - Contains control information added to the end of the PDU

Formatting Data for Transmission

Typical field types include:

Type field - The type of PDU contained in the frame

A data field -The frame payload (Network layer packet)

Following methods are used for carrying out framing.

The character count method is illustrated in Fig. 5.

Character Count Method

Starting and ending character with character stuffing :

Starting and ending character with character stuffing

Starting and ending flags, with bit stuffing :

Bit Stuffing and Destuffing