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Mobile Computing

The document outlines the syllabus for the Mobile Computing course at Vidhyaa Arts and Science College for M.Sc in Computer Science, covering various units including mobile computing fundamentals, telephony, GSM, GPRS, and wireless LAN technologies. It details the architecture, standards, and emerging technologies in mobile computing, as well as the functions and devices involved. The syllabus also includes model question papers and references for further reading.
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0% found this document useful (0 votes)
3 views

Mobile Computing

The document outlines the syllabus for the Mobile Computing course at Vidhyaa Arts and Science College for M.Sc in Computer Science, covering various units including mobile computing fundamentals, telephony, GSM, GPRS, and wireless LAN technologies. It details the architecture, standards, and emerging technologies in mobile computing, as well as the functions and devices involved. The syllabus also includes model question papers and references for further reading.
Copyright
© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
Available Formats
Download as PDF, TXT or read online on Scribd
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VIDHYAA ARTS AND SCIENCE COLLEGE

(For M.Sc(Computer Science), Semester - III)

DEPARTMENT OF COMPUTER SCIENCE

MOBILE COMPUTING

Version: 2018 – 2019 - 1.0


Contents
UNIT - I ............................................................................................................................... 1
1. 1. Mobile Computing .................................................................................................. 1
1.1.1. Dialog Control ................................................................................................... 4
1.1.2. Networks ............................................................................................................ 5
1.1.3. Middleware and Gateways............................................................................... 6
1.1.4. Application and Services .................................................................................. 8
1.1.5 Developing Mobile Computing Applications ................................................ 10
1.1.6. Standards.......................................................................................................... 12
1.1.7. Standard Bodies ............................................................................................... 13
1.1.8. Players in Wireless Space ................................................................................ 19
1.2. Mobile Computing Architecture .......................................................................... 19
1.2.1. Architecture for Mobile Computing .............................................................. 19
1.2.2. Three Tier Architecture ................................................................................... 20
1.2.3. Design Considerations for Mobile Computing ............................................. 27
UNIT - II ............................................................................................................................ 41
2.1. Mobile Computing Through Telephony.............................................................. 41
2.1.1. Evolution of Telephony................................................................................... 41
2.1.2. Multiple Access Procedure ............................................................................. 44
2.1.3. Mobile Computing Through Telephone........................................................ 45
2.1.4. Voice XML ........................................................................................................ 51
2.2. Emerging Technologies ......................................................................................... 59
2.2.1. Bluetooth .......................................................................................................... 59
2.2.2. RFID (Radio Frequency Identification).......................................................... 64
2.2.3. Mobile IP .......................................................................................................... 66
2.2.4. IPV6(Internet Protocol Version 6) .................................................................. 73
UNIT - III........................................................................................................................... 77
3.1. GSM......................................................................................................................... 77
3.1.1. Global System for Mobile Communications ................................................. 77
3.1.2. GSM Architecture ............................................................................................ 79
3.1.3. GSM Entities .................................................................................................... 82
3.1.4. Call Routing in GSM ....................................................................................... 85
3.1.5. GSM Address and Identifiers ......................................................................... 87
3.1.6. Network Aspects in GSM ............................................................................... 89
3.2. SMS ......................................................................................................................... 93
3.2.1. Mobile Computing Over SMS ........................................................................ 93
3.2.2. SMS (Short Message Service) .......................................................................... 93
3.2.3. Value Added Services through SMS .............................................................. 99
UNIT - IV ........................................................................................................................ 102
4.1. GPRS ..................................................................................................................... 102
4.1.1. GPRS and Packet Data Network .................................................................. 102
4.1.2. GPRS Network Architecture ........................................................................ 103
4.1.3. Data Services in GPRS ................................................................................... 111
4.1.4. Billing and Charging in GPRS ...................................................................... 113
4.2. WAP ...................................................................................................................... 115
4.2.1. Evolution of Wireless Data and WAP.......................................................... 115
4.2.2. GPRS Applications ........................................................................................ 116
UNIT - V.......................................................................................................................... 119
5.1. CDMA and 3G...................................................................................................... 119
5.1.1. Introduction ................................................................................................... 119
5.1.2. Architecture CDMA versus GSM................................................................. 120
5.1.3. IEEE 802.11Standards .................................................................................... 121
5.1.4. Wireless Data ................................................................................................. 122
5.2. Wireless LAN ....................................................................................................... 125
5.2.1. Introduction ................................................................................................... 125
5.2.2. Wireless Advantages ..................................................................................... 125
5.2.3. Wireless LAN Architecture .......................................................................... 129
5.2.4. Types of Wireless LAN ................................................................................. 130
5.2.5. Mobility in Wireless LAN ............................................................................. 131
5.2.6. Wireless LAN Security .................................................................................. 133
5.3. Next Generation Networks ................................................................................. 140
5.3.1. OFDM ............................................................................................................. 140
5.3.2. MPLS .............................................................................................................. 144
5.3.3. Wireless asynchronous transfer Mode ........................................................ 149
5.3.4. Multimedia Broadcast Services .................................................................... 156
MODEL QUESTION PAPER-I ...................................................................................... 159
MODEL QUESTION PAPER-II .................................................................................... 161
OBJECTIVE TEST-I ........................................................................................................ 163
MOBILE COMPUTING (17PCS11 ) - SYLLABUS

UNIT - I
Mobile Computing - Dialog Control - Networks - Middleware and Gateways -
Application and Services - Developing Mobile Computing Applications - Standards -
Standard Bodies – Players in Wireless Space. Mobile Computing Architecture:
Architecture for Mobile Computing – Three Tier Architecture - Design
Considerations for Mobile Computing

UNIT - II
Mobile Computing Through Telephony: Evolution of Telephony - Multiple Access
Procedure - Mobile Computing Through Telephone - Voice XML - TAPI - Emerging
Technologies: Bluetooth - RFID - Mobile IP - IPV6.

UNIT - III
GSM: Global System for Mobile Communications - GSM Architecture - GSM
Entities- Call Routing in GSM - GSM Address and Identifiers - Network Aspects in
GSM. SMS: Mobile Computing Over SMS - SMS - Value Added Services through
SMS.

UNIT - IV
GPRS: GPRS and Packet Data Network - GPRS Network Architecture - Data Services
in GPRS - Billing and Charging in GPRS. WAP: Evolution of Wireless Data and
WAP- GPRS Applications.

UNIT - V
CDMA and 3G: Introduction -Architecture CDMA versus GSM -IEEE
802.11Standards- Wireless Data. Wireless LAN: Introduction - Wireless Advantages-
Wireless LAN Architecture - Types of Wireless LAN - Mobility in Wireless LAN -
Wireless LAN Security. Next Generation Networks – OFDM – MPLS - Wireless
synchronous transfer Mode-Multimedia Broadcast Services.
TEXT BOOK
1. Asoke K Talukder, Roopa RYavagal, Mobile Computing , 2nd Edition,Tata MC
Graw Hill PublishingCompany Limited 2010.

REFERENCE BOOK
1. Jochen Schiller, Mobile Communications, Pearson Education, Second Edition,
2011.
2. William C.Y. Lee, Mobile Cellular Telecommunications, Second Edition,
McGraw Hill, 1995.
Vidhyaa Arts and Science College - Konganapuram

UNIT - I

1. 1. Mobile Computing

Mobile computing can be defined as a computing environment over physical


mobility. The user of a mobile computing environment will be able to access data,
information or other logical objects from any device in any network while on the
move. Mobile computing system allows a user to perform a task from anywhere
using a computing device in the public (the Web), corporate (business information)
and personal information spaces (medical record, address book). While on the move,
the preferred device will be a mobile device, while back at home or in the office the
device could be a desktop computer. To make the mobile computing environment
ubiquitous, it is necessary that the communication bearer is spread over both wired
and wireless media. Be it for the mobile workforce, holidaymakers, enterprises, or
rural population, the access to information and virtual objects through mobile
computing are absolutely necessary for optimal use of resource and increased
productivity.
Mobile computing is used in different contexts with different names. The
most common names are:

• Mobile Computing: The computing environment is mobile and moves


along with the user. This is similar to the telephone number of a GSM
(Global System for Mobile communication) phone, which moves with the
phone. The offline (local) and real time (remote) computing environment
will move with the user. In real time mode user will be able to use all his
remote data and services online.

• Anywhere, Anytime Information: This is the generic definition of ubiquity,


where the information is available anywhere, all the time.

• Virtual Home Environment: Virtual Home Environment (VHE) is defined


as an environment in a foreign network such that the mobile users can

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experience the same computing experience as they have in their home or


corporate computing environment. For example, one would like to put ones
room heater on when one is about 15 minutes away from home.

• Nomadic Computing: The computing environment is nomadic and moves


along with the mobile user. This is true for both local and remote services.

• Pervasive Computing: A computing environment, which is pervasive in


nature and can be made available in any environment.

• Ubiquitous Computing: A disappearing (nobody will notice its presence)


every place computing environment. User will be able to use both local and
remote services.

• Global Service Portability: Making a service portable and available in every


environment. Any service of any environment will be available globally.

• Wearable Computers: Wearable computers are those computers that may be


adorned by humans like a hat, shoe or clothes (these are wearable
accessories). Wearable computers need to have some additional attributes
compared to standard mobile devices. Wearable computers are always on;
operational while on move; hands free, context aware (with different types
of sensors). Wearable computers need to be equipped with proactive
attention and notifications. The ultimate wearable computers will have
sensors implanted within the body and supposedly integrate with the
human nervous system. These are part of new discipline of research
categorized by "Cyborg" (Cyber Organism).
Mobile Computing Functions
We can define a computing environment as mobile if it supports one or more
of the following characteristics:

• User Mobility: User should be able to move from one physical location to
another location and use the same service. The service could be in the home
network or a remote network.

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• Network Mobility: User should be able to move from one network to another
network and use the same service.

• Bearer Mobility: User should be able to move from one bearer to another
and use the same service.

• Device Mobility: User should be able to move from one device to another
and use the same service.

• Session Mobility: A user session should be able to move from one user-
agent environment to another.

• Service Mobility: User should be able to move from one service to another.

• Host Mobility: The user device can be either a client or server. When it is a
server or host, some of the complexities change. In case of host mobility the
mobility of IP needs to be taken care of.

The mobile computing functions can be logically divided into following major
segments

Mobile Computing Functions

• User with device: The user device, this could be a fixed device like desktop
computer in office or a portable device like mobile phone. Example: laptop
computers, desktop computers, fixed telephone, mobile phones, digital TV
with set-top box, palmtop computers, pocket PCs, two way pagers,
handheld terminals, etc.
• Network: Whenever a user is mobile, he will be using different networks at
different places at different time. Example: GSM, CDMA, iMode, Ethernet,

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Wireless LAN, Bluetooth etc.

• Gateway: This is required to interface different transport bearers. These


gateways convert one specific transport bearer to another transport bearer.

• Middleware: This is more of a function rather than a separate visible node.


In the present context middleware handles the presentation and rendering
of the content on a particular device. It will also handle the security and
personalization for different users.
• Content: This is the domain where the origin server and content is. This
could be an application, system, or even an aggregation of systems. The
content can be mass market, personal or corporate content. Origin server
will have some means to accessing the database and the storage devices.
Mobile Computing Devices
The device for mobile computing can be either a computing device or a
communication device. In computing device category it can be a desktop computer,
laptop computer, or a palmtop computer. On the communication device side it can
be a fixed line telephone, a mobile telephone or a digital TV. The human computer
interface challenges are:
 Interaction must be consistent from one device to another.
 Interaction has to be appropriate for the particular device and
environment in which the system is being used.
1.1.1. Dialog Control

In any communication there are two types of user dialogues. These are
• Long session-oriented transactions - Going through a monolithic
document page by page can be considered as a session-oriented
transaction.
• Short transaction – Going to a particular page directly through an
index can be considered as a short transaction.

Selection of the transaction mode will depend on the type of device we use. A
session may be helpful in case of services offered through computers with large
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screens and mouse. For devices with limited input/output like SMS for instance,
short transactions may be desired. Let us consider an example of bank balance
enquiry over the Internet. In case of Internet banking through desktop computer,
the user has to go through the following minimum dialogues:

 Enter the URL of the bank site.


 Enter the account number/password and Login into the application.
 Select the balance enquiry dialogue and see the balance.
 Logout from the internet banking.
1.1.2. Networks

Mobile computing will use different types of networks. These can be fixed
telephone network, GSM, GPRS, ATM (Asynchronous Transfer Mode), Frame Relay,
ISDN (Integrated Service Digital Network), CDMA, CDPD (Cellular Digital Packet
data), DSL (Digital Subscriber Loop), Dial-up, Wi-Fi (Wireless Fidelity), 802.11,
Bluetooth, Ethernet, Broadband, etc.
Wire-line Networks
This is a network, which is designed over wire or tangible conductors. This
network is called fixed or wire-line network. Fixed telephone networks over copper
and fiber-optic will be part of this network family. Broadband networks over DSL or
cable will also be part of wire-line networks. Wire-line network are generally public
networks and cover wide areas. Though microwave or satellite networks do not use
wire, when a telephone network uses microwave or satellite as a part of its
infrastructure, it is considered part of wire-line networks. When we connect to ISPs it
is generally a wire-line network. The Internet backbone is a wire-line network as
well.
Wireless Networks
Mobile networks are generally termed as wireless network. This includes
wireless networks used by radio taxis, one way and two way pager, cellular phones.
Example will be PCS (Personal Cellular System), AMPS (Advanced Mobile Phone
System), GSM, CDMA, DoCoMo, GPRS etc. WiLL (Wireless in Local Loop) networks

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using different types of technologies are part of wireless networks as well. In a


wireless network the last mile is wireless and works over radio interface. In a
wireless network other than the radio interface rest of the network is wire-line, this is
generally called the PLMN (Public Land Mobile Network).
Ad-hoc Networks
In Latin, ad hoc literally means 'for this purpose only'. An ad-hoc (or
spontaneous) network is a small area network, especially one with wireless or
temporary plug-in connections. In these networks some of the devices are part of the
network only for the duration of a communication session. An ad-hoc network is
also formed when mobile, or portable devices, operate in close proximity of each
other or with the rest of the network.
Bearers
For different type of networks, there are different types of transport bearers.
These can be TCP/IP, http, protocols for or dialup connection. For GSM it could be
SMS, USSD (Unstructured Supplementary Service Data) or WAP. For mobile or
fixed phone, it will be Voice.

1.1.3. Middleware and Gateways

Any software layered between a user application and operating system can be
termed as middleware. Middleware examples are communication middleware,
object oriented middleware, message oriented middleware, transaction processing
middleware, database middleware, behavior management middleware, RPC
middleware etc. There are some middleware components like behavior management
middleware, which can be a layer between the client device and the application. In
mobile computing context we need different types of middleware components and
gateways at different layers of the architecture. These are:

• Communication middleware
• Transaction processing middleware
• Behavior management middleware
• Communication gateways

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Communication Middleware
The application will communicate with different nodes and services through
different communication middleware. Different connectors for different services
will fall in this category. Examples could be TN3270 for IBM mainframe services, or
Javamail connector for IMAP or POP3 services
Transaction Processing Middleware
In many cases a service will offer session oriented dialogue (SoD). For a
session we need to maintain a state over the stateless Internet. This is done through
an application server. The user may be using a device, which demands a short
transaction whereas the service at the backend offers a SoD. In such cases a separate
middleware component will be required to convert a SoD to a short transaction.
Management of the Web components will be handled by this middleware as well.

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Behavior Management Middleware


For different devices we need different types of rendering. We can have
applications, which are developed specially for different types of rendering. For
example, we can have one application for Web, another for WAP, and a different one
for SMS. On the contrary, we may choose to have a middleware, which will manage
entire device specific rendering at the run time. This middleware will identify the
device properly and handle all the behavior related stuff independent of the
application. The system may be required to have some context awareness. All these
will be handled by behavior management middleware.
Communication Gateways
Between the device and the middleware there will be network of networks.
Gateways are deployed when there are different transport bearers or networks with
dissimilar protocols. For example, we need an IVR gateway to interface voice with a
computer, or a WAP gateway to access internet over a mobile phone.
1.1.4. Application and Services

Data and information, through mobile computing services are required by all
people regardless of the fact that they are mobile are not. There can be many
applications and services for the mobile computing space. These application or
services run on origin server. These are also known as content servers. Contents will
primarily be lifestyle specific. From lifestyle perspective they can be grouped into
different categories like:

• Personal—belongs to the user (wallet, life-tool, medical records, diary).

• Perishable—time sensitive and relevance passes quickly (general news,


breaking news, weather, sports, business news, stock quotes).
• Transaction Oriented-transactions need to be closed (bank transactions,
utility bill payment, mobile shopping).
• Location Specific—information related to current geographical location
(street direction map, restaurant guide).

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• Corporate—corporate business information (mail, ERP, inventory, directory,


business alerts, reminders).
• Entertainment-applications for fun, entertainment.
Here are some examples:

• News: News could be political, current affair, breaking news, business news,
sports news, community news etc.

• Youth: These are primarily messaging based application like person-to-


person messaging, chat, forums, dating etc.

• Weather: GPS-based system to locate a person (with GPS receiver) can


sometime save lives in case of natural calamity.

• Corporate application: This will include corporate mail, address book,


appointments, MIS applications, corporate Intranet, corporate ERP etc.

• Sales Force Automation: Applications will include sales order booking,


inventory enquiry, shipment tracking, logistics related applications etc

• m-broker: Getting correct and timely information related to different stocks


are very important. Also, online trading of stocks while on move is quite
critical for certain lifestyle. Stock ticker, stock alerts, stock quote, and stock
trading can be made ubiquitous so that users can check their portfolio and
play an active role in this marketplace.

• Tele-banking: We need to access our banks for different transactions. The


tele-banking can be made ubiquitous it helps everybody, both the customer
and the bank. Many banks in India are today offering banking over Internet
(web), voice and mobile phones through SMS.

• m-shopping: Application to do different types of shopping using mobile


devices like Palmtop, PocketPC, mobile phone etc. Buying of a soft
drink/soda from a vending machine in an airport or a movie theatre using a
mobile phone may be very handy especially when we do not have change.

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• GPS-based systems: Applications related to location tracking come under


this category.

• Remote monitoring: This is important for children at home where parents


monitor where their children are or what are they doing. Also, monitoring
and controlling of home appliances will be part of this application.

• Directory services: This includes information related movie theatre, public


telephone, restaurant guide, public information system and Yellow pages.

• Sports: This service offers online sports update. In India live cricket score is
the most popular mobile computing application. Live cricket score is
available in India with many service providers through Web, Voice, SMS and
WAP.

• Maps/navigation guide: This is an application, which has lot of demand for


traveling individuals. These services need to be location aware to guide the
user to use the most optimum path to reach a destination. The directions
given by these applications also take the traffic congestion, one way, etc. into
consideration.

• Digital library: Internet converted the world into a global village and global
library. Accessing this library from anywhere anytime will only help
reducing the digital divide.

• Job facilitator: These could be either proactive alerts or information related to


jobs and employment opportunities.

1.1.5 Developing Mobile Computing Applications

Any portal system today supports user mobility. If I have an Internet mail
account like hotmail or yahoo, I can access my mail from anywhere. I need a desktop
or laptop computer to access my mailbox. I may not be able to access the same mail
through some other device like a fixed phone. There are a number of factors that
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make mobile computing different from desktop computing. As a result of mobility


the attributes associated with devices, network, and users are constantly changing.
These changes imply that context and behavior of applications needs to be adapted
to suit the current environment.

The context and behavior adaptation is required to provide a service that is


tailored to the user's present situation. There are several ways in which context and
behavior can be adapted. One way is to build applications without any context or
behavior awareness. Context and behavior adaptation will be handled by a behavior
management middleware at the runtime. Other option is to build different
applications specific to different context and behavior patterns. For a long life system
or a legacy application the content behavior adaptation will need to be done
externally.
New Mobile Applications
Let us assume that in a bank, some new applications need to be built for e-
Commerce. The bank wants to offer banking through voice (telephone) and Web
(Internet). Assuming that the bank already has a computerized system in place, the
bank will develop two new applications. One will handle the telephone interface
through Interactive Voice Response (IVR) and the other through Web. At a later
point in time, if the bank decides to offer SMS and WAP, they will develop two new
applications to support SMS and WAP interfaces respectively. To protect the
investment and quick adaptation, the bank may decide to use transaction processing
middleware and RPC middleware. All these are possible only if it is a fresh
applications development.
Making Legacy Application Mobile
How do we make a long life existing legacy application mobile? We define an
application as legacy if it has one or more of the following characteristics:

• The application has moved into the sustenance phase in the software
development lifecycle.

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• An application, which cannot be modified. This could be due to unavailability


of the original development platforms, unavailability of original source code
or unavailability of expertise to make necessary changes.

• Products and packaged software where enterprise does not have any control.
This could be due to high cost of ownership for new upgrade or the vendor
does not have any plan to support the new requirement.

1.1.6. Standards

Standards are documented agreements containing technical specifications or


other precise criteria to be used consistency as rules, guidelines or definitions of
characteristics. Standards ensure that materials, products, processes and services fit
for their defined and agreed purpose. A standard begins as a technical contribution,
which is supported by a number of interested parties to the extent that they indicate
their willingness to participate in the standard's development. Standards are
available for experts to challenge, examine and validate. No industry in today's
world can truly claim to be completely independent of components, products, rules
of application that have been developed in other sectors. Without standards,
interoperability of goods and services will not be possible.

When the proposed standard or technical document is near completion, the


formulating Engineering Committee circulates the draft of the document for a ballot.
The purpose of this ballot is to identify any unresolved issues and to establish
consensus within the formulating group. Every effort is made to address and resolve
comments received.
Who makes the standards?

There are many institutes that generate and provide standards across the
world. There are standard bodies at the regional or country level; also, there are
bodies at the international level. Based on the area of operations, standard bodies
are formed by the governments, professional institutes or industry consortiums.
These standard bodies sometime also function as regulators. In India there is a

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standard body under the Government of India, which is called Bureau of Indian
Standard or simply BIS. A standards process include following steps:

• Consensus on a proposed standard by a group or 'consensus body' that


includes representatives from materially affected and interested parties.

• Broad-based public review and comment on draft standards.

• Consideration of and response to comments submitted by voting members of


the relevant consensus body and by public review commenter.

• Incorporation of approved changes into a draft standard.

• Right to appeal by any participant that believes that due process principles
were not sufficiently respected during the standards development in
accordance with the ANSI-accredited procedures of the standards developer.

1.1.7. Standard Bodies

ISO

The International Organization for Standardization (ISO) is a worldwide


federation of national standards bodies from more than 140
countries, one from each country. ISO is a non-governmental organization
established in 1947. The mission of ISO is to promote the development of
standardization and related activities in the world with a view to facilitating the
international exchange of goods and services, and to developing cooperation in the
spheres of intellectual, scientific, technological and economic activity. ISO9000
relates to Software quality. The famous 7-layer model for Open System
Interconnection (OSI) is ISO standard (IS07498). For Information security ISO has
come up with the recommendation ISO 17799.
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IETF

The Internet Engineering Task Force (IETF) is the standard-making body for
Internet and related technologies. IETF is an open international community of
network designers, operators, vendors and researchers concerned with the evolution
of the Internet architecture and the smooth operation of the Internet. The Internet
Assigned Numbers Authority (IANA) is the central coordinator for the assignment
of unique IP address. Standards defined by IETF are called Request For Comment or
RFC. The standard for email is defined in RFC821 (Simple Mail Transfer Protocol or
SMTP); RFC2616 describes the version 1.1 of Hypertext Transfer Protocol
(HTTP/1.1).
ETSI

ETSI (the European Telecommunications Standards Institute) is an


organization whose mission is to produce the telecommunications standards that
will be used for decades to come throughout Europe and possibly beyond.ETSI is
officially recognized by the European Commission. GSM Standard is created,
maintained and managed by a committee within ETSI. GSM standards document
GSM 01.04 (ETR 350): 'Digital cellular telecommunications system (Phase 2+);
Abbreviations and acronyms'. GSM 12.13 standard defines the interface Digital
cellular telecommunications system (Phase 2+); Man-Machine Interface (MMI) of the
Mobile Station (MS) (GSM 02.30 version 7.1.0 Release 1998).

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OMA

The Open Mobile Alliance (OMA) has been established by the consolidation
of the WAP Forum and the Open Mobile Architecture initiative. WAP and MMS
standards are created, maintained and managed by OMA.
ITU

ITU (International Telecommunication Union) is an organization within the


United Nations System. It was founded on the principle of cooperation between
governments and the private sector. With a membership encompassing
telecommunication policy-makers and regulators, network operators, equipment
manufacturers, hardware and software developers, regional standards-making
organizations and financing institutions. ITU's activities, policies and strategic
direction are determined and shaped by the industry it serves. ITU has three Sectors
of the Union; they are Radio communication (ITU-R), Telecommunication
Standardization (ITU-T), and Telecommunication Development (ITU-D). ITU-T was
founded in 1993, replacing the former International Telegraph and Telephone
Consultative Committee (CCITT) whose origins go back to 1865. Any telephone in
this world has a unique number (technically known as Global Title). These
numbering schemes are defined through the ITU-T standards E.164.

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IEEE

The IEEE Standards Association (IEEE-SA) is an organization that produces


standards, which are developed and used internationally. Standards for Wireless
LAN are created, maintained and managed by IEEE. These are defined through
different 802.11 standards.
EIA

The Electronic Industries Alliance (EIA) is a national trade organization


within USA that includes the full spectrum of U.S. electronics industry. EIA
comprises companies whose products and services range from the
smallest electronic components to the most complex systems used by defense, space
and industry, including the full range of consumer electronic products. One of the
most commonly used EIA standard is EIA RS-232. This is a standard for 25-pin
connector between a computer and a modem.
W3C

The World Wide Web Consortium (W3C) develops interoperable technologies


(specifications, guidelines, software, and tools) to lead the Web to its full potential.
W3C is a forum for information, commerce, communication and collective
understanding. W3C recommendations include HTML, XML, CSS (Cascading Style
Sheet), Web Services, DOM (Document Object Model), MathML (Maths Markup

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Language), PNG (Portable Network Graphics), SGV (Scalable Vector Graphics), RDF
(Resource Description Framework), P3P (Platform for Privacy Preferences) etc.
3GPP

3GPP is to produce globally applicable technical specifications and technical


reports for 3rd Generation Mobile System based on evolved GSM core networks and
the radio access technologies that they support-i.e., Universal Terrestrial Radio
Access (UTRA) both Frequency Division Duplex (FDD) and Time Division Duplex
(TDD) modes.
ANSI

The American National Standards Institute (ANSI) is the national standard


organization in the United States. ANSI Standard X3.4-1968 defines 'American
National Standard Code for Information Interchange (ASCII)' character set. ASCII
character set is used in almost every modern computer today. The same standard
has also been adapted as ISO 8859-1 standard.
UMTS

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Universal Mobile Telecommunications System (UMTS) represents an


evolution in terms of services and data speeds from today's 'second generation'
mobile networks like GSM. UMTS-Forum is the standards-making body for
WCDMA (Wideband Code Division Multiple Access) and UMTS technology.
BLUETOOTH

Bluetooth wireless technology is a worldwide specification for a small-form


factor, low cost radio solution that provides links between mobile computers, mobile
phones, other portable handheld devices, and connectivity to the Internet. The
standards and specification for Bluetooth are developed, published and promoted by
the Bluetooth Special Interest Group.
CDG

The CDMA Development Group (CDG) is an international consortium of


companies who have joined together to lead the adoption and evolution of CDMA
wireless systems around the world. The CDG is comprised of the world's leading
CDMA service providers and manufacturers.
PKCS
The Public-Key Cryptography Standards are specifications produced by RSA
Laboratories in cooperation with secure systems developers worldwide for the
purpose of accelerating the deployment of public key cryptography. Contributions
from the PKCS series have become part of many formal and de facto standards,
including ANSI X9 documents, PKIX, SET, S/MIME and SSL (Secure Sockets Layer).

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PAM

In the world of ubiquitous computing, knowing the position and context of a


device is very important. The Presence and Availability Management (PAM) Forum
is an independent consortium will define a framework for the various standards and
specifications needed for context/location aware applications.
PARLAY GROUP
The Parlay Group is a multi-vendor consortium formed to develop open,
technology-independent application programming interfaces (APIs). Parlay
integrates intelligent network (IN) services with IT applications via a secure,
measured, and billable interface.

1.1.8. Players in Wireless Space

In a wireless network there are many stakeholders. These are:


• Regulatory authorities
• The operator or the service provider
• The user or the subscriber
• Equipment vendors (network equipment and user device)
• Research organizations.

1.2. Mobile Computing Architecture

1.2.1. Architecture for Mobile Computing


The network-centric mobile computing architecture uses three-tier
architecture. In the three-tier architecture, the first layer is the User Interface or
Presentation Tier. This layer deals with user facing device handling and rendering.
This tier includes a user system interface where user services (such as session, text

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input, dialog and display management) reside. The second tier is the Process
Management or Application Tier. This layer is for application programs or process
management where business logic and rules are executed. This layer is capable of
accommodating hundreds of users. In addition, the middle process management tier
controls transactions and asynchronous queuing to ensure reliable completion of
transactions. The third and final tier is the Database Management or Data Tier. This
layer is for database access and management.
The three-tier architecture is better suited for an effective networked
client/server design. It provides increased performance, flexibility,
maintainability, reusability, and scalability, while hiding the complexity of
distributed processing from the user.

1.2.2. Three Tier Architecture

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The three-tier architecture for a mobile computing environment consist


presentation tier, application tier and data tier. Depending upon the situation, these
layers can be further sub layered.
Presentation (Tier-1)
This is the user facing system in the first tier. This is the layer of agent
applications and systems. These applications run on the client device and offer all
the user interfaces. This tier is responsible for presenting the information to the end
user. Humans generally use visual and audio means to receive information from
machines and use keyboard, pen, touch screen or voice to feed the data to the
system. In the case of the visual, the presentation of information will be through a
screen. Therefore 'Presentation Tier' includes web browsers (like Mozila, lynx,
Internet Explorer and Netscape Navigator), WAP browsers and customized client
programs. A mobile computing agent needs to be context-aware and device
independent.
In general, the agent software in the client device is an Internet browser. In
some cases, the agent software is an applet running on a browser or a virtual
machine (Java Virtual Machine for example).

There are agent software in the Internet that access the remote service through
telnet interface. There are different flavors of telnet agents in use. These are standard
telnet for Unix servers; TN3270 for IBM OS/390; TN5250 for IBM AS/400 or VT3K
for HP3000.
Application Tier (Tier-2)

The application tier or middle tier is the 'engine' of a ubiquitous application. It


performs the business logic of processing user input, obtaining data, and making
decisions. In certain cases, this layer will do the trans-coding of data for appropriate
rendering in the Presentation Tier. The Application Tier may include technology like
CGI's, Java, JSP, .NET services, PHP or ColdFusion, deployed in products like
Apache, WebSphere, WebLogic, iPlanet, Pramati, JBOSS or ZEND. The application
tier is presentation and database-independent.

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In a mobile computing environment, in addition to the business logic need to


be perform some functions relate to decisions on rendering, network management,
security, data store access etc. Most of these functions are implemented using
different middleware software. A middleware framework is defined as a layer of
software, which sits in the middle between the operating system and the user facing
software. Middleware covers a wide range of software systems, including
distributed objects and components, message-oriented communication, database
connectors, mobile application support, transaction drivers, etc. Middleware can also
be considered as a software gateway connecting two independent open objects.

We can group middleware into following major categories.

• Message-Oriented Middleware
• Transaction Processing Middleware
• Database Middleware
• Communication Middleware
• Distributed Object and Components
• Trans-coding Middleware.
Message-oriented Middleware (MOM)
Message-oriented Middleware is a middleware framework that loosely
connects different applications through asynchronous exchange of messages. A
MOM works over a networked environment without having to know what platform
or processor the other application is resident on. The message can contain formatted
data, requests for action, or unsolicited response. The MOM system provides a
message queue between any two interoperating applications. If the destination
process is out of service or busy, the message is held in a temporary storage location
until it can be processed. Example of MOM are Message Queue from IBM known as
MQ Series. The equivalent from Java is JMS (Java Message Service).
Transaction Processing (TP) Middleware
Transaction Processing Middleware provides tools and an environment for
developing transaction-based distributed applications. An ideal TP system will be
able to input data into the system at the point of information source and the output
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of the system is delivered at the point of information sink. In an ideal TP system, the
device for input and output can potentially be different. TP systems are generally
capable of providing services to thousands of clients in a distributed client/server
environment.

TP middleware maps numerous client requests through application-service


routines to different application tasks.

Communication Middleware
Communication Middleware is used to connect one application to another
application through some communication middleware, like connecting one
application to another application through telnet. These types of middleware are
quite useful in the telecommunication world.
Distributed Object and Components
An example of distributed objects and components is CORBA (Common
Object Request Broker Architecture). CORBA is an open distributed object
computing infrastructure being standardized by the Object Management Group.
CORBA is useful in many situations because of the easy way that CORBA integrates
machines from so many vendors, with sizes ranging from mainframes through minis
and desktops to hand-helds and embedded systems. One of its most important, as
well as the most frequent, uses is in servers that must handle a large number of
clients, at high hit rates, with high reliability.
Trans-coding Middleware
Trans-coding Middleware is used to trans-code one format of data to another
format to suit the need of the client.

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Internet Content Adaptation Protocol (ICAP)


Popular web servers are required to deliver content to millions of users
connected at ever-increasing bandwidths. Progressively, contents is being accessed
through different devices and agents. ICAP is a lightweight protocol to do trans-
coding on HTTP messages. This is similar to executing a 'remote procedure call' on a
HTTP request. The protocol allows ICAP clients to pass HTTP messages to ICAP
servers for some sort of transformation. The server executes its transformation
service on messages and sends back responses to the client, usually with modified
messages. The adapted messages may be either HTTP requests or HTTP responses.
For example, before a document is displayed in the agent, it is checked for virus.

Typical data flow in an ICAP environment is described below.

 A user agent makes a request to an ICAP-capable surrogate (ICAP client)


for an object on an origin server.

 The surrogate sends the request to the ICAP server.

 The ICAP server executes the ICAP resource's service on the request and
sends the possibly modified request, or a response to the request back to
the ICAP client.
 The surrogate sends the request, possibly different from the original client
request, to the origin server.

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 The origin server responds to the request.

 The surrogate sends the reply (from either the ICAP server or the origin
server) to the client.

In future, ICAP servers may be available to provide some of the following services:

• Suite content delivery based on network bandwidth.

• Suite content delivery based on device characteristics.

• Language translation based on the user's preference.

• Virus checking for the requested content.

• Content filtering based on the sensor rating like PG (Parental Guidance), R


Restricted).

• Local real time advertisement insertion like television

• Local real time advertisement elimination for premium subscribers

• Wireless protocol translation

• Anonymous Web usage profiling for a dating service

• Trans-coding or image enhancement

• Image magnification for the elderly

• Image size reduction based on device display characteristics

• Intelligent Video condensation by dropping frames

• Digest production/batch download of Web content

• Content filtering based on copyright or digital signature

• Peer-to-Peer compression and encryption of data


Web services
Web Services provide a standard means of communication and information
exchange among different software applications, running on a variety of platforms

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or frameworks. Web service is a software system identified by a URI, whose public


interfaces and bindings are defined using XML (extensible Markup Language).

The basic architecture includes Web service technologies capable of:

• Exchanging messages.

• Describing Web services.

• Publishing and discovering Web service descriptions.


Data Tier (Tier-3)
The Data Tier is used to store data needed by the application and acts as a
repository for both temporary and permanent data. The data could be stored in any
form of data store or database. These can range from sophisticated relational
database, legacy hierarchical database, to even simple text files. The data can also be
stored in XML format for interoperability with other system and data sources. A
legacy application can also be considered as a data source or a document through a
communication middleware.
Database Middleware
Database middleware allows the business logic to be independent and
transparent of the database technology and the database vendor. Database
middleware runs between the application program and the database. These are
sometimes called database connectors as well. Example of such middleware will be
ODBC, JDBC, etc. Using these middleware, the application will be able to access data
from any data source. Data sources can be text files, flat file, spreadsheets, or a
network, relational, indexed, hierarchical, XML database, object database, etc. from
vendors like Oracle, SQL, Sybase, etc.
SyncML
SyncML protocol is an emerging standard for synchronization of data access
from different nodes. SyncML is a new industry initiative to develop and promote a
single, common data synchronization protocol that can be used industry-wide. The
SyncML protocol supports naming and identification of records, common protocol
commands to synchronize local and network data. It supports identification and

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resolution of synchronization conflicts. The protocol works over all networks used
by mobile devices, both wireless and wireline. Since wireless networks employ
different transport protocols and media, a SyncML will work smoothly and
efficiently over:

• HTTP 1 (i.e. the Internet)

• WSP (the Wireless Session Protocol, part of the WAP protocol suite)

• OBEX (Object Exchange Protocol, i.e. Bluetooth, IrDA and other local
connectivity)

• SMTP, POP3 and IMAP

• Pure TCP/IP networks

• Proprietary wireless communication protocols.

1.2.3. Design Considerations for Mobile Computing


The mobile computing environment needs to be context-independent as well
as context-sensitive. Context information is the information related to the
surrounding environment of an actor in that environment. The term 'context' means,
all the information that help determine the state of an object (or actor). This object
can be a person, a device, a place, a physical or computational object, the
surrounding environment or any other entity being tracked by the system. In a
mobile computing environment, context data is captured so that decisions can be
made about how to adapt content or behavior to suit this context. Mobility implies
that attributes associated with devices and users will change constantly. These
changes mean that content and behavior of applications should be adapted to suit
the current situation. There are many ways in which content and behavior can be
adapted. Following are some examples:

 Content with context awareness Build each application with context


awareness. There are different services for different client context (devices).

 Content switch on context Another way is to provide intelligence for the


adaptation of content within the service. This adaptation happens
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transparent to the client. In this case the service is the same for Internet,
PDA and WAP.

 Content trans-coding on context Another way is to provide an underlying


middleware platform that performs the adaptation of the content based on
the context and behavior of the device. This adaptation happens transparent
to the client and the application. The middleware platform is intelligent
enough to identify the context either from the http parameters or additional
customized parameters. In this case the service may be in html or XML, the
middleware platform trans-code the code from html (or XML) to html, and
WML on the fly. It can also do the trans-coding based on policy so that the
html generated for a computer is different from a PDA.
Client Context Manager

A mobile computing application, needs to operate in dynamic conditions.


This is due to various device characteristics and network conditions. This demands a
reactive platform that can make decisions about how to respond to changes to device
capability, user preferences, enterprise policy, network policy and many other

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environmental factors. Context can be used as the basis by which an adaptation


manager or algorithm decides to modify content or application behavior. We
therefore need a Client Context Manager to gather and maintain information
pertaining to the client device, user, network and the environment surrounding each
mobile device. All these information will be provided by a set of Awareness
Modules. Awareness modules are sensors of various kinds. These sensors can be
hardware sensors or software sensors or a combination of these. A hardware sensor
can be used to identify the precise location of a user; whereas, a software sensor can
be used to determine the type of the user agent. These awareness modules can be in
the device, network, or even in the middleware. Almost any information available at
the time of an interaction can be seen as context information. Some examples are:
 Identity: The device will be in a position to communicate its identity without
any ambiguity.
 Spatial information: Information related to the surrounding space. This
relates to location, orientation, speed, elevation and acceleration.
 Temporal information: Information related to time. This will be time of the
day, date, timezone and season of the year.
 Environmental information: This is related to the environmental
surroundings. This will include temperature, air quality, moisture, wind
speed, natural light or noise level. This also includes information related to
the network and network capabilities.
 Social situation: Information related to the social environment. This will
include who you are with, and people that are nearby; whether the user is in a
meeting or in a party.
 Resources that are nearby: This will relate to the other accessible resources in
the nearby surroundings like accessible devices, hosts or other information
sinks.
 Availability of resources: This will relate to information about the device in
use. This will include battery power, processing power, persistence store,
display, capabilities related to I/O (Input/Output) and bandwidth.
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 Physiological measurements: This relates to the physiological state of the


user. This includes information like blood pressure, heart rate, respiration
rate, muscle activity and tone of voice.
 Activity: This relates to the activity state of the user. This includes
information like talking, reading, walking and running.
 Schedules and agendas: This relates to the schedules and agendas of the
user.
Three aspects of context management:
 Context sensing: The way in which context data is obtained.
 Context representation: The way in which context information is stored and
transported.
 Context interpretation: The way in which meaning is obtained from the
context representation.

W3C has proposed a standard for context information. This standard is called
Composite Capabilities/Preference Profiles (CC/PP), for describing device
capabilities and user preferences.
Composite Capabilities/Preference Profiles (CC/PP)
The CC/PP model is based on RDF (Resource Description Framework) and
can be serialized using XML. A CC/PP profile contains a number of attribute names
and associated values that are used by an application to determine the appropriate
form of a resource to deliver to a client. This is to help a client or proxy/middleware
to describe their capabilities to an origin server or other sender of resource data. It is
anticipated that different applications will use different vocabularies to specify
application-specific properties within the scope of CC/PP. However, for different
applications to interoperate, some common vocabulary is needed. The CC/PP
standard defines all these.

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CC/PP is designed in such a way that an origin server or proxy can perform
some sort of content to device matching. CC/PP is designed to suit an adaptation
algorithm. The sequence of steps in the general case would look something like the
following:
 Device sends serialized profile model with request for content.
 Origin server receives serialized RDF profile and converts it into an in-
memory model.
 The profile for the requested document is retrieved and an in-memory model
is created.
 The device profile model is matched against the document profile model.
 A suitable representation of the document is chosen. At this stage the
document to be returned can be chosen from a number of different versions of
the same document (content switch on context) or it can be dynamically
generated (content trans-coding on context).
 Document is returned to device and presented.
Policy Manager
The Policy Manager is responsible for controlling policies related to mobility.
A policy is a set of rules; these rules need to be followed under different conditions.
Semantic Web
Facilities to put machine understandable data on the Web are becoming a
necessity. The Semantic Web is targeted to address this need. The idea of having
data on the web defined and linked in a way that it can be used by machines not just
for display purposes, but for automation, security, filtering, integration and reuse of
data across various applications.
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The Semantic Web is generally built on syntaxes which use URIs to represent
data, usually in tuple-based structures: i.e. many tuples of URI data that can be held
in databases, or interchanged on the world Wide Web using a set of particular
syntaxes developed especially for the task. These syntaxes are called 'Resource
Description Framework (RDF)' syntaxes.
Security Manager
The Security Manager provides secure connection between the client device
and the origin server. Depending on the security policies of an organization, if the
security requirements are not met, some content may not be viewable. Security
manager will ensure security with respect to:
• Confidentiality-the message being transacted needs to be confidential.
Nobody will be able to see it.
• Integrity-the message being transacted needs to be tamper-resistant. Nobody
will be able to change any part of the message.
• Availability—the system will be available. Nobody will be able to stop the
service.
• Non-repudiation-the users of the system can be identified. Nobody after
using the system can claim otherwise.
• Trust-there are complex issues of knowing what resources, services or agents
to trust. The system will be trusted.
Confidentiality is managed by encryption. Using encryption techniques we
change the message to some other message so that it cannot be understood. There
are different types of encryption algorithms and standards. In a defined
environment like enterprise LAN or a VPN (Virtual Private Network), we can
standardize some encryption algorithm like 128 bits AES to be used. Integrity can be
managed using different hashing algorithms. Availability relates to peripheral
security related to Web server, firewall etc. The non-repudiation can be managed
with digital signature. For trust we may need to establish some sort of third-party
recommendation system. Third party rating system can also help establish trust. The
security manager needs to manage all these aspects.

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Platform for Privacy Preference Project (P3P)


The Platform for Privacy Preference Project (P3P) is an emerging standard
defined by W3C. P3P enables web sites to express their privacy practices in a
standardized format so that it can be retrieved and interpreted by user agents. The
goal of P3P is to increase user trust and confidence in the Web.
Adaptability Manager
The Adaptability Manager is responsible for adapting content, behavior and
other aspects according to context and policy.
Content Adaptation and Trans-coding
The content should be able to adapt to these dynamic situations. The
adaptation may be static or dynamic. Content adaptation can be performed either at
the content level at the server end or at the agent level in the client device. Content
adaptation can be done at an intermediate level in a middleware framework as well.
Content adaptation needs to consider the following attributes.
 Physical capabilities of the device, viz., screen size i.e., width and height in
pixels, color and bits/pixel.
 Logical capabilities of the device for displaying video, image and playing
audio.
 Effective Network bandwidth.
 Payload can be defined as the total amounts of bits that can be delivered to
the agent for the static parts. For streaming media this will be the initial
buffer space required before the media starts playing. For storage constrained
devices, the payload will be defined as the storage space.
Trans-coding can be classified as the following:
• Spatial trans-coding is trans-coding in space or dimension. In this trans-
coding technique a standard frame is downscaled and reduced. The frame
is changed from one size to a different size to suit the target device.
• Temporal trans-coding copes with a reduction of number of frames in the
time scale. This technique downscales the number of transferred frames to
suit the target device and the network bandwidth.

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• Color trans-coding is sometimes requested for monochrome clients. Using


less bits for pixel can reduce bandwidth and sometime modify the
perception of images.
• Code trans-coding is used to change coding from one standard to another.
One such example could be compression of the data or trans-code a BMP
file to WBMP for wireless device.
• Object or semantic trans-coding comprises some different techniques based
on computer vision techniques. The goal is to extract semantically valuable
objects from the scene and transfer them with the lower amount of
compression in order to maintain both details and speed.
Server side content adaptation can be achieved through the concept of
InfoPyramid. InfoPyramid creates context-aware content through static trans-
coding. The trans-coding is done off-line at the content creation time. InfoPyramid is
used to store multiple resolutions and modalities of the trans-coded content, along
with any associated meta-data. The resource requirement is determined by the
following attributes.

 Static content size in bits.


 Display size such as height, width and area.
 Streaming bit-rate.
 Color requirements.
 Compression formats.
 Hardware requirements, such as display for images, support
for audio and video.

In case of client-side adaptation, the adaptation is done by the agent


application. The agent application does the adaptation based on its capabilities.
The other technique of trans-coding is through a middleware. One big benefit
of the middleware approach is that it is totally transparent to the device and the
content.

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Content providers do not have to change the way they author or serve content.
However, there are a number of drawbacks to this approach:
 Content providers have no control over how their content will appear to
different clients.
 There may be legal issues arising from copyright that may preclude or
severely limit the trans-coding by proxies.
 HTML tags mainly provide formatting information rather than semantic
information.
 Trans-coding sometimes-could be difficult to apply to many media types
such as video and audio.
 Developing a general purpose trans-coding engine is very difficult if not
impossible.
Trans-coding through middleware is transparent to both device and content.
Therefore, this trans-coding technique has to be very robust and universal. That is
why this trans-coding technique is the most difficult to engineer. It is most desirable
for content aggregators and value added service providers.
Content Rating and Filtering
W3C has proposed a standard called PICS (Platform for Internet Content
Selection) for rating of web content. Filtering of the content can take place depending
on this rating. PICS specification is a set of technical specifications for labels (meta-
data) that help software and rating services to work together. Rating and labeling
services choose their own criteria for proper identification and filtering of the
content. Since rating will always involve some amount of subjective judgment, it is
left to the service provider to define the ratings. Rating can be through self-labeling
or third party labeling of content. In a third party labeling some independent rating
agency can be used
The W3C is working to extend Web meta-data capabilities generally and is
applying them specifically in the following areas:

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 Digital Signature—coupling the ability to make assertions with a


cryptographic signature block that ensures integrity and authenticity.

 Intellectual Property Rights Management-using a meta-data system to


label Web resources with respect to their authors, owners and rights
management information.

 Privacy (P3)-using a meta-data system to allow sites to make assertions


about their privacy practices and for users to express their preferences
for the type of interaction they want to have with those sites.

 Personalization-based on some policy, the content can be personalized to


suit the need of the user and the service.
Regardless of content control, meta-data systems such as PICS are going to be
an important part of the Web, because they enable more sophisticated commerce
(build and manage trust relationships), communication, indexing, and searching
services. Content filtering can take place either at the client end or at the middleware
proxy end.
Content Aggregation
There are primarily four parties involved; they are end user
(EU), the content provider (CP), the content aggregator (CA), and the ISP (Internet
Service Provider) or the wireless or wire line network operator (NO). The network
operator will have routers, cache, gateways and other nodes to offer the service. In
this scheme anybody can become a requester or a responder. There could be
different parameters, which will determine the content. These parameters are of two
types static and dynamic. The static adaptation parameters are those which can be
received before the service begins. The content is adapted, based on this parameter.
The dynamic adaptation parameters are those which are required with every
request. From the content aggregator's perspective we may classify the service into
two categories:

 Single service request—this works at user level and works for only one user.
For example, a user may request the proxy server at the NO to translate the

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page into Hindi and then deliver the same to the user. In this case, the end
user buys the content and the translation service.

 Group service request—this works for a group of users. This type of request
is initiated either at the CA level or the NO level. For example, the content
aggregator has some arrangement for advertisement. The content aggregator
examines all the HTML pages and inserts an advertisement at an appropriate
place.
Seamless Communication
The seamless communication offers users to roam across different wireless
networks. Roaming generally works within homogeneous networks, like GSM to
GSM or CDMA2000 to CDMA2000. True seamless roaming will include handoff and
roaming in a heterogeneous hybrid network. User will move from a WiFi to 3 G to
wired LAN to GSM while the session is in progress. Users will be able to
communicate using whatever wireless device is currently at hand. Thus, GPRS-
enabled cell phones, PDAs and laptops will be able to roam and communicate freely
and access the Internet across both WLANs and WWANs.
In a seamless roaming, the following aspects need to be maintained and
managed in a seamless fashion without any disruption of service:

 Authentication across network boundaries.


 Authorization across network boundaries.
 Billing and charging data collection.
 End-to-end data security across roaming.
 Handoff between wireless access points.
 Roaming between networks.
 Session migration.
 IP mobility.
Autonomous Computing
The essence of autonomous system is self-management, which is a
combination of the following functions:

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 Self-configurable: An autonomous system will configure itself


automatically in accordance with high-level policies. This will suit the
functional requirement of the user.

 Self-optimizing: An autonomous system will continuously look for ways


to improve its operation with respect to resource, cost and performance.
This will mean that an autonomous system will keep on tuning hundreds
of tunable parameters to suit the user and the environment.

 Self-healing: An autonomous system will heal detect, diagnose and


repair localized problems resulting from bugs or failures. These failures
could be the result of either software or hardware failure.

 Self-protecting: An autonomous system will be self-protecting. This will


be from two aspects. It will defend itself from external attacks; also, it will
not propagate or cascade failure to other parts of the system.

 Self-upgradeable: An autonomous system will be able to grow and


upgrade itself within the control of the above properties.
Context Aware Systems
The role of a Context Manager is to maintain information pertaining to
location, mobile devices, network, users, the environment around each mobile
device and any other context information deemed relevant. Following is a
description of these information and relevance in the mobile computing
environment.

• Location Information: This feature helps us to identify the location of the


user/device. This can be achieved in either of the two ways. One is
through the device and the other is through the network. From the device,
the best way to find the location is through GPS (Global Positioning
Systems). GPS-based systems can offer location information to a precision
of 10 feet radius. Base-station-based location information is likely to be
correct to the precision of 100 feet radius.

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• Device Information: This feature helps us to know the characteristics of


the device. The device information can be obtained from the device and
from the network. Through the User-Agent parameter of http protocol we
can get some information about the device. As this information is
provided by the browser in the device, the information is very generic.
This does not give the device properties like color, pixel capability,
display size etc. From the network side, the information about the device
can be obtained from the EIR (Equipment Identity Register) database of
the network. In all the wireless networks (GSM, GPRS, UMTS, 3G) we
have the EIR. However, we do not have any concept of EIR in wireless
LAN or WiFi.

• Network information: In a mobile computing environment, the user


moves from network to network. Sometime they are even heterogeneous
in nature. Network information is required to identify the capability of the
network. Capability information will include security infrastructure,
services offered by the networks etc.

• User information: This information is required to identify the user


correctly. From the security point of view, the system needs to ensure that
the user is a genuine user who he claims to be.

• Environment information: This includes ambient surrounding


awareness. We need to know the temperature, elevation, moisture, and
other ambient-related information. These information are necessary for
sensor-based networks.

For general mobile-computing environment we need location information,


network information, user information, and the device information.
GPS

Global Positioning System (GPS) is a system that gives us the exact position
on the Earth. GPS is funded by and controlled by the US Department of Defense.
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There are GPS satellites orbiting the Earth, which transmit signals that can be
detected by anyone with a GPS receiver. Using the receiver, we can determine the
location of the receiver. GPS has 3 parts: the space segment, the user segment, and
the control segment.
The space segment consists of 24 satellites, each in its own orbit 11,000
nautical miles above the Earth. The GPS satellites each take 12 hours to orbit the
Earth. Each satellite is equipped with an accurate clock to let it broadcast signals
coupled with a precise time message.
The user segment consists of receivers, which can be in the users' hand,
embedded in a mobile device or mounted in a vehicle. The user segment receives the
satellite signal which travels at the speed of light. Even at this speed, the signal takes
a measurable amount of time to reach the receiver. The difference between the time
the signal is sent and the time it is received, multiplied by the speed of light, enables
the receiver to calculate the distance to the satellite. To measure precise latitude,
longitude and altitude, the receiver measures the time it took for the signals from
four separate satellites to get to the receiver. If we know our exact distance from a
satellite in space, we know we are somewhere on the surface of an imaginary sphere
with radius equal to the distance to the satellite radius. If we know our exact
distance from four satellites, we know precisely where we are on the surface of the
each.

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UNIT - II

2.1. Mobile Computing Through Telephony

2.1.1. Evolution of Telephony


The first telephone system allowed a two-way voice communication between
two individuals in two locations on either side of a wire. Calling or A party (the
person who makes the call) speak into one unit of the phone at one end of the wire
and someone else hears our voice at another location known as called or B party (the
person who responds to the call) at the other end of the wire. During the long era of
analog telephony, the purpose of interconnecting two subscribers was to establish a
physical connection between their respective telephone devices by establishing a
physical circuit between two parties (A party and B party). In early days, each
telephone was connected to a central place (the exchange) and from this exchange
the operator would manually connect the call to another subscriber. The operator
was required to make a note of who placed the call, whom the call was for, and
when it started and ended. This information made it possible to charge the caller for
the call, the classic billing and charging information. If we wanted to make a call to
someone outside our own local exchange, say to the neighboring exchange, an
operator at our exchange would call an operator at the adjacent exchange and then
ask the other operator to connect through to the desired subscriber. If we wanted to
call someone much further away we had to book a trunk call. In the case of a trunk
call, the call would have to be set up with a whole chain of operators, each one
calling the next and so on.

In 1912, the Swedish engineer Gotthief Betulander patented an automatic


switching system based on a grid. This type of exchange was also electromechanical
and was called crossbar exchange. In 1960 the first Electronic Switching System (ESS)
was developed by AT&T and commissioned for testing. Finally, on 30 May 1965, the
first commercial electric central office was put into operation at Succasunna, New
Jersey. In 1976, Bell Labs developed the 4ESS toll switch for the long-distance voice
network. This was the first digital circuit switch. The idea behind a digital switch

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was that the analog voice is digitized before it is given to a switch for switching. The
1960s and 1970s saw the advent of telephone exchanges that were controlled by
processors and software (digital computers). These were called stored program
control exchanges. The primary objective of a sophisticated telephone exchange is
still the same as that of the manual exchange a century ago. These are to detect the
A-subscriber's (calling party) call attempt, connect him to the correct B-subscriber
(called party), and to save data about the call for the purpose of billing.

The digital revolution in telephony started with the introduction of electronic


switches. It was possible to encode an analog signal in digital form if the analog
signal was sampled at twice its frequency. Each sample could then be encoded and
transmitted. There would be enough information in the encoded signal for the
original voice signal to be reconstructed into an understandable analog signal at the
receiving end. Assuming that the audio voice band is 0 to 4000 Hzs, we start with a 4
KHz analog voice channel. Then we take a snapshot of the voice signal's amplitude
at l/8000th of a second (every at double the frequency of 4 KHz). Then we convert
the measured amplitude to a number (the quantization process) that is represented
by 8 bits. This type of digitization is called Pulse Code Modulation (PCM). Thus,
PCM requires 64 Kb/s of digital bandwidth (8 KHz * 8 bits). Bell Laboratory
scientists first introduced digital transmission using PCM in 1962. The Bell
Laboratory system was named Tl with a transmission rate of 1.544 Mb/s carrying 24
channels of 64 Kb/s each. In Europe, a similar system was called El, where it had a
bandwidth of 2 Mb/s and carried 32 channels of 64 Kb/s. The developments in
transmission techniques have been advancing largely to reduce network costs. We
have witnessed an evolution from systems employing open-wire lines to
multiplexed, analog systems using coaxial or radio links, on to digital fiber-optic
systems with a capacity of tens of Gbit/s per fiber pair.
In a manual switching system, an operator would be able to inform the caller
of the current status of the call. In manual exchanges, the operator's intelligence was
a control system separate from the switching mechanism. An operator, alerted to an
incoming call:
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• Listens to and remembers the desired number.

• Finds the right way to connect the caller's line to the line being called.

• Checks if the desired line is free.

• Makes the connection.

• Notes down the call details: time of call, duration of call, calling number
and called number.
Having removed the need for an operator in the automated exchange, a
system was necessary to indicate the progress of the call to the caller. A series of
distinct tones were generated by a machine called Ring Generator. The tones
produced were as follows:

• Dial Tone (DT). This is a signal applied to the line after the calling party
(A party) has lifted his handset and the switching equipment has allocated
him an available outlet (a circuit) for this call to proceed.

• Busy Tone (BT). Busy tone indicated either that the called subscriber (B
party) is already off-hook (busy) or that the route to the called subscriber
is congested.

• Ring Tone (RT). When a circuit between A party and the B party is
established, the telephone rings at B party's end and a ring tome is
generated for the A party.
A normal telephone system is called Public Switched Telephone Network
(PSTN). PSTN nodes can be subdivided into three main categories:
 Local exchanges (also known as End Office) - used for the connection of
subscribers.
 Transit exchanges (also known as Local Access Tandem) - switch traffic
within and between different geographical areas.
 International exchanges (also known as Interexchange Carrier) and
other gateway-type exchanges - switch traffic to telecommunication
networks in foreign countries and other networks.
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A physical wire (also known as local loop) is laid from the local exchange to
the telephone device at each subscriber's place. In case of a wireless network like
GSM or WILL (Wireless in Local Loop), there is no wire from the local exchange to
the telephone. The communication between the local exchange and the telephone
device is managed over the wireless radio interface. In India, there are network
operators who are offering basic or fixed telephone, WiLL and GSM.

2.1.2. Multiple Access Procedure


In a PSTN network, a separate physical wire is used to connect the
subscriber's telephone with the switch. Therefore, multiple users can have speech
communication at the same time without causing any interference to each other. The
scene is different in the case of wireless communication. Radio channel, used in a
wireless network, is shared by multiple subscribers. Unless we control simultaneous
access of the radio channel (by multiple users), collisions can occur. In a connection-
oriented communication, collision is undesirable. Therefore, every mobile subscriber
must be assigned a dedicated communication channel on demand. This is achieved
by using different multiplexing techniques.

Frequency Division Multiple Access


Frequency Division Multiple Access (FDMA) is one of the most common
multiplexing procedures. The available frequency band is divided into channels of
equal bandwidth so that each communication is carried on a different frequency.
This multiplexing technique is used in all the first generation analog mobile
networks like Advanced Mobile Phone System (AMPS) in the USA and Total Access
Communication System (TACS) in the UK.

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Time Division Multiple Access


Time Division Multiple Access (TDMA) is a more expensive technique
compared to FDMA as it needs precise synchronization between the transmitter and
the-receiver.
The TDMA technique is used in digital mobile communication. In a TDMA
system, the whole frequency bandwidth is subdivided into sub-bands using FDMA
techniques. TDMA technique is then used in each of these sub-bands to offer
multiple accesses. GSM uses such a combination of FDMA and TDMA. A frequency
range of 25 MHz holds 124 single channels of 200 KHz each. Each of these frequency
channels contains 8 TDMA conversation channel.
Code Division Multiple Access
Code Division Multiple Access (CDMA) is a broadband system. CDMA uses
spread spectrum technique where each subscriber uses the whole system
bandwidth. Unlike the FDMA or TDMA where a frequency or time slot is assigned
exclusively to a subscriber, in CDMA all subscribers in a cell use the same frequency
band simultaneously. To separate the signals, each subscriber is assigned an
orthogonal code called 'chip'.
Space Division Multiple Access
Along with TDMA, FDMA, and CDMA, we need to make use of the space
effectively. Space division multiple access (SDMA) is a technique where we use
different part of the space for multiplexing. SDMA is used in radio transmission and
is more useful in satellite communications to optimize the use of radio spectrum by
using directional properties of antennas. In SDMA, antennas are highly directional,
allowing duplicate frequencies to be used at the same time for multiple surface
zones on earth. SDMA requires careful choice of zones for each transmitter, and also
requires precise antenna alignment.

2.1.3. Mobile Computing Through Telephone


One of the early examples of mobile computing was accessing applications
and services through voice interface. This technology was generally referred to as
Computer Telephony Interface (CTI). Different banks around the world were

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offering telephone banking for quite some time using this technology. In a telephone
banking application, the user calls a number and then does his banking transaction
through a fixed telephone. In this application the telephone does many functions of
a bank teller. Input to this system is a telephone keyboard and output is a
synthesized voice. These applications can be used from anywhere in the world. The
only issue in this case is the cost of a call
To make this type of mobile computing work through voice interfaces, we use
Interactive Voice Response (IVR). In USA and Japan IVRs are commonly known as
Voice Response Unit (VRU). The technical name for this technology is CT (Computer
Telephony) or CTI (Computer Telephony Interface or Computer Telephony
Integration). IVR software can be hosted on a Windows-NT, Linux, or other
computers with the voice cards. There are many companies who manufacture voice
cards; however, one of the most popular card vendors is from Intel/Dialogic. IVR
works as the gateway between a voice-based telephone system and a computer
system. Multiple telephone lines are connected to the voice card through appropriate
telecom interfaces (El or an analog telephone extension). When a caller dials the IVR
number, a ring tone is received by the voice card within the IVR. The voice card
answers the call and establishes a connection between the caller and the IVR
application. The caller uses the telephone keyboard to input data. The switch can be
either a PSTN exchange or a local PBX in the office. For PSTN switch, the voice card
will have El interface whereas for a PBX, the voice card will have analog interface.
The IVR will have all the gateway-related functions. The server will host the
business application.

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A telephone keyboard has 12 keys (viz., 1, 2, 3, 4, 5, 6, 7, 8, 9, 0, *, and #). The


English alphabetic characters are also mapped on these 12 keys. They are mapped as
follows.

1. Alphabet A, B, C on key 2

2. Alphabet D, E, F on key 3
3. Alphabet G, H, I on key 4
4. Alphabet J, K, L on key 5
5. Alphabet M, N, O on key 6
6. Alphabet P, Q, R, S on key 7
7. Alphabet T, U, V on key 8
8. Alphabet W, X, Y, Z on key 9

It is possible to enter alphabetic data through the telephone keyboard by


pressing a key in multiple successions. For example DELHI will be entered as 3-3
(D), 3-3-3 (E), 5-5-5-5 (L), 4-4-4 (H), 4-4-4-4 (I). These key inputs are received by the
voice card as DTMF (Dual Tone Multi Frequency) inputs generated through
combination of frequencies. Following is the table of these frequencies:
1209 Hz 1336 Hz 1477 Hz
697 Hz 1 2/ABC 3/DEF

770 Hz 4/GHI 5/JKL 6/MNO


852 Hz 7/PQRS 8/TUV 9/WXYZ

941 Hz * 0 #

If we press key 1, it will generate a frequency 697 + 1209 Hz. Likewise 0 will
be 941+1336 Hz. These DTMF signals are different audio frequencies interpreted by
the voice card and passed to the IVR program as numbers through appropriate APIs.
For example the user presses '2' three times. The voice card will receive 697+1336
Hz-697+1336 Hz-697+1336 Hz. This will be interpreted by a program as 2-2-2.
Looking at the time interval between the numbers, the program can decide whether

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the user entered '222' or 'B'. When the application needs to send an output to the
user, the standard data is converted into voice either through synthesizing voice files
or through TTS (Text To Speech). In a cheque-printing software we print the amount
in both words and figure. For example an amount of 'Rs. 320,145.00' will be printed
on a cheque as 'Rupees three lacs twenty thousand one hundred forty-five only'.
Within the cheque-printing application, one function converted the numeric number
320145 into text. Likewise in the case of IVR application, we assemble a series of
prerecorded voice prompts to generate the equivalent sound response. In this case
we assemble voice data 'three' 'lacs' 'twenty' 'thousand' 'one' 'hundred' 'forty' 'five'
'only' and then give the voice card to play. We can generate the same voice response
by giving the number 320,145 to the TTS interface to convert the text into speech and
play through the IVR. TTS is a interface software which takes text and numbers as
input and generates equivalent sounds at the run time. There are different TTS
available for different languages. In India there are companies who have Hindi TTS
software. Hindi TTS software takes Devanagari text stream as input and generates
the voice as if someone is reading the same text.

Overview of the Voice Software


Voice technology encompasses the processing and manipulation of an audio
signal in a Computer Telephony (CT) system. It supports filtering, analyzing,
recording, digitizing, compressing, storing, expanding and replaying of audio voice
and also includes the ability to receive, recognize and generate specific telephone
and network tones. This fundamental technology is at the core of most IVR systems.
Voice products also offer Digital Signal Processing (DSP) technology and signal
processing algorithms, for building the core of any converged communications
system. Most of the voice cards come with industry-standard Peripheral Component
Interface (PCI) bus expansion boards. The PCI interface makes it possible to
integrate these voice products into Windows or Linux systems quite easily.

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One of the most popular voice cards used for small office interface is
D/41JCT-LS from Dialogic. The D/41JCT-LS board is a four-port analog converged
communications voice, fax, and software-based speech recognition board. This
board is ideal for building enterprise unified messaging and interactive voice
response (IVR) applications. The D/41JCT-LS provides four telephone line interface
circuits for direct connection to analog loop start lines through RJ11 (the standard
telephone jack used in homes) interface. D/41JCT-LS possesses dual-processor
architecture, comprising a digital signal processor (DSP) and a general-purpose
microprocessor, which handles all telephony signaling and performs DTMF
(touchtone) and audio/voice signal processing tasks. A voice card also has some on-
board memory and with voice store-and-forward feature.
Voice Driver and API
Dialogic is now part of Intel and one of the leading vendors on voice-based
hardware. Many IVR vendors around the world use Dialogic cards from Intel in
their IVR systems. Voice driver in an IVR system is used to communicate and control
the voice hardware on the IVR system. A voice driver can make calls, answer calls,
identify caller id, play and record sound from the phone line, detect DTMF signals
(touch-tones) dialed by the caller. It can tear down a call, detect when the caller has
hung up. It also offers APIs to record the transaction details. Transaction information
is required for audit trail and for charging. Voice boards are treated as board
devices, channels within a board are treated as channel devices or board sub-
devices by the voice driver.

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IVR Programming
There are different voice libraries provided by Dialogic to interface with the
voice driver. The voice libraries for single-threaded and multi-threaded applications
include:

• libdxxmt.lib—the main Voice Library

• libsrlmtlib-the Standard Run-time Library


These C function libraries can be used to:

• Utilize all the voice board features of call management

• Write applications using a Single-threaded Asynchronous or Multi-


threaded paradigm

• Configure devices

• Handle events that occur on the devices

• Return device information

• Gather call transaction details.


The Standard Run-time Library provides a set of common system functions
that are device independent and are applicable to all Dialogic devices.
Single-threaded Asynchronous Programming Model
Single-threaded asynchronous programming enables a single program to
control multiple voice channels within a single thread. This allows the development
of complex applications where multiple tasks must be coordinated simultaneously.
The synchronous programming model supports both polled and callback event
management.
Multi-threaded Synchronous Programming Model
The multi-threaded synchronous programming model uses functions that
block application execution until the function completes. This model requires that
the application control each channel from a separate thread or process. The
operating system can put individual device threads to sleep while allowing threads
that control other Dialogic devices to continue their actions unabated. When a

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Dialogic function is completed, the operating system wakes up the function's thread
so that processing continues. This model enables the IVR system to assign distinct
applications to different channels dynamically in real time.
Voice APIs
To use the voice board, Dialogic provides different APIs. All Dialogic APIs are
prefixed with dx_; this helps to identify them easily. APIs are available for device
management, configuration function, input output functions, play and record
functions, tone detection functions, tone generation functions, call control functions
etc. Following are some of the important functions in Dialogic voice card, which are
used quite often to develop a mobile computing application.
- dx_open( ) - open a voice channel
- dx_close( ) - close a voice channel
- dx_wtcallid( ) - waits for rings and reports Caller ID
- dx_getdig( ) - get digits from channel digit buffer
- dx_play( ) - plays recorded voice data
- dx_jplayvox () - play a single vox file
- dx playwav() - play a single wave file
- dx_rec( ) - record voice data
- dx_recvox( ) - records voice data to a single vox file
- dx_recwav( ) - records voice data to a single wave file
- dx_dial( ) - dial an ASCII string of digits.

2.1.4. Voice XML


In mobile computing through telephone, the IVR is connected to the server
through client/server architecture. It is also possible to host the IVR and the
application on the same system. Today Internet (http) is used in addition to
client/server interface between the IVR and the server. This increases the flexibility
in the whole mobile computing architecture. Http is used for voice portals as well. In
the case of a voice portal, a user uses an Internet site through voice interface. For all
these advanced features, VoiceXML has been introduced. Recent IVRs are equipped

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with DSP (Digital Signal Processing) and are capable of recognizing voice. The
output is synthesized voice through TTS (Text to Speech).
The Voice extensible Markup Language (VoiceXML) is an XML-based
markup language for creating distributed voice applications. VoiceXML is designed
for creating audio dialogs that feature synthesized speech, digitized audio,
recognition of spoken voice and DTMF key input. Using VoiceXML, we can create
Web-based voice applications that users can access through telephone. VoiceXML
supports dialogs that feature :

• Spoken input

• DTMF (telephone key) input

• Recording of spoken input

• Synthesized speech output ('text-to-speech')

• Recorded audio output

• Dialog flow control

• Scoping of input
Architectural Model

The architectural model for VoiceXML has the following components:

A Document Server (e.g. a web server) services requests from a client


application. The client side of the application runs on a VoiceXML Interpreter, and
is accessed through the VoiceXML interpreter context. The server delivers
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VoiceXML documents, which are processed by the VoiceXML Interpreter. The


VoiceXML Interpreter Context is responsible for special actions on voice escape
phrases.

For instance, in an interactive voice response application, the VoiceXML


interpreter context may be responsible for detecting an incoming call, acquiring the
initial VoiceXML document, and answering the call, while the VoiceXML interpreter
manages the dialog after answer. The implementation platform generates events in
response to user actions (e.g. spoken or character input received, disconnect) and
system events (e.g. timer expiration).
How Voice XML Fits into Web Environment?

All of us are familiar with the web as it works today. We use a visual GUI web
browser which renders and interprets http requests to present information to the
user. When the user makes a selection the web browser sends an HTTP request to
the web server. The web server responds by locating the new page and returns the
page to the user. The content server may also have to interact with a back-end
infrastructure (database, servlets, etc.) to obtain and return the requested
information.

The Voice Browser extends this paradigm. A telephone and a Voice Server
have been added to the web environment. The Voice Server manages several Voice
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Browser sessions. Each Voice Browser session includes one instance of the Voice
Browser, the speech recognition engine, and the text-to-speech engine.

VoiceXML introduces a new way of presenting the web information. Instead


of presenting the information visually (through HTML, graphics and text), the Voice
Browser presents the information to the caller in audio using VoiceXML. When the
caller says something the Voice Browser sends an HTTP request to the web server,
which accesses the same back-end infrastructure, to return information this time in
audio. This type of portal is known as voice portal. Voice portal is very useful in a
hands free situation while driving.
The Voice Browser

An audio Voice Browser is similar to a visual web browser like Netscape


Communicator or Microsoft Internet Explorer. Through voice browser, we interact
with a web server using our voice and a telephone. Instead of clicking a mouse and
using keyboard, we use our voice and a telephone (and even the phone keypad) to
access web information and services.
Dialogs

A VoiceXML application defines a series of dialogs between a user and a


computer. There are two types of dialogs that can be implemented in VoiceXML:

• Forms - an interaction that collects values for a set of fields.

• Menus - present the user with choices or options and then transition to
another dialog based on the choice.
Essential Elements of Voice XML Documents
The first line of any VoiceXML application must contain the <?xml> element.
The second line must contain the <vxml> element. And each VoiceXML <tag>, must
have an associated <Jtag>. The very last line of VoiceXML document must be the
</vxml> tag. So, a VoiceXML document looks like this:

<?xml version="l.0"?>
<vxml versions"1.0">

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.
.
.
Interesting stuff goes here
.
.
.
</vxml>

In between the <vxml> and the </vxml> tags we put all the really interesting
stuff, the VoiceXML code that defines the dialog with the user.
Prompts
In a VoiceXML application, we present information to the user through audio
prompts. These prompts can either be prerecorded audio, or they can be synthesized
speech (TTS). We use the <prompt> element in VoiceXML to generate TTS. Any text
within the body of a <prompt> element is spoken.

<prompt>
Would you like coffee, tea, milk or nothing
</prompt>
Grammars
Each dialog has one or more speech and/or DTMF grammars associated with
it. In VoiceXML, we use the <grammar> element to define what the caller can say to
the application at any given time. There are three different types of grammars
supported in VoiceXML: Inline, External, Built-in

o Inline grammars are those that are defined right in the VoiceXML
code. For example
<grammar>
credit card | credit | tuition | tuition bill
</grammar>

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In an inline grammar, the words and phrases that a caller is allowed to


say are defined within the body of the <grammar> element. Each
word or phrase is separated by a vertical bar ("|") symbol. This symbol
essentially means "or." So, the caller can say either "credit card" or
"credit" or "tuition" or "tuition bill".

o External grammars are those that are specified outside of the


VoiceXML document in another file and are referenced from within the
VoiceXML code. We use the <grammar> element to specify an external
grammar, too. For example:

<grammar>src="names.gram" type="application/x-j sgf"</grammar>


In this example, the grammar is defined in 'names.gram' file.
Form

Form is one of the ways of developing a dialog with the caller in VoiceXML.
Forms are central to VoiceXML. A VoiceXML form is a process to present
information and gather input from the caller. A form is, basically, a collection of one
or more fields that the caller fills in by saying something. A VoiceXML form is a very
similar concept to a paper or online form, except that in the case of VoiceXML, we
cannot see the field and instead of typing or writing in a field, we say something to
fill it in.
In VoiceXML, we define a form using the <form> element and fields within
the form using the <field> element. Here is a simple Voice Form.

<?xml version="l.0" ?>


<vxml version="l.0">
<form id="add_funds">

<field name="amount" type="currency">


<prompt>How much?</prompt>

</field>

<field>

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<prompt>Charge to credit card or tuition bill?</prompt>


<grammar>
credit card | credit | tuition | tuition bi11
</grammar>
</field>
</form>
</vxml>
Links
If user input matches the link's grammar, control transfers to the link's
destination URL A <link> can be used to throw an event to go to a destination URL
VoiceXML Elements
Element Purpose Page
<assign> Assign a variable a value.

<audio> Play an audio clip within a prompt.


<block> A container of (non-interactive) executable code.
<break> JSML element to insert a pause in output.

<catch> Catch an event.


<choice> Define a menu item.
<clear> Clear one or more form item variables.
<disconnect> Disconnect a session.
<div> JSML element to classify a region of text as a particular

<dtmf > type


Specify a touch-tone key grammar.
<else> Used in < if > elements.
<elseif> Used in < if > elements.
<emp> JSML element to change the emphasis of speech output.

<enumerate> Shorthand for enumerating the choices in a menu.


<error> Catch an error event.
<exit> Exit a session.
<field> Declares an input field in a form.

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<filled> An action executed when fields are filled.


<form> A dialog for presenting information and collecting data.
<goto> Go to another dialog in the same or different document.
<grammar> Specify a speech recognition grammar.
<help> Catch a help event.
<if> Simple conditional logic.
<initial> Declares initial logic upon entry into a (mixed-initiative)
<link> Specify a transition common to all dialogs in the link's
<menu> A dialog for choosing amongst alternative destinations.
<meta> Define a meta data item as a name/value pair.
<noinput> Catch a noinput event.
<nomatch> Catch a nomatch event.
<obj ect> Interact with a custom extension.
<option> Specify an option in a <field>.
<param> Parameter in <object> or <subdialog>.
<prompt> Queue TTS and audio output to the user.
<property> Control implementation platform settings.
<pros> JSML element to change the prosody of speech output.
<record> Record an audio sample.
<reprompt> Play a field prompt when a field is re-visited after an
<return> Return from a subdialog.
<sayas> JSML element to modify how a word or phrase is spoken.
<script> Specify a block of ECMAScript client-side scripting logic.
<subdialog> Invoke another dialog as a subdialog of the current.
<submit> Submit values to a document server.
<throw> Throw an event.
<transfer> Transfer the caller to another destination.
<value> Insert the value of a expression in a prompt.
<var> Declare a variable.
<vxml> Top-level element in each VoiceXML document.

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2.1.5. TAPI(Telephony Application Programming Interface)


TAPI (Telephony Application Programming Interface) and Speech
Application Programming Interface (SAPI) are two standards that can be used when
developing voice telephony applications. Using TAPI, programmers can take
advantage of different telephone systems, including ordinary PSTN, ISDN, and PBX
(Private Branch Exchange) without having to understand all their details. Use of
these API will save the programmer the pain of trying to program hardware
directly. Through TAPI and SAPI a program can "talk" over telephones or video
phones to people or phone-connected resources. Through TAPI one will be able to:

• Simple user interfaces to setup calls. This can be calling someone by


clicking on their picture or other images

• Use simple graphical interface to set up a conference call and then attend
the call at the scheduled time

• See who you're talking to

• Attach voice greeting with an email. This will allow the receiver to listen
to this greeting while opening the email

• Set groups and security measures such that a service can receive phone
calls from certain numbers (but not from others)

• Send and receive faxes

• Same set of TAPI APIs are available in many smart phones. This facilitates
accessing telephony interfaces from a mobile phone along with from a
desktop computer
2.2. Emerging Technologies

2.2.1. Bluetooth
The concept behind Bluetooth wireless technology was unifying the telecom
and computing industries. Bluetooth technology allows users to make
ad hoc wireless connections between devices like mobile phones, desktop or
notebook computers without any cable. Devices carrying Bluetooth-enabled chips

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can easily transfer data at a speed of about 720 Kbps within 50 meters (150 feet) of
range or beyond through walls, clothing and even luggage bags.

Bluetooth Protocol
The Bluetooth radio is built into a small microchip and operates in a globally
available frequency band ensuring interoperability worldwide. Bluetooth uses the
unlicensed 2.4 GHz ISM (Industrial Scientific and Medical) frequency band. There
are 79 available Bluetooth channels spaced 1 MHz apart from 2.402 GHz to 2.480
GHz. Bluetooth supports both unicast (point-to-point) and multicast (point-to-multi-
point) connections. Bluetooth protocol uses the concept of master and slave. In a
master-slave protocol a device cannot talk as and when they desire. They need to
wait till the time the master allows them to talk. The master and slaves together form
a piconet. Up to seven 'slave' devices can be set to communicate with a 'master'.
Several of these piconets can be linked together to form a larger network in an ad
hoc manner. The topology can be thought as a flexible, multiple piconet structure.
This network of piconets is called scatternet. A scatternet is formed when a device
from one piconet also acts as a member of another piconet. In this scheme, a device
being master in one piconet can simultaneously be a slave in the other one.
Bluetooth protocol is a combination of different protocols. The Bluetooth Core
protocols plus the Bluetooth radio protocols are required by most of Bluetooth
devices, while the rest of the protocols are used by different applications as needed

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Bluetooth Protocol Stack

Bluetooth protocol stack can be divided into four basic layers according to
their functions. These are:

• Bluetooth Core Protocols this comprises of Baseband, Link Manager Protocol


(LMP), Logical Link Control and Adaptation Protocol (L2CAP), and Service
Discovery Protocol (SDP).

o Baseband The Baseband and Link Control layer enables the physical
RF link between Bluetooth units forming a piconet. All audio and data
packets can be provided with different levels of CRC
(Cyclic Redundancy Code) or FEC (Forward Error Correction) for
error detection/ correction.
o Link Manager Protocol (LMP) When two Bluetooth devices come
within
each other's radio range, link managers of either device discover each
other. LMP then engages itself in peer-to-peer message exchange.
These messages perform various security functions starting from
authentication to encryption.

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o Logical Link Control and Adaptation Protocol (L2CAP) this layer is


responsible for segmentation of large packets and the reassembly of
fragmented packets. L2CAP is also responsible for multiplexing of
Bluetooth packets from different applications.

o Service Discovery Protocol (SDP) The Service Discovery Protocol


(SDP)
enables a Bluetooth device to join a piconet.

• Cable Replacement Protocol this protocol stack has only one member viz.,
Radio Frequency Communication (RFCOMM).

o RFCOMM this is a serial line communication protocol and is based on


ETSI 07.10 specification. The "cable replacement" protocol emulates RS-
232 control and data signals over Bluetooth baseband protocol.

• Telephony Control Protocol this comprises of two protocol stacks viz.,


Telephony Control Specification Binary (TCS BIN), and the AT-Commands.

o Telephony Control protocol Binary TCS Binary or TCS BIN is a bit


oriented protocol. TCS BIN defines the call control signaling protocol
for setup of speech and data calls between Bluetooth devices. It also
defines mobility management procedures for handling groups of
Bluetooth TCS devices.

o AT-Commands this protocol defines a set of AT-commands by which a


mobile phone can be used and controlled as a modem for fax and data
transfers.

• Adopted Protocols This has many protocol stacks like Point-to-Point


Protocol (PPP), TCP/IP Protocol, OBEX (Object Exchange Protocol), Wireless
Application Protocol (WAP), vCard, vCalendar, Infrared Mobile
Communication (IrMC), etc.

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o PPP Bluetooth offers PPP over RFCOMM to accomplish point-to-


point connections. Point-to-Point Protocol is the means of taking IP
packets to/from the PPP layer and placing them onto the LAN.

o TCP/IP Protocol is used for communication across the Internet.


TCP/IP stacks are used in numerous devices including printers,
handheld computers, and mobile handsets.

o OBEX Protocol OBEX is a session protocol developed by the Infrared


Data Association (IrDA) to exchange objects. OBEX, provides the
functionality of HTTP in a much lighter fashion.

o Content Formats vCard and vCalendar specifications define the


format of an electronic business card and personal calendar entries
developed by the Versit consortium, These are now maintained by
the Internet Mail Consortium. Other content formats, supported by
OBEX, are vMessage and vNote. These content formats are used to
exchange messages and notes. They are defined in the IrMC (IrDA
Mobile Communication) specification. IrMC also defines a format for
synchronization of data between devices.
Bluetooth Security
In a wireless environment where every bit is on the air, security concerns are
high. Bluetooth offers security infrastructure starting from authentication, key
exchange, to encryption.
Bluetooth Application Models
Each application model in Bluetooth is realized through a Profile. Profiles
define the protocols and protocol features supporting a particular usage model.

• File Transfer The file transfer usage model offers the ability to transfer data
objects from one device (e.g., PC, smart-phone, or PDA) to another. Object
types include .xls, .ppt, .wav, .jpg,.doc files, folders or directories or
streaming media formats.

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• Internet Bridge In this usage model, mobile phone or cordless modem acts
as modem to the PC, providing dial-up networking and fax capabilities
without need for physical connection to the PC.

• LAN Access In this usage model multiple data terminals use a LAN access
point (LAP) as a wireless connection to an Ethernet LAN. Once connected,
the terminals operate as if they were connected directly to the LAN.

• Synchronization The synchronization usage model provides a device-to-


device (phone, PDA, computer, etc.) synchronization of data. Examples
could be PIM (personal information management) information, typically
phonebook, calendar, message, and note information.

• Headset The headset can be wirelessly connected for the purpose of acting
as a remote device's audio input and output interface. This is very
convenient for hands free cellular phone usage in automobiles.

2.2.2. RFID (Radio Frequency Identification)


RFID is a radio transponder carrying an ID (Identification) that can be read
through radio frequency (RF) interfaces. These transponders are commonly known
as RFID tags or simply tags. To assign an identity to an object, a tag is attached to the
object. Data within the tag provides identification for the object. Biologists had been
using RFID for some time to track animals for the purpose of studying animal
behavior and conservation. The earliest use of RFID was for tracking farm animals.
A RFID system comprises of different functional areas like:

• Means of reading or interrogating the data in the tag

• Mechanism to filter some of the data

• Means to communicate the data in the tag with a host computer

• Means for updating or entering customized data into the tag.

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RFID tags are categorized on three basic criteria. These are based on
frequency, application area and the power level.

• On Frequency: There are 6 basic frequencies on which RFID operates.


These are 132.4 KHz, 13.56 MHz, 433 MHz, 918 MHz, 2.4 GHz and 5.8
GHz

• On Application: RFIDs are also grouped according to application and


usage.

• Power-based grouping: RFIDs can be grouped into two types based on


power requirements. These are active and passive tags. Passive tags are
generally in low frequency range, whereas tags at higher frequency range
can be either active or passive.
 Active RFID tags: Active tags are powered by an internal battery and are
typically read/write. The life of an active tag is limited by the life of the
battery. The data within an active tag can be rewritten or modified.
 Passive RFID tags: Passive tags operate without a power source of its
own. A passive tag obtains operating power from the reader's antenna.
The data within a passive tag is read only and generally cannot be
changed during operation.
A basic RFID systemconsist of three components:

 A transponder programmed with unique information (RFID tag)


 A transceiver with decoder (a reader)
 An antenna or coil.

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Areas of Application for RFID

Principal areas of application for RFID that can be currentiy identified


include:

• Transportation and Logistics

• Manufacturing and Processing

• Security

• Animal tagging

• Store in an enterprise

• Retail store

• Community library

• Time and attendance

• Postal tracking

• Airline baggage reconciliation

• Road toll management.

2.2.3. Mobile IP
A data connection between two end-points through TCP/IP network requires
a source IP address, source TCP port and a target IP address with a target TCP port.
TCP port number is application-specific and remains constant. IP address, on the
other hand, is network specific and varies from network to network. IP addresses
are assigned to a host from a set, of addresses assigned to a network. This structure
works well as long as the client is static and is using a desktop computer. Let us
assume that the user is mobile and is using a laptop with WiFi. As the user moves,
the point of attachment will change from one subnet to another subnet resulting in a
change of IP address. This will force the connection to terminate. The technology to
do so is 'Mobile IP'. The term 'mobile' in 'Mobile IP' signifies that, while a user is
connected to applications across the Internet and the user's point of attachment
changes dynamically, all connections are maintained despite the change in
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underlying network properties. This is similar to the handoff/roaming situation in


cellular network.

How does Mobile IP work?


Internet Protocol routes packets from a source endpoint to a destination
endpoint through various routers. An IP address of a host can be considered to be a
combination of network address (most significant 24 bits) and the node address
(least significant 8 bits). Let us assume a 'C class IP address 203.197.175.123 to be of
the mail server of iiitb. We can assume that the first 24 bits 203.197.175 is the address
of the network and the last 8 bits containing 123 is the address of the host. The
network portion of an IP address is used by routers to deliver the packet to the last
router in the chain to which the target computer is attached. This last router then
uses the host portion (123 in this example) of the IP address to deliver the IP packet
to the destination computer.
The mobile IP allows the mobile node to use two IP addresses. These IP
addresses are called home address and care-of address. The home address is static
and known to everybody as the identity of the host. The care-of address changes at
each new point of attachment and can be thought of as the mobile node's location
specific address. This is similar to the concept of HLR (Home Location Register) and
VLR (Visitor Location Register) in cellular networks. When the mobile node is
roaming and is attached to a foreign network, the home agent receives all the
packets for the mobile node and arranges to forward them to the mobile node's
current point of attachment. The network node that is responsible for forwarding
and managing this transparency is known as the home agent.

Whenever the mobile node moves, it registers its new care-of address with its
home agent. The home agent forwards the packet to the foreign network using the
care-of address. The delivery requires that the packet header is modified so that the
care-of address becomes the destination IP address. This new header encapsulates
the original packet, causing the mobile node's home address to have no impact on
the encapsulated packet's routing. This phenomenon is called tunneling.

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Let us take an example of IP data grams being exchanged over a TCP


connection between the mobile node (A) and another host (server X), the following
steps occur:

 Server X wants to transmit an IP datagram to node A. The home address


of A is advertised and known to X. X does not know whether A is in the
home network or somewhere else. Therefore, X sends the packet to A with
A's home address as the destination IP address in the IP header. The IP
datagram is routed to A's home network.

• At the A's home network, the incoming IP datagram is intercepted by the


home agent. The home agent discovers that A is in a foreign network. A
care-of-address has been allocated to A by this foreign network and
available with the home agent. The home agent encapsulates the entire
datagram inside a new IP datagram, with A's care-of address in the IP
header. This new datagram with the care-of-address as the destination
address is retransmitted by the home agent.

• At the foreign network, the incoming IP datagram is intercepted by the


foreign agent. The foreign agent is the counterpart of the home agent in

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the foreign network. The foreign agent strips off the outer IP header, and
delivers the original datagram to A.

• A intends to respond to this message and sends traffic to X. In this


example, X is not mobile; therefore X has a fixed IP address. For routing
A's IP datagram to X, each datagram is sent to some router in the foreign
network. Typically, this router is the foreign agent. A uses X's IP static
address as the destination address in the IP header.

• The IP datagram from A to X travels directly across the network, using X's
IP address as the destination address.
To support the operations illustrated in the example above, mobile IP needs to
support three basic capabilities:

• Discovery: A mobile node uses a discovery procedure to identify


prospective home agents and foreign agents.

• Registration: A mobile node uses a registration procedure to inform its


home agent of its care-of address.

• Tunneling: Tunneling procedure is used to forward IP data grams from a


home address to a care-of address.
Discovery

The Mobile IP discovery procedure has been built on top of an existing ICMP
router discovery and advertisement procedure as specified in RFC 1256. Using these
procedures a router can detect whether a new mobile node has entered into its
network. Also, using this procedure the mobile node determines whether it is in a
foreign network. For the purpose of discovery, a router or an agent periodically
issues a router advertisement ICMP message. The mobile node on receiving this
advertisement packet compares the network portion of the router IP address with
the network portion of its own IP address allocated by the home network. If these
network portions do not match, then the mobile node knows that it is in a foreign
network. A router advertisement can carry information about default routers and
information about one or more care-of addresses.
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Registration
Once a mobile node obtained a care-of-address from the foreign network, the
same needs to be registered with the home agent. The mobile node sends a
registration request to the home agent with the care-of address information. When
the home agent receives this request, it updates its routing table and sends a
registration reply back to the mobile node.
The registration process involves the following four steps:

 The mobile node requests for forwarding service from the foreign network
by sending a registration request to the foreign agent.

 The foreign agent relays this registration request to the home agent of that
mobile node.

 The home agent either accepts or rejects the request and sends a
registration reply to the foreign agent.

 The foreign agent relays this reply to the mobile node.


Tunneling

In the mobile IP, IP-within-IP encapsulation mechanism is used. Using IP-


within-IP, the home agent, adds a new IP header called tunnel header.

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Cellular IP

A change of access point while connectivity is maintained is called a handoff.


The mobile IP architecture comprises three functions:
 A database that contains the most up-to-date mapping between the two
address spaces (home address to care-of-address)
 The translation of the host identifier to the actual destination address
 Agents ensuring that the source and destination packets for arriving and
outgoing packets are updated properly so that routing of packets are proper
Whenever the mobile host moves to a new subnet managed by a different
foreign agent, the dynamic care-of-address will change. This changed care-of-
address needs to be communicated to the home agent. This process works for slowly
moving hosts. For a high speed mobile host, the rate of update of the addresses
needs to match the rate of change of addresses. Otherwise, packets will be

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forwarded to the wrong (old) address. Mobile IP fails to update the addressed
properly for high speed mobility. Cellular IP a new host mobility protocol has been
designed to address this issue.

In a Cellular IP, none of the nodes know the exact location of a mobile host.
Packets addressed to a mobile host are routed to its current base station on a hop-by-
hop basis where each node only needs to know on which of its outgoing ports to
forward packets. This limited routing information (referred as mapping) is local to
the node and does not assume that nodes have any knowledge of the wireless
network topology. Mappings are created and updated based on the packets
transmitted by mobile hosts.
Cellular IP uses two parallel structures of mappings through Paging Caches
(PC) and Routing Caches (RC). PCs maintain mappings for stationary and idle (not
in data communication state) hosts; whereas, RC maintains mappings for mobile
hosts. Mapping entries in PC have a large timeout interval, in the order of seconds
or minutes. RCs maintain mappings for mobile hosts currently receiving data or
expecting to receive data. For RC mappings, the timeout are in the packet time scale.

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While idle at location 1, the mobile host X keeps PCs up-to-date by


transmitting dummy packets at a low frequency. Let us assume that the host is
mobile and moved to location 2 without transacting any data. The PC mapping for X
now points to location 2. While at location 2, there are data packets to be routed to
the mobile host X, the PC mappings are used to find the host. As there is data
transmission, the mapping database to be used will be the RC. As long as data
packets keep arriving, the host maintains RC mappings, either by its outgoing data
packets or through the transmission of dummy packets.

Idle mobile hosts periodically generate short control packets, called paging-
update packets. These are sent to the nearest available base station. The paging-
update packets travel in the access network from the base station toward the
gateway router, on a hop-by-hop basis. Handoff in Cellular IP is always initiated by
the mobile host. As the host approaches a new base station, it redirects its data
packets from the old to the new base station. First few redirected packets will
automatically configure a new path of RC mappings for the host to the new base
station. For a time equal to the timeout of RC mappings, packets addressed to the
mobile host will be delivered at both old and new base stations.

2.2.4. IPV6(Internet Protocol Version 6)


IP version 6 (IPv6), the successor to today's IP version 4 protocol (IPv4),
dramatically expands the available address space. Internet Engineering Task Force
(IETF) has produced a comprehensive set of specifications (RFC 1287, 1752, 1886,
1971, 1993, 2292, 2373, 2460, 2473 etc.) that define the next-generation IP protocol
originally known as TPNg,' now renamed as 'IPv6'. IPv6 addresses both a short-term
and long-term concern for network owners, service providers and users.

Address Space
IPv6 uses 128 bit addresses for each packet, creating a virtually infinite
number of IP addresses (approx. 3.4*10**38 IP addresses), as opposed to 3758096384
IPv4 addresses (2**31 A Class address + 2**30 B Class + 2**29 C Class address). This

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also means that if we set the world population at 10 billion in 2050, there will be
3.4*10**27 addresses available per person.
In IPv6, there are global addresses and local addresses. Global addresses are
used for routing of global Internet. Link local addresses are available within a
subnet. IPv6 uses hierarchical addressing with three-level of addresses.

This includes a Public Topology (the 48 bit external routing prefix), a Site
Topology (typically a 16 bit subnet number), and an Interface Identifier (typically an
automatically generated 64 bit number unique on the local LAN segment).
IPv6 Security
One of the biggest differences between IPv6 and IPv4 is that all IPv6 nodes
are expected to implement strong authentication and encryption features to improve
Internet security. IPv6 comes native with a security protocol called IP Security
(IPSec). IPSec protocol is a standards-based method of providing privacy, integrity
and authenticity to information transferred across IP networks.

IPSec combines several different security technologies into a complete system


to provide confidentiality, integrity and authenticity. In particular, IPSec uses:
 Diffie-Hellman key exchange mechanism for deriving key between peers
on a public network.
 Public key cryptography to guarantee the identity of the two parties and
avoid man-in-the-middle attacks.
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 Bulk encryption algorithms, such as 3DES, for encrypting the data.


 Keyed hash algorithms, such as HMAC, combined with traditional hash
algorithms such as MD5 or SHA for providing packet authentication.
 Digital certificates signed by a certificate authority to act as digital ID
cards.
 IPSec provides IP network-layer encryption.
Packet payload

Each IPv6 packet payload is attached a tag which can be customized to enable
a better quality in the packet flow, or by a price of other class, such as non-real time
quality of service or 'real-time' service. Information is packetized into IPv6 packets,
with the corresponding levels of control.

Migrating from IPv4 to IPv6

The Migration from IPv4 to IPv6 is quite an involved task. This includes the
following:

• Migration of the network components to be able to support IPv6 packets.


As there is no change at the physical layer between IPv4 and IPv6,
network components like hub or switch need not change. As there is a
change in the packet header the routers need to be upgraded. However,
using IP tunneling IPv6 packets can propagate over an IPv4 envelope.
Existing routers can support IP tunneling.

• Migration of the computing nodes in the network: this will need the
operating system upgrades so that they support IPv6 along with IPv4.
Upgraded systems will have both IPv4 and IPv6 stacks. Therefore, both
the IPv4 and IPv6 applications can run without any difficulty.

• Migration of networking applications in both client and server systems:


this requires porting of the applications from IPv4 to IPv6 environment.
Interconnecting IPv6 Networks
Till all the routers/system become IPv6-compatible, the interconnection
between IPv6 networks can be accomplished by tunneling. Tunneling is one of the
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key deployment strategies for both service providers as well as enterprises during
the period of IPv4 and IPv6 coexistence. Tunneling service providers can offer an
end-to-end IPv6 service without major upgrades to the infrastructure and without
impacting current IPv4 services.

A variety of tunnel mechanisms are available. These mechanisms include:


 Manually created tunnels such as IPv6 manually configured tunnels (RFC
2893).
 IPv6 over IPv4 tunnels.
 Semiautomatic tunnel mechanisms such as that employed by tunnel
broker services.
 Fully automatic tunnel mechanisms such as IPv4-compatible and 6 to 4.
 The dual-stack routers run both IPv4 and IPv6 protocols simultaneously
and thus can interoperate directly with both IPv4 and IPv6 end systems
and routers.

Mobile IP with IPv6


IPv6 includes many features for streamlining mobility support that are
missing in IP version 4, including Stateless Address Autoconfiguration and
Neighbor Discovery. IPv6 with hierarchical addressing scheme will be able to
manage IP mobility much efficiently. IPv6 also attempts to simplify the process of
renumbering, which could be critical to the future mutability of the Internet traffic.
Mobility Support in IPv6, as proposed by the Mobile IP working group, follows the
design for Mobile IPv4. It retains the ideas of a home network, home agent and the
use of encapsulation to deliver packets from the home network to the mobile node's
current point of attachment. While discovery of a care-of-address is still required, a
mobile node can configure its a care-of address by using Stateless Address
Autoconfiguration and Neighbor Discovery. Thus, foreign agents are not required
to support mobility in IPv6.

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UNIT - III

3.1. GSM

3.1.1. Global System for Mobile Communications


GSM is Global System for Mobile Communication. Its principal use is for
wireless telephony, and messaging through SMS. It also supports facsimile and data
communication. GSM is based on a set of standards, formulated in the early 1980s.
Year Event
1982 Groupe Special Mobile (GSM) established

1987 Essential elements of wireless transmission specified


1989 GSM become an ETSI technical committee
1990 Phase 1 GSM 900 specification (designed 1987 through 1990) frozen

1991 First GSM network launched


1993 First roaming agreement came into effect

1994 Data transmission capability launched


1995 Phase 2 launched. Fax and SMS roaming services offered
2002 SMS volume crosses 24 billions/year, 750 millions subscribers

The proposed GSM system had to meet certain business objectives. These are:

• Support for international roaming

• Good speech quality

• Ability to support handheld terminals

• Low terminal and service cost

• Spectral efficiency

• Support for a range of new services and facilities

• ISDN compatibility.
GSM uses a combination of FDMA (Frequency Division Multiple Access) and
TDMA (Time Division Multiple Access). The GSM system has an allocation of 50
MHz (890-915 MHz and 935-960 MHz) bandwidth in the 900 MHz frequency band.
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Using FDMA, this band is divided into 124 (125 channels, 1 not used) channels each
with a carrier bandwidth of 200 KHz. Using TDMA, each of these channels is then
further divided into 8 time slots. Therefore, with the combination of FDMA and
TDMA we can realize a maximum of 992 channels for transmit and receive. In order
to be able to serve hundreds of thousands of users, the frequency must be reused.
This is done through cells. The essential characteristics of this reuse are as follows:

• The area to be covered is subdivided into radio zones or cells. Though in


reality these cells could be of any shape, for convenient modeling purposes
these are modeled as hexagons. Base stations are positioned at the center of
these cells.

• Each cell / receives a subset of frequencies fbi from the total set assigned to
the respective mobile network. To avoid any type of co-channel interference,
two neighboring cells never use the same frequencies.

• Only at a distance of D (known as frequency reuse distance), the same


frequency from the set fb{ can be reused. Cells with distance D from cell i, can
be assigned one or all the frequencies from the set ^ belonging to cell i

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• When moving from one cell to another during an ongoing conversation, an


automatic channel change occurs. This phenomenon is called handover.
Handover maintains an active speech and data connection over cell
boundaries.
The regular repetition of frequencies in cells result in a clustering of cells. The
clusters generated in this way can consume the whole frequency band. The size of a
cluster is defined by k, the number of cells in the cluster. This also defines the
frequency reuse distance D.

3.1.2. GSM Architecture

GSM networks are structured in hierarchic fashion. It consists at the


minimum one administrative region assigned to one MSC (Mobile Switching
Centre). The administrative region is commonly known as PLMN (Public Land
Mobile Network). Each administrative region is subdivided into one or many
Location Area (LA). One LA consists of many cell groups. Each cell group is
assigned to one BSC (Base Station Controller). For each LA there will be at least one
BSC. Cells in one BSC can belong to different LAs.

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Cells are formed by the radio areas covered by a BTS (Base Transceiver
Station). Several BTSs are controlled by one BSC. Traffic from the MS (Mobile
Station) is routed through MSC. Calls originating from or terminating in a fixed
network or other mobile networks is handled by the GMSC (Gateway MSC). It’s an
architecture of a GSM PLMN from technology point of view, whereas the below
shows the same architecture from operational point of view.

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For all subscribers registered with a cellular network operator, permanent


data such as the service profile is stored in the Home Location Register (HLR). The
data relate to the following information:

• Authentication information like International Mobile Subscriber Identity


(IMSI)

• Identification information like name, address, etc. of the subscriber

• Identification information like Mobile Subscriber ISDN (MSISDN) etc.

• Billing information like prepaid or postpaid

• Operator selected denial of service to a subscriber

• Handling of supplementary services like for CFU (Call Forwarding


Unconditional), CFB (Call Forwarding Busy), CFNR (Call Forwarding Not
Reachable) or CFNA (Call Forwarding Not Answered)

• Storage of SMS Service Center (SC) number in case the mobile is not
connectable so that whenever the mobile is connectable, a paging signal is
sent to the SC

• Provisioning information like whether long distance and international calls


allowed or not

• Provisioning information like whether roaming is enabled or not

• Information related to auxiliary services like Voice mail, data, fax services
etc.

• Information related to auxiliary services like CLI (Caller Line Identification)


etc.

• Information related to supplementary services for call routing. In GSM


network one can customize the personal profile to the extent that while the
subscriber is roaming in a foreign PLMN, incoming calls can be barred. Also,
outgoing international calls can be barred etc.

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3.1.3. GSM Entities


The GSM technical specifications define different entities that form the GSM
network by defining their functions and interface requirements. The GSM network
can be divided into four main groups:

• The Mobile Station (MS). This includes the Mobile Equipment (ME) and the
Subscriber Identity Module (SIM).

• The Base Station Subsystem (BSS). This includes the Base Transceiver Station
(BTS) and the Base Station Controller (BSC).

• The Network and Switching Subsystem (NSS). This includes Mobile


Switching Center (MSC), Home Location Register (HLR), Visitor Location
Register (VLR), Equipment Identity Register (EIR), and the Authentication
Center (AUC).

• The Operation and Support Subsystem (OSS). This includes the Operation
and Maintenance Center (OMC).
Mobile Station
Mobile Station is the technical name of the mobile or the cellular phone. MS
consists of two main elements:

• The mobile equipment or the mobile device. In other words, this is the
phone without the SIM card.

• The Subscriber Identity Module (SIM).


The handheld GSM terminals have experienced the highest evolution. The
weight and volume of these terminals are continuously decreasing. The life of a
battery between charging is also increasing. The evolution of technologies allowed
decrease of this power to 0.8 W.

The SIM is installed in every GSM phone and identifies the terminal. Without
the SIM card, the terminal is not operational. The SIM cards used in GSM phones are
smart processor cards. These cards posses a processor and a small memory. By
inserting the SIM card into the terminal, the user can have access to all the
subscribed services. The SIM card contains the International Mobile Subscriber
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Identity (IMSI) used to identify the subscriber to the system, a secret key for
authentication, and other security information. Another advantage of the SIM card is
the mobility of the users. Typically SIM cards contain 32 K bytes of memory. Part of
the memory in the SIM card is available to the user for storing address book and
SMS messages. Applications are developed and stored in SIM cards using SAT (SIM
Application Toolkit).

The Base Station Subsystem


The BSS (Base Station Subsystem) connects the Mobile Station and the NSS
(Network and Switching Subsystem). It is in charge of the transmission and
reception for the last mile. The BSS can be divided into two parts:

• The Base Transceiver Station (BTS) or Base Station in short.

• The Base Station Controller (BSC).


The Base Transceiver Station corresponds to the transceivers and antennas
used in each cell of the network. A BTS is usually placed in the center of a cell. Its
transmitting power defines the size of a cell.
Base Station Controller is the connection between the BTS and the Mobile
service Switching Center (MSC). The BSC manages the radio resources for one or
more BTS. It handles handovers, radio-channel setup, control of radio frequency
power levels of the BTSs, exchange function, and the frequency hopping.

The Network and Switching Subsystem


The central component of the Network Subsystem is the Mobile Switching
Center (MSC). It does multiple functions:

• It acts like a normal switching node for mobile subscribers of the same
network (connection between mobile phone to mobile phone within the
same network).

• It acts like a normal switching node for the PSTN fixed telephone
(connection between mobile phone to fixed phone).

• It acts like a normal switching node for ISDN.

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• It provides all the functionality needed to handle a mobile subscriber, such


as registration, authentication, location updating, handovers and call
routing.

• It includes databases needed in order to store information to manage the


mobility of a roaming subscriber.
The Operation and Support Subsystem (OSS)

The Operations and Support Subsystem (OSS) controls and monitors


the GSM system. The OSS is connected to the different components of the NSS and
to the BSC. It is also in charge of controlling the traffic load of the BSS.

Equipment Identity Register (EIR) is a database that contains a list of all valid
mobile equipment within the network, where each mobile station is identified by its
International Mobile Equipment Identity (IMEI).
Message Centre

Short message or SMS is one of the most popular services within GSM. SMS is
a data service and allows a user to enter text message up to 160 characters in length
when 7-bit English characters are used. It is 140 octets when 8-bit characters (some
European alphabets or binary data) are used, and 70 characters in length when non-
Latin alphabets such as Arabic, Chinese or Hindi are used (70 characters of 16-bit
Unicode). SMS is a proactive bearer and is an always ON network. Message center is
also referred to as Service Centre (SC) or SMS Controller (SMSC). SMSC is a system
within the core GSM network, which works as the store and forward system for SMS
messages. There are two types of SMS,

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 SMMT (Short Message Mobile Terminated Point-to-Point) - SMMT is an


incoming short message from the network and is terminated in the MS
(phone or Mobile Station).
 SMMO (Short Message Mobile Originated Point-to-Point) - SMMO is an
outgoing message, originated in the MS, and forwarded to the network for
delivery

3.1.4. Call Routing in GSM


In GSM there are many complex technologies used between the human
analog interface in the mobile and the digital network.

Digitizer and source coding

The user speech is digitized at 8 KHz sampling rate using


Regular Pulse Excited-Linear Predictive Coder (RPE-LPC) with a Long Term
Predictor loop. Each sample is then represented in signed 13-bit linear PCM value.
This digitized data is passed to the coder with frames of 160 samples. The encoder
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compresses these 160 samples into 260-bits GSM frames resulting in one second of
speech compressed into 1625 bytes and achieving a rate of 13 Kbits/sec.

Channel coding

This step introduces redundancy information into the data for error
detection and possible error correction. The gross bit rate after channel coding is
22.8 kbps (or 456 bits every 20 ms). These 456 bits are divided into eight 57-bit
blocks, and the result is interleaved amongst eight successive time slot bursts for
protection against burst transmission errors.
Interleaving
This step rearranges a group of bits in a particular way. This is to improve
the performance of the error-correction mechanisms. The interleaving decreases the
possibility of losing whole bursts during the transmission, by dispersing the errors.
Ciphering
Encrypts blocks of user data using a symmetric key shared by the mobile
station and the BTS.
Burst formatting
Adds some binary information to the ciphered block. This additional
information is used synchronization and equalization of the received data.
Modulation
The modulation technique chosen for the GSM system is the Gaussian
Minimum Shift Keying (GMSK). Using this technique the binary data is converted
back into analog signal to fit the frequency and time requirements for the multiple
access rules. This signal is then radiated as radio wave over the air. Each time slot
burst is 156.25 bits and contains two 57-bit blocks, and a 26-bit training sequence
used for equalization (Fig 5.9). A burst is transmitted in 0.577 ms for a total bit rate
of 270.8 kbps.

Multipath and equalization

At the GSM frequency bands, radio waves reflect from buildings, cars, hills,
etc. So not only the 'right' signal (the output signal of the emitter) is received by an

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antenna, but also many reflected signals, which corrupt the information, with
different phases are received. An equalizer is in charge of extracting the 'right' signal
from the received signal. It estimates the channel impulse response of the GSM
system and then constructs an inverse filter. In order to extract the 'right' signal, the
received signal is passed through the inverse filter.

Synchronization

For successful operation of a mobile radio system, time and frequency


synchronization are needed. Frequency synchronization is necessary so that
transmitter and receiver frequency match (in FDMA). Time synchronization is
necessary to identify the frame boundary and the bits within the frame (in TDMA).

3.1.5. GSM Address and Identifiers


GSM distinguishes explicitly between the user and the equipment. It also
distinguishes between the subscriber identity and the telephone number. To manage
all the complex functions, GSM deals with many addresses and identifiers. They are:

• International Mobile Station Equipment Identity (IMEI): Every mobile


equipment in this world has a unique identifier. This identifier is called IMEI.
The IMEI is allocated by the equipment manufacturer and registered by the
network operator in the Equipment Identity Register (EIR). In your mobile
handset you can type *#06# and see the IMEI.

• International Mobile Subscriber Identity (IMSI): When registered with a


GSM operator, each subscriber is assigned a unique identifier. The IMSI is stored
in the SIM card and secured by the operator. A mobile station can only be
operated when it has a valid IMSI. The IMSI consists of several parts. These are:
o 3 decimal digits of Mobile Country Code (MCC). For India MCC is
404.
o 2 decimal digits of Mobile Network Code (MNC). This uniquely
identifies a mobile operator within a country. For Airtel in Delhi
this code is 10.

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o Maximum 10 decimal digits of Mobile Subscriber Identification


Number (MSIN). This is a unique number of the subscriber within
the home network.
 Mobile Subscriber ISDN Number (MSISDN): The MSISDN number is the
real telephone number as is known to the external world. MSISDN number
is public information, whereas IMSI is private to the operator. This is a
number published and known to everybody. In GSM a mobile station can
have multiple MSISDN number. When a subscriber opts for Fax and data, he
is assigned a total of 3 numbers: one for voice call, one for fax call and
another for data call. The MSISDN categories follow the international ISDN
(Integrated Systems Data Network) numbering plan as the following:
o Country Code (CC): 1 to 3 decimal digits of country code
(USA - 1, India - 91, and Finland – 358)
o National Destination Code (NDC): Typically 2 to 3 decimal digits
(India BSNL -94 and all other operators – 98)
o Subscriber Number (SN): maximum 10 decimal digits.
(India SN – 8 Digit, Operator code – 2, Level No – 1, Subscriber
Code – 5)
Ex : In India a MSISDN number looks like 919845062050.
CC - 91
NDC – 98
SN – 45062050( OC – 45, LN – 0, SC – 62050)
 Location Area Identity: Each LA in a PLMN has its own identifier. The
Location Area Identifier (LAI) is structured hierarchically and unique. LAI
consists of 3 digits of CC, 2 digits of Mobile Network Code and maximum 5
digits of Location Area Code.

• Mobile Station Roaming Number (MSRN): When a subscriber is roaming


in another network a temporary ISDN number is assigned to the subscriber.
This ISDN number is assigned by the local VLR in charge of the mobile
station. The MSRN has the same structure as the MSISDN.
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• Temporary Mobile Subscriber Identity (TMSI): This a temporary identifier


assigned by the serving VLR. It is used in place of the IMSI for identification
and addressing of the mobile station. TMSI is assigned during the presence
of the mobile station in a VLR and can change (ID hopping). Thus, it is
difficult to determine the identity of the subscriber by listening to the radio
channel. The TMSI is never stored in the HLR. However, it is stored in the
SIM card. Together with the current location TMSI allows a subscriber to be
identified uniquely. For an ongoing communication the IMSI is replaced by
the 2-tuple LAI, TMSI code.

• Local Mobile Subscriber Identity (LMSI): This is assigned by the VLR and
also stored in the HLR. This is used as a searching key for faster database
access within the VLR.

• Cell Identifier: Within a LA, every cell has a unique Cell Identifier (CI).
Together with a LAI a cell can be identified uniquely through Global Cell
Identity (LAI+CI).

• Identification of MSCs and Location Registers: MSCs, Location Registers


(HLR, VLR), SCs are addressed with ISDN numbers. In addition, they may
have a Signaling Point Code (SPC) within a PLMN. These point codes can be
used to address these nodes uniquely within the Signaling System number 7
(SS#7) network.
3.1.6. Network Aspects in GSM

Transmission of voice and data over the radio link is only a part of the
function of a cellular mobile network. A GSM mobile can seamlessly roam nationally
and internationally. This requires that registration, authentication, call routing and
location updating functions are standardized across GSM networks. The
geographical area covered a network is divided into cells of small radius. When a
call is in progress and the user is on the move, there will be a handover mechanism
from one cell to another. These functions are performed by the Network Subsystem,

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mainly using the Mobile Application Part (MAP) built on top of the Signaling
System # 7 (SS7) protocol.

The signaling protocol in GSM is structured into three general layers,


depending on the interface. Layer 1 is the physical layer, which uses the channel
structures over the air interface. Layer 2 is the data link layer. Across the Um
interface, the data link layer is a modified version of the LAPD protocol used in
ISDN or X.25, called LAPDm. Layer 3 of the GSM signaling protocol is itself divided
into 3 sub-layers.

• Radio Resources Management: Controls the set-up, maintenance, and


termination of radio and fixed channels, including handovers.

• Mobility Management: Manages the location updating and registration


procedures as well as security and authentication.

• Connection Management: Handles general call control, similar to CCITT


Recommendation Q.931 and manages Supplementary Services and the Short
Message Service.
Handover
In a cellular network, while a call is in progress, the relationship between
radio and fixed links is dynamic. The user movements may make a user move away
or closer to a tower. When the user moves away from a tower, the radio signal
strength or the power of the signal keeps reducing. This can result in change of the

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channel or cell. This procedure of changing the resources is called handover. This
procedure is called 'handoff in North America. There are four different types of
handover in the GSM system, which involve transferring a call between:

• Channels (time slots) in the same cell

• Cells (Base Transceiver Stations) under the control of the same Base Station
Controller (BSC),

• Cells under the control of different BSCs, but belonging to the same Mobile
services Switching Center (MSC)

• Cells under the control of different MSCs.


The first two types of handover, called internal handovers, involve only one
Base Station Controller (BSC). The last two types of handover, called external
handovers, are handled by the MSC.

Mobility Management

The Mobility Management (MM) function handles the functions that arise
from the mobility of the subscriber. MM is in charge of all the aspects related to the
mobility of the user, especially the roaming, the location management, and the
security/authentication of the subscriber. Location management is concerned with
the procedures that enable the system to know the current location of a powered-on
mobile station so that the incoming call routing can be completed.
When a mobile station is switched on in a new location area (for example,
the user is roaming and has disembarked from an aircraft in a new city) or the
subscriber moves to a new location area or a different operator's PLMN, the
subscriber must register with the new network to indicate its current location. The
first location update procedure is called the IMSI attach procedure where the MS
indicates its IMSI to the network. When a mobile station is powered off, it performs
an IMSI detach procedure in order to tell the network that it is no longer connected.
Normally, a location update message is sent to the new MSC/VLR, which records
the location area information, and then sends the location information to the
subscriber's HLR. If the mobile station is authenticated and authorized in the new
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MSC/VLR, the subscriber's HLR cancels the registration of the mobile station with
the old MSC/VLR A location updating is also performed periodically. If after the
updating time period, the mobile station has not registered, it is then deregistered.
Unlike routing in the fixed network, where a terminal is semi-permanently
wired to a central office, a GSM user can roam nationally and even internationally.
When there is an incoming call for a subscriber, the mobile phone needs to be
located, a channel needs to be allocated and the call connected. A powered-on
mobile is informed of an incoming call by a paging message sent over the paging
channel of the cells within the current location area. The location updating
procedures, and subsequent call routing, use the MSC and both HLR and the VLR.
The information sent to the HLR is normally the SS7 address of the new VLR. If the
subscriber is entitled to service, the HLR sends a subset of the subscriber
information needed for call control to the new MSC/VLR, and sends a message to
the old MSC/VLR to cancel the old registration.

An incoming mobile terminating call is directed to the Gateway MSC (GMSC)


function. The GMSC is basically a switch, which is able to interrogate the
subscriber's HLR to obtain routing information and thus contains a table linking
MSISDNs to their corresponding HLR. A simplification is to have a GSMC handle
one specific PLMN. Though the GMSC function is distinct from the MSC function, it
is usually implemented within an MSC. The routing information that is returned to
the GMSC is the Mobile Station Roaming Number (MSRN), which is also defined by
the E.164 numbering plan. MSRNs are related to the geographical numbering plan,
and not assigned to subscribers, nor are they visible to subscribers.

The most general routing procedure begins with the GMSC querying the
called subscriber's HLR for an MSRN. The HLR typically stores only the SS7 address
of the subscriber's current VLR. The VLR temporarily allocates an MSRN from its
pool for the call. This MSRN is returned to the HLR and back to the GMSC, which
can then route the call to the new MSC. At the new MSC, the IMSI corresponding to
the MSRN is looked up, and the mobile is paged in its current location area. As a

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rule of thumb, HLR is referred for incoming call; whereas VLR is referred for
outgoing call.

3.2. SMS

3.2.1. Mobile Computing Over SMS


GSM supports data access over CSD (Circuit Switched Data). GSM is
digitized but not packetized. In case of CSD, a circuit is established and the user is
charged based on the time the circuit is active and not on the number of packets
transacted. GPRS (General Packet Radio Service), also known as 2.5G, which is the
next phase within the evolution of GSM, supports data over packets. WAP is a data
service supported by GPRS and GSM to access Internet and remote data services.
Other data services in GSM include Group 3 facsimile, which is supported by use of
an appropriate fax adaptor. A unique data service of GSM, not found in older analog
systems, is the Short Message Service (SMS). SMS enables sending and receiving text
messages to, and from, GSM mobile phones.

3.2.2. SMS (Short Message Service)


The SMS is the most popular data bearer/service within GSM with an
average of one billion SMS messages (at the end of 2002) transacted every day
around the world. SMS uses the free capacity of the signaling channel. Each short
message is up to 160 characters in length when 7-bit English characters are used. It is
140 octets when 8-bit characters (some European alphabets) are used, and 70
characters in length when non-Latin alphabets such as Arabic, Chinese or Hindi are
used (70 characters of 16 bit Unicode).

Strengths of SMS

Following is a list of unique characteristics of SMS, which make this an


attractive bearer for mobile computing.
 Omnibus nature of SMS: SMS uses SS7 signaling channel, which is available
throughout the world. SMS is the only bearer that allows a subscriber to send
a long distance SMS without having long distance subscription.

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 Stateless: SMS is sessionless and stateless. Every SMS message is


unidirectional and independent of any context. This makes SMS the best
bearer for notifications, alerts and paging.
 Asynchronous: In http, for every command (e.g., GET or POST) there is a
request and a response pair making it synchronous at the transaction level.
Unlike http, SMS is completely asynchronous. In case of SMS, even if the
recipient is out of service, the transmission will not be abandoned. Therefore,
SMS can be used as message queues.

 Self-configurable and last mile problem resistant: SMS is self-configurable.


In the case of Web or WAP, it is no trivial task to connect to a service from a
foreign network without any change in the configuration or preference
setting. The device needs to be configured interactively by the user or system
administrator to access the network. This makes the access dependent on the
last mile. SMS has no such constraints. While in a foreign network, one can
access the SMS bearer without any change in the phone settings. The
subscriber is always connected to the SMS bearer irrespective of the home and
visiting network configurations. While roaming in a foreign network, even if
the serving network does not have a SMSC (SMS Center) or SC (Service
Center), SMS can be sent and received.
 Non-reputable: SMS message carries the SC and the source MSISDN as a part
of the message header.
 Always connected: As SMS uses the SS7 signaling channel for its data traffic,
the bearer media is always on. User cannot switch OFF, BAR or DIVERT any
SMS-message.
SMS Architecture
SMS are basically of two types, SMMT (Short Message Mobile Terminated
Point-to-Point), and SMMO (Short Message Mobile Originated Point-to-Point).
SMMT is an incoming short message from the network side and is terminated in the
MS. SMMO is an outgoing message, originated in the user device (MS), and
forwarded to the network for delivery.
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Short Message Mobile Terminated (SMMT)

For a SMMT message, the message is sent from SC to the MS. This whole
process is done in one transaction. For the delivery of MT or incoming SMS
messages, the SC of the serving network is never used. This implies that a SMS
message can be sent from any SC in any network to a GSM phone anywhere in the
world. This makes any SM MT message mobile operator independent.

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Short Message Mobile Originated (SMMO)


SMMO is an outgoing message originated in the MS where generally the user
types in a message and sends it to a MSISDN number. For a MO message, the MSC
forwards the message to the home SC. The SC is an independent computer in the
network and works as a store and forward node.

SMS as an Information Bearer


To use SMS as a bearer for any information service, we need to connect the
services running on the Enterprise Origin server to the SC through an SME (Short
Message Entity) or ESME (External Short Message Entity). SME in any network is
generally a SMS gateway. With respect to SMS, a GSM subscriber is always in
control of the SC in the home network irrespective of the serving network. Thus, if
there is any SMS-based data service in the home network, it will be available in any
foreign network.
Operator-Centric Pull

For a SMMO to work it is mandatory that a SC is used. As a part of SMS value


added services, operators offer different information on demand and entertainment
services. These are done through connecting an Origin server to the SC via a SMS
gateway. These service providers are known as MVNO (Mobile Virtual Network
Operators). Virtual operators develop different systems, services, and applications to
offer data services using SMS. Many enterprises use these MVNOs to make their
services available to mobile phone users. There are quite a few banks in India which
offer balance enquiry and other low security banking services over SMS. For
example, if a HDFC customer wants to use these services, he needs to register for the
service. During the registration, the HDFC customer needs to mention the MSISDN
of the phone which will be used for this service. Once a user is registered for the

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service, he enters 'HDFCBAL' and send the message to a service number as a MO


message. SC delivers this MO message to the SMS gateway connected to this service
number. SMS gateway then forwards this message to the enterprise application. The
response from the enterprise application is delivered to the MS as a MT message
from the SME. Even if the subscriber is in some remote region of a foreign network
within GSM coverage, he can send the same SMS to the same service number in his
home network. This makes the home services available in the foreign network. This
also implies that an operator-centric SMS pull service is completely ubiquitous.
The connectivity between SC to SME and SME to Enterprise Origin server is
not defined by GSM. However, there are a few de facto standard protocols for this
communication. The most popular protocol is Short Message Peer to Peer (SMPP).
There are certain other protocols like CIMD from Nokia as well. The connectivity
between SME and Origin server could be anything like SOAP (Simple Object Access
Protocol), or direct connection through TCP socket. However, common practice is
through http. Http helps user to get information from the Internet via SMS. There is
an open source for SMS gateway called Kannel, which supports a multitude of
protocols and forwards the SMS enquiry as an http request and gets information
from the Internet. This is how an SMS can be converted into a simple Internet access.
Conventionally SMS queries are keywords driven like 'CRT for live cricket score, or
'RSK 2627 3 03' to get the availability of sear/berth in Indian Railways train number
2627 (Karnataka Express) for March 3. There are applications where SMS is used in
session oriented transactions. Applications like 'SMS chat' and 'SMS contests' need to
remember the user context over multiple transactions.
Operator Independent Push

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We have seen that it is possible to send a SMS to any phone in any network.
For example, a MT message can be delivered from a network in India to a MS of UK
roaming in Germany. Which in other words means that any push, which may be an
alert, notification or even response from a pull message generated by an application,
can be serviced by any network and delivered to any GSM phone in any network
without any difficulty.

Assuming that appropriate roaming tie-ups are in place, an enterprise can use
SMS to send business alerts or proactive notifications to its customer anywhere,
anytime on any GSM phone. With roaming tie-ups, operators reach an agreement on
revenue share and call forwarding mechanism. Roaming tie-ups are a commercial
issue rather than technical. Some credit card companies in India send SMS
notifications to its cardholders in different networks using operator independent
push.

Operator Independent Pull

As the SME is always connected to the home network's SC, with the
conventional framework, it is not possible to route mobile originated SMS messages
to any application or any SME of choice. There are ways by which a SMS message
can be routed to some enterprise SME connected to external SC. This is achieved
through SAT, where the SAT application running on the SIM card changes the SC
number during the transmission of the SMS and forces the SMS to recognize a
different SC of a different network as its home SC. In this case also, technically the
SMS is sent to the SME connected to the home SC. SMS has always been considered
a revenue generating tool for cellular operators. Therefore, the current framework
suits a cellular operator very well. If a SMS service is operator dependent, the
cellular operator can use this to its advantage. In today's global scenario an
enterprise or a MVNO has its customers around the world subscribing to different
GSM networks. To make this possible, enterprises need operator-independent pull
as well. Operator-independent pull services can be achieved using GSM modem

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technology described in the following sections. Also, the same can be done using
Intelligent Network Technologies.

3.2.3. Value Added Services through SMS

Value Added Services (VAS) can be defined as services, which share one or
more of the following characteristics:
• Supplementary service (not a part of basic service) but adds value to total
service offering.

• Stimulates incremental demand for core services offering.

• Stands alone in terms of profitability and revenue generation potential.

• Can sometimes stand-alone operationally.

• Does not cannibalize basic service unless clearly favorable.

• Can be an add-on to basic service, and as such, may be sold at a premium


price.

• May provide operational and/or administrative synergy between or among


other services and not merely for diversification.
Information on demand has three categories as described below.
 Static information. This type of information does not change frequently. A
good example is a restaurant guide.
 Dynamic information. This type of information changes in days. For example
the daily horoscope needs to be updated on daily basis.
 Real-time information. This type of information changes continually. Third
party contents fall in this category. For example, scores in a live cricket match
or stock quote undergo continual change.
User Interface in SMS Value Added Services
SMS will use the Short transaction model. For a SMS-based service, the user
interface is always keyword-based. This is something similar to the character-based
command interfaces, where the first word is the keyword (command) and rest are
the parameters for the command. If it is more than 160 characters, we need to split
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the response into multiple message responses. It is advised that while the message is
broken into multiple messages, it is broken at the word boundary. It is also advised
that a sequence number like ... 1/3, . . . 2/3, and ... 3/3 is added in the first, second and
third messages respectively.

VAS Examples

News/Stock Quotes Service

In a service like News or Stock Quote, we get the latest news or stock
information. This will be a Short transaction. The keyword for news will be News,
whereas the keyword for stock quote can be BSE.

Session-based Chat Application

A chat service is essentially a session oriented transaction. In a chat service the


user needs to log in. The user needs to explicitly log out or will be logged out
implicitly following a period of inactivity
Alert Services

These are proactive alert services. For a stock quote the alert services can be of
the following kind.
 Time-based: In this service, proactive alerts are sent to the mobile phone at a
pre-assigned time of the day.
 Watermark based: In this service whenever the stock price goes up or falls
down to a certain level, alerts are sent. This information will help the
subscriber to decide whether to buy or sell some particular stock.

Location-based software
Location-based services could be road direction, restaurant guide etc. Some
location-aware VAS services provide shopping alerts as well. In location-based
services only the information relevant to the current location of the mobile phone (or
the subscriber) is provided. Device-specific location awareness requires either of the
following technologies:

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 Cell ID (CID)-based system.


 Global Positioning System (GPS)-based system.
The below shows the basic value added service provisioning architecture for
SMS. The reader should try to map an application scenario and get a feeling of how
information travels across in the case of 'pull'/'push' .

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UNIT - IV

4.1. GPRS

4.1.1. GPRS and Packet Data Network

GPRS will thrive in both vertical and horizontal markets where high-speed
data transmission over wireless networks is necessary. The deployment of GPRS
networks will allow a variety of new applications ranging from mobile e-commerce
to mobile corporate VPN access.
Capacity and Other End-user Aspects
GPRS has the ability to offer data speeds of 14.4 KBps to 171.2 KBps, which
allow for comfortable Internet access. It allows for short 'bursty' traffic, such as e-
mail and web browsing, as well as large volumes of data. To support GPRS
operations, new protocols and new network devices are required. For GPRS, no dial-
up modem connection is necessary. It offers fast connection set-up mechanism to
offer a perception of being 'always on'. This is why GPRS users are sometimes
referred to as being 'always connected'.
Quality of Service (QoS)
GPRS allows definition of QoS profiles using the parameters of service
precedence, reliability, delay and throughput.

• Service precedence is the priority of a service in relation to another service.


There exist three levels of priority: high, normal, and low.

• Reliability indicates the transmission characteristics required by an


application. Three reliability classes are defined, which guarantee certain
maximum values for the probability of loss, duplication, mis-sequencing and
corruption (an undetected error) of packets.

• Delay parameters define maximum values for the mean delay and the 95-
percentile delay. The delay is defined as the end-to-end transfer time
between two communicating mobile stations or between a mobile station
and the signaling interface to an external packet data network.

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• Throughput specifies the maximum/peak bit rate and the mean bit rate.
Integral Part of the Future 3G Systems
GPRS is introduced as an intermediate step to efficiency transport high-speed
data over the current GSM and TDMA-based wireless network infrastructures.
GPRS is therefore called the 2.5G (two and half G or two and half generation) in the
evolution process of wireless cellular networks.

4.1.2. GPRS Network Architecture


GPRS uses the GSM architecture for voice. In order to offer packet data
services through GPRS, a new class of network nodes need to be introduced as an
upgrade to existing GSM network. These network nodes are called GPRS support
nodes (GSN). GPRS support nodes are responsible for the delivery and routing of
data packets between the mobile stations and the external packet data networks
(PDN). There are two types of support nodes, viz., SGSN (Serving GSN) and GGSN
(Gateway GSN). The below diagram shows the GPRS system components for data
services.

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 Serving GPRS Support Node (SGSN): A serving GPRS support node (SGSN)
is at the same hierarchical level as the MSC. Whatever MSC does for voice,
SGSN does the same functions for packet data. SGSN's tasks include packet
switching, routing and transfer, mobility management (attach/detach and
location management), logical link management, and authentication and
charging functions. SGSN processes registration of new mobile subscribers
and keeps a record of their location inside a given service area. The location
register of the SGSN stores location information (e.g., current cell, current
VLR) and user profiles of all GPRS users registered with this SGSN. SGSN
sends queries to Home Location Register (HLR) to obtain profile data of
GPRS subscribers. The SGSN is connected to the base station system with
Frame Relay.
 Gateway GPRS Support Node (GGSN): A gateway GPRS support node
(GGSN) acts as an interface between the GPRS backbone network and the
external packet data networks. GGSN's function is similar to that of a router
in a LAN. GGSN maintains routing information that is necessary to tunnel the
Protocol Data Units (PDUs) to the SGSNs that service particular mobile
stations. It converts the GPRS packets coming from the SGSN into the
appropriate packet data protocol (PDP) format for the data networks like
Internet or X.25. PDP sends these packets out on the corresponding packet
data network. In the other direction, PDP receives incoming data packets
from data networks and converts them to the GSM address of the destination
user. The readdressed packets are sent to the responsible SGSN. For this
purpose, the GGSN stores the current SGSN address of the user and his or her
profile in its location register. The GGSN also performs authentication and
charging functions related to data transfer.

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GPRS Network Enhancements


In addition to the new GPRS components (SGSN and GGSN), some existing
GSM network elements must also be enhanced in order to support packet data.
These are:
 Base Station System (BSS): BSS system needs enhancement to recognize and
send packet data. This includes BTS upgrade to allow transportation of user
data to the SGSN. Also, the BTS needs to be upgraded to support packet data
transportation between the BTS and the MS (Mobile Station) over the radio.
 Home Location Register (HLR): HLR needs enhancement to register GPRS
user profiles and respond to queries originating from GSNs regarding these
profiles.
 Mobile Station (MS): The mobile station or the mobile phone for GPRS is
different from that of GSM.
 SMS nodes: SMS-GMSCs and SMS-IWMSCs are upgraded to support SMS
transmission via the SGSN. Optionally, the MSC/VLR can be enhanced for
more efficient co-ordination of GPRS and non-GPRS services and
functionality.
Channel Coding
Channel coding is used to protect the transmitted data packets against errors.
The channel coding technique in GPRS is quite similar to the one employed in
conventional GSM. Under very bad channel conditions, reliable coding scheme is
used. In reliable coding scheme many redundant bits are added to recover from
burst errors. In this scheme a data rate of 9.05 KBps is achieved per time slot. Under
good channel conditions, no encoding scheme is used resulting in a higher data rate
of 21.4 KBps per time slot. With eight time slots, a maximum data rate of 171.2 KBps
can be achieved.
Transmission Plane Protocol Architecture
The protocol architecture of the GPRS transmission plane, providing
transmission of user data and its associated signaling.

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 Signaling Plane
The protocol architecture of the signaling plane comprises
protocols for control and support of the functions of the transmission plane.
This includes GPRS attach and detach, PDP context activation, control of
routing paths, and allocation of network resources. The signaling architecture
between SGSN and the registers like HLR, VLR, and EIR uses the same
protocols as GSM. However, they are extended to support GPRS-specific
functionality. Between SGSN and HLR as well as between SGSN and EIR, an
enhanced MAP (Mobile Application Part) is employed. MAP is a mobile
network-specific extension of the Signaling System SS#7 used in GSM. It
transports the signaling information related to location updates, routing
information, user profiles, and handovers. The exchange of MAP messages is
accomplished over the transaction capabilities application part (TCAP) and
the signaling connection control part (SCCP). The base station system
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application part (BSSAP+) is an enhancement of GSM's BSSAP. It is used to


transfer signaling information between the SGSN and the VLR.
 GPRS Backbone
GPRS backbone includes the transmission plane between SGSN and
GGSN. User data packets and related signaling information within the GPRS
network are encapsulated using the GPRS Tunneling Protocol (GTP). The
GTP protocol is used in both intra PLMN (between SGSN and GGSN within
one PLMN) and inter-PLMN (between SGSN and GGSN of different PLMNs).
In the transmission plane, GTP protocol runnels the user data packets through
the GPRS backbone by adding GPRS specific routing information. GTP
packets carry the user's data packets from both IP and X.25 data networks.
Below GTP, the standard protocols TCP or UDP are used to transport the GTP
packets within the backbone network. X.25 expects a reliable data link;
therefore TCP is used for tunneling X.25 data. For IP based user data, UDP is
used as it does not expect reliability in the network layer or below. Ethernet,
ISDN, or ATM-based protocols may be used in the physical layer in the IP
backbone. In essence, in the GPRS backbone we have an IP/X.25-over-GTP-
over-UDP/TCP-over-IP transport architecture.
 BSS-SGSN Interface
The BSS and SGSN interface is divided into the following layers:
o Sub-Network Dependent Convergence Protocol (SNDCP): The
SNDCP is used to transfer data packets between SGSN and MS. Its
functionality includes:
 Multiplexing of several connections of the network layer onto
one virtual logical connection of the underlying LLC layer.
 Segmentation, compression, and decompression of user data.
o Logical Link Control (LLC): a data link layer protocol for GPRS
which functions similar to Link Access Procedure—D (LAPD). This
layer assures the reliable transfer of user data across a wireless
network.
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o Base Station System GPRS Protocol (BSSGP): The BSSGP delivers


routing and QpS-related information between BSS and SGSN.
o Network Service: This layer manages the convergence sublayer
that operates between BSSGP and the Frame Relay Q.922 Core by
mapping BSSGP's service requests to the appropriate Frame Relay
services.
 Air Interface
The air interface of GPRS comprises the physical and data link layer.
o Data Link Layer The data link layer between the MS and the BSS is
divided into three sub-layers: the logical link control (LLC) layer,
the radio link control (RLC) layer and the medium access control
(MAC) layer.
 Logical Link Control (LLC): This layer provides a reliable
logical link between an MS and its assigned SGSN. Its
functionality is based on HDLC (High-level Data Link
Control) protocol and includes sequence control, in-order
delivery, flow control, detection of transmission errors,
and retransmission (automatic repeat request, ARQ).
Encryption is used in this interface to ensure data
confidentiality. Variable frame lengths are possible.
Both acknowledged and unacknowledged data
transmission modes are supported. This protocol is an
improved version of the LAPDm protocol used in GSM.
 Radio Link Control (RLC): The main purpose of the radio
link control (RLC) layer is to establish a reliable link between
the MS and the BSS. This includes the segmentation and
reassembly of LLC frames into RLC data blocks and ARQ^ of
uncorrectable
data.

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 Medium Access Control (MAC): The medium access control


(MAC) layer controls the access attempts of an MS on the
radio channel shared by several MSs. It employs algorithms
for contention resolution, multiuser multiplexing on a packet
data traffic channel (PDTCH), and scheduling and
prioritizing based on the negotiated QpS.
o Physical Layer The physical layer between MS and BSS is divided
into two sublayers: the physical link layer (PLL) and the physical
RF Layer (RFL).
 Physical Link Layer (PLL): This layer provides services for
information transfer over a physical channel between the MS
and the network. These functions include data unit framing,
data coding, and the detection and correction of physical
medium transmission errors. The Physical Link layer uses
the services of the Physical RF layer.
 Physical RF Layer (RFL): This layer performs the
modulation of the physical wave forms based on the
sequence of bits received from the Physical Link layer above.
The Physical RF layer also demodulates received wave forms
into a sequence of bits that are transferred to the Physical
Link layer for interpretation.
Multiple Access Radio Resource Management
On the radio interface, GPRS uses a combination of FDMA and TDMA. As in
GSM, GPRS uses two frequency bands at 45 MHz apart; viz., 890-915 MHz for
uplink (MS to BTS), and 935-960 MHz for downlink (BTS to MS). Each of these bands
of 25 MHz width is divided into 124 single carrier channels of 200 kHz width. Each
of these 200 kHz frequency channels is divided into eight time slots. Each time slot
of a TDMA frame lasts for a duration of 156.25 bit times and contains a data burst.
On top of the physical channels, a series of logical channels are defined to
perform functions like signaling, broadcast of general system information,
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synchronization, channel assignment, paging or payload transport. As with GSM,


these channels can be divided into two categories: traffic channels and signaling
channels. Traffic channel allocation in GPRS is different from that of GSM. In GSM, a
traffic channel is permanently allocated for a particular user during the entire call
period (whether any data is transmitted or not). In contrast, in GPRS traffic,
channels are only allocated when data packets are sent or received. They are
released after the transmission of data. GPRS allows a single mobile station to use
multiple time slots of the same TDMA frame for data transmission. This is known as
multi slot operation and uses a very flexible channel allocation. One to eight time
slots per TDMA frame can be allocated for one mobile station. Moreover, uplink and
downlink are allocated separately, which efficiency supports asymmetric data traffic
like Internet where the bandwidth requirements in uplink and downlink are
different.
In GPRS, physical channels to transport user data packet is called data traffic
channel (PDTCH). The PDTCHs are taken from a common pool of all channels
available in a cell. Thus, the radio resources of a cell are shared by all GPRS and non-
GPRS mobile stations located within the cell. The mapping of physical channels to
either packet switched data (in GPRS mode) or circuit switched data (in GSM mode)
services are performed dynamically depending on demand. This is done depending
on the current traffic load, the priority of the service and the multi slot class. A load
supervision procedure monitors the load of the PDTCHs in the cell. According to the
demand, the number of channels allocated for GPRS can be changed. Physical
channels not currently in use by GSM can be allocated as PDTCHs to increase the
bandwidth of GPRS.
Security
GPRS security functionality is similar to the existing GSM security. The SGSN
performs authentication and cipher-setting procedures based on the same
algorithms, keys and criteria as in GSM. GPRS uses a ciphering algorithm optimized
for packet data transmission. Like its predecessor, a GPRS device also uses SIM
card.
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4.1.3. Data Services in GPRS


A wide range of corporate and consumer applications are enabled by GPRS
services. A user is likely to use either of the two modes of the GPRS network. These
are Application mode or Tunneling mode.
 Application mode: In this mode the user will be using the GPRS mobile
phone to access the applications running on the phone itself. The phone here
acts as the end user device. All GPRS phones have WAP browser as an
embedded application. This browser allows browsing of WAP sites. Some
GPRS devices support mobile execution environment (MExE classmark 3).
These devices support development of client application that can run on the
device. The device operating execution environments supported are Symbian
and J2ME. Applications can be developed in C/C++ or Java.
 Tunneling mode: This mode is for mobile computing where the user will use
the GPRS interface as an access to the network. The end user device will be a
large footprint device like laptop computer or small footprint device like
PDAs. The mobile phone will be connected to the device and used as a
modem to access the wireless data network. For these devices, access can be
gained via a PC Card (PCMCI) or via a serial cable to a GPRS capable phone.
These 'black-box' devices do not have display, keypad and voice accessories
of a standard phone.
GPRS Handsets
A GPRS terminal can be one of three classes: A, B or C. A Class A terminal
supports GPRS data and other GSM services such as SMS and voice simultaneously.
This includes simultaneous attach, activation, monitor, and traffic. As such, a Class
A terminal can make or receive calls on two services simultaneously. In the presence
of circuit-switched services, GPRS virtual circuits will be held or placed on busy
rather than being cleared. SMS is supported in Class A terminal. Like GSM, a SMS
can be received while a voice or data call is in progress.
A Class B terminal can monitor GSM and GPRS channels simultaneously, but
can support only one of these services at any time. Therefore, a Class B terminal can

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support simultaneous attaches, activation, and monitor but not simultaneous traffic.
As with Class A, the GPRS virtual circuits will not be closed down when circuit-
switched traffic is present. Instead, they will be switched to busy or held mode.
Thus, users can make or receive calls on either a packet or a switched call type
sequentially but not simultaneously. SMS is supported in Class A terminal. Like
GSM, a SMS can be received while a voice or data call is in progress.
A Class C terminal supports only non simultaneous attach. The user must
select which service to connect to. Therefore, a Class C terminal can make or receive
calls from only the manually selected network service. The service that is not
selected is not reachable. The GPRS specifications state that support of SMS is
optional for Class C terminals.
Device Types
In addition to the three types of terminals, each handset will have a unique
form factor. Terminals will be available in the standard form factor with a numeric
keypad and a relatively small display. Other types of phones with different form
factors, color displays, with cameras are common. Smart phones with built-in voice,
non-voice and Web-browsing capabilities are common too. Smart phones have
various form factors, which may include a keyboard or an icon drive screen.
Bearers in GPRS
The bearer services of GPRS offer end-to-end packet switched data transfer.
GPRS is planned to support two different kinds of data transport services. These are
the point-to-point (FTP) service and the point-to-multipoint (PTM) service. Out of
these two, FTP is available now; PTM will be available in future releases of GPRS.
The PTP service offers transfer of data packets between two users. GPRS will
support the following types of data services:
 SMS: Short message service was originally designed for GSM network. GPRS
will continue to support SMS as a bearer. Please refer to chapter 6 for details
on SMS and application development using SMS.

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 WAP: WAP is Wireless Application Protocol. It is a data bearer service over


HTTP protocol. WAP uses WML (Wireless Markup Language) and a WAP
gateway.
 MMS: MMS is Multimedia Messaging Service. This is the next generation
messaging service. SMS supports text messages whereas MMS supports
multimedia messages.
MMS uses WAP and SMS as its lower layer transport. Video, audio pictures
or clips can be sent through MMS.

4.1.4. Billing and Charging in GPRS


For voice networks tariffs are generally based on distance and time. This in
other words means that user pays more for long distance calls. They also pay more if
they keep the circuit busy by talking for a longer period of time. In data services
have evolved without any concept of charging. In packet network keeping the
circuit busy does not have any meaning. Also, charging a customer by the distance
traversed by a packet does not make any sense. Many times due to congestions
packets traverse much longer distance than the optimum distance.
Tariffing
Decisions on charging for GPRS by packet or simply a flat monthly fee are
contentious but need to be made. Charging different packets at different rates can
make things complicated for the user. It is believed that the optimal GPRS pricing
model will be based on two variables, time and packet. Network operators will levy
a nominal per packet charge during peak times plus a flat rate. There will be no per
packet charge during non-peak times. Time and packet-related charging will
encourage applications such as remote monitoring, meter reading and chat to use
GPRS at night when spare network capacity is available. Simultaneously, a nominal
per packet charge during the day will help to allocate scarce radio resources, and
charge radio heavy applications such as file and image transfer more than
applications with lower data intensity. It has the advantage of automatically
adjusting customer charging according to their application usage.

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Billing
GPRS is essentially a packet switching overlay on a circuit switching network.
The GPRS specifications stipulate that the minimum charging information that must
be collected are:

• Destination and source addresses

• Usage of radio interface

• Usage of external Packet Data Networks

• Usage of the packet data protocol addresses

• Usage of general GPRS resources and location of the Mobile Station


Since GPRS networks break the information to be communicated down into
packets, at a minimum, a GPRS network needs to be able to count packets to
charging customers for the volume of packets they send and receive. Today's billing
systems have difficulties handling charging for today's data services. It is unlikely
that circuit switched billing systems will be able to process a large number of new
variables created by GPRS.
GPRS call records are generated in the GPRS Service Nodes. The incumbent
billing systems are often not able to handle real time Call Detail Record flows. As
such, an intermediary charging platform is a good idea to perform billing mediation
by collecting the charging information from the GPRS nodes and preparing it for
submission to the billing system. Packet counts are passed to a Charging Gateway
that generates Call Detail Records that are sent to the billing system.
The billing of the services can be based on the transmitted data volume, the
type of service, and the chosen QoS profile. It may well be the case that the cost of
measuring packets is greater than their value. The implication is that there will not
be a per packet charge since there may be too many packets to warrant counting and
charging for. The billing of roaming GPRS subscribers from one network to another
is still a challenge.

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4.2. WAP

4.2.1. Evolution of Wireless Data and WAP


In 1992, Nippon Telegraph and Telephone Corporation (NTT), a leading
telephone company in Japan spun off a wireless division and named it DoCoMo.
DoCoMo was derived combining two syllables Do and como. Do in Japanese mean
'everywhere' and como was for Communications. DoCoMo's success has been due
to i-mode, its mobile Internet service. DoCoMo developed a language called cHTML
(Compant Hyper Text Markup Language) and a gateway. Using these frameworks
an i-mode user can use the Internet.
In 1994, in USA a company named Unwired Planet was founded to develop
and market a platform for Internet access through wireless devices like PDA.
Unwired Planet developed a comprehensive framework including browser, gateway
and markup language. The markup language was called HDML (Handheld Device
Markup Language). Unwired Planet also developed HDTP (Handheld Device
Transport Protocol). Unwired Planet launched all these technologies in 1995.
In 1995, Ericsson another leading wireless company in Europe, began work
on a protocol known as ITTP or Intelligent Terminal Transfer Protocol. ITTP was
designed with the intent of making it easy for call control and add services to mobile
telephony platforms.
In 1997, Nokia, the other major wireless company in Europe developed the
TTML (Tagged Text Mark-up Language). TTML was designed to allow a mobile
phone to communicate with a World Wide Web site via gateway. Following TTML,
Nokia also introduced Narrowband Sockets (NBS) to create wireless messaging
applications for PCs communicating with GSM 'smart phones'.
Out of all these different technologies, i-mode in Japan was the most
successful service. It started growing like a bushfire. In USA and Europe, though
there were different services and protocols offered by different companies, none of
them could match the popularity of DoCoMo. Realizing that these competing
wireless protocols could fragment and possibly destroy the potential market, in June
of 1997, Ericsson, Motorola, Nokia, and Unwired Planet (now known as Phone.com)
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joined hands to launch the WAP Forum (www.wapforum.com). WAP Forum is now
known as Open Mobile Alliance. The goal of this effort was to produce a refined,
license-free protocol, which is independent of the underlying airlink standard. The
WAP inherited its main characteristics and functionality from HDML and HDTP
developed by Unwired Planet, the Smart Messaging specification based on TTML
and NBS developed by Nokia, and the ITTP specification developed by Ericsson.
The first release of the WAP 1.0 specifications was released in the spring of 1998.

4.2.2. GPRS Applications


For GPRS or WAP there are no specific services. These are the same services
over the wireless media that can run either on GSM or 3G. However, as GPRS offers
a higher bit rate, the user experience is better. Though, it is not mandatory, MMS
based contents are better suited for GPRS networks. Some of the applications that
are better suited for GPRS network are:

• Dating: This is an interactive dating system that exploits data bearer rather
than the SMS bearer.

• Games: Online games, cartoons can be better suited for GPRS network.

• Sending/Receiving Fax/Email on the Handset: Different mobile office


applications are quite well suited for GPRS network.

• Location aware applications: Location aware application combined with MAP


and other type of travel related applications will have a better reach in GPRS
network.

• VPN: It is possible to deploy wireless virtual private networks over GPRS


network.

• Multimedia downloads: Various picture, music, video clips download will be


part of this.
Digital Rights Management
MMS opens up lots of new possibilities for content providers to offer
innovative premium mobile services, such as comic services, news services, sports

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updates, movie and music clips. As these services take off, it is crucial to take into
account Digital Rights Management (DRM) for the distribution and consumption of
mobile content. With DRM, the content owner or service provider can determine if
and how his content can be distributed if at all from person to person. Of course,
some content will be available even without DRM. All the content created by mobile
users themselves such as capturing and sending of photo and video clips, or
advertising content like sponsored movie trailers that promote new films in the
cinema, will be free. Companies involved in mobile content services such as ring
tones, wallpapers, or games where the business plan is based on being able to collect
the rightful payment, will be interested in implementing DRM technology.
OMA Digital Rights Management
OMA has proposed Digital Rights Management and OMA Download
standards. By implementing OMA DRM, service providers can allow end users to
preview content before making a purchase decision. OMA DRM also allows end
users to distribute content to other users via super distribution. The OMA DRM
standard will govern the use of mobile-centric content types. The first DRM
standard, OMA DRM version 1.0, was officially approved in October 2002. The
standard provides three DRM methods as following:

• Forward-lock is intended for the delivery of subscription-based services. The


device is allowed to play, display or execute the MMS, but lit cannot forward
the MMS object. The content itself is hidden inside the DRM message that is
delivered to the terminal. Examples will be news, sports, information and
images that should not be sent on to others.

• Combined Delivery enables usage rules to be set for the media object. This
method extends Forward-lock by adding a rights object to the DRM
Message. Rights define how the device is allowed to render the content and
can be limited using both time and count constraints. This method allows
previews.

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• Separate Delivery protects higher value media and enables super


distribution. This allows the device to forward the media, but not the rights.
This is achieved by delivering the media and rights via separate channels,
which is more secure than combined delivery. The media is encrypted into
DUM Content Format (DCF) using symmetric encryption, while the rights
hold the Content Encryption Key (CEK), which is used by the DRM User
Agent in the device for decryption. Recipients of super distributed content
must contact the content retailer to obtain rights to either preview or
purchase media.

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UNIT - V

5.1. CDMA and 3G

5.1.1. Introduction

Mobile phone technology had a recreation from first generation analogue


(using FDMA) to second generation digital (using TDMA). The next recreation is
from second generation digital TDMA to third generation packet (using CDMA).
CDMA is a specific modulation technique of Spread Spectrum technology. Third
generation or 3G is more of a generic term to mean mobile networks with high
bandwidth. Looking at the success of second generation GSM (using TDMA and
roaming) and also the potential of second generation cdmaOne (IS-95 using CDMA),
it was quite apparent that the next generation networks would have to be a
combination of the best of these two technologies with incorporation of some of the
recent technology innovations.
How it started
Hedy Lamarr helped was the original patent holder of Spread-Spectrum
Technology, which is at the foundation of today's CDMA (Code Division Multiple
Access), Wireless LAN, IMT-2000 (International Mobile Telecommunication-2000),
3G (Third Generation), and GPS (Global Positioning System) technology.
With the help of an electrical engineering professor from the MIT, Hedy
Lamarr and George Antheil, a film music composer patented 'Secret Communication
System' in 1942. Like many other great technologies, the idea of 'Secret
Communication System' was ahead of its time. Electronic technologies were
beginning to develop and in the 1950s, engineers from Sylvania Electronic Systems
Division began to experiment with the ideas in the Secret Communication System
patent, using digital components. They developed an electronic spread-spectrum
system that handled secure communications for the US during the Cuban Missile
Crisis in 1962. It was in the early 1960s that the term 'spread spectrum' began to be
used. Today it refers to digital communications that use a wide frequency spreading
factor (much wider than typical voice telephone communications), and are not

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dependent on a particular type of tonality (such as a human voice) in the


transmitting waveform.
In the mid-1980s, the US military declassified spread-spectrum technology.
Immediately, the commercial sector began to develop it for consumer electronics.
Qualcomm was the first to use this technology for commercial deployment of
CDMA. 3G has been in gestation since 1992, when the International
Telecommunications Union (ITU) began work on a standard called IMT-2000. IMT
stands for International Mobile Telecommunications; the number 2000 initially had
three meanings: the year that services should become available (year 2000), the
frequency range in MHz that would be used (2000 MHz or 2 GHz), and the data rate
in Kbits/sec (2000 KBps or 2 MBps).
5.1.2. Architecture CDMA versus GSM
Functions GSM IS-95
Frequency 900MHz; 1800MHz 800MHz; 1900MHz
(DCS 80);
Channel 1900MHz
Total (PCSbandwidth
25 MHz 1900) with Total 12MHz with 1.25MHz
bandwidth 200KHz per channels, 8 for the spread spectrum
timeslots per channel
with frequency hopping
Voice codec 13Kbits/second 8 Kbits/sec or 13Kbps
Data bit rate 9.6Kbits/second expandable 9.6Kbits
Short message 160 characters of text 120 characters
service Supports
SIM card Yes No
Multipath Causes interference Used as an advantage
and
Radio interface TDMA CDMA
Handoff Hard Handover (handoff) Soft Handoff (handover)
System Capacity Fixed and limited Flexible and higher than
GSM
Economics Expensive Due to many technological
advantages, dimension of
investment per subscriber is
expected to be lower than
GSM

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5.1.3. IEEE 802.11Standards


The IEEE 802 committee was set up in February 1980 to set the standard for
local area networks. From time to time, IEEE came up with different standards in the
LAN domain. This includes all the layers from physical, media access, and data link
layer. When IEEE deliberated the standards, it was clear that wireless LAN will be
different only at the physical and media access layer.

There are quite a few types of wireless LANs. However, IEEE 802.11 is
gaining fast momentum in India and other parts of the world. When we refer to
802.11, we generally mean the generic 802.11 families of standards. There are many
standards within this family with almost all the letters of the English alphabet
starting from 'a' to 'x'. Different standards cover different aspects like bandwidths,
modulation techniques, physical media, security etc.
Standard Description Status
IEEE 802.11 Standard for WLAN operations at data Approved in July 1997.
rates up to 2 Mbps in the 2.4-GHz ISM
(Industrial, Scientific and Medical) band.
IEEE 802.11a Standard for WLAN operations at data Approved in Sept 1999.
rates up to 54 Mbps in the 5-GHz End-user products began
Unlicensed National Information shipping in early 2002.
Infrastructure (UNII) band.
IEEE 802.11b Standard for WLAN operations at data Approved in Sept 1999.
rates up to 11 Mbps in the 2.4-GHz ISM End- user products began
(Industrial, Scientific and Medical) band. shipping in early 2000.

IEEE 802.llg High-rate extension to 802.1 lb Draft standard adopted


allowing for data rates up to 54 Mbps in Nov 2001. Full ratification
the 2.4-GHz ISM band. expected in 2003.
IEEE Wireless Personal Area Network standard 802.15.1-2002
based on the Bluetooth specification, conditionally
802.15.1
operating in the 2.4-GHz ISM band. approved on March 21,
2002.
IEEE802.1x Port-based network access control defines Approved June 2001
infrastructures in order to provide a
means of authenticating and authorizing
devices attached to a LAN port that has
point-to-point connection characteristics.

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Upcoming Description
Standard
IEEE 802.lie Enhance the 802.11 Medium Access Control improve and
managed (MAC) to Quality of Service, provide classes of
service, and enhanced security and authentication
mechanisms. These enhancements should provide the quality
required for services such as IP telephony and video
streaming.
IEEE 802.1 If Develop recommended practices for an Inter-Access Point
Protocol (LAPP),which provides the necessary capabilities to
achieve multi-vendor Access Point interoperability across a
Distribution System supporting IEEE P802.11 Wireless LAN
Links.
IEEE 802. llh Enhance the 802.11 Medium Access Control (MAC) standard
and 802.11a High Speed Physical Layer (PHY) in the 5GHz
band. Objective is to make IEEE 802.1 lah products compliant
with European regulatory requirements.

IEEE 802.111 Enhance the 802.11 Medium Access Control (MAC) to enhance
security and authentication mechanisms.
IEEE 802.15 Developing Recommended Practices to facilitate coexistence
TG2 of Wireless Personal Area Networks™ (802.15) and Wireless
Local Area Networks (802.11).
IEEE 802.15 Draft and publish a new standard for high-rate (20Mbit/s or
TG3 greater) WPANs™ .
IEEE 802.15 Investigate a low data rate WPAN solution with multi- month
TG4 to multi – year battery life and very low complexity.

5.1.4. Wireless Data


Data transmission over wireless networks like CDMA or GSM is always a
challenge. Typically raw channel data error rates for cellular transmission are 10-2.
This means that one in every 100 bits has an error. This is an error rate, which can be
tolerated for voice transmission. This is because; our perception of hearing cannot
detect it. Even if our ear is sometime able to detect it, our mind is able to correct it
from the context. This error rate of 10~2 is too high for data transmission. An
acceptable BER (Bit Error Rate) for data transmission is 10-6. This means that one bit

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in a million can be tolerated as an error. In order to achieve this high level of


reliability, it requires a design of effective error correction code and Automatic
Repeat Request (ARQ). The CDMA protocol stack for data and facsimile has the
following layers.

 Application Interface Layer: This layer includes an application interface


between the data source in the mobile station and the transport layer. The
application interface provides functions like modem control, AT (Attention)
command processing, data compression etc.
 Transport Layer: The transport layer for CDMA asynchronous data and fax is
based on TCP. TCP has been modified for IS-95.

 Network layer: The network layer for CDMA asynchronous data and fax
services is based on IP. The standard IP protocol has been enhanced for IS-95.

 Sub-network Dependent Convergence Function: The SNDCF performs


header compression on the header of the transport and network layers.
Mobile station supports Van Jacobson TCP/IP header compression algorithm.
Negotiation of the parameters for header compression is carried out using
IPCP (Internet Protocol Control Protocol). The SNDCF sub-layer accepts the
network layer datagram packets from the network layer, performs header
compression and passes that datagram to the PPP (Point to Point Protocol)

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layer. In the reverse operation, it receives network layer data grams with
compressed header from the PPP layer and passes it to the network layer.

 Data Link layer: This layer uses PPP. The PPP Link Control Protocol (LCP) is
used for initial link establishment and for the negotiation of optional link
capabilities.

 Internet Protocol Control Protocol Sub-layer: This sub-layer supports


negotiation of the IP address and IP compression protocol parameters. In
general a mobile station does not have a permanent IP address. Therefore, the
IP address needs to be negotiated and obtained from the network. IPCP does
this job of leasing an IP address when the transport connection is established.
The IP address is discarded when the connection is closed. This is similar to
obtaining the IP address from a DHCP (Dynamic Host Configuration
Protocol) server in a LAN environment.

 Radio Link Protocol Layer: This layer provides octet stream service over the
air. This service is responsible for reducing the error rate over the forward
and reverse channels. There is no direct relationship between PPP packet and
the traffic channel frame. A large packet may span multiple traffic channel
frame. A single traffic channel frame may contain multiple PPP packets. RLP
frames may be transported as traffic or signaling via data burst message.

Short Message Service

SMS in IS-95 is similar to SMS in GSM. Unlike GSM, the maximum size of a
SMS in IS-95 is 120 octets. The SMS in IS-95 work the same way as in GSM. It
supports SMPP protocol and other features as in GSM. Like in GSM, the SMS in IS-
95 uses the signaling channel for data transfer. SMS administration features include
storage, profiling, verification of receipt and status enquiry capabilities.

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5.2. Wireless LAN

5.2.1. Introduction
Wireless Local Area Network (LAN) is a local area data network without
wires. Wireless LAN is also known as WLAN in short. Mobile users can access
information and network resources through wireless LAN as they attend meetings,
collaborate with other users, or move to other locations in the premises. Wireless
LAN is not a replacement for the wired infrastructure. It is implemented as an
extension to a wired LAN within a building or campus.

5.2.2. Wireless Advantages


Schools, campuses, manufacturing plants, hospitals and enterprises install
wireless LAN systems for many reasons. Some of these are:

• Mobility: Productivity increases when people have access to data and


information from any location. Decision-making capability based on real-
time information can significantly improve work efficiency. Wireless LAN
offers wire-free access to information within the operating range of the
WLAN.

• Low Implementation Costs: WLANs are easy to set up, relocate, change and
manage. Networks that frequently change, both physically and logically, can
benefit from WLAN's ease of implementation. WLANs can operate in
locations where installation of wiring may be impractical.

• Installation Speed and Simplicity: Installing a wireless LAN system can be


fast and easy and can eliminate the need to install cable through walls and
ceilings.

• Network Expansion: Wireless technology allows the network to reach


where wires cannot reach.

• Reduced Cost-of-Ownership: While presently the initial investment


required for Wireless LAN hardware is higher than the cost of wired LAN
hardware, overall installation expenses and life-cycle costs is expected to be

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significantly lower. Long-term cost benefits are the greatest in dynamic


environments requiring frequent moves, adds and changes.

• Higher User to Install Base Ratio: Wireless environment offers a higher user
to capacity ratio. For example in a wired network like telephone, physical
wire needs to be laid for each and every subscriber. Whereas, for a cellular
network the ratio between subscribers and available channel is from 10 to 25
or even more. This means that if there is capacity for 100 channels, the
network operator can safely have 2500 subscribers. Likewise in a wireless
LAN, the network can offer a very high level of return on investment.

• Reliability: One of the common causes of failure in wired network is


downtime due to cable fault. WLAN is resistant to different types of cable
failures.

• Scalability: Wireless LANs can be configured in a variety of topologies to


meet the needs of specific applications and installations. Configurations are
easily changed and range from peer-to-peer networks suitable for a small
number of users to full infrastructure networks of thousands of users that
allow roaming over a broad area.

• Usage of ISM band: Wireless LAN operates in the unregulated ISM


(Industrial Scientific and Medical) band (2.40 GHz to 2.484 GHz, 5.725 GHz
to 5.850 GHz) available for use by anyone. A user need not go to the
government to get a license to use the wireless LAN. In India 2.4 GHz band
is made free for use in WLANs. The 5.7 GHz band is not yet unregulated as
it may conflict with the C-band of satellite.
Wireless LAN is also commercially known as WiFi or Wi-Fi. Wi-Fi is an
acronym for Wireless Fidelity. The Wi-Fi™ logo is a registered trademark of the
Wireless Ethernet Compatibility Alliance a group founded by many companies that
develop 802.11 based products.

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Wireless LAN Evolution


Wireless LAN development started in an unstructured way. Some vendors
started offering wireless communication between the corporate LAN and mobile
devices (like laptop computers). This is like using a wireless keyboard or a wireless
mouse. Protocols and interfaces were proprietary. However, within a short period of
time many vendors started offering products in this space. These products were
incompatible and soon interoperability became an issue. As IEEE is responsible for
maintaining Ethernet LAN standards, IEEE assumed the responsibility of defining
the wireless Ethernet LAN standards. The initial standard was published in June
1997. All these early 802.11 systems are first generation systems.

It was not until the introduction of the 11-Mbps 802.11b standard in


September 1999 that the horizontal WLAN market achieved some semblance of
legitimacy. Also, standards like 802.11a and 802.llg offered much higher bandwidth.
All these are second generation WLANs. Second generation WLANs extended the
security through 802.Ix specifications and offered horizontal roaming. In a
horizontal roaming, a user can move from one AP to another AP seamlessly.

In third generation WLANs, vertical roaming will be possible. Vertical


roaming will provide seamless roaming between different networks. Third
generation WLANs will integrate with third generation (3G) telecom networks.
These WLANs will eliminate the boundaries between enterprise LAN (both wireline
and wireless] systems and the public wireless systems for seamless roaming. It will
extend the application of IP mobility standards. Also the security system is being
extended. These will be achieved through standards like 802.11f and 802.11i.
Wireless LAN Applications
There are many areas and applications of wireless LAN. Wireless LAN is best
suited for dynamic environments. Following are some of the examples.
 Office/Campus Environment- WLAN is very useful in office environments and
buildings with a big campus. In big buildings or in campuses people move
between floors, rooms, indoors and outdoors. In an office environment, a
person can move with his laptop to the meeting room and continue working.
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In a university campus, a student can move from the library to the cafeteria
and continue working. In a hotel, a guest can move to the pool and work. In a
hospital, a doctor can carry the patient information with him while on a
regular round.
 Factory Shop Floor -This includes environments like factory shop floor,
warehouse, exhibition sites, retail shops, labs etc. These are very dynamic
environments, where floor layouts change very frequently; objects within the
building are constantly moving. Laying cables and setting up a wired LAN in
these kinds of facilities are almost impossible. Wireless LAN can be very
useful in such situations.

 Homes - In homes WLAN can be used for convergence applications. These will
include networking of different home devices like phones, computers and
appliances.

 Workgroup Environment - Any set-up where small workgroups or teams need to


work together be it within a building or in the neighbourhood, WLAN can be
very useful. This may include a survey team on top of a hill or rescue
members after a natural disaster or an accident site. WLAN can be very useful
in civil construction sites as well.

 Heritage Buildings - There are many building of national heritage, where a data
network needs to be set up. In a very old church for example, if we need to
setup a virtual reality show, it is difficult to install a wired LAN. Wireless
LAN can solve the problem.

 Public Places - This includes airports, railway stations or places where many
people assemble and need to access information.

 War/Defense Sites - When there is a war or war game, access to networks help.
There is some major research going on in US on mobile ad hoc networks for
defense establishments.

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5.2.3. Wireless LAN Architecture


There are two major approaches today for deploying WLAN Networks in the
enterprise.
The first architecture to be presented is the so-called “Centralized” WLAN
architecture. The Centralized architecture requires one or more servers or special
purpose switches (Mobility controller) to be deployed in conjunction with wireless
access points. By Default In the centralized approach, all wireless traffic is sent
through the WLAN Switch. In either case, the centralized approach is considered to
be“Overlay” architecture. That is, it rides on top of the existing Ethernet Network.
Another approach is the “Distributed” WLAN architecture. AP have Built-in
WLAN Security, layer 2 bridging, and access control features. Depending on the
number of Aps required, Centralized management may be required. Distributed AP
vendors may provide Centralized management tools or the AP’s will act as Virtual
Mobility Controller.
Data Forwarding
The “Distributed” WLAN architecture approach is that the wireless traffic
load is literally distributed across the Aps and does not depend on a centralized
element to process all of the wireless traffic. A “Centralized” WLAN architecture
offers more choices, and thus more flexibility, than a “Distributed” WLAN
architecture model. With a controller, organizations can choose to forward traffic
locally at the APs (similar to the method used in “Distributed” WLAN architecture),
Or they can choose to tunnel certain types of traffic back to the controller for security
reasons. With a “Centralized” WLAN architecture organizations have the flexibility
to mix and match these approaches as appropriate.
Deployment Scenario
In a “Distributed” WLAN architecture, you need to reconfigure your access
layer with the addition of each new AP. Since it is necessary to configure all virtual
LANs (VLANs) on the switch port that is needed by each new AP, your network
administrator needs to configure the wiring closet switches that each new AP
connects to. For example, you may have a VLAN for guest access, a VLAN for

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corporate access, and a VLAN for special access (such as VoIP). All these VLANs
must be configured each time you add a new AP.
With a “Centralized” WLAN architecture, it is infinitely easier to add Aps
when we send traffic to controller with Tunnel mode. The access layer is configured
once at the handoff to the controller and the system manages the rest. The
centralized controller provides rich functionality for automating deployment
complexity, eliminating the need for frequent, error-prone changes to the access
layer. You simply plug in the AP and it automatically self-configures.
Still if we required some AP to be configured for (bridge mode) sending
traffic locally in “Centralized” WLAN architecture, those AP’s it is necessary to
configure all virtual LANs (VLANs) on the switch port that is needed.

5.2.4. Types of Wireless LAN


There are different types and flavors of wireless local area networks. Some of
the most popular ones are:
 802.11 - In June 1997, the IEEE finalized the initial specification for wireless
LANs: IEEE 802.11. This standard specifies a 2.4 GHz frequency band with
data rate of 1 Mbps and 2 Mbps. This standard evolved into many variations
of the specification like 802.11b, 802.11a, 802.11g, etc. using different encoding
technologies. Today these standards offer a local area network of bandwidths
going up to a maximum of 54Mbps.
 HyperLAN - HyperLan began in Europe as a specification (EN 300 652)
ratified in 1996 by the ETSI Broadband Radio Access group. HyperLAN/1,
the Network current version works at the 5 GHz band and offers up to 24
MBps bandwidth. Next version HyperLAN/2 will support a band width of 54
Mbps with QoS support. This will be able to carry Ethernet frames, ATM
cells, IP packets and support data, video, voice and image.
 HomeRF - In 1998, the HomeRF Working Group offered to provide an
industry specification to offer Shared Wireless Access Protocol (SWAP). This
standard will offer interoperability between PC and consumer electronic

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devices within the home. SWAP uses frequency hopping spread spectrum
modulation and offers 1 Mbps and 2 Mbps at 2. 4 GHz frequency band.
 Bluetooth - Bluetooth was promoted by big industry leaders like IBM,
Ericsson, Intel, Lucent, 3Com, Microsoft, Nokia, Motorola, and Toshiba. It was
named after Harold Bluetooth, King of Denmark during 952 t o 995 A.D., who
had a vision of a world with cooperation and interoperability. Bluetooth is
more of a wireless Personal Area Network (PAN) operating at 2. 4 GHz band
and offers 1Mbps data rate. Bluetooth uses frequency hopping spread-
spectrum modulation with relatively low power and smaller range (about 10
meters).

 MANET - Manet is a working group within the IETF to investigate and


develop the standard for Mobile ad hoc NETworks.

5.2.5. Mobility in Wireless LAN

When a station wants to access an existing BSS (either after power-up, sleep
mode, or physically entering into the BSS area), the station needs to get
synchronization information from the AP (or from the other stations when in ad hoc
mode).
The station can get this information by one of two means:

• Passive Scanning - In this case the station just waits to receive a Beacon Frame
from the AP, or

• Active Scanning - In this case the station tries to locate an Access Point by
transmitting Probe Request Frames, and waits for Probe Response from the AP.
The Authentication Process
Once a wireless station has located an AP and decides to join its BSS, it goes
through the authentication process. This is interchange of authentication information
between the AP and the station, where the WLAN device proves its identity.

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The Association Process


Once the station is authenticated, it then starts the association process which
is the exchange of information about the stations and BSS capabilities, and which
allows the DSS (the set of APs) to know about the current position of the station. A
station is capable of transmitting and receiving data frames only after the association
process is completed.

Roaming

Roaming is the process of moving from one cell (or BSS) to another without
losing connection. This function is similar to the cellular phones' handover, with two
main differences:
 On a packet-based LAN system, the transition from cell to cell may be per-
formed between packet transmissions, as opposed to telephony where the
transition may occur during a phone conversation.
 On a voice system, a temporary disconnection during handoff does not affect
the conversation. However, in a packet-based environment it significantly
reduces performance because retransmission is performed by the upper layer
protocols.
The 802.11 standard does not define how roaming should be performed, but
defines the basic tools. These include active/passive scanning, and a re-association
process, where a station that is roaming from one AP to another becomes associated
with the new AP. The Inter-Access Point Protocol (LAPP) specification addresses a
common roaming protocol enabling wireless stations to move across multivendor
access points. IAPP is the scope of IEEE standard 802.1 If.

IAPP defines two basic protocols, viz., Announce protocol and Handover
protocol. The announce protocol provides coordination information between access
points. This information relates to network wide configuration information about
active APs. The handover protocol allows APs to coordinate with each other and
determine the status of a station. When a station associates with a different AP, the
old AP forwards buffered frames for the station to the new AP. The new AP updates

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the necessary tables in the MAC layer to ensure that the MAC level filtering will
forward frames appropriately. This type of roaming is called horizontal roaming.

Mobile IP is another protocol that is used to allow application layer roaming.


Using Mobile IP, a mobile station can move from one type of network to another
type of network. For example in an IMT-200 situation, the station moves from
wireless LAN environment to a 3G wireless MAN environment.

5.2.6. Wireless LAN Security

In a wired network one has to be physically connected to transfer or receive


data. This implies that it is possible to control the users in the network by controlling
the physical access. Using a wireless network means using a radio transmitter and
receiver. With varying degrees, radio signals will penetrate most building materials.
Therefore, it is not possible to set up absolute physical boundary and expect that no
outsider will be able to intrude into the network. With wireless networks we have
no control of who might be receiving and listening to the transmissions. It could be
someone in the building across the road, in a van parked in the parking lot or
someone in the office above. Therefore, it is important that we understand the
vulnerabilities of the wireless LAN and take necessary precautions.

As a part of the original specification, IEEE 802.11 included several security


features, such as open system and shared key authentication modes; the Service Set
Identifier (SSID); and Wired Equivalent Privacy (WEP). Each of these features
provides varying degrees of security.
Limiting RF Transmission

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It is important to consider controlling the range of RF transmission by an


access point. It is possible to select proper transmitter/antenna combination that will
help transmission of the wireless signal only to the intended coverage area.
Antennas can be characterized by two features—directionality and gain. Omni-
directional antennas have a 360-deg coverage area, while directional antennas limit
coverage to better-defined areas.

Service Set Identifier (SSID)

According to the 802.11 standard, a mobile station has to use the SSID of the
access point for association between the NIC (Network Interface Card) in the client
and the AP. The SSID is a network name (Id of the BSS or Cell) that identifies the
area covered by an AP. The AP periodically broadcasts its SSID as a part of the
management frame (beacon packet). The broadcast of beacon packet is necessary for
clock synchronization. Unfortunately, as management frames of 802.11 are always
sent in the clear, an attacker can easily listen on the wireless media for the
management frames and discover the SSID to connect to the AP. The SSID can be
used as a security measure by configuring the AP to broadcast the beacon packet
without its SSID. The wireless station wishing to associate with the AP must have its
SSID configured to that of the AP. If the SSID is not known, management frames sent
to the AP from the wireless station will be rejected. It is also advised that the SSID of
the AP is changed from the factory set defaults to some name, which is difficult to
guess.
MAC Address Access Control
Many access points support MAC address filtering. This is similar to IP
Filtering. The AP manages a list of MAC addresses that are allowed or disallowed in
the wireless network. The idea is that the MAC address of the network card is
unique and static. By controlling the access from known addresses, the administrator
can allow or restrict the access of network only to known clients.

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Authentication Modes

Two types of client authentication are defined in 802.11: Open System


Authentication and Shared Key Authentication. Open system authentication is no
authentication at all. Shared key authentication on the other hand is based on the
fact that both stations taking part in the authentication process have the same
"shared" key.
It is assumed that this key has been transmitted to both stations through some
secure channel other than the wireless media itself. In typical implementations, this
is set manually on the client station and the AP. The authenticating station receives a
challenge text packet (created using the WEP Pseudo Random Number Generator
(PRNG)) from the AP. The station encrypts this PRNG using the shared key, and
sends it back to the AP. If, after decryption, the challenge text matches, then one-
way authentication is successful. To obtain mutual authentication, the process is
repeated in the opposite direction.
WEP (Wired Equivalent Privacy)

WEP was designed to protect users of a WLAN from casual eavesdropping


and was intended to offer following facilities:

• Reasonably strong encryption. It relies on the difficulty of recovering the


secret key through a brute force attack. The difficulty grows with the key
length.

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• Self-synchronizing. Each packet contains the information required to


decrypt it. There is no need to deal with lost packets.

• Efficient. It can be implemented in software with reasonable efficiency.

• Exportable. Limiting the key length leads to a greater possibility of export


beyond US.
The WEP algorithm is the RC4 cryptographic algorithm from RSA Data
Security. RC4 uses stream cipher technique. It is a symmetric algorithm and uses the
same key for both enciphering and deciphering the data. For each transmission, the
plaintext is bitwise XORed with a pseudorandom key stream to produce cipher text.
For decryption the process is reversed. The algorithm operates as follows:

 It is assumed that the secret key has been distributed to both the transmitting
and receiving stations by some secure means.
 On the transmitting station, the 40-bit secret key is concatenated with a 24-bit
Initialization Vector (IV) to produce a seed for input into the WEP PRNG
(Pseudo Random Number Generator).
 The seed is passed into the PRNG to produce a stream (key stream) of pseudo
random octets.
 The plaintext PDU is then XORed with the pseudo-random key stream to
produce the cipher text PDU.
 This cipher text PDU is then concatenated with the 24-bits IV and transmitted
on the wireless media.
 The receiving station reads the IV and concatenates it with the secret key,
producing the seed that it passes to the PRNG.
 The receiver's PRNG produces identical key stream used by the transmitting
station. When this PRNG is XORed with the cipher text, the original plaintext
PDU is produced.
It is worth mentioning that the plaintext PDU is also protected with a CRC to
prevent random tampering with the cipher text in transit.

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Possible Attacks
The possible security attacks on wireless LAN are:

• Passive attacks to decrypt traffic based on statistical analysis.

• Active attack to inject new traffic from unauthorized mobile stations, based
on known plaintext.

• Active attacks to decrypt traffic, based on tricking the access point.

• Dictionary-building attack that, after analysis of about a day's worth of


traffic, allows real time automated decryption of all traffic.

• Hijacking a session: Following successful authentication, it is possible to


hijack the session.
Analysis suggests that though these attacks are not common, it is possible to
perform them using inexpensive off-the-shelf equipment.
802.1X Authentication

To prevent attacks on wireless LAN, the IEEE specification committee on


802.11 included the 802.lx authentication framework. The 802.lx framework provides
the link layer with extensible authentication, normally seen in higher layers. 802.lx
requires three entities:

• The supplicant Resides on the wireless LAN client

• The authenticator Resides on the access point

• The authentication server Resides on the server authenticating the client (e.g.,
RADIUS Kerberos, or other servers)

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These are logical entities on different network elements. In a single network


there could be many points of entry. These entries are through access points. Once
the link between a supplicant (wireless station) and an authenticator (AP) is
achieved, the connection is passed to the authentication server. The AP authenticates
the supplicant through the authentication server. If the authentication is successful,
the authentication server instructs the authenticator to allow the supplicant to access
the network services. The authenticator works like a gatekeeper.

The authenticator creates one logical port per client, based on the client's
association ID. This logical port has two data paths. The uncontrolled data path
allows network traffic through to the network. The controlled data path requires
successful authentication to allow network traffic through. In order to obtain
network connectivity, a wireless client must associate with the AP. Complete
association with an AP involves three states:
 Unauthenticated and unassociated
 Authenticated and unassociated
 Authenticated and associated
IEEE 802.lx offers flexibility in authentication and possible encryption. After
the link has been established, PPP (Point-to-Point Protocol) provides for an optional
authentication phase before proceeding to the network layer protocol phase. This is
called EAP (Extensible Authentication Protocol). Through the use of EAP, support
for a number of authentication schemes may be added, including Smart cards,
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Kerberos, Public Key, One Time Passwords, CHAP (Challenge Handshake


Authentication Protocol), or some other user-defined authentication systems.

There are still some vulnerabilities in the EAP. To overcome this, a new
standard is being proposed in IETF to override the EAP proposal. This new standard
is called PEAP (Protected EAP). PEAP uses an additional phase of security over and
above EAP.
Wireless VPN

Virtual Private Network technology (VPN) has been used to secure


communications among remote locations via the Internet since the 1990s. It is now
being extended to wireless LAN. VPNs were traditionally used to provide point-to-
point encryption for long Internet connections between remote users and the
enterprise networks. VPNs have been deployed in wireless LANs as well. When a
wireless LAN client uses a VPN tunnel, communications data remains encrypted
until it reaches the VPN gateway, which sits behind the wireless AP. Thus, intruders
are effectively blocked from intercepting all network communications.
802.1 l i
Task Group T within IEEE 802.11, is developing a new standard for WLAN
security. The proposed 802.1 li standard is designed to embrace the authentication
scheme of 802.lx and EAP while adding enhanced security features, including a new
encryption scheme and dynamic key distribution. Not only does it fix WEP, it takes
wireless LAN security to a higher level.
The proposed specification uses the Temporal Key Integrity Protocol (TKIP)
to produce a 128-bit 'temporal key' that allows different stations to use different keys
to encrypt data. TKIP introduces a sophisticated key generation function, which
encrypts every data packet sent over the air with its own unique encryption key.
Consequently, TKIP greatly increases the complexity and difficulty of decoding the
keys. Intruders will not have enough time to collect sufficient data to decipher the
key.

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802.1 li also endorses the Advanced Encryption Standard (AES) as a


replacement for WEP encryption. AES has already been adopted as an official
government standard by the US Department of Commerce. It uses a mathematical
ciphering algorithm that employs variable key sizes of 128-, 192- or 256-bits, making
it far more difficult to decipher than WEP. AES, however, is not readily compatible
with today's Wi-Fi Certified WLAN devices. It requires new chipsets, which, for
WLAN customers, means new investments in wireless devices. Those looking to
build new WLANs will find it attractive. Those with previously installed wireless
networks must justify whether AES security is worth the cost of replacing
equipment.

5.3. Next Generation Networks

5.3.1. OFDM
Orthogonal Frequency Division Multiplexing or OFDM is a modulation
format that is being used for many of the latest wireless and telecommunications
standards.
OFDM has been adopted in the Wi-Fi arena where the standards like 802.11a,
802.11n, 802.11ac and more. It has also been chosen for the cellular
telecommunications standard LTE / LTE-A, and in addition to this it has been
adopted by other standards such as WiMAX and many more.
Orthogonal frequency division multiplexing has also been adopted for a
number of broadcast standards from DAB Digital Radio to the Digital Video
Broadcast standards, DVB. It has also been adopted for other broadcast systems as
well including Digital Radio Mondiale used for the long medium and short wave
bands.
Although OFDM, orthogonal frequency division multiplexing is more
complicated than earlier forms of signal format, it provides some distinct advantages
in terms of data transmission, especially where high data rates are needed along
with relatively wide bandwidths.
What is OFDM?

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OFDM
DM is a form of multicarrier modulation. An OFDM signal consists of a
number of closely spaced modulated carriers. When modulation of any form - voice,
data, etc. is applied to a carrier, then sidebands spread out either side. It is necessary
for a receiverr to be able to receive the whole signal to be able to successfully
demodulate the data. As a result when signals are transmitted close to one another
they must be spaced so that the receiver can separate them using a filter and there
must be a guard band between them. This is not the case with OFDM. Although the
sidebands from each carrier overlap, they can still be received without the
interference that might be expected because they are orthogonal to each another.
This is achieved by having the carrier spacing
spacing equal to the reciprocal of the symbol
period.

To see how OFDM works, it is necessary to look at the receiver. This acts as a
bank of demodulators, translating each carrier down to DC. The resulting signal is
integrated over the symbol period to regenerate the data from that carrier. The same
demodulator also demodulates the other carriers. As the carrier spacing equal to the
reciprocal of the symbol period means that they will have a whole number of cycles
in the symbol period and their contribution
contribut will sum to zero - in other words there is
no interference contribution.

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One requirement of the OFDM transmitting and receiving systems is that they
must be linear. Any non-linearity will cause interference between the carriers as a
result of inter-modulation distortion. This will introduce unwanted signals that
would cause interference and impair the orthogonality of the transmission.
In terms of the equipment to be used the high peak to average ratio of multi-
carrier systems such as OFDM requires the RF final amplifier on the output of the
transmitter to be able to handle the peaks whilst the average power is much lower
and this leads to inefficiency. In some systems the peaks are limited. Although this
introduces distortion that results in a higher level of data errors, the system can rely
on the error correction to remove them.
Data on OFDM
The data to be transmitted on an OFDM signal is spread across the carriers of
the signal, each carrier taking part of the payload. This reduces the data rate taken by
each carrier. The lower data rate has the advantage that interference from reflections
is much less critical. This is achieved by adding a guard band time or guard interval
into the system. This ensures that the data is only sampled when the signal is stable
and no new delayed signals arrive that would alter the timing and phase of the
signal.

The distribution of the data across a large number of carriers in the OFDM
signal has some further advantages. Nulls caused by multi-path effects or
interference on a given frequency only affect a small number of the carriers, the
remaining ones being received correctly. By using error-coding techniques, which

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does mean adding further data to the transmitted signal, it enables many or all of the
corrupted data to be reconstructed within the receiver. This can be done because the
error correction code is transmitted in a different part of the signal.
OFDM advantages
OFDM has been used in many high data rate wireless systems because of the
many advantages it provides.
 Immunity to selective fading: One of the main advantages of OFDM is that
is more resistant to frequency selective fading than single carrier systems
because it divides the overall channel into multiple narrowband signals that
are affected individually as flat fading sub-channels.
 Resilience to interference: Interference appearing on a channel may be
bandwidth limited and in this way will not affect all the sub-channels. This
means that not all the data is lost.
 Spectrum efficiency: Using close-spaced overlapping sub-carriers, a
significant OFDM advantage is that it makes efficient use of the available
spectrum.
 Resilient to ISI: Another advantage of OFDM is that it is very resilient to
inter-symbol and inter-frame interference. This results from the low data rate
on each of the sub-channels.
 Resilient to narrow-band effects: Using adequate channel coding and
interleaving it is possible to recover symbols lost due to the frequency
selectivity of the channel and narrow band interference. Not all the data is
lost.
 Simpler channel equalization: One of the issues with CDMA systems was
the complexity of the channel equalization which had to be applied across the
whole channel. An advantage of OFDM is that using multiple sub-channels,
the channel equalization becomes much simpler.
OFDM disadvantages
OFDM has been widely used, there are still a few disadvantages to its use
which need to be addressed when considering its use.
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 High peak to average power ratio: An OFDM signal has a noise like
amplitude variation and has a relatively high large dynamic range, or peak to
average power ratio. This impacts the RF amplifier efficiency as the amplifiers
need to be linear and accommodate the large amplitude variations and these
factors mean the amplifier cannot operate with a high efficiency level.
 Sensitive to carrier offset and drift: Another disadvantage of OFDM is that
is sensitive to carrier frequency offset and drift. Single carrier systems are less
sensitive.

5.3.2. MPLS
Multiprotocol Label Switching (MPLS) is a Layer-2 switching technology.
MPLS-enabled routers apply numerical labels to packets, and can make forwarding
decisions based on these labels. The MPLS architecture is detailed in RFC 3031.
MPLS reduces CPU-usage on routers, by allowing routers to make forwarding
decisions solely on the attached label, as opposed to parsing the full routing table.
Labels can based on a variety of parameters:
 Destination IP network
 Source IP address
 QoS parameters
 VPN destination
 Outgoing interface
 Layer-2 circuit
MPLS is not restricted to IP, or any specific Layer-2 technology, and thus is
essentially protocol-independent. Labels are applied to and removed from packets
on edge Label Switch Routers (edge LSRs). Only edge routers perform a route-table
lookup on packets. All core routers (identified simply as LSRs) in the MPLS network
forward solely based on the label. As a packet traverses the core MPLS network, core
routers will swap the label on hop-by-hop basis. MPLS is completely dependent on
Cisco Express Forwarding (CEF) to determine the next hop.

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Cisco Express Forwarding (CEF)


Multilayer switches contain both a switching and routing engine. A packet
must first be “routed,” allowing the switching engine to cache the IP traffic flow.
After this cache is created, subsequent packets destined for that flow can be
“switched” as opposed to “routed,” reducing latency. This concept is often referred
to as route once, switch many. Cisco refers to this type of Multilayer switching as
NetFlow switching or route cache switching.
As is their habit, Cisco replaced NetFlow multilayer switching with a more
advanced method called Cisco Express Forwarding(CEF). CEF is enabled by default
on all Catalyst multi-layer switches (at least, those that support CEF). CEF cannot
even be disabled on the Catalyst 3550, 4500 and 6500. CEF contains two basic
components:
 Layer 3 Engine – Builds the routing table and then “routes” data
 Layer 3 Forwarding Engine –“Switches” data based on the FIB.
The Layer 3 Engine builds its routing table using either static routes, or routes
dynamically learned through a routing protocol (such as RIP or OSPF). The routing
table is then reorganized into a more efficient table called the Forward Information
Base (FIB). The most specific routes are placed at the top of the FIB. The Layer 3
Forwarding Engine utilizes the FIB to then “switch” data in hardware, as opposed to
“routing” it through the Layer 3 Engine’s routing table. The FIB contains the
following information:
 Destination networks
 Destination masks
 Next-hop addresses
 The MAC addresses of each next hop (called the Adjacency Table)
To view the CEF FIB table: Switch# show ip cef
The MPLS Label
Two forms of MPLS exist:
 Frame Mode MPLS – utilizes a 32-bit label that is injected between the Layer-
2 and Layer-3 headers.
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 Cell Mode MPLS – used with ATM, and utilizes the VPI/VCI fields ATM
header as the label.
This guide will concentrate on Frame Mode MPLS. The32-bit label has the following
format:

 Label (20 bits)


 Experimental (3 bits) – This field is officially undefined, but is used by Cisco
as an IP precedence value.
 Bottom-of-Stack (1 bit) – This field indicates the last label, as multiple labels
are supported in the same packet. A value of 1
identifies the last label in the stack.
 TTL (8 bits) – This field indicates the number of router this label can ‘live’
through.
An Ethernet header is modified to indicate the presence of an MPLS label.
 0x8847 – indicates a labeled unicast IP packet
 0x8848 – indicates a labeled multicast IP packet
The MPLS Components
MPLS router designations include:
 LSR (Label Switch Router)– responsible for forwarding packets through the
provider core based on the packet’s label. Cisco refers to this as a P (provider)
router.
 Edge LSR (Label Edge Router)– responsible for adding or removing labels
from packets. Cisco refers to this as a PE (provider edge) router.
 Non-Label Routers – Cisco refers to this as C (customer) routers
 LSRs perform the following functions:
 Control Plane- exchanges routing and label information
 Data Plane- forwards actual packets based on label information

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The Control Plane, in charge of information exchange, builds and maintains the
following tables:
 Routing Table – routing information is exchanged between LSRs using a
routing protocol, such as IGRP, EIGRP, IS-IS, OSPF, or BGP.
 Label Information Base (LIB) –label information is exchanged between LSRs
using a label protocol, such as LDP (Label Distribution Protocol) or TDP (Tag
Distribution Protocol).
LDP is now default on Cisco devices, and uses TCP port 646. TDP is a Cisco-
proprietary label protocol, and uses TCP port711. Label convergence will occur after
routing convergence is completed. Label protocols require the underlying routing
infrastructure in order to function.
The Data Plane, in charge of information forwarding, maintains the following
tables:
 The CEF Forwarding Information Base (FIB) –for forwarding unlabeled
packets. Contains destination IP networks, and the Layer-2 address of the
next-hop router.
 Label Forwarding Information Base (LFIB)– for forwarding labeled packets.
The MPLS Process
There are four scenarios detailing how LSRs forward packets:
 An unlabeled IP packet is received, and is routed unlabeled to the next
hop.
 An unlabeled IP packet is received; a label is inserted in the header, and is
switched to the next hop.
 A labeled IP packet is received; the label is swapped, and is switched to
the next hop.
 A labeled IP packet is received; the label is stripped off, and is routed to
the next hop or destination.
Frame-mode MPLS performs as follows:
 An edge LSR receives a packet.

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 The edge LSR performs a routing table lookup to determine the next hop
(or exit interface).
 If destined for the MPLS network, the edge router inserts the label
between the Layer-2 and Layer-3 headers.
 The edge LSR forwards the labeled packet to the core LSR.
 Core LSRs will route solely based on the label, and will not perform a
routing table lookup.
MPLS VPNs
Cisco identifies two key categories of VPNs:
 Overlay – connections are set up and maintained by the service provider.
However, the provider has no knowledge of, and does not participate in,
the customer’s routing infrastructure.
 Peer-to-Peer– the provider directly participates in routing the customer’s
infrastructure. Routes from multiple customers are not kept separate.
This may require customers to readdress their networks.
MPLS VPNs provide the best of both words. Advantages of MPLS VPNs
include:
 The provider directly participates in routing the customer infrastructure.
 Peer-to-peer peering is not required, leading to a scalable infrastructure.
 Customer networks do not need to be readdressed
 Routes from multiple customers are kept separate.
MPLS VPNs use two labels. MPLS separates customer routes by assigning
each a unique Virtual Routing Instance (VRI), stored in a Virtual Routing and
Forwarding (VRF) table. If the addressing structure of multiple clients overlaps, each
network is assigned a unique 64-bit route distinguisher (RD). The IP network and
RD combination is called a VPNv4 address. Route targets are BGP attributes that
designate MPLS VPN membership of routes.

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5.3.3. Wireless asynchronous transfer Mode


Introduction
ATM (Asynchronous Transfer Mode) has been advocated as an important
technology for all types of services and networks. Most people believe that ATM will
be the standard for the future B-ISDN (Broadband Integrated Services Digital
Network). From the service point of view, ATM combines both the data and
multimedia information into the wired networks while scales well from backbones
to the customer premises networks. To ensure the success of ATM, lots of the design
issues have been standardized by ATM Forum. Wireless personal communication
networks (PCN) has been growing very fast in the last decade. Now a day, laptop,
cellular phone, and pagers are very popular. Many systems have been developed to
provide different services, such as, Personal Communications Service (PCS), Portable
Telephone Systems, and Satellite Communications System. Usually, these services
do not guarantee QoS (Quality of Services) so they are not suitable for the fast
growing multimedia applications. Due to the success of ATM on wired networks,
wireless ATM (WATM) is a direct t result of the ATM "everywhere" movement.
WATM can be viewed as a solution for next-generation personal
communication networks, or a wireless extension of the B-ISDN networks, which
will support integrated data transmission (data, voice, video) with guaranteed QoS
WATM architecture and protocols are already being well discussed. The ATM
Forum are currently involved in defining the baseline of WATM system. The
specification for both mobility control in ATM infrastructure networks, and seamless
radio extension of ATM to mobile devices are under development.
Wireless ATM Reference Models

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A system reference model for WATM is the system consists of a fixed ATM
network infrastructure and a radio access segment. In the fixed ATM network, the
switches, which communicate directly with wireless station or wireless end user
devices, are mobility enhanced ATM switches. These switches setup connections on
behalf of the wireless devices. They serve as the "entrance" to the infrastructure
wired ATM networks. The other ATM switching elements in the wired ATM
networks remain unchanged. Based on the different types of wireless applications,
the radio access segment falls into a number of areas which may need different
solutions.
 Fixed Wireless Components - in fixed wireless LANs or network
interconnection via satellite or microwaves links, the end user devices and
switching devices are fixed. They establish connections with each other via
wireless channel, not through cable. In these kinds of applications, the data
transmissions are wireless, yet without mobility. Since the user devices do not
roam around, some design issues, e.g. handover, location management, and
re-routing, are not presented.
 Mobile End Users - In digital cellular, PCS, and wireless LANs, the
end user devices, which are mobile, communicate directly with the fixed
network switching devices via wired or wireless channels. To support the
ATM connections, the end user devices are required to be equipped with a
Wireless Terminal Adaptor which communicates with the Wireless Access
Point in the fixed switching elements.
 Mobile Switches with Fixed End Users - End user devices are
connected to switches via wired or wireless channels. The end user device
and the switch, as a unit, are mobile. There can be more than one end user
devices attach to one switch. An end user device is fixed to one switch instead
of roaming around different switches. The switch is responsible to establish
connections with the fixed infrastructure network component, either through
wired channel or wireless channel. In this case, Wireless Access Points and

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Wireless Terminal Adapters are needed by the fixed mobility enhanced ATM
switches and the mobile switches.
 Mobile Switches with Mobile End Users - In this case, end user devices
are mobile. There are also some mobile switching elements. When the end
user wants to establish a connection, it first setups a connection with a mobile
switch, which then setups a connection with the fixed network switches,
either directly, or via another mobile switches. Wireless Access Points and
Wireless Terminal Adapters are also needed to support the mobility.
 Interworking with PCS - In PCS networks, the users are PCS terminals.
PCS terminals send data to proper PCS base stations via wireless link, which
then establish connections to the fixed network switching elements through a
base station controller. The base station controller is a logical element which
function as the ATM<->PCS
 Wireless Ad Hoc Networks - An Ad Hoc network is the cooperative
engagement of a collections of mobile terminals without the required
intervention of any centralized access point. An auto-configuration of a
wireless ATM network will be required for this kind of application. In
wireless Ad Hoc Networks, an end user can communicate with the mobility
enhanced ATM switches either directly, or via a central controller.
WATM Design Issues
WATM adopts ATM to provide the data communications services so the
overall architecture is based on the ATM protocol stack. To support mobility,
appropriate extensions need to be added to the ATM protocol stack. The wireless
segment of the network will require new mobility functions.

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Wireless ATM Protocol Architecture

The WATM items are divided into two distinct parts: Mobile ATM (Control
Plane), and Radio Access Layer (Wireless Control). Mobile ATM is dealing with the
higher-layer control/signaling functions needed to support mobility. These
control/signaling include handover, location management, routing, addressing, and
traffic management. Radio Access Layer is responsible for the radio link protocols
for wireless ATM access. Radio Access Layers consists of PHY (Physical Layer),
MAC (Media Access Layer), DLC (Data Link Layer), and RRC (Radio Resource
Control).
Radio Access Layer
To support wireless communication, new wireless channel specific physical,
medium access and data link layers are need to be added below the ATM network
layer. These layers are called Radio Access Layer in the WATM network. The
following sections address the design issues of the Radio Access Layer.
 Physical Layer (PHY) - While a fixed station may own an 25 Mbit/s up to
155 Mbit/s data rate ATM link, a 25 Mbit/s data link in a wireless
environment is currently difficult to implement. A several GHz spectrum
would be required to provide high speed wireless transmission. Currently, 5
GHz band is considered to be used to provide 51 Mbit/s channel with
advanced modulation and special coding techniques. Although 155 Mbit/s is
unreachable due to the limitation of today's techniques, people believe that it

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will soon be available in the 60 GHz band and 622 Mbit/s would be reached
in the not-too-distant future. Based on this belief, two separate PHY layer
specifications are recommended, one for 5 GHz band, one for the 60 GHz
band since they will require different operations.
Low speed wireless PHY High speed wireless PHY
Frequency band 5.15 – 5.35GHz. 59 GHz – 64 GHz
5.725 – 5.875GHz
Cell radius 80m 10 – 15 m
Transmit power 100 mW 10-20mW
Frequency reuse factor Upto 12 7
Channel band width 30 MHz 150/700 MHz
Data rate 25 Mbit/s 155/622 Mbit/s
Modulation 16 tone DQPSK 32 tone DQPSK
MAC interface Parallel, transfer speed Parallel, transfer speed
3.27Mbyte/s 87.5Mbyte/s
Fixed packet length PHY reader+ MAC reader + 4*ATM cells

 Media Access Control (MAC) - WATM MAC is responsible for providing


functionally point to point links for the higher protocol layer to use. To
identify each station, both IEEE 48 bit address and local significant address,
which is assigned dynamically within a cell, are allowed. Each station
registers it's address to it's hub during a hub initiated slotted-ALOHA content
period for new registration so that make itself know by others. In a shared
environment, there must be some control over the usage of the medium to
guarantee QoS. An extended TDMA, which satisfies PCR (Peak Cell Rate),
SCR (Sustainable Cell Rate), and MBS (Maximum Burst Size) requests, is
suggested. Each station may use the media only when it's informed by the
central control elements (hub). Each can send out several packets at a time. To
minimum overhead, the MAC should support multiple ATM cells in a packet.
Another design issue of MAC layer is to support multiple PHY layers.
Currently, people are interested in different wireless bands, which includes
infra-red medium, 5 GHz radio band, and 60 GHz band. Different PHY will
be need for different medium. WATM MAC should support all of them. Some

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other design issues like error recovery, support for sleep are also under
consideration
 Data Link Control (DLC) - Data Link Control is responsible for providing
service to ATM layer. Mitigating the effect of radio channel errors should be
done in this layer before cells are sent to the ATM layer. In order to fulfill this
requirement, error detection/retransmission protocols and forward error
correction methods are recommended. Currently, the DLC protocol and
syntax, interface to MAC layer, and interface to ATM layer have not been
proposed yet.
 Radio Resource Control (RRC) - RRC is needed for support of control
plane functions related to the radio access layer. It should support radio
resource control and management functions for PHY, MAC, and DLC layers.
The design issues of RRL will include control/management syntax for PHY,
MAC and DLC layers; meta-signaling support for mobile ATM; and interface
to ATM control plane
Mobile ATM
To support mobility, new higher layer control/signaling functions are needed
to handling handover, location management, routing, addressing, and traffic
management. The item, which defines the design the functions of control/signaling,
are called Mobile ATM.
 Handover - In WATM networks, a mobile end user establish a virtual
circuit (VC) to communicate with another end user (either mobile or ATM
end user). When the mobile end user moves from one AP (access point) to
another AP, proper handover is required. To minimize the interruption to cell
transport, an efficient switching of the active VCs from the old data path to
new data path is needed. Also the switching should be fast enough to make
the new VCs available to the mobile users. When the handover occurs, the
current QoS may not be support by the new data path. In this case, a
negotiation is required to set up new QoS. Since a mobile user may be in the
access range of several APs, it will select the one which can provides the best
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QoS. During the handover, an old path is released and a new path is then re-
established. There is a possibility that some cells will get lost during this
process (when the connection is broken). In case no cell lost is allowed. Cell
buffering is used to guarantee that no cell is lost and cell sequence is
preserved. Cell buffering consists of Uplink Buffering and Downlink
Buffering. If VC is broken when the mobile user is sending cells to APs,
Uplink Buffering is required. The mobile user will buffer all the outgoing
cells. When the connection is up, it send out all the buffered cells so no cells
are lost unless the buffers is overflowed. Downlink Buffering is performed by
APs to preserve the downlink cells for sudden link interruptions, congestion,
or retransmissions. It may also occur when handover is executed. There are
severing options for downlink buffering based on different handover
situations.
 Location Management - When a connection is needed to be established
between an mobile ATM end point and another ATM end point, the mobile
ATM end point is needed to be located. There are two basic location
management schemes:
o The mobile PNNI scheme and
o The location register scheme.
In the mobile PNNI scheme, when a mobile moves, the reach ability update
information only propagates to the nodes in a limited region. The switches
within the region has the correct reachable information for the mobiles. When
a call is originated by a switch in this region, it can use the location
information to directly establish the connection. If a call is originated by a
switch outside this region, a connect is established between this switch and
the mobile's Home Agent, which then forward the cells to the mobile. This
scheme decreases the number of signaling messages during a local handover.
In the location register scheme, an explicit search is required to prior to the
establishment of connections. A hierarchy of location registers, which is
limited to a certain level, is used.
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 Routing - Due to the mobility feature of mobile ATM, routing signaling is


a little bit different from that for the wired ATM network. First, mapping of
mobile terminal routing-id's to paths in the network is necessary. Also
rerouting is needed to re-establish connection when the mobiles move
around.
 Addressing - Addressing issue of WATM focuses on the addressing of the
mobile terminal (or mobile end user device). The current solution is that each
mobile terminal has a name and a local address. The name of the mobile
terminal is a regular ATM end system address. It's constant and doesn't
change while the terminal moves. When a terminal is up, its name is
advertised by the switch it attaches to other switches. An local (or temporary)
address is assigned when the mobile terminal attaches to a different switch
during roaming. This switch will assign a local significant address to the
terminal. A mapping have to be take place in order to map the terminal's
name to its temporary address.
 Traffic and QoS Control - The mobility feature puts additional impact on
traffic control and QoS control. Currently a reference model for resource
allocation in WATM is still unavailable. Support for dynamic QoS
renegotiation and extensions to ABR control policy to deal with handover and
other related design issues have not been proposed yet.
 Wireless Network Management - In wireless networks, the topology is
changing in time. This, as well as other mobility feature, presents a unique set
of network management challenges. A specific method must be designed to
maintain the dynamic nature of the network topology. Other issues, like
network and user administration, fault identification and isolation, and
performance management, are also need to be considered.

5.3.4. Multimedia Broadcast Services


This is the case with MBMS that deliver content to multiple users
simultaneously with a fraction of the resources required by normal data services.

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Economies of scale will yield a greater value to mobile users who will be able to
receive rich content, such as video, at a fraction of the cost.
For example, during major sporting events and concerts, instead of a single mobile
user receiving related video services at a very high premium and using dedicated
resources, LTE MBMS will be able to provide the same video services to multiple
users at a much lower cost per user and using the same amount of resources.
Architecture
MBMS are delivered by the collaboration of four main network components:
 Broadcast multicast service center (BMSC) — located at the core of the
network, managing the interface with content providers including billing and
the content to be transmitted over the wireless network
 MBMS gateway (MBMS-GW) — a logical element that delivers MBMS traffic
using IP-multicast reaching multiple cell sites in a single transmission
 Multi-cell/multicast coordination entity (MCE) responsible for the
administration of radio resources for MBMS to all radios that are part of the
MBMS service area
 The mobility management entity (MME) which performs the MBMS session
control signaling including session start, update, and stop, as well as
delivering additional MBMS information to the MCE including QoS and
MBMS service area MBMS provides broadcast multimedia services through
the LTE network combining unicast (PDSCH) and multicast (PMCH) services
in the same LTE frame.
LTE-MBMS radiates the same content to multiple users located within a
predefined MBMS service area, allowing all the users subscribed to MBMS to
simultaneously receive the same multimedia content.
Delivery
Transmissions from all the sectors of the MBMS service area are time
synchronized and transmitted over the same frequency in the network, also referred
as single frequency network (SFN), therefore the resulting MBMS-SFN signal is
received from users as if the signal would be transmitted for a single point.

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This MSFN synchronization area is defined by 3GPP as an area of the network


where all radios are synchronized and capable of supporting one or multiple
MBSFN areas, over the same carrier; however, each radio can only serve one MBSFN
synchronization area.
The multicast channels within the same MBSFN area occupy a pattern of LTE
sub-frames, not necessarily adjacent in time, indicated by the common subframe
allocation (CSA) pattern that is periodically repeated.
Performance
MBMS-SFN services are static and do not vary over time. This provides
several benefits, including:
 Reduced interference
 Increased diversity
MBMS-SFN is broadcasted by physical multicast channels (PMCH). This
provides the following unique characteristics:
 The physical multicast channel can be transmitted in QPSK, 16 QAM, or 64
QAM
 No transmit diversity (MIMO) scheme is specified
 Layer mapping and pre-coding shall be done assuming a single antenna port
and the transmission shall use antenna port 4
 The PMCH can only be transmitted in the MBSFN region of an MBSFN sub-
frame
 The PMCH shall use extended cyclic prefix
 The PMCH schedule cannot be dynamically adjusted by the radio

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VIDHYAA ARTS AND SCIENCE COLLEGE


DEPARTMENT OF COMPUTER
COM SCIENCE
MODEL QUESTION PAPER-I
PAPER
M.SC (CS) DEGREE EXAMINATION
Third semester
Computer Science
MOBILE COMPUTING

Time: Three hours Maximum marks: 75 marks

SECTION - A

Answer ALL the Questions. (5 X 5 = 25 )

1. a) Discuss about network


network.
(OR)
b) Discuss about mobile computing architecture.
2. a) Write short note on evolution of telephony.

(OR)
b) Discuss about Bluetooth security.
3. a) List and explain the value added services through SMS.

(OR)
b) Write about GSM.
4. a) Discuss about GPRS architecture.

(OR)
b) Describe about evolution of WAP.
5. a) Explain the difference between CDMA and GSM.
GSM
(OR)
b) Explain about wireless data.

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Vidhyaa Arts and Science College - Konganapuram

SECTION - B

Answer ALL the Questions. ( 5 X 10 = 50 )

6. a) Explain in detail about middle wares & gateways.

(OR)
b) Write about standard bodies.
7. a) Write about multiple access procedure.

(OR)
b) Explain in detail about Bluetooth.
8. a) Discuss in detail about GSM entities.

(OR)
b) Describe in detail about SMS architecture.
9. a) Write in detail about data services in GPRS.

(OR)
b) List and explain in detail about GPRS applications.
10. a) Write in detail about wireless LAN security.

(OR)
b) Explain in detail about MPLS.

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Vidhyaa Arts and Science College - Konganapuram

VIDHYAA ARTS AND SCIENCE COLLEGE


DEPARTMENT OF COMPUTER SCIENCE
MODEL QUESTION PAPER-II
M.SC (CS) DEGREE EXAMINATION
Third semester
Computer Science
MOBILE COMPUTING

Time: Three hours Maximum marks: 75 marks

SECTION - A

Answer ALL the Questions. (5 X 5 = 25 )

1. a) Explain about the standard.

(OR)
b) List and explain the categories of middleware.
2. a) Discuss about elements of XML document.

(OR)
b) Write in detail about IPV6.
3. a) Discuss about handover.

(OR)
b) Describe about GSM address and identifiers.
4. a) Discuss about billing and charging in GPRS.

(OR)
b) Describe about multiple access radio resource management.
5. a) List and explain the IEEE 802.11 standard.

(OR)
b) Write about the types of wireless LAN.

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Vidhyaa Arts and Science College - Konganapuram

SECTION - B

Answer ALL the Questions. ( 5 X 10 = 50 )

6. a) Write in detail about design consideration of mobile computing.

(OR)
b) Discuss about GPS.
7. a) Explain about voice software.

(OR)
b) Discuss in detail about mobile IP.
8. a) Discuss in detail about GSM architecture.

(OR)
b) Explain about call routing in GSM.
9. a) Discuss in detail about transmission plane protocol architecture.

(OR)
b) Write in detail about GPRS network enhancement.
10. a) Discuss in detail about OFFDM.

(OR)
b) Describe in detail about WATM.

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VIDHYAA ARTS AND SCIENCE COLLEGE


DEPARTMENT OF COMPUTER SCIENCE, CLASS: II M.SC. (CS)

OBJECTIVE TEST-I

Title: MOBILE COMPUING

1. Delay in handoffs is caused due to_______________.


a) Week signal conditions b) High traffic conditions
c) Unavailability of the channel d) All of the above
2. The shape of the cellular region for maximum radio coverage is ____________.
a) Circular b) Square c) Oval d) Hexagon
3. A Bluetooth network is called_____________.
a) Wireless Network b) WAN c) Piconet d) LAN
4. IPv6 addressed has a size of_____________.
a) 32 bits b) 64 bits c) 128 bits d) 265 bits
5. GSM is a digital cellular phone system using _____________.
a) FDMA b) TDMA c) CDMA d) Both FDMA and TDMA
6. SMS uses_____________ signaling channel.
a) SS4 b) SS7 c) SS5 d) SS6
7) WAP is designed for__________________.
a) Internet explorer b) Web browser c) Micro browser d) Macro browser
8) A deck contains a set of________________.
a) Pages b) Cards c) Decks d) Decks
9) ________ is the family of specifications for WLAN developed by a working group
of IEEE.
a) 3G b) 802.11 c) 802.3 d) 802.5
10) A wireless network interface controller can work in___________.
a) Infrastructured mode b) Ad hoc mode
c) Both infrastructure & ad hoc d) None

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