Master Thesis in Electrical

Engineering with Emphasis

on Telecommunications
Thesis no:4350
May 11, 2010

Evaluation of Video Quality of Service in

3G/UMTS Wireless Networks
as Succor for B3G/4G Wireless Network

AL-IMRAN
alic08@student.bth.se

AJAYI OLUWAFEMI SAMUEL

oaaj08@student.bth.se

A Dissertation Submitted in Partial Fulfillment of the

Requirements for the Masters Degree in Electrical Engineering

Blekinge Institute of Technology

May 2010

Blekinge Institute of Technology

School of Computing
Supervisor: Tahir Nawaz Minhas
Examiner: Dr. Patrik Arlos
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 ABSTRACT

Nowadays cellular technology is going to be emerged rapidly. Based on the demand of the users,
next generation cellular system is being able to provide a variety of applications for users’
satisfaction. The Universal Mobile Telecommunication (UMTS) System is one of the features of
cellular communication that is able to yield the different types of services as per users’
satisfaction. Video and voice conferencing have a great deal of demand over the cellular
network. The quality of service (QoS) is the major concern for real time application in this area.
For suitable service to fulfill the users’ demand, it is necessary to improve the QoS. Next
generation/4G network will be intelligent and common platform to integrate all other wireless
technologies. To attain better performance for voice and video conferencing in the heterogeneous
network, proper adaption of QoS is required. In this paper, we consider the 3G/UMTS network
to implement the QoS and analyze the results. The simulation result is evaluated based on some
parameters such as Packet Loss, End to End Delay, Packet Delay Variation, Throughput and
Jitter. Investigation of the simulation result shows that the proper adaption of QoS provides the
better performance regarding these parameters. In order to improve the performance of existing
parameters in UMTS network, we analyze the simulation results. Finally on the basis of results
in UMTS/3G network; we study the QoS performance for B3G/4G wireless network and propose
integration scheme to develop the QoS performance based on existing QoS parameters for
heterogeneous network in 4G.

Keywords: UMTS, QoS, B3G.

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 ACKNOWLEDGEMENTS

First of all, my thanks and respect to almighty ALLAH for blessing me with the capability to do
this work.

I am especially grateful to my supervisor Tahir Nawaz Minhas for helping me greatly to extend
my educational experience. Without his guidance, it was impossible for me to complete this
work. I am also greatly indebted to my Examiner Dr. Patrik Arlos to grant me this opportunity.

Last but not least I would like to thank Michael Åsman and Lena Magnusson for helping me in
different manner during study period.

AL-IMRAN

Thanks to almighty for the strength, favor and fortitude for completion of this thesis work.

I would also appreciate and regards to my supervisor, Tahir Minaz for his enormous guidance
and understanding, I am very thankful for his continuous support.

Thanks to my program coordinator, Mikael Åsman for every form of assistance during my course
of study and stay in Sweden.

AJAYI OLUWAFEMI SAMUEL

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 Dedication

To our parents and families

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 TABLE OF CONTENTS
List of Figures ............................................................................................................................................viii
List of Tables.................................................................................................................................................x
List of Acronyms .........................................................................................................................................xi

1. Introduction ........................................................................................................1
1.1 Introduction……………………………………………………………………………………………..1
1.2 generation of Cellular Networks………………………………………………………………………..1

1.2.1 First generation (1G)……….….…………………………………………………………………..1

1.2.2 Second generation (2G)…………….….………….………………………………………………1

1.2.3 Enhance second generation (2.5G)………….………………………………………………….....2

1.2.3.1 General Packet Radio Service (GPRS)….….…………………………………….…….......2

1.2.3.2 Enhanced Data rates for GSM Evolution (EDGE)….…….……………………………......2

1.2.4 3G Mobile Networks……………………………….…………………………………………........2

1.2.5 B3G/4G Networks…………………….….……………………………………………...................3

1.3 Background and related work….……………………………………………………………………….3

1.4 Objective and main contribution………………………………………………………………………..4

1.5 Thesis outlines…………………………………………………………………………………………..4

2 Different features of 3G Network .......................................................................5

2.1 Different features of third generation (3G) technologies…...……………………………….………….5
2.2 CDMA 2000….........................................................................................................................................5
2.3 3G W-CDMA (UMTS)………………….…………….….….….….…………………………………..6

2.3.1 Protocol and Interfaces…..................................................................................................................6

2.3.1.1 Signaling Protocols….............................................................................................................7

2.3.1. 2 Transport Protocols................................................................................................................8

2.3.2 User Plane Control ….……………………………………………………………………………..9
3 UMTS and Quality of Service ...........................................................................11
3.1 UMTS Architecture……………………………………………………………………………………11
3.2 UMTS Domain………………………………………………………………………………………...12
3.2.1 User Equipment Domain ................................................................................................................12
3.2.1.1 UMTS Subscriber Identity Module (USIM)............................................................……….13

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 3.2.1.2 Mobile Equipment (ME)...........................................................................................................13
3.2.2 Infrastructure Domain ........................................................................................................................13
3.2.3 Access Network Domain .................................................................................................................. 13
3.2.3.1 Universal Terrestrial Radio Access Network............................................................................13
3.2.3.1.1 Radio Network Controller (RNC)……....................................................................... 13
3.2.3.1.2 Node B…….…………………………………………………………………………15

3.2.4 Core Network Domain ..................................................................................................................16

3.3Circuit Switched (CS) elements of Core Network…..............................................................................17

3.4 Packet Switched (PS) elements of Core Network................................................................................. 17

3.5 Circuit switched and Packet switched in Core Network ……………………………………………...18

3.6 UMTS Network Interfaces …………………………………………………………………………....18

3.7 Quality of Service in 3G network……………………………………………………………………..19

3.8 UMTS/3GPP –Defined QoS…………………………………………………………………………..20

3.9 Benefits of QoS ……………………………………………………………………………………….20

3.10 UMTS QoS Basic Classes……………………………………………………………………………20

3.10.1 Conversational Class…………………………………………………………………………...21

3.10.2 Interactive Class…………………………..……………………………………………………21

3.10.3 Streaming Class………………………………………………………………………………...21

3.10.4 Background Class….…………………………………………………………………………..21

3.11 UMTS QoS Parameter and Attributes ……………………………………………………………....22

4 B3G/4G Network and Quality of Service.........................................................23

4.1 Introduction ...........................................................................................................................................23
4.2 B3G/4G concepts...................................................................................................................................23
4.3 Features and characteristics of 3G and4G Network technologies.........................................................24
4.3.1Orthogonal Frequency-Division Multiplexing.................................................................................25
4.3.2 Advantage of OFDM......................................................................................................................25
4.4 Quality of Service in 4G…………………………………………........................................................25

4.4.1Integrated Service (IntServ)……………………………………………………………………….26

4.4.2Differential Service (DiffServ)……………………………………………………………………26

4.4.3 Quality of Service (QoS) Manager.................................................................................................26

5 Simulations and Analysis...................................................................................27
5.1 Introduction ...........................................................................................................................................27

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5.2 performance metrics .............................................................................................................................27
5.3 Different Scenarios ...............................................................................................................................27
5.4 Elements of the network........................................................................................................................27
5.5 Voice and Video traffic scenario before implementing QoS………….………………………...…….28

5.6 Experiment and Simulation result without QoS………………………………………………………29

5.7 Voice and Video traffic scenario after implementing QoS …………………………………………...30

5.8 Experiment and Simulation result with QoS …………………………………………………………31

5.9 Simulation result analysis……………………………………………………………………………..51

6 Conclusion and Future Work…………………………………………………53

6.1 Conclusion and Future Work…………………………………………………………………………53

References ..............................................................................................................54

vii
 LIST OF FIGURES
2.1 CDMA2000 architecture.....................................................................................................................6
2.2 Model of protocol stack in UMTS.................................................................................................... 6
2.3 operation of the RRC Protocol in the UE capability enquiry procedure...........................……...7
3.1 UMTS model architecture..................................................................................................................11
3.2 UMTS domain architecture................................................................................................................12
4.1 4G-architecture....................................................................................................................................24
5.1 Scenario before Implementing QoS..................................................................................................28
5.2: Total sent and received packet before Implementing QoS for video conference...........…….29
5.3 Total sent and received packet before Implementing QoS for voice conference.............……..30
5.4 Scenario after Implementing QoS…..................................................................................................31
5.5 Total sent and receive packet after Implementing QoS for video conference …………….…..32

5.6 Total sent and receive packet after Implementing QoS for voice conference ….……………..33

5.7 Total End To End delay comparison for video conference......................................................34

5.8 comparison of total packet delay Variation for video conference ………..…………………….35

5.9End To End delay comparison for low quality video conference……………..……….………36

5.10 End To End delay comparison for High quality video conference..………………………......37

5.11 Comparison of packet delay variation for low quality video conference..……........................38

5.12 Comparison of packet delay variation for high quality video conference…………………………...39

5.13 Comparison of throughput for low quality video conference ………………………………....40

5.14 Comparison of throughput for high quality video conference……………………………….............41

5.15 End to End delay comparison for low quality video streaming……………………………………...42

5.16 End to End delay comparison for high quality video streaming……………………………………..43

5.17 Comparison of packet delay variation for low quality video streaming……………………..............44

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5.18 Comparison of packet delay variation for high quality video streaming………………...........45

5.19 Throughput comparison for low quality video streaming……………………………………………46

5.20 Throughput comparison for high quality video streaming…………………………………………...47

5.21 Total End To End delay comparison for voice conference…………………………………………..48

5.22 comparison of total Jitter for voice conference………………………………………………............49

5.23 comparison of throughput for voice conference…………………………………………..................50

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 LIST OF TABLES
2.1 List of signaling protocols in UMTS, according to their task………..........................................9
2.2 List of user plane protocols in UMTS, according to the task........................................................10
2.3 List of transport protocols in UMTS according to the task...........................................................10
3.1 UMTS QoS requirement....................................................................................................................19
3.2 Qualitative QoS requirements for different applications.........................................................20
4.1 Features and Characteristics of 3G and 4G Network Technologies.....................................................24

5.1 Statistics of the Nodes……………………………………………………………………………….. 52

x
List of Acronyms

1G 1st Generation
2G 2nd Generations
3G 3rd Generations
3GPP 3rd Generation partnership Project
4G 4th Generations
AMPS Advanced Mobile Phone Service
BS Base Station
BSS Base station Subsystem
BTS Base Transceiver Station
CDMA Code Division Multiple Access
CN Core Network
EDGE Enhanced Data Rates for Global Evolution
FDD Frequency Division Duplex
GGSN Gateway GPRS Support Node
GMSC Gateway Mobile Switching Centre
GSM Global System for Mobile communication
GPRS General Packet Radio Service
HLR Home Location Register
IMT International Mobile Telecommunications
LTE Long Term Evolution
MTS Mobile Telephone Service
MSC Mobile Switching Centre
MSE Mobile Switching Equipment

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OFDM Orthogonal Frequency Division Multiplexing
OSI Open System Interconnection
QoS Quality of Service
PSTN Public Switch Telephone Network
RNC Radio Network Controller
RAN Radio Network Access
SS Subscriber Station
SGSN Serving GPRS Support Node
TACS Total Access Communication System
TDD Time Division Duplex
UE User Equpment
UTRA Universal Terrestrial Radio Access
UTRA UMTS Terrestrial Radio Access Network
TDMA Time Division Multiple Access
UMTS Universal Mobile Communication System
WIMAX Worldwide Interoperability for Microwave Access
WCDMA Wideband Code Division Multiple Access

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Chapter-1: Introduction

1.1 Introduction.

In this chapter we describe the motivation for this frame work and explain its scope, starting from the
inception of wireless networks, its attributes and the contribution of this thesis work.

Telecommunication has been in existence for ages and has undergone numerous changes in the recent
years, most especially in the development of mobile broadband. Day by day communication is going
to be popular due to wireless communication. Cellular networks are also emerging with an improved
level of Quality of Service (QoS) and better mobility and improvement in voice and video
transmission over different technologies. Successively below is a brief description of generations of
technological developments in wireless networks.

1.2 Different Generation of Cellular Networks

1.2.1 First Generation (1G)

The first generation (1G) system was analog, which is radio frequency (RF) transmission could be sent
in a thread-like form. For 1G maximum data throughput of a network was 9.6 kbps. 1G was fairly
dependable, but the disadvantage is that the limited number of services for the user and roaming
between the networks could not be possible. First generation used Frequency Division Multiple
Access (FDMA). 1G technology is based on the Advanced Mobile Phone Service (AMPS). AMPS
operate in the frequency range of 800-900 MHz’s. Each channel is 30 kHz wide, with a bandwidth of
45 kHz (the additional space on each side of the transmission band). In order to transmit, there are 832
frequencies are available. For the purpose of voice traffic, 790 frequencies are assigned while the
remaining 42 are assigned for the control channel. The first generation (1G) network operates by using
circuit switching scheme. While the call is setup, a session is established between the caller and the
recipient by using the company’s switch. During the call in progress, the link remains active only
between these two users. It cannot be possible to make another call during the ongoing call session and
the busy signal will be appeared in the phone for the new call. The switch breaks the connection when
the session of the call is over [31].

1.2.2 Second Generation (2G)

The second generation (2G) cellular system was launched at the beginning of 1990s. It was the
inception to use the digital radio channel for voice control. Different modulation techniques are
offered by 2G system such as GSM, CDMA, and TDMA [31]. The data transmission rate in 2G
system is 9.6 kbps. The similarities between 1G and 2G cellular systems are that both uses circuit
switched scheme. In 2G system radio channels are divided into many logical channels. Few of these
channels are responsible for control and few are used for voice transmission. Different kinds of
multiple access technologies are allowed in 2G systems such as FDMA, TDMA, and CDMA [31]. In
FDMA technology the frequencies are divided among the multiple users. In TDMA system, time slots
are divided among the users for a particular time interval. In this method a user occupies a time slot for
a fixed time interval. In CDMA system users are get access on the basis of code. Second generation
offers digital data transmission like short message service (SMS) and web browsing.

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1.2.3 Enhance Second Generation (2.5G)

To overcome some drawbacks of 2G system the 2.5G is evolved. The major difference between 2G
and 2.5G is that 2.5G is eligible for packet switched network. Though circuit switching is ultimate for
voice data but it is not suitable for packet data transmission. 2.5G is enhanced digital technology
which is efficient and flexible for packet data transmission. For high speed data transmission 2.5G is
developed for GPRS and EDGE.

1.2.3.1 General Packet Radio Service (GPRS)

GPRS provides high speed data transmission. In GSM network, GPRS dynamically assign the time
slot for efficient transmission of packet data. It provides the web access features for the users. It is
faster and flexible to access the wireless web. It makes the wireless web service user friendly and
efficient. GPRS is able to operate data rate between 144kbps-170kbps. For GPRS service GSM
network requires some supplementary equipment like Gateway GPRS Support Node (GGSN), Serving
GPRS Support Node (SGSN).

1.2.3.2 Enhanced Data Rates for GSM Evolution (EDGE)

It is the upgraded version and provides higher-speed than GSM. This technology has been developed
based on the GSM. It is able to provide different kinds of real time and non-real time service such as
voice and video streaming, http service and many other effective applications. It can support voice and
video conference as well though the quality is low. EDGE can operate up to data rate at 384kbps.

1.2.4 3G Mobile Networks

International Telecommunication Union (ITU) planned in order to implement a frequency band of

2000MHz globally. The International Mobile Telephone IMT2000 supports technical analysis for
high-speed telephone solutions. The world of wireless communication development arrives at GSM.IS
– 136/PDC AND CDMA. The 3G evolution for CDMA system brings CDMA2000. THE 3G
evolution for GSM, IS-136 and PDC system leads to Wideband CDMA (W- CDMA), also known as
Universal Mobile Telecommunication Service (UMTS). W-CDMA is based on the network
fundamentals of GSM with the same improvement also implemented in GSM and IS-136 through
EDGE [17].

Past communication was mostly depending only on 2G but due to advancement in technology, there
was an introduction of latest technology such as Wireless Internet Access (Wi-Fi and Video telephony
which require universal standards at higher user bit rates [18].

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1.2.5 B3G/4G network

The 4G mobile technology has many advantages over 3G networks, 4G networks operates up to
100Mbps, and an efficient use of spectrum then the normal 3G networks, low latency and low cost, 4G
mobile technology provides ever new and better ways of providing higher level of graphic user
interface and it provides high level of online gaming, multimedia and better video quality. According
to the working groups which will use this technology, the infrastructure and terminals that will be used
in 4G network will use all standards from 2G to 4G but the legacy system will be placed to adopt users
in the existing infrastructure but the 4G network will all be based on packets i.e. it will all be IP based
[18] [20].

1.3 Background and Related work

The success of 3G/UMTS and next generation wireless technologies depend on the users’ satisfaction
and affordability for the different types of services such as interactive communications, web surfing,
entertainments and so on. The real time applications such as voice and video conferencing are highly
delay and loss sensitive. Implementation of QoS in the UMTS network has a great role in order to
meet the bounded delay and packet loss for the real time applications. In the recent years, some related
research works have been done in the same area. Defining and monitoring QoS metrics can provide a
framework which addresses the existing SLAs and their monitoring methodologies in the network [5].
QoS is more essential for real time application such as voice and video communication in the 3G
network in order to improve the QoS performance and the behavior of the wireless link [10]. It is
important to optimize the Transmission Control Protocol (TCP) for wireless IP network to meet the
requirements in order to improve the end-to-end IP network [11].

The next generation wireless networks will be heterogeneous in nature and will rely on intelligent
network selection decision strategies to support QoS [6]. The heterogeneous wireless networks
integrate different wireless network technologies, e.g. IEEE 802.16 (WMAN), WCDMA, CDMA,
GPRS/EDGE, IEEE 802.11, IEEE 802.15 (WPAN), SATELLITE NETWORKS, WLAN [1] [2] [10].
Therefore the practical mobility and QoS solutions are necessary in the existing 3G and B3G/4G
wireless network [2] [3] [4]. Call admission is another important issue in order to get access in the
heterogeneous wireless networks. Efficient access in the heterogeneous networks can be provided by
designing proper call admission control scheme [7].

The main motivation of our thesis work is to evaluate and analyze the QoS performance of video and
voice conferencing based on the performance metrics such as packet loss, end to end delay,
throughput, delay variation and jitter over the UMTS network. This thesis work will show how well
the QoS performs over the 3G/UMTS wireless network. Moreover the architectural framework of our
proposed UMTS network model will be the effective deployment of proper QoS adaption for voice
and video communication.

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1.4 Objective and main contribution

At present 3G/UMTS networks support different types of services for the real time applications. The
purpose of this thesis work is to evaluate the QoS performance of video and voice conferencing in the
3G/UMTS network based on the performance metrics such as packet loss, end to end delay, packet
delay variation, throughput and jitter. On the basis of the simulation results of QoS metrics for
3G/UMTS network, we analyzed the adaptability of the existing metrics for B3G/4G wireless
networks. From the analytical observation, we propose the QoS improvement scheme which can be
effective to develop the existing QoS parameters for both 3G and 4G networks.

This research work has contributed to knowledge, simulating the 3G (UMTS) in almost new
environment, i.e. OPNET modeler 16, gives new set of simulation results and creates avenue for more
research works. We have designed a model for the simulation, which was simulated with one scenario
without QoS support and its control scenario (another scenario with QoS support). We learnt the
operation of the new OPNET tool and the result was analyzed. We have evaluated and showed the
effects of QoS on video and voice traffics in the UMTS network and realize how effective it is for
combining DiffServ and MPLS in improving QoS parameters for both the UMTS and B3G networks.
Finally we have explained a model for common QoS in the heterogeneous network for efficient
service. However this contribution will be effective to extend our educational experience.

1.5 Thesis outlines

Chapter 1:

This chapter contains the introduction, different generations of cellular networks, background and
related works, objective and main contributions and the thesis outlines.

Chapter 2:

This chapter is about different features of the third generation (3G) technologies.

Chapter 3:

This chapter include with details about UMTS network and its quality of service.

Chapter 4:

This chapter Contains about beyond 3G/4G network and its quality of service.

Chapter 5:

This chapter is about Simulation and simulation result analysis.

Chapter 6:

This chapter contains conclusion and future work.

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Chapter-2: Different Features of 3G Network

2.1 Different Features of third Generation (3G) Technologies.

Third Generation (3G) technology is standard for mobile communication. Different standards are required
to cooperate for working together. The solution is provided by standardization bodies and promoted to the
third Generation partnership program (3GPP).

The 3GPP consists of the following components

 The access Network

The access network depends on the radio interface of Universal Terrestrial Radio Access (UTRA)
which has two operation modes, Frequency Division Duplex (FDD) and Time Division Duplex
(TDD).
 The core Networks
The core network developed for 3GPP is evolved from the GSM core network with the addition
of some new technology like Gateway Mobile Location Center (GMLC).

3GPP adopted two new approaches in developing new radio scheme which was based on Wide band
CDMA (WCDMA), it provides FDD and TDD mode of operations, and the realization of the network
elements from 2G and 2.5G like Visitor Location Register (VLR), Authentication Unit (AuC), Equipment
register (ER), Home Location Register (HLR), Gateway GPRS Support Node (GGSN) e.t.c, which
resulted into a new mobile technology with upgraded software and hardware and higher bandwidth usage
[23].

2.2 CDMA2000

CDMA2000 network is able to provide data rates of 144kbps and the voice quality is twice better than the
CDMA one system. The system architecture that makes up the CDMA2000 is the same as the CDMA one
with the fundamental difference of the introduction of the packet data services. The introduction of data
service meant that Base Transceiver Station (BTS) and Base Station Controller (BSC) should be upgraded
to handle this packet data services. The network of CDMA2000 consists of three major parts which are
the Radio access network (RAN), Core Network (CN) and Mobile Station (MS), the CN can then be
further divided into two parts, one part is that which interacts with the PSTN and the other is connected to
the internet [20].

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 Figure 2.1: CDMA2000 architecture [27].

2.3 3G W-CDMA (UMTS)

This technology supports up to 14.0 Mbps data transfer rate, the mobile phone supports 384Kbits/s which
is more than 9.6kbps of a single GSM circuit switched channel. UMTS has a frequency band of 1885-
2025MHz and 2110-2200MHz for uplink and downlink respectively; it uses a pair of 5MHz channel over
W-CDMA [26].

2.3.1 Protocol and interfaces

There exist a number of interfaces in UMTS network with its respective protocol stack, the figure below
shows a basic pattern of protocol stacks which vary from one interface to another [24].

Figure 2.2: Model of protocol stacks in UMTS [25].

6
The application layer creates and interprets the UMTS signaling messages and also manipulates data
streams, while the transport layer is responsible for transferring the data streams from one network
component to another. We can roughly say that in the OSI model the application layer are from 5-7 and
transport layer contains OSI layers 1-4. The three planes for protocol stack are user plane which carry
information from the user such as data packets or voice, while signaling messages are carried by control
plane. The transport control plane carries internal signaling messages if the data transports using ATM. In
application layer the control plane contains signaling protocols use by the network elements to
communicate with each other. The user plane protocol manipulates the date, i.e. compression and
decompression [25].

2.3.1.1 Signaling Protocols

The application layer protocol shows how they operate Radio Resource Controls (RRC) (which lies
between mobile equipment a serving radio network controller SRNC) figure 2.3 shows the signaling
procedure, The SRNC can be used to find mobile capabilities. In the first step the RRC message is
composed by SRNC known as UE capability enquiry, which is sent to the mobile, the mobile replies with
the capability information that include different parameters that describes its capabilities. (e. g. maximum
data rate, number of data rate stream simultaneously). It can handle a whether to support or not to support
GSM [23].

The SRNC replies with conformation in formation. When time expires before receiving SRNC’s
conformation, it retransmits its capability information.

Figure 2.3: operation of the RRC Protocol in the UE capability enquiry procedure

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The MAP mobile application part handles signaling communications across different interfaces in the
core network. If a call from the mobile to gateway MSC arrives, the MAP message will be sent to the
HLR by GMSC and will ask for mobile current location. So it will be able to forward the call to correct
MSC. The radio access network application part (RANAP) and radio network subsystem application part
(RNSAP), Node B application part (NBAP) and radio network subsystem application part (RNSAP) have
similar role in the radio access network on the Iub, Iu and Iur interfaces respectively [25].

The air interface has two levels non access stratum (NAS) and access stratum (AS) Protocol, which lies in
the non access stratum exchange messages between the core network and the mobile. The four of these
are call control (CC) protocol which runs in the circuit switch domain and the mobile, and manages setup
and tear down data transfer. The mobility management (MM) and the GPRS mobility management
(GMM) protocols handle bookkeeping messages which only effect the internal operation in the system
and not related to any data stream. The RCC protocol lies in the access stratum and use for exchanging
messages between the radio access network and mobile [25].

2.3.1.2 Transport Protocols

In the air interface access stratum, information is transported using unique protocols to UMTS. The
physical layer air interface is most important. The physical layer is assisted by two layer protocols, the
Medium Access Control (MAC) protocol and Radio Link Control (RLC) protocol. The MAC controls the
physical layer e.g. at a particular time how much data is transmitted to and from the mobile station. While
RLC manages the data link between the radio access network and the mobile by task. As retransmitting of
data packets in case of incorrect arrival, we can roughly say that the physical layer is implemented in the
Node B and in the mobile. While RLC and MAC are implemented in the mobile and it’s serving RNC.
The circuit switch domains transmit voice calls using pulse code modulation PCM. This transport
mechanism is used in digital fixed line phone network. The analogue speech signal is digested with 8 bit
resolution at a sample rate of 8 KHz to give 64 Kbps bit rate. The resultant signal is converted into
symbols, then mixed with a carrier and multiplexed with other PCM signals, before transmission there is
no processing lie compression or error correction. In other parts of the network, the protocols use for data
transportation or Asynchronous Transfer Mode (ATM), internet protocol IP and Message Transfer Part
(MTP) of SS7 protocol stack [24][25].

8
 Location Protocol Description

CS domain BICC Bearer independent call control protocol

ISUP ISDN User part

MEGACO Media Gateway Control Protocol

TUP Telephone User Part

CS and PS domain BSSAP+ Base station Subsystem application part plus

MAP Mobile application part

PS domain GTP-C GPRS Tunneling protocol control part

UTRAN NBAP Node B Application part

RANAP Radio access network application part

RNSAP Radio network subsystem application part

Uu non-access stratum CC Call control

GMM GPRS Mobility Management

MM Mobility Management

SM Session Management

Uu Access Stratum RRC Radio resource Control

UE AT Attention commands

USIM UMTS subscriber identity Module

TABLE 2.1 List of signaling protocols in UMTS, according to their task [25].

2.3.2 User Plane Control

It manipulates the data of user interest and carries small number of signaling messages. In example we
take Adaptive Multi Rate (AMR) codec. The voice calls are transmitted at 64Kbps by circuit switch. For
air interface, it is too fast because Signal to Noise (SNR) ratio is rather low, so low maximum data rate.
The code AMR compresses the information on the path between MSC and mobile work in a data rate

9
between 4.7kbps and 212kbps, this increase the number of mobiles in the cell. The ARM codec is
implemented in the core network and mobile compressed information from using ATM and IP [25].

Location Protocol Description Service

CS domain Nb UP Nb user plane protocol

PS domain GTP-U GPRS tunneling protocol user part GPRS

UTRAN Iu UP Iu User plane protocol

Iub UP Iub User plane protocol

Iur UP Iur User plane protocol

Uu Non-access AMR Adaptive multi rate Codec Voice

Stratum RLP Radio Link protocol CS Data
SMS Short message service protocol SMS

Uu access BMC Broadcast multicast control CBS

Stratum PDCP Packet data convergence protocol GPRS

UE USAT USIM application toolkit

Table 2.2 List of user plane protocols in UMTS according to the task [25].

Location Protocol Description

CS domain PCM Pulse code modulation

CS domain ALCAP Access link control application protocol

PS domain
UTRAN ATM Asynchronous transfer mode
IP Internet Protocol
MTP Message transfer part

Uu access stratum MAC Medium access control

PHY Air interface physical layer
RLC Radio link control

Table 2.3 List of transport protocols in UMTS according to the task [25].

10
Chapter 3: UMTS Network and Quality of Service

3.1 UMTS Architecture

Universal Mobile Telecommunications System (UMTS) is characterized by the development in

terms of various services and bandwidth. The evolution of these networks is based on the GSM/GPRS
networks. UMTS networks support all types of applications like data, voice and video. While IP is the
driving technology, UMTS introduced a new radio access technology based on radio access network
without any major change to the core network [13].

Figure 3.1: UMTS model architecture

The public land mobile network has two major divisions:

 UMTS Terrestrial Radio Access Network (UTRAN)

It handles all radio related functionalities.

 Core Network
It is responsible for maintaining subscriber data and for switching voice and data
connections.

UMTS architecture physically contains two main domains:

 User equipment (UE)

It is the area where users acquire the services of UMTS
 Infrastructure domain (PLMN)
This domain includes the physical nodes responsible for termination of the interfaces to
provide end-to-end service to the user.

Each characterizes a maximum level group of physical segments

11
 The detailed description of the UMTS domain architecture is depicted below [13].

Figure 3.2: UMTS domain architecture [13]

3.2 UMTS domain

UMTS domain is classified mainly in two parts.

 User Equipment Domain

 Infrastructure Domain

3.2.1 User Equipment Domain

This domain contains various mobile terminals and electronic smart card that is removable as well as
portable. This domain is divided into two parts.

 UMTS Subscriber Identity Module(USIM)

 Mobile Equipment(ME)

12
3.2.1.1 UMTS Subscriber Identity Module (USIM)

It is an electronic smart card that contains the information about the user to identify in the network. To
enable the user in the network the SIM should be connected in the ME.

3.2.1.2 Mobile Equipment (ME)

It is the Terminal equipment that the subscriber can access the network by using this equipment. It is
capable both for transmission and reception. It is a sort of transceiver.

3.2.2 Infrastructure domain

This domain is the principle part of the UMTS Architecture. It is classified into two parts.

 Access Network domain

 Core Network domain

3.2.3 Access Network domain: It is used to connect all the nodes with core network. It is as like as
gateway to connect with core network for node-B.

3.2.3.1 Universal Terrestrial Radio Access Network (UTRAN)

The radio access network of UMTS is called UTRAN/Universal Terrestrial Radio Access Network, which
is responsible for radio resource, data, signaling traffic, exchange between UE and CN, it also handles
with drawl and allocation of radio bearers required for traffic support and control to some extent UE
mobility and network access technology is based on WCDMA.

UTRAN is combination of two parts.

 Radio Network Controller(RNC)

 Node-B

3.2.3.1.1 Radio Network Controller (RNC)

Radio Network Controller is control unit of UTRAN, it performs various tasks and controls all radio
resources within RNC, it is same as BSC in GSM. Most of protocols between RAN and UE are
implemented in the RNC, it communicates over Iu interface with a maximum of one fixed network nodes
SGSN and MSC. Each RNC is allocated to an SGSN and MSC; it also has an option of using Iur-
interface to communicate over core network CN with neighboring RNC‟s [20] [21].

RNC is responsible for the following:

13
 Call Admission Control

In contrast to GSM, CDMA in UMTS provides a large number of possible channels at the
radio interface, but not all of these can be used at the same time; this is because of
interference that results from increasing the number of channels used. Therefore RNC for
individual call must calculate traffic load for each cell. On the basis of this information
Call Admission Control (CAC) decides whether the interference level after the channel
request is occupied, acceptable and, if necessary, rejects the call [21].

 Radio Resource Management

The RNC manages the radio resources in all the cells attached to it. Utilization level,
priority control and interface calculation are responsibilities of RNC.

 Radio Bearer set up and Release

The radio bearer in UMTS is the radio data channel within access stratum above the Link
Control (RLC) sub layer. The RNC is setup. The responsibility of RNC is setting up,
maintaining and ultimately releasing radio bearers as required.

 CODE Allocation

In UMTS CDMA codes are managed in code tree. The RNC allocates part of these codes
to each mobile and during the course of a connection the allocation can be changed.

 Power Control

CDMA network works efficiently when the transmitting power of all mobile users is
controlled. The actual fast control process takes place in Node B but the target control
values are established in RNC. The RNC only perform (counter loop) loose power control
to minimize the cell interference between the adjacent cell within the RNS and between
nearby RNS [20].

 Packet Scheduling

The same resource at the radio interface is shared by the mobile users in the packet data
transmission. The cyclical allocation of transmission capacity to mobile station
individually is the responsibility of RNC.

 Handover control and RNS relocation

Based on the signal strength of the Node B and UE, the RNC decides another suitable
cell connection where the signal strength is strong during the UE moves out of the range
of one RNC. A new RNC is chosen for the user this is called RNS allocation.

14
  Encryption

The data is encrypted in the RNC, which is arrived for transmission from fixed network.

 Protocol Conversion

RNC is responsible for protocol conversion between core network and Node B.

 ATM Switching

The communication link between RNC and Node Bs and between CN and RNC is
generally on ATM router. The RNC connects and switch ATM connection to communicate
between different nodes. Iu core network (Iucs MSC/GSM, Iups SGSN/GPRS), Iur (other
RNCs), Iub (Base station are different interfaces).

 O&M

The data available are transmitted over interface defined to an OMC operations and
maintenance center.

3.2.3.1.2 Node B

Node B corresponds to the Base Transceiver Station (BTS) in UMTS. Node B can manage one or more
cells connected to RNC over interface Iub. Node B includes CDMA receiver which convert the radio
interface signal into data stream and then forward to RNC over Iub interface. The CDMA transmitter
prepares incoming data for transport over radio interface and routes it to power amplifier, and because of
the large distance between the RNC and the Node B time critical tasks are not stored in RNC. The RNC
knows the exact picture of the cell current situation and makes a sensible decision on power control,
handover and call admission control. The mobile station and Node B continuously measures the quality
of connection and interference level and the result is transmitted to RNC. Node B, in some cases handles
the splitting and combining data streams of different sectors [20].

 Node B cell Types and Cell Hierarchy

UMTS supports different cell types.

 Macro cells

These are large cells and coverage area with radius less than 250m, it has low traffic
and roaming for UEs in rural areas and highways.

 Micro cells

These types of cells have medium coverage area and cover radius of less than 150m, it
has medium traffic load and modest roaming in city parks etc.

15
  Pico cells

These are called hot spots and having lower coverage area, the radius is less than 50m;
it has high traffic and limited roaming on malls, offices, e.t.c.

Different hierarchy levels use different frequency bands (5 MHz).

Routing Area (RA)

Group of cells, efficient localization of UEs (paging…), RAs contains fewer cells than
Pico cells. UMTS cells predicted to be more crowded.

Node -B functions
 Uplink signal measurement

This involves the measurement of signal strength or quality reporting to RNC.

 Soft handover

UE receives the signal from two different antenna elements of Node B, and combines
received signals using RAKE- receiver. It contains multiple antenna elements.

 Power control (Inner loop)

The inner loop power-control functions are to reduce „near-far terminal and
interference‟ sharing among UEs in cells.

3.2.4 Core Network domain

The core network controls the connection when the user initiates or terminates the call to access Packet
Switched (PS) or Circuit Switched (CS) services. The core network also provides interworking with
external networks, it also manages mobility of the user in a home and visited network, also responsible
for location updating, authentication, control charging and accounting. The core network of the UMTS is
the combination of GSM network subsystem (NSS) and backbone of GPRS with a complete new radio
network (UTRAN). The main difference between UMTS & GSM is the radio interfaces and access
technique [20]. This domain consists of three parts

 Home Network Domain

 Serving Network Domain
 Transit Network Domain

16
3.3 Circuit Switched (CS) elements of Core Network

The core network consists of the following elements:

 The Mobile-Services Switching Center (MSC)

It connects the UTRN and other MSC‟s, and also manages the mobility and
registration of the subscriber, allocates the physical resources in combination with
UTRN. Call routing, call roaming, call handover, charging and accounting are also
controlled by MSC [20].

 The Gateway MSC (GMSC)

The GMSC collects location information of the UE call to the MSC serving the UE
current instant. It also ensures the internet functionality with other networks like
PSTN and N-ISDN.

 The Home Location Register (HLR)

This is the database which is shared by CS and PS domains; it contains static
information like MSISDN, IMSI, and UMTS subscription information. In the CS
domain, dynamic information like current VLR address is used to route incoming
calls towards MSC. In the PS Domain the address of the serving SGSN contains in
HLR [20].

 The Visitor Location Register (VLR)

This is a database which stores the information of the user equipment which is
located in the Local Area (LA) covered by VLR, which stores MSNR (Mobile
subscriber roaming number), TMSI, LA and supplementary services like IMSI,
MSISDN, and HLR address. One or more MSC can be linked with VLR and may
be embedded within the same MSC equipment [20].

 The Equipment identity Register (EIR)

The EIR shared database by CS and PS which maintains a list of mobile
equipments to prevent calls from stolen or unauthorized mobile.

 The Authentication Center (AuC)

This protected database accessed by HLR contains USIM secret keys of each
subscriber.

3.4 Packet switched (PS) elements of Core Network

The modified version of the GPRS network is UMTS PS domain. In the core network the PS domain co-
exist with a CS domain and sharing HLR, EIR and Auc database. Two additional physical nodes are
required to support PS domain services that are serving GPRS support Node (SGSN) and the Gateway
GPRS Node (GGSN)

17
  Service GPRS Support Node (SGSN)
The SGSN responsibility is to communicate between UMTS user and PS domain
within the serving area. It is also responsible for user authentication, ciphering,
Integrity, Charging and mobility management procedure for UMTS users, SGSN also
have embedded VLR functionality, data transfer and routing [20].

 Gateway GPRS Support Node (SGSN)

The GGSN responsibility is logical interface to external packet data network (PDN).
The protocol used by PDN is named PDP. The data packet from SGSN is converted
to PDP format by GGSN and sent to external network. Network address (IP) can also
be allocated dynamically by GGSN [20].

GGSN and SGSN in the core network are connected by IP based GPRS backbone which can be intra
PMLN. Roaming agreement is required in case of connecting with different PLMN or if it connects the
GGSN/SGSN of the same GPRS provider or inter-PLMN or when connecting GGSN/SGSN to different
PLMNs [20] [26].

3.5 Circuit Switched (CS) and Packet Switched (PS) in the core network.

The core network of UMTS is based on Circuit Switched (CS) domain and Packet Switched (PS) domain.
CS domain provides real time constraints such as Video telephony and voice while the PS domain
provides services like web browsing, email, MMS/SMS [20]. The core circuits switch and packet switch
services can be accessed simultaneously. Before access the mobile equipment has to register with the
required domain, the phase called IMSI attach, for registration with the CS domain and “GPRS attach” for
registration with the PS domain. TMSI (Temporary Mobile Subscriber Identity) is assigned for access to
CS domain and PTMSI (Packet TMSI) for packet services [20]. The terminations to the domains are done
by “IMSI detach” and “GPRS detach” respectively. When the registration occurs, the user equipment is
tracked by its location management procedures. Location Area (LA) in the CS domain and routing Area
(RA) in the PS domain [20]. After registration, the CS & PS can be accessed in one of the following
configurations

3.6 UMTS network interfaces.

 lub: Responsible for connection between RNC and Node-B
 lur: This interface is used to connect two RNCs.
 Uu: UTRAN and UE are connected through this interface.
 Iu: This inter face help to establish link between RNC and 3G core network.
 Iu-CS: Circuit switched domain connects with the RNC using this interface.
 Iu-PS: Packet switched domain connects with the RNC through this interface.

18
 3.7 Quality of Service in 3G Network.

The second generation of Global System for Mobile Communications (GSM)/code division multiple
access (CDMA) is necessary for one QoS option , which is speech transmission at its full rate coding in
GSM. Thereafter a half-rate service and thus introduced a new QoS option. However the effect of this on
communication was to save network capacity, so it can serve more users in congested hotspots rather than
providing a new grade of service to all level of users. The offer of full or half-rate was never given to the
user; subscribers with half-rate capable mobile phones were put onto half-rate, without the subscriber
knowing that the Quality of speech was deliberately lowered by the network being used. In the evolution
of 2.5G networks e.g. General Packet Radio Service (GPRS), an attempt was made to introduce
mechanism whereby the subscriber can request a different QoS (throughput, packet delay, e.t.c). From
study, the QoS metrics are established at the beginning of the data transfer session at the PDP CONTEXT
setup. For instance, a user using an interactive service, might opt for a service with faster reaction
time/lower round –trip delay, that might be suitable for a smaller packet delay at PDP context setup time,
and the network decide if its allowed or denied [21].

Medium Application Degree of Data Key performance parameters and target

symmetry Rate values
(Kbps)
End-to- Delay Information
end one variation loss
way delay within a call
(ms) (ms)
Audio Conversational Two-way 4-25 < 150, < 1 (jitter) < 3% FER
voice preferred
< 400
limit.
Video Video phone Two-way 32-384 <150, <1 < 1% FER
preferred
<400
limit.
Lip-
synch.
< 100ms
Data Telemetry two- Two-way < 28.8 < 250 N/A Zero
way control
Data Interactive Two-way 1 < 250 N/A Zero
games
Data Telnet Two-way 1 < 250 N/A Zero
(asymmetric)

Table 3.1 UMTS QoS requirement [20]

19
3.8 UMTS/3GPP –Defined QoS

All IP-based data services have been standardized by the Third Generation Partnership Project (3GPP)
with a common QoS framework. The QoS framework was defined for end-to-end QoS covering all
subsystems in a UMTS network, i.e. the core network, wireless and universal terrestrial access networks.
And this marks the first wireless data service that offers a better QoS specification in wireless wide area
network infrastructure. This is one of the prerequisite for the provisioning of multimedia application
support [20].

3.9 Benefits of QoS

QoS enables a network to deliver classes of service (CoS), which means different classes of treatment are
given to different group of services and or group of users. QoS allocates network capacity according to
the type of service while CoS do the provisioning of the preferred allocation of required network
resources in a way as of DiffServ for IP-based services. CoS is used in QoS policy associated with the
user and also used by the network to provide QoS treatments to different services. 3GPP end-to-end QoS
parameters, which also include the identification and definition of UMTS architecture, bearer services,
and recommendations for supporting QoS mechanisms, it establishes four UMTS QoS traffics for mobile
and wireless data [20].

3.10 UMTS QoS Basic Classes

The following traffic classes are available in UMTS/3GPP

 Conversational
 Interactive
 Streaming
 Background

Conversational voice Voice messaging Streaming audio and Fax

and video video

telnet, interactive e-commerce, www FTP, still image, paging Email arrival
games browsing information
Conversational (delay Interactive (delay Streaming (delay< 10 Background
<< 1 sec) approx. 1sec) sec) (delay > 10 sec)

Table 3.2 Qualitative QoS requirements for different applications [20]

20
3.10.1 Conversational Class
This is the class that involves real-time communication e.g. audio, video calls. Generally the
requirements for speech are low delay, low jitter, and clarity with no reverberation. Failure to
meet the basic requirement can result in lack of quality and unacceptable. Subjective evaluations
have shown that the end-to-end delay has to be less than 400 ms for video and audio
conversation. Video application is another application that will use conversational class for
UMTS transport [20].

3.10.2 Interactive class

This class mostly occurs in the internet, it allows smooth interaction of humans and machines
with other devices. Asymmetric type is used and buffering is allowed, there is no guarantee of bit
rate because it uses the best effort service model while the delay factor is also kept at the
minimum, example of applications using this class are web browsing, server access, and database
retrieval, also is the emerging M-commerce application e.g. wireless auction and online games
[20].

3.10.3 Streaming Class

This class consists of real-time applications that exchange information between viewer and
listener; all multimedia services are present in this class. Asymmetric traffic type is used in it and
guaranteed bit rate is provided, here we use the requirement for low jitter and media
synchronization such as Multimedia streaming [20].

3.10.4 Background Class

This class includes all applications that either receive data passively or actively request it, but
without any immediate need to handle the data. Email is the suitable example for this class, the
traffic type is asymmetric, buffering is allowed and high variable delay exists, bit rate does not
provide any guarantee because it relates to the best effort service model [20].

21
3.11 UMTS QoS Parameters and Attributes

The following QoS parameters are defined for UMTS that are essential to maintain of multimedia
services [21].

 Guaranteed bit rate (Kbps)

 Maximum bit rate (Kbps)
 Maximum service data size (octets)
 Transfer delay (ms)
 Delivery order (yes/no)
 Service data unit size error ratio
 Service data unit size format information (bits)
 Residual bit error ratio
 Delivery of erroneous service data units (yes/no)

22
Chapter 4: B3G/4G Network and Quality of Service

4.1 Introduction

The 4G mobile technology has many advantages over 3G networks, 4G networks operates up to 100
Mbps, and an efficient use of spectrum then the normal 3G networks, low latency and low cost, 4G
mobile technology provides ever new and better ways of providing higher level of graphic user interface
and it provides high level of online gaming, multimedia and better video quality. Long Term evolution
(LTE) known as super 3G or 3.9G has been developed by 3GPP project as an improvement to the UMTS
system. LTE uses orthogonal Frequency Division Multiple Access (OFDMA) and LTE can provide us
download data rates up to 150Mbps for multi antenna (2x2) multiple input-multiple output (MIMO)
which are for the highest terminal category and for upload of data up to 50 Mbps. LTE makes efficient
use of the spectrum with the available bandwidth of 1.25 MHz to 200 MHz [32].

According to the working groups which can use this technology, the infrastructure and terminals that
would be used in 4G networks will use all standards from 2G to 4G but the legacy system will be placed
to adopt users in the existing infrastructure however the 4G network will all be based on packets i.e. it
will all be IP based [32].

This technology has been under assessment, even before the 3G networks was deployed. The major
question posed to the providers is why 4G technology is needed when 3G networks seems to be sufficient
for users’ high data rates and quality of service.

The 3G network has less features and applications as compare to 4G. Moreover the Main focus is cost
minimization. It should be much cheaper and affordable for the user.

4.2 B3G/4G Network concepts

 The bandwidth is high and data rate of 100 Mbps to subscribers for better communication.
 Possibility of WLAN and WAN integration and many other networks may get accessed.
 Mobility of services will be efficient and reliable for the users.

It is also expected that 3G might not meet the quality of services like video-conferencing, full motion
pictures.

23
 Figure 4.1: 4G-architecture

4.3 Features and Characteristics of 3G and 4G Network Technologies

Key features 3G Networks 4G Networks

Data rate 384kbps – 2Mbps 20 – 100 Mbps
Frequency band 1.8 – 2.4Ghz 2 – 8Ghz
Bandwidth 5MHz About 100MHz
Switching technique Circuit- and packet switched Fully digital with packet voice
Radio access technology WCDMA,CDMA2000,e.t.c. OFDMA,MC-CDMA, e.t.c
IP IPv4.0, IPv5.0, IPv6.0 IPv6.0
QoS and security Not supported Supported
Multi-antenna techniques Very Limited support Supported
Multicast/broadcast services Not supported supported

Table 4.1 Features and Characteristics of 3G and 4G Network Technologies [15] [19].

24
The major technologies are considering for the 4G networks are OFDM and MC-CDMA for the physical
interface while All-IP and WLAN are vying for the upper layers.

4.3.1 Orthogonal Frequency-Division Multiplexing

OFDM is a frequency division multiplexing technique that is used to transmit large amounts of data
through the radio signal and the spectrum lies between 200MHz and 3.5GHz, with spectral efficiency of
approximately 1 bit/s/Hz

4.3.2 Advantage of OFDM

The main advantage of the OFDM technique is the mutual orthogonality of its carriers which provides
high spectral efficiency. There is no interference because the carriers are orthogonal.

4.4 Quality of service in 4G

Quality of service (QoS) is an important factor for heterogeneous network in terms of managing the
resource reservation scheme. QoS is effective to improve the overall performance of the network. The
term communication arise when the data flows between sender and receiver while there is reliable
magnitude of packet loss, delay jitter, delay and data rate for proper guarantees of QoS. In order for real
time application such as voice and video conference, video streaming QoS is essential.

In heterogeneous network resource reservation and traffic congestion are important issues in terms of the
performance of overall network. QoS is effective in the network to avoid the traffic congestion and
resource reservation. Fourth generation network is completely heterogeneous network where different
types of network can get access efficiently. It is the common platform for all other wireless networks and
provides Always Best Connected (ABC) idea [32].

QoS implementation is not so easy in 4G as well as heterogeneous network. It is very difficult to manage
different network resources for various applications that are from different kinds of wireless networks.
When the users are in motion that means when the users change their position regarding coverage area, it
is very challenging task to provide the appropriate QoS under this circumstances. In this case the users are
required to manage the communication link seamlessly moreover for the video and voice conference or
streaming , the users would like to get the service in an efficient manner as well as seamlessly.

In order to minimize the delay and packet losses for mobile user in the heterogeneous network a few
protocols have been designed to support the users’ mobility in such network. The designed protocols are
mobile IPv6 (MIP6), Fast MIPv6 and Hierarchical MIPv6 [32]. These protocols are very effective for
mobile users to manage the seamless communication during mobility. It is quite impossible to continue
the communication by traditional IP for mobile users when the users move frequently form one network
to other network. In this case mobile IP can detect new wireless network to establish new link in order to
maintain the session without any interference when the user is disconnected from old connection.

25
Hierarchical Mobile IPv6 (HMIPv6) is responsible to reduce latency during handover to improve the
handover management. Fast Mobile IPv6 (FMIPv6) is classified into two types. These are defined as
predictive and reactive. Predictive is used when there is adequate time to process for handover. Other one
is utilized when there is inadequate time for processing the handover [32]. In order to maintain the session
of communication, session initiation protocol (SIP) is very effective to improve performance of QoS.

4.4.1 Integrated Service (IntServ)

Integrated service (IntServ) is responsible to provide the service for end user. It is suitable for and simple
network. IntServ work per flow basis. The major responsibility of Intserv is resource reservation and call
setup. It is used to occupy the network resources for the users. The resource with highest priority is served
first, the lowest priority gets low opportunity and the similar priorities are provisioning into queue [30]. It
has different kind of services such as Best effort service, Controlled load service and guaranteed service.
But the main drawback is that it is not scalable to work in large network.

4.4.2 Differential Service (DiffServ)

Differential service (DiffServ) works in core network. It is scalable and suitable for large network.
DiffServ does not deal with per flow basis; on the contrary it evaluates the service based on the class. The
flows that are received get treatment by class. It is more efficient, scalable in order to provide service in
large network [32]. In DiffServ network DS (Differentiated service) field is used instead of ToS(Type of
Service) field. In DS field there are two types of forwarding such as Expedited Forwarding (EF) and
another is Assured Forwarding (AF).

4.4.3 Quality of Service (QoS) Manager

Fourth generation network is heterogeneous network. In heterogeneous network different types of

applications are available from different type of networks. Different traffic flows are available from
different networks. In order to manage the various traffic flows appropriate QoS is required for proper
resource management. In order to manage the resources QoS manager is necessary. It can allocate the
resources dynamically for appropriate application. It is able to manage the network resources for different
application that comes from other network as well as for it’s own network application. Qos manager can
play an important role in different types of handovers. Due to mobility of user QoS guaranty is required
[32]. QoS manager can affect to reduce the packet losses and delay by provisioning the appropriate
resources during handover. Every network/domain contains QoS manger which is defined as Domain
QoS (DQoS) manager [32]. IP core network also hold a QoS manager which interact and share
knowledge with Domain QoS manager.

26
Chapter 5: Simulation and Analysis

5.1 Introduction

To obtain the performance based on the simulation, OPNET Modeler 16.0 has been used to implement
different scenarios. We have used student version of OPNET Modeler 16.0. This tool has well defined
user interface and rich set of modules where users can efficiently create suitable environment for
simulation by dragging the objects modules. Different types of technologies can be used from start up
wizard of OPNET [28].

5.2 Performance Metrics

Jitter: If two sequential packets transmit from the sender with time x1 and x2 are arrived at the receiver
at time x3 and x4. Then, jitter = (x4- x3)-(x2- x1) [28].

Packet Loss: Packet loss is the failure of the packet arrival in the destination [28].

End to End Delay: The delay between source and destination [28].

Packet delay variation: The variation of end to end delay of the packets [28].

Throughput: Throughput is the data rate in the network for a certain period of time [28].

5.3 Different Scenarios

We have created different scenarios for different types of UMTS traffic classes. Conversational class has
been used for voice conferencing and Background, Interactive and Streaming classes have been used for
video conferencing and video streaming respectively. The name of the scenarios is listed below.

 Voice and Video Traffic Scenario before Implementing QoS

 Voice and Video Traffic Scenario after Implementing QoS

5.4 Elements of the network: Different sorts of elements have been used to create the network in
order to fulfill the requirements of our scenarios. We have used following entities to create the network.

 Application Definition
 Profile Definition
 IPQoS Attribute Definition
 Umts_ggsn_atm8_ethernet8_slip8_adv
 Umts_sgsn_ethernet_atm_slip9_adv
 Umts_rnc_ethernet2_atm2_slip2_adv
 Ethernet_server_adv
 Umts_node_b_adv

27
  Umts_wkstn_adv
 PPP_DS3
 ATM_OC3
 10BaseT

The above network elements have been used to create the scenarios as per basis on requirement. Different
scenarios are described subsequently in the next sections.

5.5 Voice and Video Traffic Scenario before Implementing QoS: In the following scenarios
different traffic classes have been considered for different users. For voice users conversational class has
been used. Background and Interactive traffic classes have been considered for low and high resolution
video conference. In order to support streaming users, a video server has been configured and streaming
traffic class has been considered for these users in this network. Different results are collected for
different scenarios and their comparisons are described below.

Figure 5.1: Scenario before Implementing QoS

28
5.6 Experiment and Simulation Result without QoS: It can be seen from the above scenario
(figure 5.1) that there are two voice users namely voice_user1 and voice_user2 which are conferencing
with each other by using conversational traffic class. Ten different types of video users have been
considered. Four of them are low resolution video users, interacting (UE1, UE2, UE3, UE4) between each
other by using Background traffic class. Interactive traffic class is considered for another four video users
(UEH1, UEH2, UEH3 and UEH4) in order to support high resolution video conference. The remaining
two video users (UESL, UESH) are responsible for video streaming which interact with streaming server.
One of these video users is for high resolution video streaming and other one is for low resolution video
streaming. Both of the video streaming users work with streaming traffic class. To clarify the packet loss,
we have created traffic congestion in the link by assigning the background load. We have allotted the load
in the several steps. Initially (in 0 second) we assign 20% background load .similarly in 100,200,300,400
and 500 seconds we assign 50% 70% 80% 85% and 90% background load respectively. The statistics of
the above scenario is given below.

Figure 5.2: Total sent and received data before implementing QoS for video conferencing

In figure 5.2, the X axis of the graph represents the simulation time and the Y axis determines the sent
and received bytes per second. From the graph, it can be seen that the received video traffic is gradually
decreased while the traffic congestion increases. That means the packet loss increases while the traffic
congestion goes high. From 500 seconds due to high traffic congestion, the packet loss increases slowly
and at about 15th minute, the packet loss increases significantly.

29
 Figure 5.3: Total sent and received data before Implementing QoS for voice conference

In figure 5.3, the X axis shows the simulation times and Y axis represents the sent and received voice data
in bytes per second. From figure, we can observe that the received voice traffic is decreased slowly while
the traffic congestion is increased. The packet loss in voice conference increases due to increase the
traffic congestion and at the last stage the loss is remarkable. The statistic of the above graph is given in
table 5.1

5.7 Voice and Video Traffic Scenario after Implementing QoS: To achieve the better
performance, we have considered QoS in our network. The statistics of the scenario after implementing
the QoS are described as follows.

30
 Figure 5.4: Scenario after implementing QoS

5.8 Experiment and Simulation Result with QoS: We have implemented QoS to obtain better
performance. Weighted Fair Queuing (WFQ) mechanism has been considered for this service. In order to
optimize the performance, differentiated service (DiffServ) has been used. In order to enable DiffServ,
Differentiated Services Code Point (DSCP) has been used for the various applications such as voice and
video traffic. After implementing the QoS, different statistics are explained below.

31
 Figure 5.5: Total sent and received data after Implementing QoS for video conference

In this graph, the simulation time represents along the X axis and the Y axis determines the sent and
received video data in bytes per second. From the above figure 5.5, it is apparent that the total sent and
received traffic for video conference is almost equal although there is high traffic congestion (90%) at 500
seconds. It means that the packet loss is much less than the previous scenario (without QoS). At 14th
minute the received traffic is little bit lower which is difficult to distinguish (packet loss<1%).

32
 Figure 5.6: Total sent and received data after Implementing QoS in voice conference

The X axis of the figure 5.6 determines the simulation time and the Y axis represents the sends and
receives voice data in bytes per second. This graph represents the traffic sent and received in voice
conference. from the above graph, it can be seen that the received traffic is very close to the transmitted
traffic and close to 15th minute, there is very negligible packet loss (<1%) in spite of the maximum traffic
load (90%) which is tolerable. Figure 5.2, 5.3, 5.5, and 5.6 indicate that the packet loss can be reduced by
implementing the proper QoS mechanism in the network. So the simulation results turn out that in the
context of packet loss, it is possible to enhance the performance of the network by applying proper
queuing scheme.

33
 Figure 5.7: Total end to end delay comparison for video conference

The X axis of figure 5.7 represents the simulation time in minute and the Y axis determines the end to end
delay in second. End to end delay is another concern for data transmission of voice and video conference.
The above figure demonstrates the end to end delay comparison for video traffic before and after
implementing the QoS in the network. It can be seen that when the traffic load is less than 90% both End
to End delay are almost equivalent. But when the traffic is high (90%) at 500 seconds, the delay without
QoS increases slowly whereas initially the delay with QoS is fluctuated and then decreased gradually up
to 300 seconds and then becomes almost stable till end. Finally at 15 minutes the delay with QoS reaches
around 144 milliseconds whereas the delay without QoS becomes around 6 seconds. So the End to End
delay can be reduced by implementing the QoS in the network. From figure 5.7, it can be seen that the
End to End delay with QoS is tiny (<150ms) despite the traffic load is high.

34
 Figure 5.8: Comparison of total packet delay variation for video conference

The X axis of figure 5.8 represents the simulation time in minute and the Y axis determines the packet
delay variation in second. Figure 5.8 shows the comparison of packet delay variation for video
conference. The packet delay variation without employing the QoS is higher than the delay variation with
employing the QoS. Initially the packet delay variation without QoS is stable up to 9 minutes. After 9
minutes the delay variation without QoS increases exponentially due to high background load (90%) and
at about 15th minute, it reaches almost 28 seconds. Whereas initially the packet delay variation with QoS
increases sharply and it reaches till 4 milliseconds. Then after 2.5 minutes the delay variation gradually
decreases and at the end it gets to about 1 millisecond.

35
For efficient measurement, two types of video users (UE1 & UEH4) from the network have been
considered in order to accumulate statistics. One of them is low quality video user and another is high
quality video user. The results of these nodes are described below and the statistics of these nodes are
given in the table 5.1.

Figure 5.9: End to end delay comparison for low quality video conference

Figure 5.9 shows the statistic of end to end delay for low quality video user in the network. In the above
figure X axis represents the simulation time in minute and Y axis shows the end to end delay in second.
From figure 5.9, we can observe that initially the end to end delay without QoS for low quality video
conference is increased very slowly and after 8 minutes, the end to end delay increases significantly due
to high background load and near to 15 minutes it reaches at 2.77 seconds. On the contrary, initially the
end to end delay with QoS increases gradually and arrives up to 166 milliseconds during 3 minutes then it
fluctuates slightly till end. The statistics of this graph are listed in the table 5.1.

36
 Figure 5.10: End to end delay comparison for high quality video conference

Figure 5.10 illustrates the statistics for high quality video user in the network. In figure, X axis shows
the simulation time in minute and Y axis determines the end to end delay in second. From the above
graph it can be observed that at the beginning, the delay without QoS increases very slowly but after 7
minutes the delay increases significantly due to high background load (90%) and at about 15th minute, the
delay becomes almost 11 seconds. In contrast, at earlier state, the end to end delay with QoS is ineptly
increased and fluctuated between states. During 3 minutes the delay with QoS reaches below 144
milliseconds and it prolongs up to end with slight fluctuation.

From figure 5.9 and 5.10, it can be illustrated that the end to end delay with QoS in both figure is low.
On the other hand, the delay with QoS for low quality video is lower than the delay with QoS of high
quality video. That means when the quality of the video conference is high then the delay is increased.
The reason is that the high quality video requires large packet size. On the other hand, the end to end
delay with QoS is lower than the end to end delay without QoS in spite of high traffic load.

37
 Figure 5.11: Comparison of packet delay variation for low quality video conference

In the graph 5.11, X axis stands for the simulation time in minute and the Y axis determines the packet
delay variation in second. From the above figure we can demonstrate that the delay variation before and
after implementing the QoS for low quality video user, initially the packet delay variation without QoS is
almost linear and after 8 minutes the delay increases gradually due to high traffic congestion and at 15th
minute it reaches 1.8 seconds. Whereas, at the beginning the packet delay variation with QoS is slowly
increased and at 5th minute, the delay arrives at 0.51 milliseconds, then after 5 minutes the delay with QoS
increases very slowly though there is high traffic congestion. At 15th minute the delay variation reaches to
0.58 millisecond that are much lower than the delay variation without QoS.

38
 Figure 5.12: Comparison of packet delay variation for high quality video conference

The above graph depicts the delay variation without and with the QoS implementation for high quality
video user. The graph shows that at the beginning the packet delay variation without QoS is almost
constant and after 9 minutes the delay increases slowly because of high background load. At 15th minute
the delay becomes 27.5 seconds. Whereas initially the packet delay variation with QoS is 0.5
milliseconds. However the delay decreases sharply to 0.3 milliseconds at 3rd minute and it starts
increasing again. At 5th minute the delay arrives 0.4 millisecond. However at 15th minute the delay
variation reaches 0.47 milliseconds.

From figure 5.11 and 5.12, it can be seen that initially the packet delay variation with QoS is high.
Especially the delay variation for high quality video user is high due to large packet size. On the other
hand the delay variation with QoS is lower than the delay variation without QoS though there is high
traffic load.

39
Throughput is another important parameter for quality of service in a network in terms of determining the
capability of the channel to transfer data. The statistics of the throughput for high and low quality video
users are specified below.

Figure 5.13: Comparison of throughput for low quality video conference

The X axis of the above graph represents the simulation time in minute and the Y axis determines the
received throughput in bits per second. This graph shows the average received throughput for low quality
video users before and after employing the QoS in the network. Initially we can see the both throughputs
(with QoS and without QoS) are alike. However around 2 minutes both throughputs increase slowly
though the throughput with QoS is higher than the throughput without QoS till 15 minutes. The difference
of the received throughput is 1332 bits per second and the statistic of this throughput is given in the table
5.1.

40
 Figure 5.14: Comparison of throughput for high quality video conference

Figure 5.14 presents the average throughput for high quality video user in the proposed network. In this
figure, the X axis shows the simulation time in minute and Y axis represents the received throughput in
bits per second. At the beginning of the graph, both throughputs are alike, after 2 minutes both
throughputs increase exponentially. At around 15th minute, the throughput without QoS becomes almost
stable. On the contrary, the throughput with QoS increases slowly which is higher than the throughput
without QoS. The statistic of both throughputs in this graph is listed in the table 5.1.

By observing graph 5.13 and 5.14, high throughput can be achieved by applying the appropriate QoS
scheme for application.

41
Streaming users are being considered in the network for video streaming. In this case two types of
streaming users have been considered. One of the users is used for low quality video streaming and other
is for high quality video streaming. The statistics of these two video streaming users are described in
detail as follows.

Figure 5.15: End to end delay comparison for low quality video streaming

Figure 5.15 illustrates that the end to end delay comparison for low quality video streaming user in the
proposed network. This graph shows the simulation time in minutes along the X axis and the Y axis
determines the end to end delay in second. At the beginning, the end to end delay without QoS is almost
stable, after 8 minutes the end to end delay increases slowly due to high background load (90%) and at
15th minute, the delay arrives becomes 2.67 seconds. Whereas, initially the end to end delay with QoS
increases sharply and at 3rd minute, it starts decreasing. However, after having little bit fluctuation at the
15th minute, the delay reaches to 110 milliseconds though there is high traffic congestion.

42
 Figure 5.16: End to end delay comparison for high quality video streaming

Figure 5.16 represents the end to end delay for high quality video streaming user in the network. Along
the X axis shows the simulation time in minute and the Y represents the end to end delay in second. From
the above graph, we can see that initially the end to end delay without QoS is linear but at 8th minute, the
delay starts increasing slowly due to 90% background load and after 15 minutes it reaches 8 seconds.
Whereas, the end to end delay with QoS decreases significantly and at about 3rd minute, the end to end
delay starts increasing again, after having little bit fluctuation, the end to end delay arrives 111
milliseconds despite high traffic congestion. The statistic of the graph is depicted in table 5.1.

From graph 5.15 and 5.16, we can see that the end to end delay without QoS is higher and the end to end
delay for high quality video streaming is higher than the low quality video streaming because of large
packet size.

43
 Figure 5.17: Comparison of packet delay variation for low quality video streaming

The X axis of the graph represents the simulation time in minute and the Y axis determines the packet
delay variation in second. Figure 5.17 shows the comparison of packet delay variation for low quality
video streaming users without and with the QoS. From figure 5.17, we can see that the packet delay
variation without QoS is constant but after about 10 minutes, the delay increases significantly due to high
traffic congestion and at 15th minute it reaches 1.7 seconds. Whereas, initially the packet delay variation
with QoS increases and then decreases again slowly. At 15th minute, the delay variation reaches 0.7
milliseconds in spite of high traffic congestion. The statistic of the figures is given in the table 5.1.

44
 Figure 5.18: Comparison of packet delay variation for high quality video streaming

Figure 5.18 demonstrates the comparison of packet delay variation for high quality video streaming users.
The X axis of this graph represents the simulation time in minute and the Y axis demonstrates the delay
variation in second. From the above graph, we can see that primarily the packet delay variation without
QoS is almost steady, after 9 minutes the delay starts increasing exponentially and at 15th minutes, it
arrives at 14.8 seconds. Whereas, at the beginning the packet delay variation with QoS decreases
gradually. After 5th minute, it becomes almost steady. At 15th minute, this delay variation attains 0.8
millisecond.

In figures 5.17 and 5.18, it can be apprehended that the packet delay variation with QoS is lower than the
packet delay variation without QoS. On the other hand, the packet delay variation for high quality video
streaming user is high because of large packet size.

45
 Figure 5.19: Throughput comparison for low quality video streaming

Figure 5.19 demonstrates the comparison of received throughput for low quality video streaming user
prior and after the QoS implementation. The X axis shows the simulation time in minute and the Y axis
represents the received throughput in bits per second. From graph, we can see that initially both
throughputs are equal. But in 2 minutes, it is remarkable that both throughputs increase slowly till 11
minutes, during this period the throughput without QoS is higher than the throughput with QoS. After 11
minutes the throughput with QoS becomes higher than the throughput without QoS. The statistic of both
throughputs is given in table 5.1.

46
 Figure 5.20: Throughput comparison for high quality video streaming

Figure 5.20 illustrates the comparison of received throughput of high quality video user without and with
QoS. The X axis of the above graph determines the simulation time in minute and the Y axis shows the
received throughput in bits per second. From the graph, it can be seen that initially both throughputs are
same. But at 2nd minute it is appeared that the throughput without QoS is little bit higher than the other
one. However, during 8 minutes the throughput with QoS goes higher and increases slowly till 15 minutes
whereas the throughput without QoS becomes almost stable. The statistics of both throughputs in this
graph is shown in table 5.1.

47
 Figure 5.21: Total end to end delay comparison for voice conference

In the above graph 5.21, X axis shows the simulation time in minute and the Y axis determines the end to
end delay in second. Figure 5.21 depicts end to end delay comparison for voice conferencing. From figure
it can be demonstrated that initially the end to end delay without QoS is almost stable, however at 7th
minute the end to end delay starts increasing very slowly and during 8 minutes it increases significantly
due to high traffic congestion (90%). Finally at 15th minute, the delay becomes about 11 seconds.
Whereas, at the beginning the end to end delay with QoS decreases sharply and during 5 minutes it
arrives 0.23 seconds. Finally this end to end delay decreases to 0.22 seconds which is much lower than
the delay without QoS.

48
 Figure 5.22: Comparison of total jitter for voice conference

Figure 5.22 represents the comparison of jitter for voice conference. The X axis of this graph shows the
simulation time in minute and the Y axis represents the jitter in second for voice conferencing. Initially
both jitters are same and increase sharply. During 7 minutes, it can be seen that the jitter without QoS
goes higher than the jitter with QoS. Finally at 15th minute, the jitter without QoS arrives 0.0021 second
whereas the jitter with QoS is close to 0 second at that time. By implementing the QoS, low jitter can be
attained. It can be noted that the negative jitter means the packet arrives earlier than the expected time.

49
 Figure 5.23: Comparison of Throughput for voice conference

Figure 5.23 shows the comparison of received throughput for voice users during voice conference. The X
axis represents the simulation time in minute and the Y axis shows the received throughput in bits per
second. From the graph, we can see that up to 7 minutes the both throughputs are same. Afterwards the
throughput with QoS goes higher than the throughput without QoS. Finally at 15th minute, the throughput
with QoS keeps increasing, whereas the throughput without QoS becomes almost steady and much lower
than the throughput with QoS.

50
5.9 Simulation Result Analysis

Having experienced from the resultant graphs it can be seen that all the traffic with QoS performs better
than the other one. In our simulation different performance metrics have been considered. Packet loss is
one of the key parameters to measure the performance both for video and voice transmission in the
network. Frame size of the packet is an important factor which has a great impact on the Packet loss. The
packet loss becomes high if the frame size of the packet is large. Another cause of packet loss is disparity
of QoS configuration between end user and provider edge device. It is possible to reduce the packet loss
by taking proper adaption QoS into account.

Another significant factor is End to End delay. It has great impact on the video and voice transmission to
maintain the quality of communication. End to End delay is inversely proportional to the quality of the
communication. The quality degrades while the delay elevates. One of the causes of high delay is large
packet size and another cause is high traffic congestion. By deploying the proper queuing scheme, it is
potential to satisfy for reliable delay in terms of transmission.

Packet delay variation is imperative in the context of packets arrival to the end user. From simulation it
can be observed that the delay variation can be degraded by employing appropriate QoS. The throughput
depends on the data rate between the end user in the network. In our simulation it can be seen that finally
the received throughput with QoS of the end user is high in order for the video user despite the traffic
congestion is high. Apposite QoS application adaption, sufficient bandwidth, packet segmentation can
satisfy the requirement for reliable throughput from the user aspect. Jitter is another key issue in terms of
voice conference. The performance of jitter depends on the encoding scheme and number of frames per
packet. In the simulation it can be seen that the jitter without QoS is higher than the jitter with QoS.
Proper adaption of QoS and adequate bandwidth can provide to meet the requirement for reliable jitter in
voice conference.

In order to measure the statistics, few nodes have been considered for different types of traffic. In order to
manifest the statistics of the simulation in terms of different parameters for both scenarios are
provisioning in the following table (5.1).

51
 Parameter Without QoS With QoS
Low Quality Video
Packet loss (%) 1.243% 0.019%
Conference(UE1)
End to End Delay(sec) 2.770 0.16325
Delay Variation(sec) 1.799 0.00058510
Packet loss (%) 7.448% 0.008%
High Quality Video
Conference(UEH4) End to End Delay(sec) 10.992 0.14384
Delay Variation(sec) 27.522 0.00047102
Packet loss (%) 1.215% 0%
Low Quality Video
Streaming(UESL) End to End Delay(sec) 2.672 0.11155
Delay Variation(sec) 1.733 0.00072329
Packet loss (%) 6.490% 0%
High Quality Video
Streaming(UESH) End to End Delay(sec) 8.090 0.11728
Delay Variation(sec) 14.807 0.00080686
Voice Conference Packet loss (%) 7.258% 0.0414%
End to End Delay(sec) 11.407 0.22191
Jitter(sec) 0.0021371 -0.0000937

Table5.1: Statistics of the Nodes

52
Chapter 6: Conclusion and Future Work

6.1 Conclusion and Future Work

Wireless communication is going to be a common platform on which all the technologies can be
accessed and interacted efficiently. In order to fulfill the SLA (Service Level Agreement) especially in
terms of video and voice services in UMTS network, it is required to consider enhanced Quality of
service. To improve the existing quality of service (QoS) parameters, it is potential to take proper
initiative in different levels of QoS implementation. Appropriate resource allocation in the network can be
one of the ways to improve the QoS. In order to proper allocation of resources, QoS manager should be
considered for each network. For more expedient, the combination of DiffServ and Multiprotocol Label
Switching (MPLS) can be considered. Diffserv can provide service for scalable network by proper
utilization of network resources. But Diffserv can only support the intra domain service. In this case the
router between two networks may get congested. To overcome this limitation MPLS can play an effective
role. By employing traffic Engineering of MPLS, it is possible to avoid the congestion. The MPLS is
potential to decrease the transmission overhead in the network. The combination of DiffServ and MPLS
can improve the existing QoS parameters in UMTS network.

The B3G/4G network will be heterogeneous and able to serve a large number of subscribers. It will be
able to handle Different technologies in a common platform. For this reason the existing QoS parameters
should be upgraded for large number of subscriber. The service should be improved in different levels of
the large networks. Besides the QoS manager Adaptive Resource Management (ARM) should be consider
for proper adaption of the network resources in the heterogeneous networks. The integration of IntServ,
Diffserv and MPLS can be the potential to improve the service as well as the existing QoS parameters in
4G network. The RSVP of IntServ can make resource reservation in the network. It can enable the end
user to reserve the resources for utilization in the network whereas Diffserv can provide better support in
the core network. The integration of two services can maintain the traffic flow and reduce the packet loss
which may improve the other quality of service (QoS) parameters in the network.

To support IPv6 based networks, MPLS can be efficient to improve the service in the large network. The
Label Switched Path (LSP) of MPLS can make fast forwarding decision with the support of Diffserv.
Moreover the MPLS- traffic engineering and the RSVP can be provisioning the constraint based LSP to
avoid the congestion as well as improve the traffic flow to meet the requirement for improving the
Existing parameters such as packet loss, end to end delay, delay variation, throughput and jitter in 4G
network. From the above analysis, it can be concluded that the performance of these QoS parameter can
be improved both in UMTS and B3G/4G wireless network by adapting and integrating proper QoS
scheme.

The 4G is the common platform for all other technologies to interact properly. In order to make efficient
services for the heterogeneous network, it is important to design a common QoS scheme for the common
platform. Our next concentration will be designed a model of common QoS scheme in heterogeneous
network to improve the overall service for the end user in the network.

53
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