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OSUAGWU, HENRY ONYEMAUCHE
PG/M.ENGR/14/68120
DYNAMIC BANDWIDTH SCHEDULING FOR WCDMA
UPLINK TRANSMISSION
DEPARTMENT OF ELECTRONIC ENGINEERING
FACULTY OF ENGINEERING
Godwin Valentine
Digitally Signed by: Content manager’s Name
DN : CN = Webmaster’s name
O= University of Nigeria, Nsukka
OU = Innovation Centre
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TITLE
DYNAMIC BANDWIDTH SCHEDULING FOR WCDMA UPLINK TRANSMISSION
BY
OSUAGWU, HENRY ONYEMAUCHE
PG/M.ENGR/14/68120
DEPARTMENT OF ELECTRONIC ENGINEERING
FACULTY OF ENGINEERING
UNIVERSITY OF NIGERIA,
NSUKKA
MARCH, 2016
3
APPROVAL PAGE
DYNAMIC BANDWIDTH SCHEDULING FOR WCDMA UPLINK
TRANSMISSION
OSUAGWU HENRY ONYEMAUCHE
PG/M.ENGR/14/68120
A THESIS SUBMITTED IN PARTIAL FULFILLMENT OF THE
REQUINEERIREMENTS FOR THE AWARD OF MASTER OF ELECTRONIC
ENGINEERING (COMMUNICATION) IN THE DEPARTMENT OF ELECTRONIC
ENGINEERING, UNIVERSITY OF NIGERIA, NSUKKA
OSUAGWU, HENRY ONYEMAUCHE SIGNATURE………… DATE………… PROF. COSMAS I. ANI SIGNATURE………… DATE………… (PROJECT SUPERVISOR) EXTERNAL EXAMINER SIGNATURE………… DATE………… ENGR. DR. M.A. AHANEKU SIGNATURE………… DATE………… (AG. HEAD OF DEPARTMENT) PROF. E.S. OBE SIGNATURE………… DATE………… (CHAIRMAN, FACULTY POSTGRADUATE COMMITTEE)
4
CERTIFICATION
This is to certify that OSUAGWU HENRY ONYEMAUCHE, a postgraduate student in
the Department of Electronic Engineering with Registration Number
PG/M.ENGR/14/68120 have satisfactorily completed the requirements for the course and
thesis work for the degree of Master of Engineering (Communications).
_____________________ ____________________
PROF. COSMAS I. ANI ENGR. DR. M.A AHANEKU
(PROJECT SUPERVISOR) (AG. HEAD OF DEPARTMENT)
________________________________
PROF. E.S. OBE (CHAIRMAN, FACULTY POSTGRADUATE COMMITTEE)
5
DECLARATION
I, Osuagwu Henry Onyemauche, a postgraduate student of the department of Electronic
Engineering, University of Nigeria, Nsukka, declare that the work embodied in this
dissertation is original and has not been submitted by me in part or in full for any other
diploma or degree of this University or any other Universities.
_______________________ _______________
OSUAGWU HENRY ONYEMAUCHE DATE
PG/M.ENGR/14/68120
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DEDICATION
This work is dedicated to God and to my father Edmond Osuagwu (late)
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ACKNOWLEDGEMENTS
I wish to express my profound gratitude to my supervisor, Prof. Cosmas I. Ani for his
guidance and attention throughout the duration of this research work. I must acknowledge
in a special way all the staff of the Department of Electronic Engineering for making the
realization of this research work a success.
My gratitude also goes to my mother, Mrs Angela Osuagwu for her prayers and support. I
must acknowledge my brothers and sisters for their support and encouragement. I equally
want to acknowledge my good friends Agashi Boniface, Ajibo Augustine and Anike
Uchenna for their concerns throughout the duration of the program.
I thank everyone who has contributed in one way or the other to ensure the successful
completion of this research work.
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ABSTRACT
Providing quality of service is a challenging issue in UMTS mobile networks for multimedia traffic (video, voice and data). Critical services such as real-time audio, voice and video are given priority over less critical ones, such as file transfer and web surfing. One of the approaches that efficiently provides standard quality of service for multimedia traffic in wireless networks is to dynamically allocate bandwidth to varying traffic load and channel conditions. There are several of such dynamic bandwidth allocation approaches developed in the recent time by researchers. The choice of which one to implement at an instance and for a specific condition is an issue in mobile communication networks. In this work, the popular Code-Division Generalized Processor Sharing (CDGPS) was analyzed. The CDGPS variations – priority and non-priority – were compared, the two techniques were modelled and simulated using MATLAB Simulink object oriented environment. Simulation results show that priority CDGPS provides the best performance and improvement in the delay and loss rate, while still maintaining a high bandwidth utilization of percentage value of 98.2%.
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TABLE OF CONTENTS
Title page - - - - - - - - - - i
Approval page - - - - - - - - - ii
Certification - - - - - - - - - - iii
Declaration - - - - - - - - - - iv
Dedication- - - - - - - - - - - v
Acknowledgement - - - - - - - - - vi
Abstract - - - - - - - - - - vii
Table of contents - - - - - - - - - viii
List of Figures - - - - - - - - - xi
List of Tables - - - - - - - - - xiii
List of Acronyms - - - - - - - - - xiv
CHAPTER 1: INTRODUCTION - - - - - - 1
1.1 Background of the study - - - - - - - 1
1.2 Statement of problem - - - - - - - - 2
1.3 Aim and Objectives - - - - - - - - 2
1.4 Scope of the work - - - - - - - - 2
1.5 Significance of Study - - - - - - - - 3
1.6 Methodology - - - - - - - - - 3 1.7 Thesis outline - - - - - - - - 3
CHAPTER 2: LITERATURE REVIEW - - - - - 5
2.1 Overview and Third Generation Technology- - - - - 5
2.2 Requirements for Third-Generation system - - - - - 6
2.2.1 Wideband Code Division Multiple Access - - - - 7
2.3 Third Generation GSM objectives and capabilities - - - - 8
2.4 UMTS Multi-radio evolution path- - - - - - - 9
2.5 UMTS Network Architecture - - - - - - - 10
2.5.1 User Equipment (UE) - - - - - - - 11
2.5.2 UMTS Terrestrial Radio Access Network (UTRAN) - - - 11
2.5.3 The Core Network - - - - - - - - 13
2.6 UMTS protocol of operation - - - - - - - 14
2.6.1 Radio Interface protocol structure - - - - - - 16
2.6.2 User Plane - - - - - - - - - 18
2.6.3 Control Plane - - - - - - - - - 18
10
2.7 Radio Interface protocol reference layer - - - - - 18
2.7.1 Physical (PHY) layer - - - - - - - 18
2.2.2 Medium Access Control (MAC) layer - - - - - 24
2.7.3 Radio Link Control (RLC) protocol - - - - - 26
2.7.4 Packet Data Convergence Protocol (PDCP) - - - - 27
2.7.5 Radio Resource Control (RRC) layer - - - - - 28
2.8 Radio Resource Management (RRM) - - - - - 29
2.8.1 Radio Resource Management (RRM) Function - - - - 31
2.8.2 Radio Resource Management (RRM) Function Interaction - - 33
2.9 Scheduling Schemes - - - - - - - - 34
2.9.1 First-In-First-Out Scheduling - - - - - - 35
2.9.2 Weighted Round Robin Scheduling - - - - - 36
2.9.3 Priority Scheduling - - - - - - - - 37
2.9.4 Earliest-Due-Date Scheduling - - - - - - 38
2.9.5 Rate-Controlled Scheduling - - - - - - 38
2.10 Requirements of a Scheduler - - - - - - 38
2.11 Related Works - - - - - - - - 41
2.12 Conclusion - - - - - - - - - 50
CHAPTER THREE: RESEARCH METHODOLOGY - - - 51
3.1 System Model - - - - - - - - 51
3.2 Generalized Processor Sharing (GPS) - - - - - 53
3.3 The Code-Division Generalized Processor Sharing (CDGPS) scheme - 53
3.4 Traffic Source Model - - - - - - - - 59
3.4.1 Voice Source Modeling - - - - - - - 59
3.4.2 Video Source Modeling - - - - - - - 60
3.4.3 Data Source Modeling - - - - - - - 61
3.5 Model Validation - - - - - - - - 62
3.6 Conclusion - - - - - - - - - 63
CHAPTER FOUR: SIMULATION AND RESULT ANALYSIS - 64
4.1 Introduction - - - - - - - - - 64
4.2 MATLAB Simulation Framework - - - - - - 64
4.3 Performance metrics - - - - - - - - 70
4.4 Simulation Results - - - - - - - - 71
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CHAPTER FIVE: CONCLUSION AND RECOMMENDATION- - 78
5.1 Conclusion - - - - - - - - - 78
5.2 Recommendation for future work - - - - - - 78
5.3 Contribution to knowledge - - - - - - - 79
REFERENCE - - - - - - - - - 80
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LIST OF FIGURES
Figure 2.1: UMTS multi-radio network - - - - - - 9
Figure 2.2: UMTS Network Architecture - - - - - - 11
Figure 2.3: UTRAN architecture - - - - - - - 13
Figure 2.4: UMTS protocols - - - - - - - 15
Figure 2.5: Radio interface protocol reference architecture - - - 16
Figure 2.6: Protocol termination for a common channel - - - - 17
Figure 2.7: Physical layer for transmitting situation - - - - 19
Figure 2.8: Frame structure for downlink DPCH - - - - - 21
Figure 2.9: Frame structure for downlink PDSCH - - - - 22
Figure 2.10: Structure of the random-access transmission - - - 23
Figure 2.11: MAC layer architecture - - - - - - 24
Figure 2.12: RLC sub-layer architecture - - - - - 27
Figure 2.13: Location of RRM functions - - - - - 32
Figure 2.14: Radio Resource Management Functions Interaction - - 33
Figure 2.15: FIFO Scheduling - - - - - - - 36
Figure 2.16: Weight Round Robin Scheduling - - - - - 37
Figure 2.17: Priority Queuing Scheduler - - - - - - 37
Figure 3.1: Network Structure - - - - - - - 52
Figure 3.2: A queuing model of the CDGPS scheme - - - - 54
Figure 3.3: Priority CDGPS flowchart - - - - - - 57
Figure 3.4: Non-priority CDGPS flowchart - - - - - 58
Figure 3.5: On-Off model - - - - - - - - 59
Figure 3.6: On-Off voice packetization - - - - - - 60
Figure 3.7: Model validation with CDGPS scheme - - - - 63
Figure 4.1: Simulation Framework for WCDMA systems - - - 65
Figure 4.2: Multimedia IP Traffic (voice, video and data) - - - 66
Figure 4.3: Buffer queuing Model - - - - - - - 67
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Figure 4.4: The system Server - - - - - - - 68
Figure 4.5: CDGPS computational model - - - - - - 69
Figure 4.6: A Scope of entities generated - - - - - - 70
Figure 4.7: Throughput as a function of Traffic intensity for multimedia IP traffic 72
Figure 4.8: Throughput per flow as a function of traffic intensity - - 73
Figure 4.9: Average delay as a function of Traffic intensity - - - 74
Figure 4.10: Loss rate as a function of traffic intensity - - - - 75
Figure 4.11: Backlogged flow loss rate as a function of traffic intensity - 76
Figure 4.12: Bandwidth utilization as a function of traffic intensity - - 77
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LIST OF TABLES
Table 2.1: Main differences between WCDMA and GSM air interfaces - 7
Table 2.2: Relationship between spreading factor and bit rate - - - 20
Table 2.3: Four UMTS service class - - - - - - 30
Table 3.1: Simulation parameters - - - - - - - 62
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LIST OF ACRONYMS
1G First Generation 2G Second Generation 3G Third Generation 3GPP Third Generation Partnership Project AC Admission Control ADV Access Delay Variation AM Acknowledgement Mode AS Access Stratum AuC Authentication Center BCCH Broadcast Control Channel BER Bit Error Rate BMC Broadcast/Multicast Control BSC Base Station Controller BTS Base Transceiver Station CCCH Common Control Channel C-CDGPS Credit-based Code-Division Generalized Processor Sharing CCTrCHs Coded Composite Transport Channel CDGPS Code-Division Generalized Processor Sharing CDMA Code Division Multiple Access CM Connection Management CN Core Network CPCH Common Packet Channel CS Circuit Switch CTCH Common Traffic Channel DBA Dynamic Bandwidth Allocation DCA Dynamic Code Assignment DCH Dedicated Channel DFS Delay Fair Scheduling DPA Dynamic Priority Allocation DPCCH Dedicated Physical Control Channel DPCH Downlink Dedicated Physical Channel DRR Deficit Round Robin DRS Dynamic Resource Scheduling DS-CDMA Direct Sequence-Code Division Multiple Access DSCH Downlink Shared Channel DTCH Dedicated Traffic Channel EDD Earliest Due Date EDF Earliest Deadline First EDGE Enhance Data Rates for GSM Evolution EIR Equipment Identity Register ETSI European Telecommunication Standard Institute
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FACH Forward link Access Channel FDD Frequency Division Duplex FIFO First-In-First-Out FLC Fuzzy Logic Controller FTP File Transfer Protocol GERAN GSM/EDGE Radio Access Network GGSN Gateway GPRS Support Network GMM GPRS Mobility Management GMSC Gateway Mobile Switching Center GPRS General Packet Radio Service GPS Generalized Processor Sharing GSM Global System for Mobile Communication HC Handover Control HLR Home Location Register IMT-2000 International Mobile Telecommunication – 2000 IP Internet Protocol IPv4 Internet Protocol version 4 IPv6 Internet Protocol version 6 ITU International Telecommunication Union LC Load Control MAC Medium Access Control MDRR Multi-flow Deficit Round Robin ME Mobile Equipment MM Mobility Management MMS Multimedia Message Service MSC Mobile Switching Center MWF2Q+ Multi-flow Worst-case Fair Weighted Fair Queuing Plus NAS Non-Access Stratum OVSF Orthogonal Variable Spreading Factor PC Power Control PCCH Paging Control Channel PCH Paging Channel PDCP Packet Data Convergence Protocol PDSCH Physical Downlink Shared Channel PDU Packet Data Unit PRACH Physical Random Access Channel PS Packet Scheduling PS Packet Switch QPSK Quadruped Phase Shift Keying RACH Random Access Channel RLC Radio Link Control RM Resource Manager RNC Radio Network Controller
17
RRC Radio Resource Control RRM Radio Resource Management RTBS Real Time Bandwidth Scheduling RTCS Real Time Code Assignment RTE Real Time Emulator RTGS Real Time Generic Scheduling SAD Service Access Delay SF Spreading Factor SGSN Servicing GPRS Support Network SIM Subscriber Identity Module SM Session Management SMS Short Message Service STFQ Start-Time Fair Queuing TBs Transport Block TDD Time Division Duplex TDMA Time Division Multiple Access TFCI Transport Format Control Indicator TFCS Transport Format Combination Set TM Transport Mode TPC Transmit Power Control UE User Equipment UM Un-acknowledgement Mode UMTS Universal Mobile Telecommunication System USIM UMTS Subscriber Identity Module UTRAN UMTS Terrestrial Radio Access Network VLR Visitor Location Register WCDMA Wideband Code Division Multiple Access WFQ Weighted Fair Queuing WF2Q Worst-case Fair weighted Fair Queuing WRR Weighted Round Robin
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CHAPTER ONE
INTRODUCTION
1.1 Background of the study
Today, mobile communications play a central role in the voice/data network arena. From
the early analog mobile first generation (1G) to the third generation (3G) the standard has
changed. The new mobile generations do not pretend to improve the voice communication
experience but try to give the user access to a new global communication reality [1]. The
aim is to reach communication universality and to provide users with a new set of services.
The cellular networks are evolving through several generations; the first generation (1G)
wireless mobile communication network was analog system which was used for public
voice service with the speed up to 2.4kbps. The second generation (2G) is based on digital
technology and network infrastructure. As compared to the first generation, the second
generation can support text messaging [2]. Its success and the growth of demand for online
information via the internet prompted the development of cellular wireless system with
improved data connectivity, which ultimately leads to the third generation systems (3G). It
is now time to explore new demands and to find new ways to extend the mobile concept.
The first steps have already been taken by the 2.5G, General Packet Radio Service (GPRS)
and Enhanced Data Rates for GSM Evolution (EDGE), which gave users access to a data
network (e.g. Internet access, Multimedia Message Service).
However, users and applications demanded more communication power. As a response to
this demand a new generation with new standards has been developed-third generation
(3G). Third generation (3G) networks offer greater security than their 2G predecessors. By
allowing the UE (User Equipment) to authenticate the network it is attaching to, the user
can be sure the network is the intended one and not an impersonator [3]. With all its
enhancements, Global System for Mobile Communication (GSM) will represent the
19
mainstream of mobile communication systems for the next several years. However it is
obvious due to technical and economic reasons, GSM will be followed by third generation
(3G) mobile communication system. Third generation (3G) mobile communication system,
called Universal Mobile Telecommunication System (UMTS) within European
Telecommunication Standard Institute ETSI/Europe, aim to support a wide range of voice
and data services, focusing on mobile packet switched data services based on Internet
Protocol (IP) technology [4]. Moreover, UMTS will give the mobile user performance
similar to the fixed network and will stimulate the development of new mobile multimedia
applications.
1.2 Statement of problem
Integrated services networks support multiple services and are faced with problem of
resource sharing among applications. Providing quality of service (��) and resource
allocation is a challenging issue especially in mobile networks with applications of
multimedia traffic (video, voice and data).
1.3 Aim and objectives
The aim of this work is to design a dynamic bandwidth scheduling framework which can
improve the overall performance of radio resource management strategy in the UMTS.
The specific objectives includes the following:
� Development of scheduling scheme that would support differentiated quality of
service (��) for Universal Mobile Telecommunication System (UMTS) traffic.
� Develop a scheduling scheme that would optimized bandwidth utilization.
� Develop a scheduling scheme that would introduced dynamic bandwidth sharing
mechanism for backlogged flows.
1.4 Scope of the work
20
The scope of this work is the third generation (3G) mobile communication system, called
Universal Mobile Telecommunication System (UMTS). It is specific to the modelling of
the UMTS uplink scheduler.
1.5 Significance of Study
As the Internet evolves into the global infrastructure, there is a growing need to provide a
broad range of quality of service guarantee for different applications, which bring forth the
necessity of traffic management. This research will be significant to wireless network
providers and researchers in finding an effective means of utilizing the available scarce
resource in a heterogeneous traffic environments. Also, it will help service provider in
putting into consideration the distribution of residual bandwidth among backlogged (active
user) session in an equitable manner.
1.6 Methodology
To realize the objectives of this work, the following methodology was adopted:
� Review of UMTS uplink transmission techniques and resource allocation schemes.
� Review of existing uplink and downlink scheduling scheme in UMTS.
� Propose a scheme following the best UMTS scheduling scheme from the review.
� Development of choice computer models of scheduling scenarios and implementing
the proposed scheme.
� Validation of the results of the analysis with performance of existing schemes.
� Simulation of choice model and obtain data.
� Analyze data in terms of performance metrics.
� Compare the performance of the propose scheme in terms of performance
improvement.
1.7 Thesis outline
The remainder of the thesis is organized as follow:
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In chapter two, a review of UMTS quality of service (QoS) architecture was carried out
and radio resource management service functions that provide the background knowledge
for the design of resource allocation scheme for the UMTS system. Chapter three is focused
on the general system model requirements for achieving the dynamic radio resources
allocation. In chapter four, a MATLAB simulation framework is created for analyzing the
proposed scheduling algorithm. The simulation results are presented and analyzed. Chapter
five provides the conclusion of the thesis, some recommendations for future work and the
contribution to knowledge made by this research work.
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CHAPTER TWO
LITERATURE REVIEW
2.1 Overview and Third Generation Technology
Many packets scheduling algorithms have been extensively studied in the wired networks,
such as weighted fair queuing (WFQ), worst-case fair weighted fair queuing (WF2Q),
deficit round robin (DRR) and start-time fair queuing (STFQ). These results have also been
extended to local wireless networks by several researchers. However, due to several unique
features of 3G networks, scheduling algorithms proposed for wired and wireless networks
in the literature are not directly applicable to 3G networks. Several scheduling schemes
have been proposed in the literature for IP-based radio access networks in WCDMA to
efficiently utilize radio resources. The review of these literatures are detailed in (section
2.11).
Third Generation Technology was developed in order to face up to the new requirements
of services that were coming, as high-quality images and video or to provide access to the
Web with higher data rates. Third-generation radio access technologies aim to provide the
commercial market with high quality, efficient and easy-to-use wireless mobile multimedia
services [5]. All 2G wireless systems are voice-centric, most 2G systems also support some
data over their voice paths, but at painfully slow speeds usually 9.6 Kb/s or 14.4 Kb/s. So
in the world of 2G, voice remains fundamental while data is already dominant in wire-line
communications. And, fixed or wireless, all are affected by the rapid growth of the Internet.
Planning for 3G started in the 1980s. Initial plans focused on multimedia applications such
as videoconferencing for mobile phones. When it became clear that the real killer
application was the Internet, 3G thinking had to evolve [6].
23
Since the third-generation (3G) mobile radio systems will provide us from low to high data
rate services with a maximum data rate of 2 Mbps, it can be used in several multimedia
applications such as voice, audio/video, graphics, data, Internet access, and e-mail. These
services, regardless of based on packet switched or circuit switched, have to be supported
by the radio interface and the network subsystem [7]. In January 1998, the 3GPP (Third-
Generation Partnership Project) has agreed on the UMTS (Universal Mobile
Telecommunication System) for 3G mobile radio systems.
2.2 Requirements for Third-Generation system
The second generation systems were built mainly to provide speech services in macro cells
[8,9]. To understand the background to the differences between second and third generation
systems, we need to look at the new requirements of the third generation systems which
are listed below:
� Bit rates up to 2 Mbps;
� Variable bit rate to offer bandwidth on demand;
� Multiplexing of services with different quality of service requirements on a single
connection, e.g. speech, video and packet data;
� Delay requirements from delay-sensitive real-time traffic to flexible best-effort
packet;
� Quality requirements from 10% frame error rate to 10-6 bit error rate;
� Coexistence of second-generation and third-generation systems and inter-system
handovers for coverage enhancements and load balancing;
� Support of asymmetric uplink and downlink traffic, e.g. web browsing causes
more loading to downlink than to uplink;
� High spectrum efficiency.
� Co-existence of FDD and TDD modes
The table 2.1 lists the main differences between WCDMA and GSM. In this comparison
only the air interface is considered.
24
Table 2.1 Main differences between WCDMA and GSM air interfaces [9]
The differences in the air interface reflect the new requirements of the third-generation
systems. For example, the larger bandwidth of 5 MHz is needed to support higher bit rates.
2.2.1 Wideband Code Division Multiple Access (WCDMA)
This section introduces the principles of the WCDMA air interface. Brief explanations for
most of the main system design parameters of WCDMA were presented [8]. Special
attention is drawn to those features by which WCDMA differs from GSM.
� WCDMA is a wideband Direct-Sequence Code Division Multiple Access (DS-
CDMA) system, i.e. user information bits are spread over a wide bandwidth by
multiplying the user data with quasi-random bits (called chips) derived from CDMA
spreading codes. In order to support very high bit rates (up to 2 Mbps), the use of a
variable spreading factor and multi-code connections is supported.
WCDMA GSM
Carrier spacing 5 MHz 200 kHz
Frequency reuse factor 1 1–18
Power control frequency 1500 Hz 2 Hz or lower
Quality control Radio resource management Network planning (frequency algorithms planning)
Frequency diversity 5 MHz bandwidth gives Frequency hopping multipath diversity with Rake receiver
Packet data Load-based packet scheduling Time slot based scheduling with GPRS
Downlink transmit diversity Supported for improving Not supported by the standard, but downlink capacity can be applied
25
� WCDMA supports highly variable user data rates, in other words the concept of
obtaining Bandwidth on Demand (BoD) is well supported. The user data rate is kept
constant during each 10 � frame. However, the data capacity among the users can
change from the frame to frame. This fast radio capacity allocation will typically be
controlled by the network to achieve optimum throughput for packet data services.
� WCDMA supports two basics modes of operation: Frequency Division Duplex
(FDD) and Time Division Duplex (TDD). In the FDD mode, separate 5 MHz carrier
frequencies are used for the uplink and downlink respectively, whereas in TDD only
one 5 MHz is timeshared between the uplink and downlink. Uplink is the connection
from the mobile to the base station, and downlink is that from the base station to the
mobile.
� The WCDMA air interface has been crafted in such a way that advanced CDMA
receiver concepts, such as multiuser detection and smart adaptive antennas can be
deployed by the network operator as a system option to increase capacity and/or
coverage. In most second generation systems, no provision has been made for such
receiver concepts and as a result they are either not applicable or can be applied only
under severe constraints with limited increases in performance.
� WCDMA is designed to be deployed in conjunction with GSM. Therefore,
handovers between GSM and WCDMA are supported in order to be able to leverage
the GSM coverage for the introduction of WCDMA.
2.3 Third Generation GSM objectives and capabilities
� 3G GSM (UMTS) is an upgrade from GSM via GPRS or EDGE.
� IMT-2000 is an ITU’s umbrella name for 3G which stands for International Mobile
Telecommunications 2000
� 144 Kbps data rate available to users in high-speed motor vehicles over large areas.
� 384 Kbps available to pedestrians standing or moving slowly over small areas.
� Support (to be phased in) for 2.048 Mbps for office use.
� Support for both packet-switched and circuit-switched data services.
26
� More efficient use of the available spectrum in general.
� Support for a wide variety of mobile equipment.
� Flexibility to allow the introduction of new services and technologies.
2.4 UMTS Multi-radio evolution path
3GPP (Third Generation Partnership Project) is a global project aiming to develop open
standards for the UMTS third Generation Mobile System based on evolved GSM core
networks. This multi-radio mobile system comprises two different third generation (3G)
radio access networks, GERAN (GSM/EDGE Radio Access Network) and UTRAN
(UMTS Terrestrial Radio Access Network), which are based on different radio access
technologies, GSM/EDGE and WCDMA (Wideband Code Division Multiple Access),
respectively [5]. Initially, Universal Mobile Telecommunication System (UMTS) defined,
within the scope of UMTS Terrestrial Radio Access Network (UTRAN) standardization,
new radio access network architecture, with protocols, interfaces and quality of service
architectures specifically designed for the efficient provision of third generation
multimedia services. GSM/EDGE Radio Access Network (GERAN) has adopted all these,
hence becoming an integral part of the UMTS third generation frame. Furthermore, 3GPP
standards ensure that an efficient integration between UTRAN and GERAN can be
accomplished so that they can be merged under a single UMTS multi-radio network. This
concept is illustrated in the Figure 2.1.
Integrated radio resource
management base on QoS management
3G Core Network
UMTS 3G multi-radio
UMTS 3G multi-radio access network
GERAN
UTRAN USIM
ME
Circuit
switch core
Packet core
Network
27
Figure 2.1 UMTS multi-radio network [5]
UMTS is based on Wideband Code Division Multiple Access (WCDMA) radio
technology, which offers higher throughput, and better real-time services. The UMTS radio
access network offers multimedia applications like simultaneous transfer of speech, data,
text, pictures and audio a maximum data rate of 2Mbps, which is a result of using 5MHz
bandwidth of the radio channels in UMTS instead of 200 kHz in GSM [10]. The 3G
WCDMA air interface has been designed to provide a packet based wireless service, by
which different computing and telephone devices all share the same wireless network and
may be connected to the Internet anytime and anywhere.
2.5 UMTS Network Architecture
Universal Mobile Telecommunications System (UMTS) is a 3G cellular
telecommunication system. It will be the successor of GSM. UMTS is designed to cope
with the growing demand of mobile and internet applications with required quality of
service parameters [11]. WCDMA is used for the radio interface of UMTS. The UMTS
network has three subsystems to address different operations [12]. They are UMTS
terrestrial random access network (UTRAN), core network (CN) and user equipment (UE).
Figure 2.2, is a UMTS network architecture with its basic domains and this figure also
show its external reference points and interfaces with the UTRAN. UTRAN is connected
the core network (CN) via Iu interface. Between the radio networks controller (RNC) and
Core Network, there is Iu UTRAN interface. The UTRAN interface that is between the CN
and the radio network controller (RNC) is called Iu-PS and also UTRAN interface between
the RNC and circuit switched domain of CN is known as Iu-CS. Radio interface between
User equipment UE and UTRAN is known as Uu interface. These interfaces are also known
as reference.
28
Figure 2.2 UMTS Network Architecture [13]
2.5.1 User Equipment (UE)
The user equipment is the physical device which enables the user to have access to network
services. The UE consists of ME (Mobile Equipment) and USIM (UMTS Subscriber
Identity Module). The ME is a radio terminal used for communication over Uu interface.
The ME consists of the Mobile Termination (MT), which performs the radio transmission,
and Terminal Equipment (TE), that enables end-to-end application, e.g., a laptop that is
connected to a mobile phone [14]. The USIM is a smartcard that holds the subscriber
identity, performs authentication algorithms, and stores authentication and encryption keys
and some subscriber information that is needed at the terminal.
2.5.2 UMTS Terrestrial Radio Access Network (UTRAN)
UTRAN consists of one or more Radio Network Sub-systems (RNSs). An RNS is a sub-
network within UTRAN and consists of one RNC and one or more Node Bs. RNCs may
Mobile station Base Station Subsystem Network Subsystem Other Networks
Uu Iu
RNS
UE UTRAN Core Network
M E
ME
BTS
BSC
Node
B
RNC
MSC/
VLR
GMSC
SGSN GGSN
SIM
USIM
EIR HLR AuC
PSTN
PLMN
Internet
29
be connected to each other via an Iur interface. RNCs and Node Bs are connected with an
Iub Interface. During Release 7, work study on the support of small RNSs was done,
meaning the use of collocated RNC and Node B functionalities in a flat architecture, and
that was found feasible without mandatory specification changes [8].
The Node B
The Node B converts the data flow between the Iub and Uu interfaces. It also participates
in radio resource management. It logically corresponds to GSM Base Station but the term
“Node B” was initially adopted as a temporary term during the standardization process and
then never changed.
The Radio Network Controller (RNC)
The RNC is the network element responsible for the control of the radio resources of
UTRAN. It interfaces the CN (normally to one MSC and one SGSN) and also terminates
the Radio Resource Control (RRC) protocol that defines the messages and procedures
between the mobile and UTRAN. It logically corresponds to the GSM BSC.
UTRAN Interfaces
The UTRAN interfaces are as follows:
� Iub interface - The Iub connects a Node B and a RNC.
� Iur interface - The open Iur interface allows soft handover between RNCs from
different manufacturers and, therefore, complements the open Iu interface.
Iu interface - This connects UTRAN to the CN and is similar to the corresponding
interfaces in GSM, the open Iu interface gives UMTS operators the possibility of acquiring
UTRAN and CN from different manufacturers. The enabled competition in this area has
been one of the success factors of GSM. UTRAN interfaces are shown in figure 2.3.
30
Figure 2.3 UTRAN architecture [8]
2.5.3 The Core Network
� HLR (Home Location Register):This is a database located in the user’s home
system that stores the master copy of the user’s service profile [15]. The HLR also
stores the UE location on the level of MSC and SGSN.
� MSC/VLC (Mobile Switching Center/Visitor Location Register): The MSC
function is used to switch the CS (Circuit Switch) transactions, and VLR function
holds a copy of the visiting user’s service profile, as well as more precise
information on the UE’s location within the serving system.
� GMSC (Gateway MSC): The Switch at the point where UMTS is connected to
external CS networks. All incoming and outgoing CS connections go through
GMSC.
� SGSN (Serving GPRS Support Node): Similar to that of MSC / VLR but is used
for Packet Switched (PS) services. The part of the network that is accessed via the
SGSN is often referred to as the PS domain. It is an upgraded version of serving
GPRS support node.
U u lub
Node B
l ub l ur lu MSC
Node B SGSN
l ub lu
Node B
RNC
RNC
UE
31
� GGSN (Gateway GPRS Support Node): Functionality is close to that of GMSC
but is in the relation to PS services. It is an upgraded version of gateway GPRS
support Node
2.6 UMTS Protocol of operation
The communication among the different entities of the UMTS architecture involves several
protocol stacks that are defined for each interface and are depicted in figure 2.4. A protocol
stack defines a set of layers that specify the communication procedures between two
network entities [16]. Each layer in a network entity (e.g. the UE) communicates with the
same layer of the network entity (e.g. the node B) by means of a specific protocol that
includes a set of procedures involving a number of messages transferred between both
entities. From a vertical perspective, a given layer provides the means for the transfer of
the messages originated at the above layers. In turn, from a horizontal perspective, the
concatenation of several protocol stacks allows the communication between non-adjacent
entities (e.g. between the User Equipment (UE) and the Core Network).
The unifying principle in the UTRAN development work has been to keep the mobility
management (MM) and connection management (CM) layers independent of the air
interface radio technology [17]. This idea has been realized as the access stratum (AS) and
non-access stratum (NAS) concepts (Figure 2.4). The access stratum (AS) is a functional
entity that includes radio access protocols between the UE and the UTRAN. These
protocols terminate in the UTRAN. The NAS includes core network (CN) protocols
between the UE and the CN itself. These protocols are not terminated in the UTRAN, but
in the CN; the UTRAN is transparent to the NAS. The Mobility Management (MM) and
Connection Management (CM) protocols are GSM Core Network protocols; GPRS
Mobility Management (GMM) and Session Management (SM) are GPRS Core Network
protocols.
32
Figure 2.4 UMTS Protocols [16]
Just as the NAS tries to be independent of the underlying radio techniques, so also have
the MM, CM, GMM, and SM protocols tried to remain independent of their underlying
radio technologies. The Connection Management (CM) and Session Management (SM)
protocols – responsible for the establishment and release of connections or sessions for an
UE, respectively – or the Mobility Management (MM) and GPRS Mobility Management
(GMM) protocols, responsible for dealing with mobility functions at the network layer (e.g.
location area updating, routing area updating, paging, etc.). In turn, in the user plane, the
main NAS protocol at the network layer for packet switched services is the IP protocol,
while for circuit services information comes directly from the source without the need for
a network protocol.
In the UMTS architecture, the access stratum (AS) includes three different protocol stacks,
namely the radio interface protocols, the Iub interface protocols and the Iu interface
protocols. In particular, the radio interface protocol stack allows communication between
the UE and the UMTS access network (UTRAN). Note that the protocols at the upper
layers terminate in the UE and RNC, while the lower layers terminate in the UE and Node
33
B. With respect to the Iub interface protocols, they involve the communication of the lower
layers of the RNC and the Node B. Finally, the Iu interface protocols allow communication
between the RNC and the CN, distinguishing between the Iu-CS for communication
between RNC and MSC and the Iu-PS for communication between RNC and SGSN.
2.6.1 Radio Interface Protocol structure
In this thesis, this work is focused on the management of the resources at the radio interface,
whose scarcity constitutes in most cases the bottleneck for a proper communication to be
carried out. In Figure 2.5 the UMTS radio interface protocol stack is shown. The
termination of each protocol can be seen in Figure 2.6. The radio interface protocol is
comprised of three layers:
� The physical layer PHY (layer 1)
� The data link layer (layer 2)
� Network layer (layer 3).
Figure 2.5 Radio interface protocol reference architecture [16]
34
Figure 2.6 Protocol termination for a common channel [16]
Layer 1
The layer 1 is the physical layer which is based on WCDMA technology with a chip rate
of 3.84 ��ℎ���/�. It offers data transport services to the MAC layer via transport channels.
Transport channels are characterized by how the information is transferred to the radio
interface. They are divided into common and dedicated transport channels.
Layer 2
Layer 2 is split into the following sub layers: Medium Access Control (MAC), Radio Link
Control (RLC), Packet Data Convergence Protocol (PDCP) and the Broadcast/Multicast
Control (BMC) layer. The BMC layer is responsible for the management of broadcast and
multicast messages like the SMS Cell Broadcast Service. The MAC layer offers logical
channels to the RLC layer. A logical channel is defined by what type of information is
transferred. A general classification of logical channels is into control channels and traffic
channels. Control channels are used to transfer higher layer signaling messages. Traffic
channels are used for transfer of user information. There is a Dedicated Traffic Channel
(DTCH) and a Common Traffic Channel. The CTCH is used for broadcasting messages.
The DTCH is a point-to-point channel, dedicated to one user.
35
Layer 3
Layer 3 consists of one protocol in the control plane. This protocol is the radio resource
Control (RRC). It handles all messages required to set up, modify and release layer 1 and
layer 2 entities.
The radio interface protocol is also divided into control and user plane. The control plane
provides services for transmitting signaling messages. The user plane is responsible for
user data transmission [14].
2.6.2. User Plane
� Radio Link Control (RLC): Presents a reliable channel to higher layers by
retransmitting erroneous packets
� Medium Access Control (MAC): Channel access, multiplexing traffic streams,
scheduling priority flows
� Physical Layer (PHY): Measurements, power control algorithms
2.6.3 Control Plane
� Radio Resource Control (RRC): Connection and �� management.
� Radio Resource Management (RRM): Algorithms for admission control, handovers.
2. 7 Radio Interface protocol reference layer
2.7.1 Physical (PHY) layer
This section presents the characterization of the physical layer, whose mission is to
transform the flow of information coming from the different transport channels into
physical radio signals transmitted by the antenna [18]. The physical layer at the transmitter
side receives Transport Blocks (TBs) from the MAC layer. These transport blocks may
belong either to one or to several transport channels that are simultaneously multiplexed.
Then, the physical layer executes a set of procedures over the received transport blocks to
36
generate the radio signal that is sent to the antenna. At the receiver side, the reverse
procedures are carried out to recover the transport blocks from the received physical signal
at the antenna and to deliver them to the MAC. Figure 2.7 shows an overview of the
activities in the physical layer for a transmitting situation.
Figure 2.7 Physical layer for transmitting situation [14]
The first baseband signal processing entity includes channel coding, rate matching,
interleaving and others. Channel coding can be applied as either convolutional coding or
turbo coding with rate 1/2 or 1/3. It's also possible that no coding is done. Further the data
stream is segmented into 10ms blocks which are multiplexed with blocks of other transport
channels. The transport channel multiplexing entity combines all coded transport channels
into one special channel, called Coded Composite Transport Channel (CCTrCH), which is
split up into one or several physical channels later on. These physical channels are
separately spreaded and scrambled. Spreading is done with an Orthogonal Variable
Spreading Factor (OVSF) code which ensures orthogonality between the different physical
channels of a user and which enables different data rates for them. In addition to spreading,
scrambling is provided in order to separate terminals or base stations from each other. The
symbol rate is not affected by the scrambling operation anymore. After scrambling all the
37
data sequences are summed up and modulated. In the downlink direction, Quadruped Phase
Shift Keying (QPSK) modulation is applied.
Physical Channel
Physical channels are defined by a certain carrier frequency, scrambling code and
spreading code. Spreading of the low-bandwidth data signal to produce the wideband
CDMA signal consists of two steps:
� Channelization or spreading code to reach channel rate of 3.84 ��ℎ���/�;
� Scrambling – to provide separation of transmissions.
UMTS uses variable spreading and power levels to provide different user data rates. In
FDD mode 10 ��� frames are used. The number of chips per bits is called the Spreading
Factor (SF) and it defines the data service required for the user: ���� = �� × �� �! (2.1)
For UMTS: $%& '(&) × �� = 3.84 *+,%-./. (/ℎ�� 012�) (2.2)
The Spreading Factor (SF) can change in every 10 ��� frame
Table 2.2 Relationship between spreading factor and bit rate [15]
Service Bearer Data Rate (Kbps) SF Modulation Rate (*+,%-./.)
Speech 30 128 3.84
Packet 64 Kbps 120 32 3.84
Packet 384 Kbps 960 4 3.84
Furthermore, in this section the structure of two physical channels in the downlink and
random access channel in the uplink will be explained, the downlink dedicated physical
channel (DPCH) and the physical downlink shared channel (PDSCH). Physical channels
are typically structured into radio frames. A radio frame is a unit which consists of 15 slots.
38
The length of a radio frame corresponds to 38400 chips, which equals 10ms. Therefore one
slot takes 2560 chips.
Downlink Dedicated Physical Channel (DPCH)
Figure 2.8 Frame structure for downlink DPCH [14]
Figure 2.8 shows the frame structure of the DPCH. The DPCH transmits Layer 2 user data
and physical layer control information in a time multiplexed manner. The control
information transmitted on the downlink Dedicated Physical Control Channel (DPCCH)
consists of TPC, TFCI, and pilot bits. This control information is generated at Layer 1 and
provides Transmit Power Control (TPC) and Transport Format Indication (TFCI). The pilot
is used for channel estimation. The exact number of bits of each field may vary and it is
fixed when the connection is established. The spreading factor ranges from 512 down to 4.
Physical Downlink Shared Channel (PDSCH)
Figure 2.9 is the frame structure of the PDSCH. A PDSCH is always associated with a
downlink DPCH. So each user which shares a PDSCH requires an active DPCH. The
PDSCH doesn't carry Layer 1 information, all this information is transmitted on the
DPCCH part of the associated DPCH. Since the PDSCH is shared among several users, a
user has to be informed that it should listen to the PDSCH. This is done by the TFCI field
DPDCH DPCCH DPDCH DPCCH
2560 chips, SF = 4...512
1 radio frame = 10 ms
DATA 1 TPC TFCI DATA 2 PILOT
Slot 0 Slot 1 Slot i Slot 14
39
in the associated DPCCH. Power control for both DPCH and PDSCH is performed by the
TPC field. The spreading factor for this channel ranges from 256 to 4.
Figure 2.9 Frame structure for downlink PDSCH [14]
Random Access Channel (RACH)
The RACH is an uplink transport channel which is always received from the entire cell.
RACH is characterized by a limited size data field, collision risk and the use of open loop
power control. The Physical Random Access Channel (PRACH) is used to carry the
RACH. Its operation is based on a Slotted ALOHA approach with fast acquisition
indication [19]. The user equipment (UE) can start the transmission at a number of well-
defined time offsets, which are denoted as access slots. There are 15 access slots per two
frames, and they are spaced 5120 chips apart, i.e. 1.25 ms. Information on what access slots
are available in the current cell is given by higher layers. The structure of the random-
access transmission is shown in figure 2.10. The random access transmission consists of
one or several preambles of length 4096 chips, which is 1 ms, and a message of length 10
ms. The preamble part of the random-access burst consists of 256 repetitions of a signature.
There are a total of 16 different signatures, based on the Hadamard code set of length 16.
The 10 ms message is split into 15 slots, each of length �345� = 2560 chips. Each slot
consists of two parts, a data part that carries Layer 2 information, and a control part that
PDSCH
2560 chips, SF=4…256
1 radio frame = 10 ms
Data
Slot 0 Slot 1 Slot i Slot 14
40
carries Layer 1 control information. The data and control parts are usually transmitted in
parallel.
Figure 2.10 Structure of the random-access transmission [19]
The data part consists of 10*2k bits, where k=0,1,2,3. This corresponds to a spreading factor
of 256, 128, 64 and 32, respectively, for the message data part. The control part consists of
8 known pilot bits to support channel estimation for coherent detection and 2 TFCI bits.
This corresponds to a spreading factor of 256 for the message control part. The total number
of TFCI bits in the random-access message is 15*2 = 30. The TFCI value corresponds to a
certain transport format of the current random-access message.
Data N
data bit
Data
Pilot TFCI N
pilot bits N
TFCI bits
Control
Tslot
= 0.667 ms, 10·2k bits (0..3)
Slot 0 Slot 1 Slot 14 Slot i
TRACH
= 10 ms
Preamble 1 Preamble n Message sent
TPreamble= 1
ms TMessage= 10 ms
41
2.7.2 Medium Access Control (MAC) layer
The main role of the transmission network is to transport MAC frames between RNCs and
Node Bs [20]. The UTRAN MAC is not the same protocol as the GPRS MAC, even though
they both have similar names and handle similar tasks in similar ways. The UTRAN MAC
can even contain different functionalities depending on whether it supports FDD, TDD, or
both modes. The Medium Access Control (MAC) layer is responsible for the handling of
the logic channels and most of the priority and multiplexing issues [21]. The MAC layer is
also responsible for selecting an appropriate transport format (TF) for each transport
channel depending on the instantaneous source rate(s) of the logical channels. The transport
format is selected with respect to the transport format combination set (TFCS) which is
defined by the admission control for each connection. Figure 2.11 shows the MAC layer
architecture.
Figure 2.11 MAC layer architecture [9]
The MAC layer consists of three logical entities:
� MAC-b which handles the broadcast channel (BCH). There is one MAC-b entity
in each UE and one MAC-b in the UTRAN (located in Node B) for each cell.
� MAC-c/sh which handles the common channels and shared channels – paging
channel (PCH), forward link access channel (FACH), random access channel
42
(RACH), uplink Common Packet Channel (CPCH) and Downlink Shared Channel
(DSCH). There is one MAC-c/sh entity in each UE that is using shared channel(s)
and one MAC-c/sh in the UTRAN (located in the controlling RNC) for each cell.
� MAC-d is responsible for handling dedicated channels (DCH) allocated to a UE in
connected mode. There is one MAC-d entity in the UE and one MAC-d entity in the
UTRAN (in the serving RNC) for each UE.
MAC Logical Channels
Control channels:
� Broadcast control channel (BCCH)
� Paging control channel (PCCH)
� Dedicated control channel (DCCH)
� Common control channel (CCCH)
Traffic channels:
� Dedicated traffic channel (DTCH)
� Common traffic channel (CTCH)
MAC Services
The services MAC provides to the upper layers include the following:
� Data transfer;
� Reallocation of radio resources and MAC parameters;
� Reporting of measurements to RRC.
MAC Functions
MAC functions include the following:
� Mapping between logical channels and transport channels;
� Selection of the appropriate transport format for each transport channel depending
on the instantaneous source rate;
43
� Priority handling between data flows of one UE;
� Priority handling between UEs by means of dynamic scheduling;
� Identification of UEs on common transport channels;
� Multiplexing/demultiplexing of higher-layer PDUs into/from transport blocks
delivered to/from the physical layer on common transport channels;
� Multiplexing/demultiplexing of higher-layer PDUs into/from transport block sets
delivered to/from the physical layer on dedicated transport channels;
� Traffic-volume monitoring;
� Transport-channel type switching;
� Ciphering for transparent RLC;
� Access service class selection for RACH and CPCH transmission.
2.7.3 Radio Link Control (RLC) protocol
The Radio Link Control sub-layer is located in both the UE and the RNC immediately
above the MAC sub-layer according to the radio interface protocol architecture. In the
control plane, it provides services directly to layer 3, while in the user plane it may also
provide services to the PDCP and BMC sub-layers [22]. This layer provides three different
transfer modes to higher layer data flows: transparent mode (TM), unacknowledged mode
(UM) and acknowledged mode (AM). Each mode is associated with a different Service
Access Point (SAP) for upper layers, denoted as TM-SAP, UM-SAP and AM-SAP,
respectively, and with different RLC entities, as shown in Figure 2.12. All these three
modes provide buffering of higher layer messages. The TR and the UM entity have
separated transmitting and receiving entities. The AM entity is realized as one combined
transmitting and receiving entity due to retransmission management.
44
Figure 2.12 RLC sub-layer architecture [16]
In general, the RLC layer is in charge of the actual data packet (containing either control
or user data) transmission over the air interface. It makes sure that the data to be sent over
the radio interface is packed into suitably sized packets. The RLC task maintains a
retransmission buffer, performs ciphering, and routes the incoming data packets to the right
destination task (RRC, BMC, PDCP, or voice codec).
2.7.4 Packet Data Convergence Protocol (PDCP)
Packet Data Convergence Protocol (PDCP) sub layer is standardized in [23]. The Packet
Data Convergence Protocol only exists in the user plane and is specifically for Packet
Switched services. Its main functionality is to improve the efficiency in the radio
transmission by means of executing header compression of the IP data packets coming
from upper layers. UMTS supports several network layer protocols providing protocol
transparency for the users of the service. At the moment, IPv4 and IPv6 are supported.
Introduction of new network layer protocols to be transferred over UTRAN shall be
possible without any changes to UTRAN protocols. Therefore all functions related to
transfer packets from higher layers shall be carried out in a transparent way by the UTRAN
45
network entities. One task of the PDCP layer is this transparent transmission. Therefore,
the functions the PDCP shall perform include the following:
� Header compression and decompression of IP data streams;
� Transfer of user data;
� Maintenance of PDCP sequence numbering.
2.7.5 Radio Resource Control (RRC) layer
The Radio Resource Control (RRC) layer standardized within [24], handles the control
plane signaling of layer 3 between the UE's and UTRAN. As shown in Figure 2.5, Radio
Resource Control is attached to all logical channels that transfer control information.
Further the RRC layer is connected to all entities within the UTRAN in order to exchange
signaling information. The RRC protocol handles a large number of signaling tasks. The
functions of RRC are as follows:
� Broadcast of information related to the non-access stratum (Core Network)
� Broadcast of information related to the access stratum
� Establishment, maintenance and release of an RRC connection between the UE and
UTRAN
� Establishment, reconfiguration and release of Radio Bearers
� Assignment, reconfiguration and release of radio resources for the RRC connection
� RRC connection mobility functions
� Control of requested Quality of Service (��)
� User Equipment measurement reporting and control of the reporting
� Outer loop power control
� Control of ciphering
� Paging
� Initial cell selection and cell re-selection
� Arbitration of radio resources on uplink Dedicated Channel (DCH)
� Timing advance (Time Division Duplex mode)
46
2.8 Radio Resource Management (RRM)
Radio Resource Management (RRM) techniques are used to improve the utilization of
radio resources of the wireless network [25]. RRM operations include essential functions
like admission control, congestion control, power control, handover management, radio
resource allocation and transmission parameters management [26]. The main theme behind
the UMTS is to deliver the multimedia services characterized by stringent real time
requirements, great sensitivity to delivery delay and packet loss and the need for
considerable wireless resources. There are four basic classes of service in UMTS for
quality of service (��) provisioning. These classes are:
Conversational Class: This class is for the most delay sensitive traffic. This class is used
for voice over IP, video conference or any type of real-time interactive traffics. The transfer
delay and delay variation are very strict. However, there are loose requirements on error
tolerance.
Streaming Class: This class is used for real-time voice and video streaming applications.
Because it is unidirectional, it does not have stringent transfer delay compared with the
Conversational Class. However, a maximum bound on delay variation is given to this class.
There is no strict upper limit for the packet loss rate.
Interactive Class: This class is used for web browsing, database retrieval and any kind of
human interaction with remote equipment’s applications. A short response time is expected
for interactivity thus the round trip delay time is important in this class. This class requires
low bit error rate transport.
Background Class: This class is reserved for most delay insensitive applications. This is
because the destination does not have to accept data within a certain time limit. The class
is mainly used for email and database download. It requires low bit error rate transport.
Table 2.3 summarizes the UMTS classes defined by 3GPP.
47
Table 2.3 Four UMTS service class [27] Service Class Class Description Example Application 67� requirement
Conversational � Preserve time relation between entities
� Conversation pattern � Real time
� Voice over IP � Video conferencing � Interactive game
� Low jitter � Low delay
Streaming � Preserve time relation between entities
� Unidirectional continuous stream
� Real time video � FTP � Still image
� Low jitter
Interactive � Bounded response time
� Preserve the payload content
� Web browsing � E-commerce
� Round trip delay time
� Low BER
Background � Preserve the payload content
� Email � Fax
� Low BER
As it is clearly seen from the table above, all service class impose different quality of
service requirements. So to maintain these requirements during communication,
management of radio resources of network is necessary [25]. The main objectives of radio
resource management are to:
� Maximize the performance of all users with coverage and capacity;
� Guarantee the quality of service for different applications;
� Maintained the planned coverage;
� Optimized the system capacity.
Radio resource management (RRM) is divided into two phases as follows:
� Radio resource configuration: This is responsible for allocating the proper
resources to new requests coming into the system as a result it will not cause
network to become overloaded thus compromising stability of network. However
the congestion might occur, thus affecting quality of service (��) due to the
mobility of users.
48
� Radio resource re-configuration: This is responsible for re-allocating the
resources within the network when load is building up or congestion starts to occur
to maintain �� for different applications throughout the network. It should change
overloaded system back to target system by rearranging the resource between
various applications on the same network. Thus Radio Resource Reconfiguration is
also very essential part of RRM and infect of UMTS.
2.8.1 Radio Resource Management (RRM) Function
When taking into account the constraints imposed by the radio interface, Radio Resource
Management functions are responsible for taking decisions regarding the setting of the
different parameters influencing the air interface behavior [16]. The following elements
have been identified to be responsible for taking decisions in RRM:
� The number of active users.
� The number of simultaneous users transmitting
� The corresponding transmission rates for each user.
� The transmitted power levels corresponding to every simultaneous user.
Radio Resource Management schemes can also include a set of service control functions,
which are categorized into network based functions and connection based functions [28].
Network based functions include admission control (AC), load control (LC), packet
scheduler (PS) and resource manager (RM); whereas connection based functions include
power control (PC) and handover control (HC).
Network based functions:
Admission control (AC) -
� Handles all new incoming traffic. Check whether new connection can be admitted
to the system and generates parameters for it.
� Occurs when new connection is set up as well during handovers and bearer
modification.
49
Load control (LC) -
� Manages situation when system load exceeds the threshold and some counter
measures have to be taken to get system back to a feasible load.
Packet scheduler (PS) -
� Handles all non-real time traffic, (packet data users). It decides when a packet
transmission is initiated and the bit rate to be used.
Resource Manager (RM) -
� Controller over logical resources in Base Transceiver Station (BTS) and Radio
Network Controller (RNC) and reserves resources in terrestrial network.
Connection based functions:
Handover Control (HC) -
� Handles and makes the handover decisions.
� Controls the active set of Base Station (BS) of Mobile Station (MS).
Power Control (PC) -
� Maintains radio link quality.
� Minimize and control the power used in radio interface.
Figure 2.13 shows the radio resource management functions implementation on different
areas of a UMTS network.
Figure 2.13 Location of RRM functions [29]
Uu lub
Power Control
UE
Power Control Load Control
Node B
Power Control Handover Control
Admission Control Load Control
Packet Scheduling
RNC
50
2.8.2 Radio Resource Management (RRM) Functions Interaction
Radio Resource Management functions are highly interrelated and coupled as long as they
are all influencing the air interface. Since the objectives of the Radio Resource
Management scheme are to achieve acceptable �� levels for the user application traffic
and to design an efficient radio resource utilization.
Figure 2.14 Radio Resource Management Functions Interaction [27]
In order to achieve an efficient utilization of radio resource, it is very important to clearly
identify the �� requirements of services and the characteristics of user traffics, (table 2.3).
Based on these, the overall performance can be improved by efficiently combining
different Radio Resource Management functions. Admission Control (AC) is the function
that handles all new incoming traffics and checks whether new connection requests can be
admitted to the system subject to a set of admission criteria such as the �� requirements
and the subscriber profile. The AC sends the most up to date system load information to
the Load Control (LC) function, which monitors the load condition of the system. LC also
provides system load information that will enable AC to decide whether to admit a
connection request without violating the system load limit. When the system load exceeds
Resource Manager Admission
Control
Packet Scheduling
Power Control Handover Control Load Control
Connection
establish request
51
the threshold, LC may decide to release existing connections to other lightly loaded
coverage areas or to divert the connection request to another lightly loaded coverage region
or to ‘borrow’ resource from other resource pools in order to accommodate new
connections. Packet Scheduler (PS) decides when a packet is to be transmitted and the bit
rate that is to be used based on the connections quality of service (��) parameters
provided by Admission Control. Resource Manager (RM) is responsible for the logical
radio resource configuration and status, such as the available resources and codes. It
reserves the proportion of the available radio resources according to the resource request
from Admission Control for each connection. Power Control (PC) maintains the radio link
quality to minimize and control the power used and to satisfy the target bit error rate (BER)
and Signal to Interference Ratio (SIR) specified by the Admission Control. Lastly is the
Handover Control (HC), whose function is to handle and make handover decisions.
2.9 Scheduling Schemes
Packet scheduling is a very important aspect of radio resource management in packet
switched wireless networks. It interacts with other RRM control functions in order to
ensure that the user quality of service (��) requirements are respected. The nature of a
scheduling framework can greatly impacts the �� levels that can be provided in the
system.Based on dynamic changes in the network topology and different types of
heterogeneous access networks, next-generation wireless networks must be able to support
the multimedia communications of multiple �� requirements, and simultaneously ensure
high system throughput and low transmission delay [30]. These require a scheduling
technology of wireless networks with very high specific performance.
There will be different kinds of users in wireless networks, which do have distinct quality
of service demands. Some applications require certain characteristics from the assigned
radio resources in order to work, while others are more insensitive.This arises the need for
assigning resources in a smart way, to meet the requirements of the users and also to utilize
52
the available resources most efficiently. Scheduling schemes can be classified into two
groups based on the type of applications they can support. They are:
� Best-effort applications: These applications don't require certain performance in
order to work, they accept whatever resources the network assigns to them. For
example, a file or a web page download of course would prefer high bandwidth and
low end-to-end delay, but it also works with little resources. Scheduling schemes
for serving best-effort applications are: First-In-First-Out scheduling scheme
(FIFO), Weighted Round Robin (WRR) scheduling scheme, Priority scheduling
scheme, Queue Length Dependent scheduling scheme and Channel State Dependent
scheduling scheme.
� Guaranteed-service applications: These applications need a certain amount of
resources in order to work well, e.g., interactive multimedia requires a certain
bandwidth as well as a small round-trip delay. Scheduling scheme for serving
guaranteed-service applications are: Earliest-Due-Date scheduling, Rate-Controlled
scheduling.
2.9.1 First-In-First-Out Scheduling
First-in-first-out (FIFO) is the simplest type of scheduling scheme. The incoming packets
are placed in a single queue and are served in the order as they were received. This
scheduling scheme shown in figure 2.15, requires very little computation and its behavior
is very predictable, i.e. packet delay is a direct function of the queue size [31]. There are
many undesirable properties related to this queuing policy, due to the simplistic nature.
� It is impossible to offer different services for different packet classes since all
packets are inserted into the same queue.
� If an incoming flow suddenly becomes bursty, then it is possible for the entire buffer
space to be filled by this single flow and other flows will not be serviced until the
buffer is emptied.
53
Figure 2.15 FIFO Scheduling [31]
2.9.2 Weighted Round Robin Scheduling
The simplest form of fair scheduling scheme is Round Robin. In a Round Robin scheduling
scheme packets are stored in different classes. Packets of each class have the same chance
to be transmitted in a scheduling period [27]. Therefore every user requires its own queue.
The scheduler assigns the same amount of resources to all users successively in a cyclic
manner, empty queues are skipped. If a user provides less data, the remaining part is shared
among all the others. Simple Round Robin only can serve equal users. If users have
different requirements, Weighted Round Robin (WRR) shall be applied. In WRR, packets
are first classified into various service classes and then assigned to a queue that is
specifically dedicated to that service class. Each of the queues is then serviced in a round
robin (RR) order. The weight indicates how many packets have to be sent in each cycle
from each queue [32]. The WRR scheduler doesn’t take the size of the transmitted packets
into account. As a result, it is difficult to predict the actual bandwidth that each queue
obtains, but it ensures that all service classes have access to at least some configured
amount of network bandwidth
Flow 1
FIFO queue
Flow 2
Flow 3
1
2
3 1 2 3
Scheduler
54
Figure 2.16 Weight Round Robin Scheduling [33]
2.9.3 Priority Scheduling
In priority scheduling, packets are slotted into different queues according to their quality
of service requirements. These queues have different priorities and packets in the higher
priority queues have a higher chance to be transmitted than the packets in the lower priority
queues. Priorities can either be assigned statically to services, or dynamically to single
packets according to their delay and rate requirements. Even though this queuing strategy
is a good way of providing differentiated service, it also has some shortcomings, like large
continuous flow of high priority traffic into the queue, equals excessive delay, and perhaps
even service starvation for lower priority packets [31].
Figure 2.17 Priority Queuing Scheduler [33]
55
2.9.4 Earliest-Due-Date Scheduling
The Earliest Due Date (EDD) scheduling, also known as Earliest Deadline First (EDF)
scheduling, is a classic example of a deadline-based scheme where packets are scheduled
based on the earliest-deadline-first principle [34]. EDD was originally designed for serving
individual flows, but it can also be applied to class based differentiation. Working with the
assumption that the traffic arriving in each class is periodic and using 8� to denote the
period for class �, the EDD algorithm works simply as follows: upon arrival of the 9�
packet of class � at the router at time 1�: , the packet is stamped with a deadline i.e. the sum
of its arrival time and period. <�: = 1�: + 8� (2.3)
The packets are then served in the numerical order of their deadlines. Notice that, in reality,
the arriving traffic is not periodic; the purpose of the period is only to describe the expected
inter arrival time of packets.
2.9.5 Rate-Controlled Scheduling
A rate-controlled scheduler consists of a regulator and a scheduler. The regulator is
responsible for shaping the traffic of each service in order to guarantee conformance with
the desired traffic pattern. Hence, the scheduler receives packets with a predefined rate.
The scheduler then controls the transmission order of packets belonging to different
services. With this approach it is possible to assign to one service simultaneously a lower
bandwidth with higher delay requirements.
2.10 Requirements of a Scheduler
There are some general desirable properties common to all scheduling disciplines. A
scheduling discipline must satisfy four, sometimes contradictory, requirements:
performance bounds, fairness and protection, flexibility and ease of implementation.
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� Performance bounds: This can be expressed deterministically or statistically. A
deterministic bound holds for every packet. Whereas a statistical bound is either
expressed as a mean value or a 95-percentile. This percentile expresses that the
bound is met by 95% of all packets [14]. Deterministic bounds of course require
more network resources to be reserved than statistical bounds. General performance
parameters are bandwidth, delay, delay jitter, and packet loss due to transmission
errors or full queues.
� Fairness and protection: Some notion of fairness is incorporated in many network
mechanisms used today [35]. Fairness is a desirable property of scheduling
algorithms serving equal services. If there are several classes of services, fairness
should be provided within every class. Fairness means that resources are shared
equally among all services which are ready to send. Since real network environment
is not static [34], scheduling discipline should be able to protect and satisfy the
performance requirements of well-behaved users, also in the presence of different
sources of variability, such as best effort traffic, badly behaved users and network
load fluctuations.
� Flexibility: The scheduling discipline must not optimize performance from a single
application’s point of view but should rather be able to accommodate applications
with varying traffic characteristics and performance requirements.
� Ease of implementation: The scheduling algorithm has to do its decisions in real
time, so the complexity of the algorithm determines the hardware requirements.
Therefore it is necessary to compose fast and easy implementable algorithms in
order to keep hardware requirements and time for computation low. Schedulers are
capable of improving transmission for certain services. However, this improvement
is at the expense of worse performance for other services. This fact is revealed by
the conservation law. Consider a set of N connections at a scheduler. Traffic at � arrives at a mean rate >� and the mean service time for a packet from connection � is ?�.
57
Then �� = >� ∙ ?� is the mean utilization of the link due to connection �. If the mean waiting
time for packets of connection � is denoted as A�, then the conservation law becomes:
B -%C
%DE ∙ F% = +7G.& (H. I)
For schedulers which are only idle if all queues are empty. Since this equation is
independent of the scheduling discipline, reduction of the delay for a certain connection
results in a higher delay of other connections.
In WCDMA, packet scheduling algorithms can be done in two ways, in a time or code
division manner.
Time Division Scheduling
� One user is allocated a channel at a time (10 � frame);
� All available capacity can be allocated to that user;
� High data rate for a short period of time;
� Increase more users, each user has to wait longer.
Advantages of Time Division Scheduling
� High bit rate required less energy per bit;
� Less interference;
� Shorter delay due to high bit rate.
Disadvantages of Time Division Scheduling
� High unused physical resources due to short transmission time and relatively long
set up and release time;
� High variations in the interference levels due to high bit rate and bursty traffic;
� Limited uplink range of high bit rate due to mobile’s limited transmission power.
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Code Division Scheduling
� Many users are allocated the channels simultaneously;
� The capacity is shared with all users;
� Low data rate for a long period of time;
� Increase more users, each user’s bit rate is decreased.
Advantages of Code Division Scheduling
� Resources are in full usage due to longer transmission time;
� Small variation in interference level;
� Longer uplink range due to lower bit rate.
Disadvantages of Code Division Scheduling
� Longer transmission delay due to low bit rate;
� High interference due to high energy per bit;
� Low total throughput.
2.11 Related works
IP-based network entities integrated voice and data on unified IP backbone, which can
increase the resource utilization over existing mobile networks. WCDMA radio access
network must manipulate the delay-sensitive real-time packets to provide IP multimedia
service. Several scheduling schemes have been proposed in the literature for IP-based radio
access networks in WCDMA to efficiently utilize radio resources.
In [36], a credit based scheduling algorithm is proposed for use in the forward link of a
CDMA system. This algorithm dynamically assigns an OVSF code to a mobile user on a
timeslot-by-timeslot basis. On connection set-up the mobile user negotiates with the
network management module a guaranteed bandwidth, denoted as GBW. Throughout the
duration of a connection, the base station keeps track of a priority variable called the credit
C:
/ = 2�21J��KL × MNO − 2�21J_RST�_�U_V����W�8_�1�9�2�_��_U1V (2.5)
59
In this scheduling algorithm connections with more credits are scheduled to receive more
packets. This type of credit-based prioritization does not provide low packet delays and �� differentiation. In [37], Delay Fair Scheduling (DFS) scheme is proposed. This
scheme has low computational complexity and provides fair distribution of the available
shared capacity to the connections according to their delay requirements. According to
DFS each connection �at the start of each frame n, is characterized by its priority X�. X� = <�(R�3)�� ≥ 0, R = 0,1,2, … (2.6)
Where, ��= threshold for the acceptable data packets delays, defined during connection setup, <�= head-of-line packet delay for queue i and �3= 10ms, is the scheduling period of the DFS scheme.
Subsequently, the connections are sorted and served in decreasing order of their priorities.
This scheduling scheme fail to utilize efficiently the scarce radio resource. The Delay Fair
Scheduler with prediction (DFS_PRED) was proposed in [38]. The main idea of
DFS_PRED is to prioritize the connections, not only according to their delay requirements
but also according to their predicted error probability during the next frame. As with DFS
each connection �at the start of each frame n, is characterized by its priority X�. X� = <�(R�3)�� \1 − XLK(�)] ≥ 0, R = 0,1,2, … (2.7)
However, in this case the priority of a connection is proportional to its probability of
successful transmission (1⎼XLK(�)). Consequently, between two connections with the
same head-of-line packet delay and delay threshold, the one with the higher probability for
successful transmission will also have higher priority. Thus, DFS_PRED is able to
encounter the variable capacity of the wireless interface better than DFS. DFS_PRED
serves the connections not only according to their delay sensitivity, but also according to
the predicted state of their wireless channel. The simulation results shows the efficiency of
the scheme in terms of average packet delay and bandwidth utilization.
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The scheduling schemes presented in [39,40] are also developed for WCDMA systems.
These schemes have a scheduling discipline that resembles to the Wireless Fair Queuing
as they directly assign to each connection � a guaranteed bandwidth 0`,� instead of a weight
that corresponds to a fraction of the available capacity. Furthermore, they use a priority
variable called the credit /�, which is the difference between the error-free service (2�21J_2�� × 0`,�) and the actual number of packets a,� that each flow has received so
far. Therefore, the credit /� of flow � is defined as follows: /� = \2�21J_2�� × 0`,�] − a,� (2.8)
Connections with more credits are scheduled to receive more packets. Assuming perfect
power control, credit based prioritization provides, in the long run, a data rate guarantee to
each accepted connection. The above mention credit based schemes supporting multimedia
traffic either do not consider channel condition or fail to address the exact code position in
the code tree, which may result in inefficiency in resource utilization. A credit-based
scheduler which considers channel condition and explores the concept of compensation
codes is proposed in [41]. With this channel-sensitive scheduling algorithm, a user with
more credits will have more chance to transmit without compromising to the transmission
quality. Simulation results justify that this scheme work as claim.
Many of the proposed schemes take into account the throughput and fairness. However,
the environment, which were considered in those schemes, is that, at each frame, the base
station can transmit to at most one mobile node on a separate spectrum. This environment
is very different from the UMTS network in which all the mobiles share the same spectrum.
Sallents et al. [42] proposed a packet scheduling algorithm, the real time emulator (RTE)
with transmission power constraint for UMTS. This scheduler serves the packet based on
the priority value, but the fairness property was not considered. Thus, the amount of service
time allocated to the ill-behaved users can be more than that of the well-behaved users. In
[43], a scheduling scheme that select packet based on the value of the service credit is
proposed. This scheme allocates the data rate based on the time scheduling strategy or the
61
code scheduling strategy. Using this scheduling algorithm, the soft �� can be guaranteed.
However, this scheduling algorithm has an undesirable effect that the backlogged flow can
be starved for an arbitrary period of time as a result of excess bandwidth it received from
the server when other flows are idle.
Min-Xiou Chen, Ren-Hung Hwang in [44], proposed two scheduling algorithms, Multi-
flows Worst-case fair Weighted Fair Queuing plus (MWF2Q+) and Multi-flow Deficit
Round-Robin (MDRR), for multiple classes of service over the same spectrum in the
forward link of the UMTS network. MWF2Q+ and MDRR, are based on the WF2Q+ and
DRR algorithms, respectively. These two algorithms are well studied in wired broadband
networks. The WF2Q+ is known for its excellent fairness properties and DRR is known for
its low computational complexity while maintaining reasonable fairness properties.
The algorithm of MWF2Q+
In MWF2Q+ scheduler, each time a packet arrives or a packet of backlogged flows gets
served, the virtual time will be updated. The MWF2Q+ scheduler is based on the
Generalized Processor Sharing scheduling (GPS) discipline [45]. In MWF2Q+ scheduler,
if all flows in U are all in backlogged mode (have data to send in their sending queues), the
data rate allocated to flow � is,
b�/Kcd∑ bff∈h (2.9)
Where / is the system capacity. As long as ∑ b� ≤ 1,f∈h flow � can be guaranteed with a
minimum rate of, V� = b�/ (2.10) V� is the minimum guaranteed rate for flow �, and can be any positive number.
In MWF2Q+, j�(2) is determine based on the virtual time of flow �. j�(2) denotes the
amount of data served at frame 2 for flow �. For the UMTS system, a new constraint should
be added, that is, j�(2) ∈ {special rates provided in UMTS}. The virtual time of flow � can
be derived from the GPS discipline. If ��: is the 9� packet of flow �. The arrival time and
62
packet length of the packet are denoted as k(��:) and j�: respectively. If \X�:] and l\X�:]
also represent the virtual start tag (time) and virtual finish tag (time) of the packet,
respectively. Then, m(2) becomes the system virtual time. \X�:], l(X�:) and m(2) are
defined as follows:
\X�:] = n l\X�:op], �U ��[k(X�:)o] ≠ 0max {l\X�:op], m[k\X�:]]}, �U ��[k(X�:)o] = 0 ( 2.11)
l\X�:] = \X�:] + j�:V� (2.12)
m(2 + y) = max {m(2) + O(2, 2 + y), min�∈|(�)[\X� (�)]]} (2.13)
Where ��[k(��:)o] and ��[(2 − 1)o] are the queue length of flow � at the time just before k(��:) and the (2 − 1)2ℎ frame, respectively. O(2, 2 + y) is the total amount of service
during the period [2, 2 + y], N(2) is the set of backlogged flow at time 2, and X� (�) is the
packet at the head of flow �′� queue at time2. At initial, m(2) = 0. In WF2Q+, only one
flow at a time transmits a packet and that packet must be transmitted without interruption.
The system serves those packets based on the increasing order of their virtual time. In
UMTS, multiple flows can transmit their packets simultaneously on the same spectrum.
Furthermore, the definition of the virtual time for each flow in MWF2Q+ is very different
to that in WF2Q+. The virtual time of flow �, m�(2) is further define as follows:
m�(2) = ~ m�(2 − 1) + ��(�op)� , �U ��[(2 − 1)o] ≠ 0\X�:], �U ��[(2 − 1)o] = 0 1R8 ��[(2)o] ≠ 0 (2.14)
63
Assuming ��� denotes the data rate of layer � in the OVSF code tree, and .�% denotes the
minimum data rate for flow % in the UMTS, and .�% can be defined as .�% =��� {��%|��% ∈{Special rates provided in UMTS}, and �% ≥ ��%}. As long as ∑ �%%∈� ≤�and∑ .�% < �%∈$(&) . The residual bandwidth (RB), should be distributed to the
backlogged flows in a fair and equitable manner. Where,
0N = 1?�S ���2� �1�1��2� − B �V� �∈|(�) (2.15)
In [46], an experimental study of a new scheduling policy for achieving fairness,��, and
optimal use of resources is proposed. The proposed scheduling policy Code Division
Multiple Access based on Generalized Processor Sharing with Dynamic Weights
(CDMA/GPS-DW) is an improvement of a previous GPS policy. This scheme utilizes
dynamic weights for bandwidth assignment, the weights are calculated as a function of the
number of active Mobil Terminals (MTs). Simulation results show that the proposed policy
achieves fairness of the specified �� and makes efficient use of the network resources.
This scheme is not flexible in the traffic management, that is, it is restricted only for three
traffic types. In addition, admission control is not considered and it is designed without
considering the effect of multipath fading in the cell.
Mendez, Panduro, Covarrubias and Romero [47] proposed a rate scheduling scheme which
is based on GPS. This scheme treats different traffic type according to their quality of
service requirements in the uplink of CDMA cellular network. Multiple traffic with
variable traffic rate can be served simultaneously. This is consider a drawback of GPS
because the classical packet-based systems are based in TDMA. With this scheme, the
flexibility in bandwidth allocation of CDMA system is exploited. The analysis and
simulation results shows that this scheme is an improvement of the CDMA-GPS in [46].
The improvement is based on the dynamic weight assignment of bandwidth allocation for
each type of service provided (e.g., video, voice and WWW-data), as compared to the static
weight assignment of bandwidth allocation utilized by CDMA-GPS scheme.
64
Skoutas and Rouskas [48] proposed a Dynamic Priority Allocation scheduling algorithm,
which is designed to operate within a cross-layer framework that provides Dynamic
Priority Allocation (DPA) with the necessary information in order to take into account the
variations of the wireless channel. The proposed scheme is designed for �� provisioning
in the Downlink-Shared Channel (DSCH) in WCDMA 3G systems. The ��
differentiation between connections is based on their delay sensitivity and head-of-line
(HOL) packet delay. The DPA scheme has low computational complexity and provides
fair distribution of the available DSCH capacity to the connections. By providing a
guaranteed rate per traffic flow at each scheduling period, DPA is able to offer a
deterministic delay bound to each session when the transmission is constantly reliable and
a stochastic delay bound for identical DSCH connections with certain constraints.
Simulation results demonstrate Dynamic Priority Allocation (DPA) fairness property and
its efficiency.
Wigard, Madsen and Gutiérrez [49] proposed a packet scheduling algorithm, that can
differentiate the �� among user and service classes in WCDMA. At the same time a
parameter is introduced, which gives the possibility of adjusting the packet scheduling
algorithm from signal to interference ratio (C/I) based scheduling to inverse signal to
interference ratio (C/I). The algorithm can be tuned from signal to interference ratio (C/I)
based scheduling to Round Robin and beyond. The algorithm can upgrade the priority of
the users in the queue, in order to avoid unacceptable delays for low priority users. This
part can be used to cut the tails of the packet call delay distributions. Retransmissions are
given a relatively high priority in order to avoid timeout and unnecessary retransmissions
caused by higher layer protocols.
Wan, Shih and Chang [50] proposed three real-time scheduling algorithms to support
quality-of-service at IP-based radio access networks for the UMTS. The real-time generic
scheduling (RTGS) algorithm applies the functionalities of the radio management
framework to establish new data sessions for real-time service requests. The real-time
bandwidth scheduling (RTBS) algorithm implements the early-deadline-first (EDF)
65
scheme to do the schedulability analysis and to schedule the data sessions to reduce power
consumption. The real-time code scheduling (RTCS) algorithm (RTCS) applies Dynamic
Code Assignment (DCA) scheme to improve radio resource utilization. Experimental
results show that, under various traffic loads, RTCS performs best in terms of power
consumption, session drop rate and bandwidth utilization. It also shows that RTBS
outperforms RTGS.
Chandramathi, Raghuram, Srinivas and Singh [51] proposed a fuzzy logic (FL)-based
dynamic bandwidth allocation algorithm for multimedia services with multiple quality of
service ��: Probability of blocking (PB), Service access delay (SAD), Access delay
variation (ADV) and the arrival rate requirements. In this algorithm, each service can
declare a range of acceptable quality of service levels (e.g. high, medium, and low). As
quality of service demand varies, the proposed algorithm allocates the best possible
bandwidth to each of the services. This maximizes the utilization and fair distribution of
resources. Simulation results show that the required quality of service can be obtained by
appropriately tuning the fuzzy logic controller (FLC).
Xu, Shen and Mark [52] proposed a code-division generalized processor sharing (CDGPS)
fair scheduling dynamic bandwidth allocation (DBA) scheme for WCDMA systems. The
scheme exploits the capability of the WCDMA physical layer by allowing channel rates to
be dynamically and fairly scheduled by varying the spreading factor and/or using multiple
code channels, rather than allocating service time to each packet. Analysis and simulation
results of the model shows that bounded delay can be provisioned for real-time application
by using generalized processor sharing (GPS) service discipline, while high utilization of
system resources is achieved.
Gürbüz and Owen [53] proposed Dynamic Resource Scheduling (DRS) Scheme as a
framework that will provide quality of service provisioning for multimedia traffic in W-
CDMA systems. This scheme is an extended DRS family that is aimed at examining the
temporal quality of service in terms of delays. The DRS framework monitors the traffic
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variations and adjusts the transmission powers of users in an optimal manner to
accommodate different service classes efficiently. Variable and optimal power allocation
is also suggested to provide error requirements and maximize capacity, while prioritized
queuing is introduced to provision delay bounds. Simulations of this scheme shows that
the delay performance can be provisioned for guaranteed services by multiple queues.
Xu, Shen and Mark [54] proposed a credit-based CDGPS (C-CDGPS) scheme to further
improve the utilization of the soft capacity by trading off the short-term fairness. With the
C-CDGPS scheme, the soft uplink capacity is allocated by using a combination of credit-
based scheduling and CDGPS fair scheduling. The model considered a frequency division
duplex (FDD) Wideband DS-CDMA network supporting a large number of multimedia
users. Packetized multimedia traffic is considered. Simulation results shows that bounded
delays, increased throughput, and long-term fairness can be achieved for both
homogeneous and heterogeneous traffic.
Xu, Shen and Mark [55] proposed a dynamic fair resource allocation scheme to efficiently
support real-time and non-real-time multimedia traffic with guaranteed statistical quality
of service (QoS) in the uplink of a wideband code-division multiple access (CDMA)
cellular network. The scheme provides a trade-off between the Generalized Processor
Sharing (GPS) fairness and efficiency in resource allocation to maximize the radio resource
utilization under the fairness and quality of service constraints. Analysis and simulation
results show that, in a multipath fading environment, the proposed scheme can reduce the
inter-cell interference, increase the network capacity, guarantee a statistical delay bound
for real-time traffic and a statistical fairness bound for non-real-time users.
Salman [56] proposed a Multi-operators Code Division Generalized Processor sharing (M-
CDGPS) scheme for supporting Multiservice in the uplink of WCDMA cellular networks
with multi-operators. The scheme employs both adaptive rate allocation to maximize the
resource utilization and Generalized Processor Sharing (GPS) techniques to provide fair
services for each operator. The simulation results show that the proposed scheme improve
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both system utilization and average delays. The proposed scheme allows for a flexible
trade-off between the GPS fairness and efficiency in resource allocation and is an effective
way to maximize the radio resource utilization under the fairness and QoS constraints.
2.12 Conclusion
Universal Mobile Telecommunication System (UMTS) quality of service (QoS)
architecture has been summarized in this chapter. This QoS architecture is used for
designing the Radio Resource Management (RRM) scheme for the UMTS system under
consideration in this research work. Radio resource management service functions, such as
power control, admission control, load control, handover control and packet scheduling are
also presented. Various algorithms, strategies and classification for the RRM frameworks
have been reviewed in order to provide the background knowledge for the design of
resource allocation scheme for the UMTS compatible system.
From the related works above on resource allocation techniques, the classes of techniques
based on the Generalized Processor Sharing (GPS) provide more flexibility in bandwidth
allocation as they ensure fairness while dynamically allocating resources. But, in all the
techniques outline in section 2.11, the issue of backlogged flow loss rate has not been well
addressed. In the next chapter, the Code-Division Generalized Processor Sharing (CDGPS)
scheme will be discussed in detail.
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CHAPTER THREE
RESEARCH METHODOLOGY
3.1 System Model
This section gives the design of the scheduler functional architecture for providing quality
of service requirements, while also achieving efficient utilization of radio resources. A
code-division generalized processor sharing (CDGPS) is proposed for WCDMA systems,
to support differentiated quality of service (QoS) with a central controller that can
dynamically allocate bandwidth to mobile users according to the variation of channel
condition and traffic load. The CDGPS scheduler makes use of both the traffic
characteristics in the link layer and the adaptivity of the WCDMA physical layer to achieve
efficient utilization of radio resources. It adjusts only the channel rate (service rate) of each
traffic flow in the WCDMA system by varying the spreading factor and/or using a multiple
of orthogonal code channels, rather than allocating service time to each packet. This results
in lower implementation complexity of the CDGPS scheme than for a conventional GPS-
based time scheduling scheme.
The system model considered in this work, is the frequency division duplex UMTS cellular
network (UMTS-FDD) where user equipment (UE) are interconnected with the Internet
through Node B, Radio Network Controller (RNC) and core network, as shown in figure
3.1. The radio link in the UMTS-FDD system can be characterized by orthogonal channels
in the downlink (from Node B to UE) and multiple access channels in the uplink (from UE
to Node B). A pair of bandwidth schedulers are assumed to reside in each Node B. The
schedulers allocate the power and respective rate of the channels in the downlink and
uplink to all UEs in the same cell covered by Node B. Although the capacity of the
downlink is equal to the uplink capacity, the discussion in this chapter only focuses on the
uplink.
69
Figure 3.1 Network Structure
In this work, the network architecture illustrated in Figure. 3.1 was employed. The physical
data channels in the uplink comprises a small number of random access channels and a
large number of dedicated channels. Each mobile user is assigned to at least one dedicated
data channel, and shares the random access channels with other users in the same cell.
While signaling and short messages may be transmitted freely through the random
channels, most IP data flows are scheduled for transmission on the dedicated channels.
Multimedia IP traffic (e.g., voice, video, and data) are supported by this network. In a
multimedia IP traffic, the quality of service requirements general consist of two parts:
� Delay
� Loss rate
UE
UE
UE
UE
Node B
Node B
RNC
RNC
Core Network
Wireless backbone (Internet)
Node B
Downlink
Scheduler
Uplink Scheduler
UE UE UE
Radio Link
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3.2 Generalized Processor Sharing (GPS)
A Generalized Processor sharing (GPS) server is work conserving server, which implies,
the server must be busy if there are packets waiting in the system. GPS sever also operates
at a fixed rate V, and is characterized by positive real number bp, b�, … . b� . These
numbers denote the relative amount of service in each session, that is, if �(y, 2) is defined
as the amount of session � traffic served during an interval (y, 2), a session is backlogged
at a time 2 if a positive amount of that session’s traffic is queued at that time 2. Then a GPS
server is defined as one for which the following inequality holds for any session � that is
continuously backlogged in any interval (y, 2) [45], equation 3.1 will holds with equalities,
and the allocated bandwidth of each user is exactly proportional to its weight.
�(y, 2)f(y, 2) ≥ b�bf , � = 1,2, … . � (3.1)
A basic principle of GPS is to assign a fixed positive real number (namely weight), instead
of a fixed bandwidth, to each flow, and to dynamically allocate bandwidth for all flows
according to their weights and traffic load. Due to the burstiness of packet traffic,
sometimes a user may not have packets to transmit and gives up its bandwidth for a while.
The excess bandwidth can be distributed among all backlogged sessions at each instant in
proportion to their individual weight b�. This makes the GPS server efficient and fair in
bandwidth allocation.
3.3 The Code-Division Generalized Processor Sharing (CDGPS) scheme
The proposed scheme in this research work is the code-division generalized processor
sharing (CDGPS) fair scheduling scheme. This scheme uses the GPS fair scheduling
discipline to dynamically allocate channel rates. The model of figure 3.2 comprise of a
server of capacity, C Mbps. The input traffic is from varied sources comprising of voice,
video, and data traffic, which are bundled into flow classes. Each flow maintains a
71
connection with link rate /�(9) during the 9� time slot. The sum of /�(9) over all users
should not exceed C.
Figure 3.2 A queuing model of the CDGPS scheme
For each slot, the scheduler allocates adequate service rates to the N flows, using the
following scheduling procedure:
� Let the pre-assigned weight for flow � be b� , � = 1,2, … , � and �(9) to denote the
amount of session � traffic that would be served during time slot k. According to the
GPS scheduling discipline, Eq. (3.1) should hold for any flow � that is continuously
backlogged in the time slot k. Then, the proposed CDGPS server allocates
each /�(9) using the following steps:
• Step 1: Let N�(9) be the total amount of backlogged traffic of flow � during time slot
k. Estimate N�(9), � = 1,2, … . �, as follows: N�(9) = ��(y:) + V�(9)� (3.2)
Where y: is the end time of slot (9 − 1), ��(y:) = Backlogged traffic at time y:, V�(9) = Estimated traffic arrival rate of flow � during time slot k
Flow 1
Flow 2
Flow N
Scheduler
Connection 1 (/p(9))
Connection 2 (/�(9))
Connection N (/�(9))
C
Radio Link (Capacity = C)
72
The estimated traffic arrival rate V�(9) of flow � during time slot 9 can be estimated from
past traffic measurement using the following two approaches:
1. One-step estimation – the estimated traffic arrival rate is thus:
V�(9) = 1�(9 − 1)� (3.3)
Where 1�(9 − 1) is the total amount of the arrival traffic (in bits) during time slot (9 − 1). 2. Exponential averaging –Let 2�� and J�� be the arrival time and length of
the R� packet of flow �, respectively. The estimated rate of flow �, V� , is updated
every time a new packet arrives:
V��L� = \1 − �oa�� �⁄ ] j����� + �oa�� �⁄ · V�54� (3.4)
Where ��� = 2�� − 2��op and � is a constant. An approximate value for � would be
between 100 and 500 �. ��� = Inter packet arrival time.
Step 2: Based on the estimated N�(9), � = 1,2, … . , �, the expected amount of
service �(9) received by �th user is determine thus:
�(9) = � 0, �U N�(9) = 0���, �U N�(9) > 0
where � is the scheduling period in CDGPS scheme,
��(9) = b�/∑ bf�fDp (3.5)
is the minimum guaranteed rate of flow � and C is the network capacity.
If ∑ �(9) < /�,��Dp then the remaining network resource is distributed
proportionally to users who expect more than their guaranteed service rate. The
distribution of the remaining network resources should be in proportion to each
73
user’s weight b� according to the GPS service discipline. The allocated channel rate
to user � can then be determined by equation (3.6).
/�(9) = �(9)� (3.6)
The CDGPS scheme weights are related as (bp = b� 2� = b 3� ), that is, different
priority values, with b ≥ b� ≥ bp, where b corresponds to highest priority
and bp corresponds to the lowest priority [52]. These values (1, 1/2, 1/3) do not
guarantee the maximum data transmission under UMTS platform (384 Kbps).
Therefore, a different set of values (1/5, 1/3, 1/2) that better utilize the available
bandwidth in UMTS is presented. Where 1/5 corresponds to the lowest priority and
1/2 corresponds to the highest priority.
The priority CDGPS and non-priority CDGPS scheme flowchart are depicted in
figure 3.3, and figure 3.4 respectively. Both figures 3.3 and 3.4 illustrate the
operation of the proposed scheduling scheme. The only difference between priority
and non-priority CDGPS flowcharts is the different ways in which the backlogged
flows N�(9), are sorted. For priority CDGPS, the backlogged flows are sorted by
decreasing order of weight b� , while for non-priority CDGPS, the backlogged flows
are sorted by first-come-first-serve (FCFS) order of weight b� . After the sorting
operation, the first flow from the sorting list is removed. If the total amount of traffic
of flow � is greater than zero and the condition in equation 3.2 satisfied, then the
estimated rate of flow � is updated every time a new packet arrives, otherwise the
sorting list is empty. The expected amount of service �(9) by �2ℎ user is received
based on the estimated amount of traffic of flow �. If the sorting list is empty and
the total amount of traffic of flow � is equal to zero, then the amount of service
received by �2ℎ user becomes zero.
74
Figure 3.3 Priority CDGPS flowchart
1. Sort the backlog flow N�(9) by decreasing
order of weight b�
2. Remove the first flow from the sorting list
If N�(9) > 0, and N�(9) = ��(y9) + V�(9)� ?
Update the estimated rate V�(9)
of flow �, V�(9) = 1�(9 − 1)/�
3. Assign �(9) = ¡��, ¡� = b�/
∑ b���=1
If
B �(9) < /��
?
Repeat step 2 and 3
Sorting list not
empty?
No
Yes
Yes
Yes
If N�(9) = 0
?
No
Then, �(9) = 0
Yes
No 1
Flow i is not backlogged
No
1
Idle
Flow generation
Active
2
2
75
Figure 3.4 Non-priority CDGPS flowchart
1. Sort the backlog flow
N�(9) by first-come-first-
serve order of weight b�
2. Remove the first flow from the sorting list
If N�(9) > 0, and
N�(9) = ��(y9) + V�(9)� ?
Update the estimated rate V�(9)
of flow �, V�(9) = 1�(9 − 1)/�
3. Assign �(9) = ���, �� = b�/
∑ b���=1
If
B �(9) < /��
?
Repeat 2 and 3
Sorting list not
empty?
No
Yes
Yes
Yes
If N�(9) = 0
?
No
Then, �(9) = 0
Yes
No 1
Flow � is not backlogged
No
1
Idle
Flow generation
Active
2
2
76
3.4 Traffic Source Model
In traffic source model, while the characterization of voice users is fairly straight forward,
the traffic generated by packet data users is highly dependent on the application and has a
high degree of burstiness [57]. Multimedia traffic is very bursty in nature and simple
models as the Poisson process do not capture the important characteristics of this sources.
To model bursty traffic sources different approaches are available [58], many of them using
Markov modulated processes (MMP). These are doubly stochastic processes in which each
state of N states of embedded Markov chain originates another stochastic process. If this
originated process is a Poisson process the MMP is called Markov Modulated Poisson
Process (MMPP), if it is deterministic it is a Markov Modulated Deterministic Process
(MMDP).
3.4.1 Voice Source Modeling
In this work, the voice sources are simulated using the ON–OFF model for a single source
and aggregating many such sources. This is a Markov Modulated Deterministic Process
with only two state as shown in figure 3.5.
OFF ON
α
β
Figure 3.5 On-Off Model
m!
77
In the voice on-off model, the duration times of burst and silent period is exponentially
distributed with mean2¢� � 1/£ and2¢¤¤ � 1/¥ respectively. Some studies have
proposed2¢� � 0.4���and 2¢¤¤ � 0.6���[59], setting the transition rates to:
¥ �1
2¢¤¤�
1
0.6� 1.671R8,
£ �1
2¢��
1
0.4� 2.5
The voice source generation follows exponential inter-arrival times ∆t with constant arrival
rate of 16 Kbps. The voice activity factor2¢�is assumed to be 0.4 sec. In the active state
packets are generated with a constant speedm¦ � j ∆2⁄ , withjas the packet length
and∆2as the packet inter-arrival time. This is depicted in figure 3.6.
3.4.2 Video Source Modeling
An accurate traffic model of VBR video is necessary for evaluating the performance of a
network design. A major component of multimedia networking is the data compression
(source coding) of multimedia data sources i.e. speech, audio, image and video. The
process of reducing the amount of data required to represent a digital video signal, prior to
transmission or storage is called video compression or video encoding [60]. Once the data
is compressed, the bit stream is packetized and sent over the Internet. In this work the
variable bit rate (VBR) video sources are simulated by Markov Modulated Poisson Process
2¢�=0.4 sec
2¢¤¤ � 0.6���
2¢� + 2¢¤¤ = 1���
Time
Figure 3.6 On-Off voice packetization
78
(MMPP). Each video source is governed by an m-state Markov chain with probability
transition matrix P = ��f, where,
��f �numberoftransitionsfromstate�tostate�
numberoftransitionsoutofstate�(3.7)
Where�, � = 0,1,2, … , − 1,when a source is in state, it generates rate, 0KT��.
The average duration in each state and/or the length of video transmissions is assumed to
be exponentially distributed with a mean chosen to be 40 ms, which is equivalent to the
length of one frame of the video sequence with a frame rate of 25 frames/s.
3.4.3 Data Source Modeling
The data sources are simulated using the ON–OFF model for a single source and
aggregating many such sources. The ON-OFF model implies, there will be only 2-state
model (source silent period or source producing data at a chosen peak rate). Data source
generation too follows exponential inter-arrival times, but unlike the case of voice, where
bit rate is constant, for data source it varies randomly. A Poisson process is used to generate
the data traffic with packet size 2560 bits and average arrival rate 256 kb/s.
79
Table 3.1 Simulation parameters
3.5 Model Validation
The proposed CDGPS model is validated using the work in [54]. Both works are related
because, they made use of dynamic bandwidth allocation mechanism to serve each flow or
user according to their estimated arrival rate in the uplink direction. Figure 3.7 shows the
throughput comparison of the proposed CDGPS and Liang Xu CDGPS. The comparison
shows that, the proposed model attained an average performance level of 97.9%to that of
Liang Xu CDGPS.
Parameter
Radio access mode WCDMA (FDD) uplink
Chip rate 3.84 Mcps
Spread spectrum 5.0 MHz
WCDMA channel rate 2.0 Mbps
Slot duration 0.667 ms
Frame duration 10 ms
Voice source rate 16 Kbps
VBR video source rate 16 to 384 Kbps
Data source rate 256 Kbps
Voice active factor 0.4
Packet arrival Poisson
Packet generation type Exponential
Queue type FIFO
TTI 100
Value
80
Figure 3.7 Model validation with CDGPS scheme
3.6 Conclusion
This chapter has investigated the general system model requirements for achieving the
dynamic radio resources allocation. The proposed CDGPS scheduling algorithm, presents
a different set of priority values that better utilize the available bandwidth in UMTS. The
proposed CDGPS has been implemented with MATLAB and validated.
0.7
0.75
0.8
0.85
0.9
0.95
1
1.05
0 1000 2000 3000 4000 5000 6000 7000 8000 9000
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Proposed CDGPS
Liang Xu CDGPS
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CHAPTER FOUR
SIMULATION AND RESULT ANALYSIS
4.1 MATLAB Simulation Framework
MATLAB modeler version R2010b is chosen as the simulation method to evaluate the
performance of the designed dynamic bandwidth allocation algorithms for the WCDMA
system. The computer simulation model in figure 4.1 was simulated in the Simulink
environment to evaluate the performances of the dynamic scheduling algorithms for
multimedia IP traffic (e.g. voice, video and data).
Figure 4.1 is the entire system model and is summarized thus; the model simply consist of
traffic sources, buffer, UMTS server and the scheduler computational model. Each source
block is an aggregation of voice, video and data traffic source with different quality of
service requirements. These source blocks are served on a first-in-first-out basis in the
buffer block, which queue is serviced by the UMTS server at dynamic rate for all the source
distribution. The path-combiner was employed in the simulation model in order to ensure
that the arriving traffic has equal probability of being served in a random manner. The
system server block uses a service discipline of first-in-first-out (FIFO) to service voice,
video, or data traffic source at a service rate based on their individual quality of service
requirements. The computational model block does the actual scheduling and bandwidth
allocation processes and also ensuring that, the scarce resource is utilized efficiently.
82
Figure 4.1 Simulation Framework for WCDMA systems
83
Figure 4.2 Multimedia IP traffic (voice, video and data)
Markov-modulated Poisson process typically models the ON-OFF traffic pattern for the
services supported by a UMTS network, as shown in figure 4.2. Services such as voice,
video and data are supported. Each service is model and generates its entities using a
Markov-modulated Poisson process, whose rate depends on the state of the Markov chain.
The model in figure 4.2 includes three independent modulated Markov sources whose
behavior depends on the rate of the Poisson process when the Markov chain is in the “ON”
state. The Path Combiner block aggregates the output of all the On-Off Modulated Markov
Source subsystems.
84
Figure 4.3 Buffer queuing Model
Figure 4.3 showing the buffer queuing model. This model was developed using a FIFO
buffer, were entities are served every one second observation period through an entity
departure counter that is triggered by a repeating sequence block. These entities are stored
in the FIFO Queue block, released and translated back into a signal by the Get Attribute
block. A FIFO buffer stores data as a part of data exchange between processors. The FIFO
Queue block enables simulation of such buffers in software. Each processor is driven by a
clock. Because each clock synchronizes the processor hardware, the clock appears
synchronous to that processor.
85
Figure 4.4 The system Server
The single server block as shown in figure 4.4 serves one entity for a period of time, and
then attempts to output the entity through the OUT port. The server uses first-in-first-out
service discipline to service voice, video, or data traffic at a service rate that is based on
the individual quality of service of each service class. The Embedded MATLAB Function
block allows one to add MATLAB functions to Simulink models for deployment to
embedded processor, the quality of service of individual class of service is computed and
compared before attempting to output the entity and feed it as the new updated service rate
into the server.
86
Figure 4.5 CDGPS Computational model
Figure 4.5 shows the detailed simulation model of the rate scheduling procedure. From the
rate scheduling model, the constant block divides the service rate coming from the server
for each flow, i.e., voice, video, or data, and are scheduled slot by slot. By the end of each
time slot, the used bandwidth capacity is then subtracted from the total UMTS capacity (2
Mbps) and the remaining capacity is stored into the system memory block. Upon receiving
bandwidth requests from all backlogged flows (active users), the next service rate
computation is done in the form of written MATLAB function using the embedded
MATLAB function block. The scheduler then allocates the service rate for the next slot,
taking into account the individual quality of service of each flow and the available uplink
87
capacity. The total channel rate used by different service class is summed up by the add
block and subtract from the total UMTS channel rate by the subtract block to aid the
computation of the service utilization.
Figure 4.6 A Scope of entities generated
Figure 4.6 is a scope showing the number of entities (traffic) generated by the server at
any given time.
4.2 Performance metrics
When evaluating the quality of service, several metrics are used in this work such as
Throughput, Average delay, Packets loss rate.
� Throughput: This is the amount of successfully transmitted packets for each flow
divided by the amount of total sent packets. It is computed as
�ℎV�S¡ℎ�S2 = ∑ �923L��� . (1 − X4533�)�∑ �923L����
(4.1)
88
Where �923L��� is the number of packets sent by the flow i, X4533�is the packet loss
probability of the flow i, and �923L��� . \1 − X4533� ] is the number of successfully
transmitted packets by the flow i.
� Average delay: This measure the average simulation time taken for a packet to be
transmitted between RNC and UE
� Loss rate: This is the ratio of the number of traffic lost to the total number of traffic
offered.
4.3 Simulation Results
In this section, simulation results are presented to demonstrate the performance of the
proposed CDGPS in terms of delay, throughput, loss rate, and utilization. In the simulation,
two different scenarios (priority and non-priority) CDGPS scheme were compared under
heterogeneous traffic environment. The uplink capacity is assumed to be a constant / =2 �T��. Voice, video, and data traffic were considered for the two scenarios. In priority
CDGPS, different set of weight b� values (1/2, 1/3, 1/5) are assigned to voice, video and
data respectively, while in the non-priority CDGPS, equal weight 1/3 is assigned to voice,
video and data. In the simulation results, a percentage value is used to compare the
performance of the two scenarios.
89
Figure 4.7 Throughput as a function of Traffic intensity for multimedia IP traffic
Throughput for priority and non-priority CDGPS
Figure 4.7 shows the throughput comparison of priority and non-priority CDGPS. The
traffic intensity is the sum of the average arrival rate of the three service class. It is shown
that an increase in traffic intensity results in an increase in throughput but the priority
CDGPS do not offer any improvement on the uplink throughput.
0.75
0.8
0.85
0.9
0.95
1
0 1 0 0 0 2 0 0 0 3 0 0 0 4 0 0 0 5 0 0 0 6 0 0 0 7 0 0 0 8 0 0 0 9 0 0 0
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Priority
Non-priority
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Figure 4.8 Throughput per flow as a function of traffic intensity
Throughput per flow
In figure 4.8, the throughput per flow as a function of traffic intensity is shown. It can be
seen that CDGPS scheduler can fairly allocates service rate to different flows, according
to their assigned weight. This demonstrates the weighted fairness property of code-division
generalized processor sharing (CDGPS).
0.75
0.8
0.85
0.9
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1
1.05
0 1 0 0 0 2 0 0 0 3 0 0 0 4 0 0 0 5 0 0 0 6 0 0 0 7 0 0 0 8 0 0 0 9 0 0 0
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Flow a
Flow b
Flow c
91
Figure 4.9 Average delay as a function of Traffic intensity
Average delay
Figure 4.9 shows the average delay of a heterogeneous traffic (voice, video and data) as a
function of traffic intensity. The delays considered in this heterogeneous traffic mixed are
the queuing and transmission delays. As shown in the figure 4.9, the average delay of
priority CDGPS is better than that of non-priority CDGPS by a percentage value
of 52.8% from the point when the average delay remains constant. This constant delay
results from the CDGPS scheduler ability to distribute the unused resource more effectively
among the backlogged flows (active user). Therefore, resulting to an efficient bandwidth
utilization.
1.00E-06
1.50E-06
2.00E-06
2.50E-06
3.00E-06
3.50E-06
4.00E-06
4.50E-06
0 1 0 0 0 2 0 0 0 3 0 0 0 4 0 0 0 5 0 0 0 6 0 0 0 7 0 0 0 8 0 0 0 9 0 0 0
AV
ER
AG
E D
ELA
Y
TRAFFIC INTENSITY
Priority
Non-Priority
92
Figure 4.10 Loss rate as a function of traffic intensity
Heterogeneous traffic loss rate
Figure 4.10 compares the performance achieved by priority and non-priority CDGPS for
multimedia IP traffic (voice, video and data). The priority CDGPS provides best
performance by a percentage value of 3.5% as the scheduler tends to allocates many bits
per frame for higher priority users in priority CDGPS, regardless of the system traffic. The
traffic loss is independent of the packet size but only depends on traffic arrival rate.
0
0.05
0.1
0.15
0.2
0.25
0.3
0.35
0 1000 2000 3000 4000 5000 6000 7000 8000 9000
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Priority
Non-priority
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Figure 4.11 Backlogged flow loss rate as a function of traffic intensity
Backlogged flow loss rate
Figure 4.11 compares the backlogged loss rate for both priority and non-priority CDGPS.
To verify backlogged loss rate, if all flows or users are in backlogged mode (i.e. have data
to send in their sending queues). A metric called inter-service time, which is the interval
that a backlogged users experience, measured in time frames between two successive
transmissions. It can be observed that, as the N number of backlogged flows increases with
respect to the inter-arrival time, the loss rate also increases. The observation from figure
4.11 shows that, priority CDGPS outperform non-priority CDGPS by a percentage values
of 3.5% irrespective of the number of backlogged flows. This implies that, the backlogged
loss rate can be improved on by prioritization so that more packets can be served.
0
0.1
0.2
0.3
0.4
0.5
0.6
0.7
0.8
0.9
1
0 2000 4000 6000 8000 10000
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Backlogged 1 (Priority)
Backlogged 1 (Non-priority)
Backlogged 2 (Priority)
Backlogged 2 (Non-priority)
Backlogged 3 (Priority)
Backlogged 3 (Non-priority)
94
Figure 4.12 Bandwidth utilization as a function of traffic intensity
Utilization
Another important measure for network provider is the service utilization, shown in figure
4.12. Non-priority CDGPS outperform priority CDGPS by a percentage value of 0.92%. As expected, the bandwidth utilization decreases with prioritization of multimedia IP
traffic (i.e. voice, video and data). This is due to the extra work done by priority CDGPS,
by dynamically controlling priority level of queued calls and thus preventing one traffic
class from being adversely affecting other service class. Furthermore, the priority CDGPS
was still able to maintain a high bandwidth utilization of 98.2%.
0.00E+00
2.00E-01
4.00E-01
6.00E-01
8.00E-01
1.00E+00
1.20E+00
0 1000 2000 3000 4000 5000 6000 7000 8000 9000
UT
ILIZ
AT
ION
TRAFFIC INTENSITY
Priority
Non-priority
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CHAPTER FIVE
CONCLUSION AND RECOMMENDATIONS
5.1 Conclusion
The objective of this project is to design a dynamic bandwidth scheduling framework
which can improve the overall performance of radio resource management strategy in the
UMTS, and at the same time maintaining a good degree of dynamism and fairness in
service provision. A dynamic bandwidth scheduling using code-division generalized
processor sharing (CDGPS) scheme has been proposed for supporting multimedia IP traffic
in the uplink WCDMA cellular network. The proposed CDGPS scheduling algorithm,
presents a different set of priority values that better utilize the available bandwidth in
UMTS system. Results showed that the design satisfied the requirements and fulfills the
following scheduling objectives:
� The support of simultaneous operation of different types of services to the same
terminal according to their QoS requirements;
� Fair distribution of resources in the network, within the same traffic class
connections but also between different traffic class connections;
� Possibility to prioritize resource allocation to connections that have not been
allocated resources in the previous scheduling connections due to insufficient
resources;
� Optimum bandwidth utilization.
The analysis and simulation results are presented to demonstrate the performance of the
proposed scheme in terms of the delay, throughput, and loss rate. Simulation results show
that bounded delay can be establish by given priority for real time application when GPS
service discipline is used, while high utilization of the bandwidth can still be achieved.
96
5.2 Recommendation
While significant improvement have been made by using different set of priority value in
CDGPS scheduling approach in terms of delay and loss rate, further research can be carried
out on the following area.
� Balancing between service fairness and bandwidth utilization efficiency among the
backlogged flows for different service class.
� Effect of traffic intensity on individual service class (voice, video and data) in a
heterogeneous traffic environment.
5.3 Contribution to Knowledge
This thesis contributed to knowledge in the area of bandwidth scheduling. An efficient
dynamic scheduling scheme to support QoS of multimedia traffic in the uplink of WCDMA
cellular network has been proposed. With the capability of dynamically varying user
channel rates, WCDMA systems can provide more flexibility in bandwidth allocation. In
order to maximize the utilization of uplink capacity, thus, delay and backlogged flows loss
rate can be minimized by prioritization so that more packet can be served. The CDGPS
variation – priority and non-priority – that has not been reported were evaluated and
compared.
97
REFERENCE
[1] Tachikawa, Keiji, “A perspective on the Evolution of Mobile Communications,” IEEE
[2] H. Kaaranen, “Evolution from GSM to UMTS Multi-Access,” UMTS Networks
Archit. Mobil. Serv. Second Ed., pp. 15–27, 2005.
[3] C. Blanchard, “Security for the Third Generation (3G) Mobile System,” Inf. Secur.
Tech. Rep., vol. 5, no. 3, pp. 55–65, 2000.
[4] C. B. Jorg Eberspacher, Hans-Jorg Vogel, GSM switching, Service and Protocol. John Wiley & Son Ltd, 2001.
[5] T. Halonen, J. Romero, and J. Melero, GSM, GPRS and EDGE Performance:
Evolution Towards 3G/UMTS. 2003.
[6] V. Pereira and T. Sousa, “Evolution of Mobile Communications: from 1G to 4G,” no. July, 2004.
[7] Y.-F. Huang and T.-Y. Chiu, “Radio resource management for a mobile network with TD-CDMA,” Comput. Commun., vol. 30, no. 1, pp. 132–143, 2006.
[8] J. V. Gómez, “Third Generation Mobile Technology and its evolution towards Fourth Generation,” 2009.
[9] H. Holma and A. Toskala, WCDMA FOR UMTS: Radio Access for Third
Generation Mobile Communications, Third Edit. Finland: John Wiley & Son Ltd, 2004.
[10] F. Sallabi, K. Shuaib, and M. Shafique, “Quality of Service provisioning for UMTS network users with immediate and advance radio resource reservations,” Met.
Finish., vol. 104, no. 6, pp. 468–479, 2006.
[11] A. A. Shaikh, B. S. Chowdhry, a K. Baloch, and a H. Pathan, “Radio Resource Management Strategies in 3G UMTS Network,” no. November, pp. 48–51, 2004.
[12] J. Yao, L. Xiao, C. Nie, D. T. C. Wong, and Y. H. Chew, “Resource allocation for end-to-end QoS provisioning in a hybrid wireless WCDMA and wireline IP-based DiffServ network,” Eur. J. Oper. Res., vol. 191, no. 3, pp. 1139–1160, 2008.
[13] M. Al-Fares, M. Johnsson, P. Johansson, and A. Vahdat, “Flexible resource allocation and composition across gsm/3g networks and wlans,” Third Int. Work.
Mobil. Evol. Internet Archit., p. 73, 2008.
98
[14] W. Perndl, “Scheduling Algorithms for UMTS FDD Downlink,” 2001.
[15] D. Tipper, “UMTS Overview,” pp. 1–42, 2010.
[16] Miquel Angel Díaz-G. Jordi Pe´rez-Romero, Oriol Sallent, Ramon Agustı, Radio
Resources Management Strategies in UMTS. John Wiley & Son Ltd, 2005.
[17] 3GPP TS 23.122, “Wideband CDMA Air Interface : Protocol Stack,” pp. 131–202, 2001
[18] 3GPP TS 25.201, “Physical layer - General description.”
[19] M. Kubik, “Uplink Packet Scheduling in WCDMA Systems,” 1999.
[20] S. Malomsoky, S. Rácz, and S. Nádas, “Connection admission control in UMTS radio access networks,” Comput. Commun., vol. 26, no. 17, pp. 2011–2023, 2003.
[21] C. Network, “An overview of the 3G network.”
[22] 3GPP TS 25.322, “Radio Link Control (RLC): protocol specification.”
[23] 3GPP TS 25.323, “Packet Data Convergence Protocol (PDCP) specification.”
[24] 3GPP TS 25.331, “Radio Resource Control (RRC): protocol specification.”
[25] Ammar Ahmad Kirmani, Muhammad Abdur Rahman Haider, Abu Bakar Bhatti, “Radio Resource Management In 3G UMTS Networks,” Blekinge Institute of Technology, 2007.
[26] R. Ferrus, X. Reves, a. Umbert, and F. Casadevall, “Real-time emulation of RRM strategies for UMTS bearer services,” Proc. IEEE 56th Veh. Technol. Conf., vol. 2, 2002.
[27] Kai Xu, “Radio Resource Management for Satallite UMTS,” 2009.
[28] M. I. Silventoinen and H. Posti, “Radio resource management in a novel indoor GSM base station system,” Proc. 8th Int. Symp. Pers. Indoor Mob. Radio Commun.
- PIMRC ’97, vol. 3, 1997.
[29] Sallent Oriol, “QoS and Radio Resource Management in 3G and Beyond Systems,” IEEE Commun. Mag., 2006.
99
[30] Z. Deng-yin, J. Juan, A. Adi, and L. I. Hai-bo, “QoS-guaranteed packet scheduling in wireless networks,” J. China Univ. Posts Telecommun., vol. 16, no. 2, pp. 63–67, 2009.
[31] A. Kumar and a K. Garg, “Queuing Algorithem Based Quality of Service (Qos) For Scheduling Environment Model in Wimax Network with Opnet Modeler,” vol. 11, no. 8, 2011.
[32] S. Zoric, “Comparative Analysis of Scheduling Algorithms for UMTS Traffic in Case of DiffServ Network Congestion,” no. c, pp. 39–44, 2012.
[33] J. Siltanen, “Quality of Service and Dynamic Scheduling for Traffic Engineering in Next Generation Network,” University of Jyväskylä, 2007.
[34] J. N. (neé Antila), “Adaptive Scheduling in Cellular Access, Wireless Mesh and IP Networks,” Aalto University, 2011.
[35] A. Matinlauri, “Fairness and Transmission Opportunity Limit in IEEE 802 . 11e Enhanced Distributed Channel Access,” 2008.
[36] Kai-Yeung Siu, Anthony C. Kam, Thit Minn, “Supporting Rate Guarantee and Fair Access for Bursty Data Traffic in W-CDMA,” IEEE J. Sel. Area Commun., vol. 19, 2001.
[37] Angelos N. Rouska, Dimitrios N. Skoutas, “A dynamic traffic scheduling algorithm for the downlink shared channel in 3G WCDMA.,” IEEE Int. Conf.
Commun., vol. 1, pp. 2974 – 2979, 2004.
[38] S. M. Abd El-atty, D. N. Skoutas, A. N. Rouskas, and G. T. Karetsos, “A Cross Layer Scheduling Framework for Supporting Bursty Data Applications in WCDMA Networks,” Wirel. Pers. Commun., vol. 46, no. 1, pp. 33–45, 2008.
[39] K.-Y. Siu, Anthony C. Kam, Thit Minn, “Supporting rate guarantee and fair access for bursty data traffic in W-CDMA.,” IEEE J. Sel. Area Commun., vol. 19, pp. 2121–2130, 2001.
[40] F. Casadeval, Oriol Sallent, Jordi Pérez-Romero,Ramon Agustí., “Provisioning multimedia wireless networks for better QoS: RRM strategies for 3G W-CDMA,” IEEE Commun. Mag., pp. 100–106, 2003.
[41] C. Chao, Y. Tseng, S. Member, and L. Wang, “Dynamic Bandwidth Allocation for Multimedia Traffic with Rate Guarantee and Fair Access in WCDMA Systems,” Computer (Long. Beach. Calif)., vol. 4, no. 5, pp. 420–429, 2005.
100
[42] O. Sallent, J. Perez-Romero, F. J. Casadevall, and R. Agusti-Gomes, “An Emulator Framework For A New Radio Resource Management For Qos Guaranteed Services In W-cdma Systems,” Jsac, vol. 19, no. 10, pp. 1893–1904, 2001.
[43] Jordi perez-Romero, Luis Almajano, “Packet scheduling algorithms for interactive and streaming services under QoS guarantee in a CDMA system,” pp. 1657–1667, 2002.
[44] M. Chen and R. Hwang, “Fair and efficient scheduling for UMTS forward link,” Int. J. Commun. Syst., vol. 18, no. 4, pp. 409–426, 2005.
[45] a. K. Parekh and R. G. Gallager, “A generalized processor sharing approach to flow control in\nintegrated services networks: the multiple node case,” IEEE/ACM
Trans. Netw., vol. 2, no. 3, pp. 344–357, 1994.
[46] A. Mendez, D. Covarrubias, and C. Brizuela, “Fair Scheduling with dynamic resource allocation in CDMA/GPS System for IP-Multimedia wireless networks” J circuit sys. comput. (JCSC), vol. 13, no. 2, pp. 253–269, 2004.
[47] A. Mendez, M. Panduro, D. Covarrubias, R. Dominguez, and G. Romero, “Quality of service support for multimedia traffic in mobile networks using a CDMA novel scheduling scheme,” Comput. Electr. Eng., vol. 32, no. 1–3, pp. 178–192, 2006.
[48] D. N. Skoutas and A. N. Rouskas, “A Scheduling Algorithm With Dynamic Priority Assignment For Wcdma Systems,” Tmc, vol. 8, no. 1, pp. 126–138, 2009.
[49] J. Wigard, N. a H. Madsen, and P. A. Gutiérrez, “Packet Scheduling with QoS Differentiation,” Wirel. Pers. Commun., pp. 147–160, 2002.
[50] C. S. Wan, W. K. Shih, and R. C. Chang, “Real-time packet scheduling in next generation radio access system,” Comput. Commun., vol. 26, no. 17, pp. 1931–1943, 2003.
[51] S. Chandramathi, S. P. P. Raghuram, V. S. Srinivas, and H. S. Singh, “Dynamic bandwidth allocation for 3G wireless systems—A fuzzy approach,” Appl. Soft
Comput., vol. 8, no. 1, pp. 274–284, 2008.
[52] L. X. Xuemin Shen, Jon. W. Mark, “Dynamic bandwidth allocation with fair scheduling for WCDMA systems,” IEEE Commun. Mag., 2002.
[53] Ö. Gürbüz and C. Systems, “Dynamic Resource Scheduling Schemes for W-CDMA Systems,” no. October, pp. 80–84, 2000.
101
[54] L. Xu, Xuemin Shen, and J. W. Mark, “Dynamic Fair Scheduling With QoS Constraints in Multimedia Wideband CDMA Cellular Networks,” vol. 3, no. 1, pp. 60–73, 2004.
[55] L. Xu, Xuemin Shen, and J. W. Mark, “Fair Resource Allocation with Guaranteed Statistical QoS for Multimedia Traffic in Wideband CDMA Cellular Network,” vol. 4, no. 2, pp. 166–177, 2005.
[56] Salman A. AlQahtani and A. S. Mahmoud, “Dynamic radio resource allocation for 3G and beyond mobile wireless networks,” Comput. Commun., vol. 30, no. 1, pp. 41–51, 2006.
[57] Cosmas I. Ani, Simiat O. Isimijola “EFFECT OF DATA TRAFFIC PATTERNS ON QoS PARAMETERS,” Niger. J. Technol., vol. 28, pp. 23–28, 2009.
[58] K. D. Hackbarth, A. Brand, and R. Lehnert, “Analytical Model for Voice over IP traffic characterization 2 Voice over IP.”
[59] M. Schwartz, “Broadband Integrated Network,” Prentice Hall, 1996.
[60] S. Tanwir and H. Perros, “A Survey of VBR Video Traf fi c Models,” IEEE
Commun. Surv. Tutorials, vol. PP, no. 99, pp. 1–25, 2013.
