A Sheffield Hallam University Thesis
A Sheffield Hallam University Thesis
DOGMAN, Aboagela A.
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REFERENCE
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Multimedia Computer Networks Quality of Service
Techniques Evaluation and Development
February 2014
ABSTRACT
Quality of Service Evaluation and Improvement of Multimedia Computer
Networks
The growth in the transmission of time-sensitive applications over computer networks means
that Quality of Service (QoS) needs to be managed in an efficient manner. Network QoS
management in this thesis refers to evaluation and improvement of QoS provided by integrated
wired and wireless computer networks. Evaluation of QoS aims to analyse and quantify network
performance with respect of meeting multimedia applications' transmission requirements. QoS
improvement involves the ability to take actions to change network performance toward
improved operation. Therefore, the main aims of this thesis are: (i) to develop techniques for
evaluation QoS in multimedia computer networks, (ii) to develop techniques that uses the
information from (i) to manage and improve network performance.
Multimedia traffic generates a large amount of data. Collecting this information poses a
challenge as it needs to be sufficiently fast and accurate. A contribution of this thesis is that
adaptive statistical sampling techniques to sample multimedia traffic were developed and their
effectiveness was evaluated. Three different adjustment mechanisms were incorporated into
statistical sampling techniques to adjust the traffic sampling rate: simple linear adjustment,
quarter adjustment, and Fuzzy Inference System (FIS). The findings indicated that the
developed methods outperformed the conventional non-adaptive sampling methods of
systematic, stratified and random.
The data collected included important QoS parameters, i.e. delay, jitter, throughput, and packet
loss that indicated network performance in delivering real-time applications. An issue is that
QoS needs evaluation in an informative manner. Therefore, the second contribution of this
thesis is that statistical and Artificial Intelligent (AI) techniques were developed to evaluate QoS
for multimedia applications. The application’s QoS parameters were initially analysed either by
Fuzzy C-Means (FCM) clustering algorithm or by Kohonen neural network. The analysed QoS
parameters were then used as inputs to a regression model or Multi-Layer Perceptron (MLP)
neural network in order to quantify the overall QoS. The proposed QoS evaluation system
differentiated the network’s QoS into a number of levels (Poor to Good QoS) and based on this
information, the overall network’s QoS was successfully quantified. In order to facilitate QoS
assessment, a portable hand-held device for assessing the QoS in multimedia networks was
designed, regression model was implemented on the microcontroller board and its performance
was successfully demonstrated.
Multimedia applications transmitted over computer networks require a large bandwidth that is a
critical issue especially in wireless networks. The challenge is to enable end-to-end QoS by
providing different treatments for different classes of traffic and efficient use of network
resources. In this thesis, a new QoS enhancement scheme for wireless-wired networks is
developed. This scheme consisted of an adaptive traffic allocation algorithm that is incorporated
into the network's wireless side to improve the performance of IEEE 802.1 le Enhanced
Distributed Channel Access (EDCA) protocol, and a Weighted Round Robin (WRR) queuing
scheduling mechanism that was incorporated into the wired side. The proposed scheme
improved the QoS for Multimedia applications. The average QoS for voice, and video
applications were increased from their original values by 72.5%, and 70.3% respectively.
DEDICATION
o To the soul o f my mother, I still owe to her. Without her encouragement when
she was a live, I could not be able to reach this stage.
o My father, fo r his invaluable support throughout the years, who devoted his life
to the achievement o f this dream,
o My lovely wife whose unconditional love makes everything possible,
o My sons Yousf, Ahmed, and Abdul-Aziz who make every day new and precious.
o My brothers, sisters, and friends who shared with me my dream.
ACKNOWLEDGMENTS
All thanks to Almighty ALLAH for giving me the guidance and the strength to achieve
my aim of obtaining PhD degree.
First, I would like to express my deep thanks to my director of studies, Dr. Reza
Saatchi, for his guidance, support, encouragement, valuable suggestions and comments
on various aspects throughout my PhD study. The initial help I have received from Dr.
Reza in setting up my research focus and his subsequent professional support
throughout my research process were very valuable. I really appreciate your support!
Special thanks to my supervisor, Dr. Samir Al-Khayatt for being very helpful and
supportive. His valuable guidance, advice, and expertise throughout my research were
really helpful. Dr. Samir Al-Khayatt taught me how a supervisor can be a sincere friend.
I will never forget your help and support!
I also wish to thank researchers, who have been, or still are, PhD candidates within the
C3RI as well as many of the departmental staff, our friendly discussions helped me gain
confidence in my research.
LIST OF PUBLICATIONS
v
Communications and Mobile Computing Conference (IWCMC11), 4th - 8th July,
Bahcesehir University, Istanbul, Turkey, UK, IEEE Xplore, pp. 1731 - 1736.
Dogman, A., Saatchi, R., and Al-Khayatt, S. (2010), An Adaptive Statistical
Sampling Technique for Computer Network Traffic, In Proceedings of the 7th
IEEE, DET International Symposium on Communication Systems Networks and
Digital Signal Processing (CSNDSP10), 21st -23rd July, University of
Northumbria, Newcastle, UK, IEEE Xplore, pp. 479 - 483.
Dogman, A., Saatchi, R., and Al-Khayatt, S. (2010), Evaluation of Adaptive
Statistical Sampling versus Random Sampling for Video Traffic, In Proceedings of
the 11th International Arab Conference on Information Technology (ACIT10),
14th-16th December, University of Benghazi, Benghazi, Libya.
TABLE OF CONTENTS
ABSTRACT............................................................................................................................................... H
DEDICATION.......................................................................................................................................... HI
ACKNOWLEDGMENTS....................................................................................................................... IV
LIST OF PUBLICATIONS.......................................................................................................................V
TABLE OF CONTENTS........................................................................................................................VII
LIST OF FIGURES................................................................................................................................. XI
LIST OF TABLES........................................ XV
GLOSSARY TERMS...........................................................................................................................XVII
LIST OF VARIABLES...........................................................................................................................XX
CHAPTER 1 INTRODUCTION...............................................................................................................1
1. 1 R e se a r c h M o t iv a t io n s ....................................................................................................................................1
1.3 R e se a r c h C o n t r ib u t io n s .............................................................................................................................. 4
2.1 I n t r o d u c t io n .............................................................................
3.1 I n t r o d u c t io n .......................................................................................................................................................41
N e t w o r k .......................................................................................................................................................................... 56
3.7 S u m m a r y ............................................................................................................................................................... 58
4.1 I n t r o d u c t io n ....................................................................................................................................................... 59
4.2 N e t w o r k E v a l u a t io n A p p r o a c h e s ............................................................................................................59
5.1 I n t r o d u c t io n .......................................................................................... 71
5.2 R e l a t e d W o r k .................................................................................................................................................. 72
5.4.1 Throughput........................................................................................................................... 83
5.4.2 Delay....................................................................................................................................90
5.4.3 Jitter.....................................................................................................................................95
5.4.4 Packet Loss Ratio............................................................................................................... 100
5.5 S u m m a r y ............................................................................................................................................................105
6.2 R e l a t e d W o r k s ...............................................................................................................................................106
7.1 I n t r o d u c t io n ...................................................................................................................................................136
7.2 R e l a t e d W o r k ..............................................................................!..................................................................136
C H A P T E R 8 M IC R O C O N T R O L L E R B O A R D IM P L E M E N T A T IO N O F Q U A L IT Y O F
S E R V IC E A S S E S S M E N T S Y S T E M ....................................................................................................................... 157
8.2 R e l a t e d W o r k .................................................................................................................................................158
C H A P T E R 9 C O N C L U S IO N S , A N D F U T U R E W O R K ................................................................................. 175
9.1 C o n c l u s io n s .....................................................................................................................................................175
R E F E R E N C E S .................................................................................................................................................................. 180
x
LIST OF FIGURES
F ig u r e 1-1. T h e sc h e m a t ic o v e r v ie w o f t h e t h e s is ............................................................................................. 10
F ig u re 2-9. T h e o pe r a t io n o f W R R .............................................................................................................................27
F ig u r e 2-11. B l o c k d ia g r a m o f fu z z y in fe r e n c e sy st e m . ............................................................................... 30
F ig u re 2-13. T h e pr o c e ss o f f u z z y in fe r e n c e sy st e m ........................................................................................ 33
F ig u r e 2-14. M u l t i -L a y e r P e r c e pt r o n : ( a ) T h e a r c h it e c t u r e , ( b ) M LP o p e r a t io n ............................. 38
F ig u r e 2-15. T h e st r u c t u r e o f k o h o n e n n e u r a l n e t w o r k ..............................................................................39
OF ADAPTIVE SAMPLING............................................................................................................................................43
F ig u re 4-4. G O P s e q u e n c e in M P E G -4 .......................................................................................................................67
A d a pt iv e s a m pl in g b a se d o n FIS, (c ) A d a pt iv e s a m pl in g l in e a r a d ju st m e n t , ( d ) A d a pt iv e
SAMPLING QUARTER ADJUSTMENT, (E) SYSTEMATIC, (F) STRATIFIED, (G) RANDOM SAMPLING...........84
WITH BIAS OBTAINED USING: (A) ADAPTIVE SAMPLING BASED ON FIS, (B) ADAPTIVE SAMPLING
F ig u r e 5-7. R S E o f sa m ple d t h r o u g h pu t u s in g c o n v e n t io n a l s a m pl in g v e r s u s : ( a ) A d a p t iv e
SAMPLING BASED ON FIS, (B) ADAPTIVE SAMPLING BASED ON LINEAR ADJUSTMENT, (C) ADAPTIVE
SAMPLING BASED ON FIS, (C) ADAPTIVE SAMPLING LINEAR ADJUSTMENT, (D) ADAPTIVE SAMPLING
BIAS OBTAINED USING: (A) ADAPTIVE SAMPLING BASED ON FIS, (B) ADAPTIVE SAMPLING BASED ON
F ig u r e 5-10. R S E o f sa m pl e d d e l a y u s in g c o n v e n t io n a l sa m pl in g v e r s u s : ( a ) A d a p t iv e
SAMPLING BASED ON FIS, (B) ADAPTIVE SAMPLING BASED ON LINEAR ADJUSTMENT, (C) ADAPTIVE
SAMPLING BASED ON F IS , (C) ADAPTIVE SAMPLING LINEAR ADJUSTMENT, (D) ADAPTIVE SAMPLING
BIAS OBTAINED USING: (A) ADAPTIVE SAMPLING BASED ON FIS, (B) ADAPTIVE SAMPLING BASED ON
BASED ON FIS, (B) ADAPTIVE SAMPLING BASED ON LINEAR ADJUSTMENT, (C) ADAPTIVE SAMPLING
SAMPLING BASED ON FIS, (C) ADAPTIVE SAMPLING LINEAR ADJUSTMENT, (D) ADAPTIVE SAMPLING
QUARTER ADJUSTMENT, (E) SYSTEMATIC, (F) STRATIFIED, (G) RANDOM SAMPLING............................. 101
SAMPLING WITH BIAS OBTAINED FROM ADAPTIVE SAMPLING BASED ON: (A) FIS, (B) LINEAR
A d a pt iv e sa m pl in g b a se d o n FIS, ( b ) A d a pt iv e sa m pl in g b a se d o n lin ea r a d ju st m e n t , ( c )
A d a pt iv e sa m pl in g b a se d o n q u a r t e r a d ju st m e n t ......................................................................... 104
( a ) V o IP , ( b ) V id e o a p p l ic a t io n ............................................................................................................................ 120
F ig u r e 6 -7 . T h e p r o g r e s s o f o b j e c t iv e f u n c t io n s d u r in g F C M a n a l y s is o f : ( a ) V o IP , ( b ) V id e o
APPLICATION......................................................................................................................................................................121
F ig u r e 6 -8 . C l u s t e r in g Q o S p a r a m e t e r s o f V o I P .................................................................................................. 122
F ig u r e 6 -9 . C l u s t e r in g Q o S p a r a m e t e r s o f v id e o ...............................................................................................123
F ig u r e 6 -1 0 . T h e Q o S o f V o IP u s in g r e g r e s s io n m o d e l ...................................................................................... 124
F ig u r e 6 -1 1 . T h e Q o S o f v id e o u s in g r e g r e s s io n m o d e l .............................................. 125
F ig u r e 6 -1 2 . C l a s s if y in g Q o S p a r a m e t e r s o f V o IP u s in g k o h o n e n n e t w o r k : ( a ) D e l a y , ( b ) J it t e r ,
(c ) P a c k e t l o s s r a t io , ( d ) Q o S p a r a m e t e r s l a b e l s , ( e ) S O M s a m p l e h i t s .....................................128
F ig u r e 6 -1 3 . C l a s s if y in g Q o S p a r a m e t e r s o f v id e o a p p l ic a t io n u s in g k o h o n e n n e t w o r k : ( a )
D e l a y , ( b ) J it t e r , ( c ) P a c k e t l o s s r a t io , ( d ) Q o S p a r a m e t e r s l a b e l s , ( e ) S O M s a m p l e h i t s .
................................................................................... 129
F ig u r e 6-1 4 . T h e p r o g r e s s o f t r a in in g M L P t o a s s e s s t h e Q o S f o r : ( a ) V o IP , ( b ) V id e o ..................130
F ig u r e 6-1 5 . C o m p a r is o n b e t w e e n n o r m a l is e d a c t u a l Q o S a n d c a l c u l a t e d o u t p u t s f r o m M L P
in c a s e o f : ( a ) V o IP Q o S a s s e s s m e n t , a n d ( b ) V id e o Q o S a s s e s s m e n t ............................................. 131
F ig u r e 6 -1 6 . T h e Q o S a s s e s s m e n t o f : ( a ) V o IP , a n d ( b ) V id e o a p p l ic a t io n u s in g M u l t i -L a y e r
P e r c e p t r o n .......................................................................................................... 132
F ig u r e 7 -1 . A d a p t iv e t r a f f ic a l l o c a t io n a l g o r it h m f l o w c h a r t ............................................................... 139
F ig u r e 7 -2 . B a r c h a r t r e p r e s e n t a t io n o f a v e r a g e d e l a y w it h a n d w it h o u t Q o S e n h a n c e m e n t
s c h e m e f o r : (a ) V o IP , ( b ) V id e o , ( c ) B e s t e f f o r t t r a f f ic , ( d ) B a c k g r o u n d t r a f f ic ................ 145
F ig u r e 7 -3 . B a r c h a r t r e p r e s e n t a t io n o f a v e r a g e j it t e r w it h a n d w it h o u t Q o S e n h a n c e m e n t
s c h e m e f o r : (a ) V o IP , ( b ) V id e o , ( c ) B e s t e f f o r t t r a f f ic , ( d ) B a c k g r o u n d t r a f f ic ................ 148
F ig u r e 7 -4 . B a r c h a r t r e p r e s e n t a t io n o f a v e r a g e p a c k e t l o s s w it h a n d w it h o u t Q o S
ENHANCEMENT SCHEME FOR: (A) V o IP , (B) VIDEO, (C) BEST EFFORT TRAFFIC, (D) BACKGROUND
TRAFFIC.......................................................................................... 151
F ig u r e 7 -5 . B a r c h a r t r e p r e s e n t a t io n o f o v e r a l l a s s e s s e d Q o S w it h a n d w it h o u t Q o S
ENHANCEMENT SCHEME FOR: (A) V o IP , (B) VIDEO, (C) BEST EFFORT TRAFFIC, (D) BACKGROUND
TRAFFIC.............................................................................................................................................................................. 154
F ig u r e 7 -6 . V is u a l c o m p a r is o n o f r e c o n s t r u c t e d F o r e m a n v id e o u s in g : ( a ) Q o S e n h a n c e m e n t
F ig u r e 8 -4. Q o S A s s e s s m e n t C o d e ...................................................................................................................................164
F ig u r e 8 -6 . T h e U L IN K -M E A d a p t e r ..............................................................................................................................167
xiii
F ig u r e 8-7. T h e c o n n e c t io n b e t w e e n t h e PC a n d t h e MCB2300 u sin g ULINK-ME JTAG A d a p t e r .
......................................................................................................................................................... 167
F ig u r e 8-8. ^V isio n IDE so ft w a r e t o o l ..................................................................................................................168
xiv
LIST OF TABLES
T a b l e 4-3. W R R P a r a m e t e r s .........................................................................................................................................66
T a b l e 4-5. Q oS r eq u ir e m e n t s fo r v o ic e , v id e o , a n d d a t a as r e c o m m e n d e d b y IT U group (Z h a i e t
a l , 200 5 ).......................................................................................................................................................................68
T a b l e 5-1. M ea n a n d s t a n d a r d d e v ia t io n o f o v e r a l l st a t ist ic in pu t fu z z y m e m b e r sh ip
f u n c t io n s ....................................................................................................................................................................78
T a b l e 5-2. M e a n a n d st a n d a r d d ev ia tio n o f c u r r e n t sa m pl e in t e r v a l in pu t fu z z y m e m b e r s h ip
fu n c t io n s ....................................................................................................................................................................78
T a b l e 5-3. M ea n a n d s t a n d a r d d e v ia t io n o f sa m pl e in t e r v a l d iffe r e n c e o u t pu t fu z z y
MEMBERSHIP FUNCTIONS.......................................................................................................................................... 78
SAMPLING BASED ON FIS, (B) ADAPTIVE SAMPLING BASED ON LINEAR ADJUSTMENT, (C) ADAPTIVE
SAMPLING BASED ON QUARTER ADJUSTMENT, (D) SYSTEMATIC, (E) STRATIFIED, (F) RANDOM
SAMPLING......................................................................................................................................................................87
SAMPLING BASED ON FIS, (B) ADAPTIVE SAMPLING BASED ON LINEAR ADJUSTMENT, (C) ADAPTIVE
SAMPLING BASED ON QUARTER ADJUSTMENT, (D) SYSTEMATIC, (E) STRATIFIED, (F) RANDOM
SAMPLING........................................................................... 92
SAMPLING BASED ON FIS, (B) ADAPTIVE SAMPLING BASED ON LINEAR ADJUSTMENT, (C) ADAPTIVE
SAMPLING BASED ON QUARTER ADJUSTMENT, (D) SYSTEMATIC, (E) STRATIFIED, (F) RANDOM
SAMPLING......................................................................................................................................................................97
T a b l e 5-9. P a c k e t lo ss r a t io m e a su r e m e n t r e su l t s u s in g d iff e r e n t s a m pl in g m e t h o d s : ( a )
A d a pt iv e s a m pl in g b a se d o n FIS, ( b ) A d a p t iv e s a m pl in g b a s e d o n l in e a r a d ju st m e n t , ( c )
A d a pt iv e s a m pl in g b a sed o n q u a rte r a d ju st m e n t , ( d ) S y s t e m a t ic , ( e ) S t r a t if ie d , ( f )
T a b le 6-4. Q oS pa r a m e t e r s o f v id e o a n d e x pe c t e d Q o S u s in g : F u z z y I n fe r e n c e S y st e m (FIS),
P e r c e pt r o n (MLP)................................................................................................................................................ 173
xvi
GLOSSARY TERMS
AC Access Category
ACR Absolute Category Rating
ADC Analog to Digital Converter
AI Artificial Intelligent
AIFS Arbitration Inter Frame Space
AIFSN Arbitration Inter Frame Space Number
ANN Artificial Neural Network
AODV Ad hoc On-Demand Distance Vector
AP Access Point
CAC Call Admission Control
CAN Controller Area Network
CAP Controlled Access Phase
CBR Constant Bit Rate
CFP Contention Free Period
CF-Poll Contention Free Poll
cwmax Maximum Contention Window
CW m in Minimum Contention Window
DAC Digital to Analog Converter
DCF Distributed Coordination Function
DiffServ Differentiated Service
DIFS Distributed Inter Frame Space
DoS Denial of Service
DSDV Destination Sequenced Distance Vector
EDCA Enhanced Distributed Channel Access
FCM Fuzzy C- means
FIFO First In First Out
FIS Fuzzy Inference System
FLC Fuzzy Logic Controller
FQ Fair Queuing
FTP File Transfer Protocol
GloMoSIM Global Mobile Information System Simulator
GOP Group Of Picture
GUI Graphical User Interface
HC Hybrid Coordinator
HCCA HCF Controlled Channel Access
HCF Hybrid Coordination Function
HID Human Interface Devise
IC Integrated Circuit
IDE Integrated Development Environment
IDS Intrusion Detection System
IEEE Internet of Electrical and Electronic Engineering
IntServ Integrated Service
IP Internet Protocol
IPTV Internet Protocol Television
ISSI Inter-Sampling Section Interval
xvii
ITU International Telecommunication Union
KSOM Kernel SOM
LCD liquid crystal display
LED light-emitting diode
LSP Label Switched Path
MAC Medium Access Control
MANET Mobile Ad hoc Network
MCB Microcontroller Board
MDK Microcontroller Development Kit
MLP Multi- Layer Perceptron
MOS Mean Opinion Score
MSC Massage Storage Class
MSDU MAC Service Data Unit
MSE Mean Square Error
NAM Network Animator
NS-2 Network Simulator -2
OMNET++ Optical Micro Networks Plus Plus
OPNET Optimized Network Evaluation Tool
OSI Open System Interconnection
OTcl Object Tool Command Language
OWD One Way Delay
PAR Packet Arrival Rate
PCF Point Coordination Function
PCM Plus Code Modulation
PESQ Perceptual Evaluation of Speech Quality
PIFS PCF Inter Frame Space
PLCP Physical Layer Convergence Procedure
PLR Packet Loss Ratio
PQ Priority Queuing
PSNR Peak Signal to Noise Ratio
PSQM Perceptual Speech Quality Measure
QoS Quality of Service
RAM Random Access Memory
RED Random Early Detection
RNN Random Neural Network
ROM Read Only Memory
RSE Relative Standard Error
RST Request to Send
SD Secure Digital
SI Service Interval
SID Sample Interval Difference
SIFS Short Inter Frame Space
SOM Self Organising Map
TCL Tool Command Language
TCP Transmission Control Protocol
TD Transmission opportunity Duration
TORA Temporally Ordered Routing Algorithm
TSW Time Sliding Window
TXOP Transmission Opportunity
UDP User Datagram Protocol
xviii
USB Universal Serial Bus
VBR Variable Bit Rate
VoIP Voice over Internet Protocol
VVoIP Voice and Video over Internet Protocol
WFQ Weighted Fair Queuing
WLAN Wireless Local Area Network
WRR Weighted Round Robin
xix
LIST OF VARIABLES
XX
Mean value of post- sampling section for throughput of
mean2
traffic being sampled
Median value of pre- sampling section for throughput of
medianl
traffic being sampled
Median value of post- sampling section for throughput of
median2 5.1
traffic being sampled
Standard deviation value of pre- sampling section for
std l
throughput of traffic being sampled
Standard deviation value of post- sampling section for
std2
throughput of traffic being sampled
Updated ISSI Updated length of Inter- Sampling Section Interval
5.2
Ai l Update the length of ISSI by increasing it linearly
p2 Update the length of ISSI by decreasing it linearly 5.3
V Update the length of ISSI in quarter adjustment mechanism 5.4 - 5.5
VMM Gaussian membership of value (x) of ith fuzzy set A1
Ct The mean of the ith fuzzy set A1 5.6
Standard deviation of the ith fuzzy set A1
Y Deffuzzification output using centroid method
Vi Centroid of fuzzy region i
5.7
Hi Output membership value
m Number of fuzzy sets after implication process
SID Sample Interval Difference 5.8
Shows the difference between mean of sampled version of
Bias
QoS parameter and the mean of its original population
Mt Mean of QoS parameters for the sampled version
M Mean of QoS parameters for original population 5.9
N Number of simulation runs
RES Relative Standard Error
SE Standard Error of sampled version 5.10
n Sample size
xxi
Chapter 1 Introduction
The purpose of this chapter is to: clarify the rationales behind this research which are
presented in section 1.1, outline the aim and objectives in section 1.2, summarise the
contribution of this research in section 1.3, and present the outline and organisation of
this thesis in section 1.4.
(i) Computer networks carry different traffic types containing video, audio and data.
Analysis and interpretation of all traffic packets can be time consuming. Also,
collecting all packets poses a challenge as the process needs to be sufficiently fast,
should not load the network and interfere with the operation of the protocols
responsible for managing the network. Therefore, ways need to be found to
Chapter 1 Introduction
accurately sample packets and present them to the network management entities
responsible for their interpretation and decision making. Most of current sampling
techniques use a predetermined and fixed sampling rate irrespective of the extent
of traffic fluctuation with time. The sampling therefore is not the optimal as the
multimedia traffic is time varying. In a fixed rate sampling method, two situations
could occur: (a) If the traffic fluctuation (i.e. frequency of variation) is high, there
is a risk of losing important information (caused by under sampling), (b) If the
traffic fluctuation is low, resources would be under utilised (by over sampling)
(Giertl et al, 2006). In order to reduce the biasness presented in conventional fixed
rate sampling and to enhance the process o f gathering Quality of Service (QoS)
information, traffic should be sampled adaptively. In other words, a small sample
interval is required during the period of high activity, whereas a larger sample
interval is required during the period of low traffic fluctuation.
2
Chapter 1 Introduction
intensive because their operations are at the pixel level and they cannot take into
the account all the affected network parameters (Mohammed et al, 2001). These
limitations highlight the need for further development in evaluation of QoS. In
order to evaluate QoS effectively, the proposed methods need to combine analysis
and measurement techniques. The methods require evaluation of QoS in a manner
similar to human subjects and quantify the QoS without the necessity for complex
mathematical models, taking into the account the QoS requirements of every type
of multimedia applications. In addition, the methods should not add a significant
extra load to the network as is the case with active approaches, nor should it
depend on collection of all traffic packets.
(iv) The existing QoS monitoring techniques are limited in scope. They obtain overall
network QoS indirectly and are not stand-alone. For example, the QoS monitoring
tool proposed by Graham et al (1998) was used to assess packet latency and loss
as an indication of network performance. The Surveyor tool proposed by Zseby
and Scheiner (2004) to assess end-to-end delay and packet loss required a further
analysis using a server to measure the network performance. The disadvantage of
these tools is that network managers have to do a variety of operations to assess
the overall network QoS. From these limitations, it can be obvious that the
process of monitoring QoS can be complicated, expensive, and time consuming.
3
Chapter 1 Introduction
The overall aim of this study is to develop techniques to evaluate QoS in multimedia
networks and to use this information as part of a network management process to
improve its performance. The study uses IEEE 802.1 le standard for the wireless side of
the network. The wireless is connected to a wired network. The objectives of the study
are to:
4
Chapter 1 Introduction
the contributions of this research in line with its objectives. The research contributions
of this study are included in relevant chapters. These contributions are outlined below.
5
Chapter 1 Introduction
(iii) Develop techniques to allow the QoS information to be used as part o f network
management to improve its performance. A scheme that improves QoS in both
wired and wireless domains of the network is proposed in (Dogman, et al.,
2012c). A description of the proposed QoS enhancement scheme is provided in
Chapter 7. The scheme uses a combination of MAC layer QoS control in the
wireless domains and network layer QoS control in the wired domains. A novel
aspect of the scheme is that an adaptive access category (AC) traffic allocation is
devised and incorporated into wireless access point (AP) in order to improve the
QoS of IEEE 802.l i e Enhanced Distributed Channel Access (EDCA) protocol,
and Weighted Round Robin (WRR) queuing scheduling mechanism is
implemented into congestion point (i.e. router) in wired networks to support fair
distribution of bandwidth among different traffic classes. The adaptive traffic
allocation algorithm determines the packet arrival rate (PAR) of the up- and
down-link traffic for each access category (AC). It then dynamically allocates
traffic of a lower priority AC to the next higher AC, when the higher AC is not
receiving traffic at the time. Whereas, WRR shares the network resources, based
on the traffic’s QoS requirements. The results explained in Chapter 7 show the
improvement in QoS provided by wired and wireless sides when the QoS
enhancement scheme was implemented.
(iv) Develop an electronic portable device that facilitates accurate QoS assessment
o f multimedia networks. A hardware QoS monitoring system is designed
(Dogman, et al., 2013). The system is used regression modelling, implemented
on the MCB2300 KEIL ARM microcontroller board. More details about QoS
monitoring system are provided in Chapter 8. The hardware QoS monitoring
system analysed the QoS requirements (i.e. delay, jitter and packet loss ratio) of
multimedia applications to determine their overall QoS. The evaluation o f QoS
monitoring system was carried out by comparing the results obtained with other
QoS assessment methods to indicate the effectiveness of the developed system
in monitoring multimedia QoS accurately.
6
Chapter 1 Introduction
Chapter 2 covers the theoretical background essential for this research. This includes the
definitions of QoS and its parameters, QoS requirements of multimedia applications, the
service levels of QoS, and QoS components. Chapter 2 also provides a detailed
description of IEEE 802.1 le as an emerging WLANs standard to provide QoS, and
packet scheduling mechanisms as the most commonly mechanisms implemented in
wired network to support QoS. In addition to that, the theories o f statistical and A l
techniques used in this study are provided. Regression model as one of the most widely
employed statistical analysis methods is described. The fundamental principles of three
important paradigms in A l system: fuzzy logic, fuzzy clustering, and neural network
with their operational steps are also discussed.
In Chapter 3, the previous studies relevant to the aforementioned issues o f managing the
QoS in multimedia networks in section 1.1 are critically analysed and discussed. The
aim is to identify the potential limitations associated with these issues which in turn are
further developed. Chapter 3 is organised to review the sampling techniques used to
gather information from network traffic, and discuss the state-of-the-art o f QoS analysis
and assessment techniques used to evaluate network QoS. The relevant studies
considering QoS support in the context of wireless-wired networks are also analysed,
and the exiting monitoring tools used to assess the network performance are reviewed.
Moreover, the applications of statistical and A l techniques used in this study into the
field of network QoS management are reviewed in that chapter.
7
Chapter 1 Introduction
The main results of this research are provided in Chapters 5-8. In Chapter 5, statistical
adaptive sampling techniques to adjust sampling rate based on traffic’s statistics are
introduced. A detailed description of the proposed adaptive statistical sampling
techniques which are based on three adjustment mechanisms: quarter adjustment
mechanisms, simple linear adjustment mechanisms, and fuzzy inference system is
provided. An implementation of conventional sampling techniques (i.e. systematic
sampling, stratified sampling, and random sampling) is explained. The experimental
results in Chapter 5 show the effectiveness of statistical sampling techniques by
carrying out a comparison of devised statistical sampling technique versus the
conventional sampling techniques using a simulated computer network.
Chapter 6 introduces the use of statistical and A l techniques in the area o f QoS
management. Two innovative QoS evaluation approaches are explained. The first
approach combines Fuzzy C-Means (FCM) and regression model to analyse and assess
QoS of multimedia applications in a simulated network, whereas the other approach
analyses and assesses QoS in multimedia applications using a combination of
supervised and unsupervised neural networks. The transmitted application’s QoS
parameters are initially analysed either by FCM clustering algorithm or by the
unsupervised learning Kohonen neural network (i.e. Self-Organising Maps (SOM)). The
analysed QoS parameters are then used as inputs to a regression model or supervised
learning Multi-Layer Perceptron (MLP) neural network in order to quantify the overall
QoS. Results in Chapter 6 show how the proposed QoS evaluation systems provide
information about the network’s QoS patterns and based on this information, how the
overall network’s QoS is quantified.
Chapter 9 discusses the overall findings of this research, provides the conclusions, and
highlights future research directions.
9
Chapter 1 Introduction
Chapter2:
Theory and Thesis organisation
Fuzzy logic background
Sampling techniques for
computer network traffic
Fuzzy clustering
2.1 Introduction
The aim of this chapter is to provide the background related to the main issues of this
study. Section 2.2 of this chapter gives an overview of QoS in multimedia networks.
This includes definitions of QoS and its parameters, QoS requirements of multimedia
applications, the service levels required of QoS, and QoS components. Section 2.3
discusses the QoS in wireless and wired networks. This section includes a detailed
description of IEEE 802.l i e as an emerging WLAN standard to provide QoS, and
packet scheduling mechanisms as the most commonly used mechanisms implemented
in wired network to support QoS. Section 2.4 introduces the theory of statistical and A l
techniques used in this study. This includes the basic concepts o f regression analysis as
one of the most widely employed statistical modelling methods and the fundamental
principles of three important paradigms in A l system: fuzzy logic, fuzzy clustering, and
neural network.
In the world of telecommunications, each of the terms: “quality” and “service” has its
own definition. The latter term means the ability o f network as specified by the service
provider to transmit information to the end user, whereas the term “quality” is used to
assess the ability of the service whether it satisfies the stated and implied needs
(Gozdecki et al, 2003). The interrelation between the two terms and the importance of
their associations with three different parts: service provider, network manager, and the
end user have produced the concept of QoS. Therefore, QoS could have a number of
definitions. From the user point of view: QoS is “the collective effect o f service
Chapter 2 Relevant Theory and Background
performance, which determines the degree of satisfaction of the user o f the service”
(ITU, 1994). While from the technical aspects, QoS can be defined as the ability of
network to provide a better service to selected traffic over a variety of technologies.
QoS is thus "a collection of technologies, which allow network-aware applications to
request and receive predictable service levels in terms of data throughput capacity
(bandwidth), latency variations (jitter) or propagation latency (delay)" (Saliba et al,
2005). From service provider perspective, QoS is "a set of service requirements to be
met by the network, while transporting a flow." (Crawley et al, 1998). In this study, the
term QoS refers to the ability of network to assess and provide desired QoS
requirements in terms of delay, jitter, throughput, and packet loss for transmitted
multimedia applications.
In packet-switched networks, there are a number of QoS parameters that can be assessed
and measured to determine QoS. These parameters can express how well the network
treats packets during their journey from the source, throughout the network, and finally
to their destinations. However, as different applications require different QoS
parameters, a correct set of QoS parameters for a particular application being
transmitted should be determined in order to effectively evaluate the QoS provided by
the network for that particular application (Cheong and Lai, 1999). For instance, time
sensitive applications such as video and Voice over IP (VoIP) are susceptible to
transmission parameters such as packet delay, jitter, loss and throughput whereas time
insensitive operations such as a file transfer are more sensitive to packet loss, but can
tolerate transmission jitter (Kurose and Ross, 2005). In this study, multimedia
applications will be considered. Therefore, the most important QoS parameters would
be examined and quantified are throughput, delay, jitter, and packet loss ratio. The
following subsections explain these parameters.
2.2.2.1 Throughput
Throughput is used to assess the capability of network to transmit data over a given
period of time (Heckmann et al, 2002). It could be defined as the maximum
transmission rate of packets that can be sustained between two endpoints. Wang et al
(2000) defined the throughput as the amount of successfully received packets in a
predefined time. Equation (2.1) is used to calculate the throughput:
12
Chapter 2 Relevant Theory and Background
T, (t) = (2.1)
li
Where T* is the measured throughput in bit per second (bps) during the ith interval,
£ P i( t ) is the total bits of all successfully received packets during the ith interval, and
t t is the time duration of the ith interval.
2.2.2.2 Delay
Delay isdefined as the elapsed time for a packet to travel from its source to its
destination. It can be also defined as the total amount of time that the packet takes to be
sent from its source, through the network, until it reaches its destination (Heckmann et
al, 2002). However, as the packet travels from its source to its destination, it suffers
from various types of delay such as queuing delay, transmission delay, and processing
delay. A detailed explanation of these types of delay can be found in (Kurose and Ross,
2005). According to Wang et al (2000), the delay can be calculated using equation
(2 .2):
Di= Rt — St (2.2)
Where Dt is the delay of the ith packet arrived, Rt and St are the arrival and sending
timestamps of the ith packet. The average delay can be calculated using equation (2.3):
Most of real-time applications are delay sensitive. For example, video conferencing and
voice over IP (VoIP) have high sensitivity to delay because the transmitted packets need
to be replayed back at the receiver in real time (Nortal Networks, 2003).
2.2.2.3 Jitter
Jitter is defined as the variation of delay between two consecutive packets for a given
traffic flow (Nortal Networks, 2003). Jitter has a significant effect on real time
applications (video and audio) because they require packets to be received with a fixed
delay. For example, VoIP requires packets to be arrived at a constant rate, since the
received packets will be played back at real time. Therefore, a small amount of jitter
might be acceptable but when the jitter increases, its effect becomes obvious and might
lead to a stuttering communication with pops and clicks (Heckmann et al, 2002). Wang
13
i
Chapter 2 Relevant Theory and Background
et al (2000) pointed out that jitter can be calculated as the difference between the delay
of two consecutive packets using equation (2.4):
Where Ji is the jitter of the ith packet, Dt is the delay of packet i, and is the delay
of the previous packet. Average jitter can be calculated as in equation (2.5):
l
Average jitter = —Y i=i Ji (2.5)
Packets can be lost during their journey from source to destination. Packet loss occurs
due to many effects such as inadequate physical transmission medium, collisions
between packets, queuing overflow, and hardware failure (Nortal Networks, 2003). The
effect of packet loss leads to "distortion, which results in a stuttering and snatch
communication” (Heckmann et al, 2002). Packet loss can be defined as the percentage
of transmitted packets that failed to reach their destinations as given by (Wang et al,
2000).
PL K t) = 1 0 0 X ( l - § m ) (2.6)
Where PL\ is the loss ratio in percentage (%) during the ith interval, and 2 Ri (t ) and
2 St (t) are the total number of received and transmitted packets with the ith interval
respectively.
problems in wireless networks are critical issues to provide service guarantee for
transmitted multimedia applications. Therefore, the acceptable range of QoS parameters
for multimedia applications must be identified. Table (2-1) indicates the sensitivity of
some common applications to QoS parameters (Zhai et al, 2005) and (ITU-T, 2001).
Table 2-1. The sensitivity of some common applications to their QoS parameters.
Typical Packet
Class Application Delay Jitter
bandwidth loss ratio
The service levels of QoS can be defined as the actual capability provided by a network
to deliver service required by specific application. Computer network could provide
three main levels of QoS agreements: best effort service (Floyd and Allman, 2008),
Differentiated Service (DiffServ) (Black et al, 1998), and Integrated service (IntServ)
(Braden et al, 1994). In the best effort service, the network QoS is unspecified. The
packet delivery rate varies depending on the current network load. In this case, the
network does not provide any guarantees or priority to the transmitted applications
although the diversity of their QoS requirements. The DiffServ architecture provides a
service differentiation mechanism to manage network resources based on QoS
requirements of transmitted applications. In DiffServ, the transmitted applications are
aggregated to a number of classes to receive a particular level of service. In IntServ
architecture, end-to-end QoS guarantee per flow is provided. A flow requiring QoS
guarantees must acquire sufficient network resources during its transmission to ensure
its QoS requirements are met. This is achieved by a reservation of network resources for
a specific flow (Shah, 2001) and (Kurose and Ross, 2005).
There are several components of QoS. These include: QoS mapping (Al-Kuwaiti et al,
2007), admission control (Georgoulas et al, 2004), traffic shaping (Lekcharoen, 2007),
15
Chapter 2 Relevant Theory and Background
policing (Koutsakis, 2009), packet queue scheduling (Yu and Meng, 2009), and
prioritisation (Senkindu and Chan, 2008).
The conversion of QoS representation between OSI model layers is referred to QoS
mapping. In admission control, the network take a set of actions during traffic
establishment phase to check if that particular traffic can be admitted or should be
rejected. A new traffic is admitted to the network when its desired QoS can be satisfied,
without causing any QoS violation to the already established services. The function of
traffic shaping mechanism is to ensure that there is a smooth data rate in order to meet
the specified QoS agreement. Traffic policing is used to monitor the transmitted traffic
by discarding or remarking traffic that exceed limits in order to protect the network
from malicious behaviour. Packet queue scheduling determines bandwidth allocation
among transmitted packets and the manner to service various applications with different
QoS requirements. In a prioritisation scheme, applications are classified based on their
QoS requirements and resources are assigned according to classes of priority depending
on network resource availability.
WLANs have grown rapidly over the last decade, offering a range o f practical and
beneficial services for both home users and businesses. This growth relates to the fact
that WLANs allow users to communicate without using physical medium. Also, the cost
of installing WLANs is lower than wired networks because WLANs can be expanded
by just adding access points (AP) rather than installing cables as in wired networks.
Another influential factor in the growth of WLANs is the emergence the of IEEE
802.11 standard in 1997 and its subsequent amendments (1999 - 2005). Its cost
effectiveness, ease of deployment, and mobility support made IEEE 802.11 WLANs to
be used widely and became the dominating WLAN technology. IEEE 802.11 has
16
Chapter 2 Relevant Theory and Background
reached an unprecedented maturity in providing ever-growing bit rates (Lin et al, 2009).
However, with an emergence of multimedia communications over WLAN, the
provision of QoS in WLANs that is capable of guaranteeing QoS requirements of
multimedia applications becomes important. The demand for supporting QoS for
various applications with different QoS requirements has led to the development of a
WLAN standard so called IEEE 802.1 le.
The original standard of IEEE 802.11 was not designed to support QoS (IEEE
Computer Society, 1999). In the legacy IEEE 802.11, applications with different QoS
requirements were treated the same, and the service differentiation in terms of
guaranteed bandwidth, bounded delay, and jitter for particular applications was
disregarded. Due to the growth transmission of time-sensitive and time-insensitive
applications over WLANs, the demand for supporting QoS for various applications has
increased. Thus, the enhancement version of IEEE 802.11 standard called IEEE 802.1 le
was proposed to provide QoS support for applications with different QoS requirements
(IEEE Computer Society, 2005). The IEEE 802.l i e introduces Hybrid Coordination
Function (HCF) which defines two medium access mechanisms: HCF Controlled
Channel Access (HCCA) and Enhanced Distributed Channel Access (EDCA).
HCCA is an enhanced version of Point Coordination Function (PCF) in the legacy IEEE
802.11. Similar to PCF, HCCA has centralised access scheme which uses Hybrid
Coordinator (HC) implemented in the AP to access wireless medium, transmit at the
Contention Free Period (CFP), and the QoS Contention Free (CF-Poll) frame is used to
schedule the uplink traffic. However, HCCA has new operation parameters. These
include: Service Interval (SI), Controlled Access Phase (CAP), and Transmission
Opportunity (TXOP). SI is an interval between two successive TXOPs. CAP is a time
period when the HC maintains control over the medium after gaining medium access by
sensing the channel to be idle for the PCF Inter Frame Space (PIFS) period. TXOP is
time period where stations can transmit a number of MAC Service Data Unites
(MSDUs). Figure 2-1 shows the operation parameters of HCCA (Lee et al, 2011).
17
Chapter 2 Relevant Theory and Background
CFP ! CP
4------------------------------------------- >'<----------------------------------------------------------------------- N
SI [ SI
n ------------------------------------------------------- 1-------------- ►H----------------------------------------------------------------------- ►
i i
E HCCA HCCA HCCA ... EDCA HCCA HCCA HCCA ... EDCA
TXOP, TXOP2 TXOP3 TXOP TXOP, TXOP2 TXOP3 TXOP
In HCCA, each wireless station receives QoS CF-Poll frame from the HC before
transmitting data. While the medium is idle for PDFS, the HC can start a CAP by
sending the QoS CF-Poll frame to the station that is requesting to transmit data in order
to inform that particular station of the time allocated for its transmission. The station
should reply to this poll within a time interval equal to Short Inter Frame Space (SEFS).
For downlink transmission, the HC waits for PIFS and then start its transmission. In
HCCA, the HC has higher priority than other wireless stations. Thus, MSDUs for
downlink traffic can be transferred faster than uplink traffic (Villalon et al, 2007).
Figure 2-2 shows the operation of HCCA for uplink and downlink transmission (Lee et
al, 2011).
TXOP
Figure 2-2. Uplink and downlink transmission between AP and wireless station.
A limitation of HCCA operation is its adaption to traffic rate. The allocation of time
resources to the flows must consider the calculation of scheduling parameters such as
18
Chapter 2 Relevant Theory and Background
SI, and TXOP duration (TD) as these parameters significantly affect the performance of
the scheduler. Another limitation of HCCA is the maximum TD which was set to 8160
pS. This value is an acceptable duration if the traffic rate is relatively small. However, a
sudden increase in traffic rate can exceed the limit of TXOP which in turn affects the SI
to be set to the optimal value. The computational complexity of HCCA and its overhead
can deteriorate the performance of high priority traffic in heavily loaded networks
(Rashid et al, 2007) and (He and Ma, 2011). Due to its limitations, the HCCA
mechanism has not been implemented widely. Thus, the investigation of HCCA
mechanism will be excluded from this study.
Transmission attempt
Figure 2-3 shows that each AC forms an independent backoff entity with its own queue
and corresponds to a different level of transmission priority. To simplify the concept,
the four access categories (AC0 - AC3) assigned to different traffic priority. The traffic
with the highest priority is assigned to AC0, whereas AC3 is assigned to the traffic with
the lowest priority. The priority of an AC is determined by a set of parameters called
19
Chapter 2 Relevant Theory and Background
EDCA parameters. These parameters are: Arbitration Inter Frame Space (AIFS),
Minimum Contention Window (CWmin), Maximum Contention Window (CWmax),
and Transmission Opportunity (TXOP). The highest priority AC corresponds to the
smallest AIFS, CWmin, CWmax and largest TXOP (Lin et al, 2009).
The AIFS is a replacement of the Distributed Inter Frame Space (DIFS) in IEEE 802.11
DCF access method. The value of AIFS can be determined based on the Arbitration
Inter Frame Space Number (AIFSN) used to determine the length o f AIFS, the time unit
dictated by the physical layer characteristics SlotTime, and the Short Inter Frame Space
period (SIFS) used to manage and control frames. AIFS for particulate AC can be
calculated using equation 2.6 (Alahmadi and Madkour, 2008):
The values of CWmin and CWmax determine the range of contention window for each
AC. These values determine the range of random backoff slots which in turn control the
waiting period for traffic before accessing the channel (Abeysekera et al, 2009). The
backoff value of a particular AC can be determined using equation 2.7:
Backoff [AC] = random [0, min (2K (CWmin [AC] + ! ) - ! , CWmax [A C ]] (2.7)
where K is the number of collisions for the currently transmitted frame, CWmin, and
CWmax are the minimum and maximum contention windows respectively. The high
priority traffic with small CW values have small waiting period before accessing the
medium whereas low priority traffic with a large CW values have a long waiting time to
access the channel.
The transmission opportunity (TXOP) defines the transmission holding time for each
AC. Each AC can transmit for a certain time interval whose length is determined by
TXOP Limit.
During the operation of IEEE 802.1 le EDCA as shown in Figure 2-4 (Lin et al, 2009),
each AC contends to access the channel as an individual virtual station and start its
backoff procedure after detecting the channel is idle for an AIFS period. AC with the
smallest AIFS has the highest priority and needs to defer for its corresponding AIFS
interval. When a particular station can initiate its transmission, it will be allowed to
transmit multiple data frames from the same AC continuously during time interval
defined by TXOP. The highest priority AC has a largest TXOP period, while the lowest
20
Chapter 2 Relevant 1 neory ana nacKgrouna
priority AC has a smallest TXOP period. Similar to a real packets collision, when an
internal collision occurs among the ACs within the same station, the higher priority AC
has the rights to transmit whereas the lower priority AC suffers from a virtual collision.
AIFS [AC2]
Backoff Counter
Window Frozen
AIFS [AC1]
Backoff
Immediate access when medium Counter
Window
is idle >= AIFS fACl Frozen
AIFS [ACO] Contention Window [0,CW[AC]]
AIFS [AC]
Busy Medium Backoff Next
Window Frame
Slot Time
Defer Access Select Slot and decrement backoff
as long as medium stays idle
IEEE 802.l i e EDCA aims to ensure better service to high priority traffic and offer a
reduced service for low priority traffic. The simplicity of EDCA operation as compared
with HCCA means it is implemented widely. However, although EDCA mechanism
improves the QoS of time sensitive applications, its performance is not optimal because
EDCA parameters are not adjustable according to the network conditions. EDCA
mechanism is reported to be unable to guarantee a good performance when the network
traffic load was high. The main reason was the excessive number of packet collisions,
which in turn was due to the fixed transmission parameters values assigned to the ACs
(Villalon et al, 2007) and (Lin et al, 2009).
21
Chapter 2 Relevant Theory and Background
Most current wired networks that are based on IEEE 802.3 have error rates when
transmitting multimedia applications as compared with wireless networks. This is due to
a higher bandwidth which ranges between 10Mbps - 100 Mbps. However, with the
growth in transmission of real-time applications over wired networks, over-provisioning
which refers to enhancing the network capability by simply providing the network with
enough bandwidth, in order for all traffic to meet their QoS requirements might not be
an optimal solution. This is because over-provisioning approach can be difficult and
costly (Fraleigh et al, 2003). In addition, over provisioning bandwidth in the wired
network may not prove effective in dealing with QoS requirements of multimedia
applications (as bandwidth is a costly resource).
Multimedia applications must have priority over elastic applications because of their
higher sensitivity to QoS parameters. For example, high delay variation o f VoIP packets
would affect its quality which in turn leads to a stuttering communication with pops and
clicks.
An important issue that affects the QoS in wired networks is traffic prioritisation. When
multiple packets are serviced through a bottleneck such as a router in the same manner,
this would negatively affect the QoS, as it ignores the QoS requirements of transmitted
applications. The transmitted packets could for instance experience several types of
delay such as queuing delay, which occurs in the output buffer of a router. Also, when
the buffer of the congested router is overflowed, the transmitted packets could be
dropped regardless to their QoS requirements.
There are several mechanisms to support QoS in wired networks. These include: Call
Admission Control (CAC), bandwidth reservation, congestion-management,
congestion-avoidance, traffic policing, and shaping (Szigeti and Hatting, 2005). In this
study, the network traffic prioritisation will be investigated as one of the most important
issues that affect the QoS in wired networks. Therefore, packet scheduling mechanisms
which are the most commonly congestion-management tools will be considered.
22
Chapter 2 Relevant Theory and Background
Conceptually, the phrase “scheduling” refers to a set of rules that determine how a
frame or packet is treated when congestion or bottleneck occurs at the convergence
point. When bottleneck occurs, devices such as routers have buffers that allow packets
to be stored temporarily in order to be scheduled subsequently. This process is known as
“queuing”.
The two terms “scheduling” and “queuing” are complementary and their processes are
intertwined. Queuing process is engaged only when congestion occurs and deactivated
after the congestion is cleared. Similar to queuing, scheduling takes place when packets
experience congestion. However, the scheduler has to decide which packet should be
transmitted next, even when there is no congestion. Packet queue scheduling determines
bandwidth allocation among transmitted packets and the manner to service various
applications with respect to different QoS requirements (Szigeti and Hatting, 2005).
FIFO queuing mechanism is the most basic queue scheduling algorithm. It is also
known as First-Come-First-Served (FCFS). The incoming packets are accepted in order
of arrivals. Figure 2-5 shows the process of FIFO scheduling mechanism.
FIFO serves the first packet in the queue regardless of any prioritisation or even fairness.
This feature makes it to be the simplest scheduling mechanism in terms of
implementation. However, FIFO is drop-tail based, when the buffer at the router
implementing FIFO mechanism becomes full, the arrived packets are dropped.
Therefore, FIFO can be insufficient mechanism in meeting QoS requirement for
particular applications such as real-time applications (Semeria, 2001) and (Hasegawa et
al, 2002).
23
Chapter 2 Relevant Theory and Background
Flow 1
Flow 2
FIFO
Flow 3 Port
Flow 4
Flow 5
Flow 6
Flow 1
Highest
Flow 2 □HUD priority
Flow 3
Flow 4
mm Middle
priority
Flow 5
mm Lowest
priority
Flow 6
24
Chapter 2 Relevant Theory and Background
FQ scheduling mechanism was proposed by John Nagle in 1987 (Miaji and Hassan,
2010). It is designed to ensure that each flow has a fair distribution of the bandwidth
regardless of the traffic transmission rate. In FQ, packets are first classified into flows
and then each flow is assigned to a queue dedicated for that particular flow. During the
scheduling process, flows are serviced one packet at a time in round robin order and
empty queues are skipped. If a packet arrives at an empty queue after the scheduler is
visited, the packet has to wait in that queue until the next visit of the scheduler. Figure
2-7 shows the principle of FQ (Semeria, 2001).
Packet classification
\ Scheduler
\
Flow 1
Flow 2
Flow 3 Port
Flow 4
□U ED
Flow 5
Flow 6 1
WFQ was proposed by Zhang, Demers, Keshav and Schenke in 1989 to address the
limitations of PQ, and FQ mechanisms. WFQ allocates bandwidth to different flows
according to their assigned weights in order to satisfy the QoS requirements for
different applications (Balogh and Medvecky, 2011). However, in WFQ, flows with
large packet size are not allocated more bandwidth than flows with small packet size.
Therefore, the distribution of bandwidth among variable length packets is carried out by
a weighted bit-by-bit round robin scheduling. This approach supports fair distribution of
bandwidth because it takes into the account the length of transmitted packets. Figure 2-8
shows a weighted bit-by-bit round robin scheduling serving three flows. 50% of the
bandwidth is assigned to flow 1 whereas the remaining bandwidth are allocated to flows
25
Chapter 2 Relevant Theory and Background
2 and 3 (i.e. 25% to each flow). During scheduling, 2 bits are transmitted from flow 1, 1
bit from flow 2, and 1 bit from flow 3. This causes the packet with 600 byte to be
transmitted before the packet with 350 byte, which in turn is transmitted before the
packet with 450 byte (Semeria, 2001).
Flow 2
25% bw 350
Flow 1
50% bw 600
WRR is also known as Class Based Queuing (CBQ). The operation of WRR is shown in
Figure 2-9. The packets sent throughout the outgoing port are first classified into
different classes and then assigned to a queue that is particularly dedicated to that class.
The queues are in turn serviced using the weights associated with them. The weights
indicate the number of packets to be sent for each class in a single service round. The
number of packets transmitted for queue (/) is calculated using equation (2.8)
Wi
Number o f serviced packets^ ——— X R (2.8)
Where W, is the associated weight for queue (/), n determines the number of the queues,
and R is the link capacity (Senkindu and Chan, 2008).
26
Chapter 2 Relevant Theory and Background
single packet each time it is visited but high priority queues are visited multiple times
during a single service round (Semeria, 2001).
Packet classification
Queue2 ,Weight2
Port
Queue n-1,Weight n-1
Queue n ,Weight n
Packets sent through the port
Sent packets
FIFO does not support QoS because it treats all traffic equally. It serves the first packet
in the queue regardless of any prioritisation or fairness. PQ provides premium service to
the high priority traffic at the expense of the low priority traffic. Low priority traffic are
denied access to the buffer space, whenever high priority traffic is transmitted, which in
turn causes the low priority traffic to experience excessive delay and high packet
dropping. FQ is not designed to support traffic with different QoS requirements, as it
allocates the same amount of bandwidth among multiple traffic. The computational
complexity of WFQ algorithm affects its scalability to support lager traffic with
different requirements at the edge of the network. WRR is designed to address the
limitations of FIFO, PQ, and FQ by classifying traffic based on their QoS requirements,
and ensuring that low priority traffic has access to buffer space and output port
bandwidth. The implementation of WRR is more popular and its operation is less
complex comparing with WFQ. Although WRR does not take packet size into account,
an accurate bandwidth allocation could provide an optimal algorithm for usage in
modem multimedia networks (Balogh and Medvecky, 2011). Therefore, in this study,
WRR was considered to provide traffic prioritisation because of its practicality and low
complexity.
27
Chapter 2 Relevant Theory and Background
The aforementioned techniques were used in this study to fulfil the main objectives of
this thesis. The use of these techniques is as follows:
Different regression models are used for prediction; they can be classified into linear
and nonlinear model. Linear models include Auto Regressive (Box and Jenkins, 1976),
Moving Average (Vandaele, 1983), and mixed of Auto Regressive and Moving Average
(Box and Jenkins, 1976) and (Ljung, 1999). The nonlinear models include Bilinear
Model, Threshold Auto Regressive Model, and Exponential Auto Regressive Model
(Priestly, 1988). A detailed description about linear and nonlinear regression models is
provided by (Box and Jenkins, 1976), (Vandaele, 1983), (Ljung, 1999, and (Priestly,
1988).
Due its simplicity and effectiveness, multi linear regression model is a commonly used
method for prediction purposes (Chatteijee and Hadi, 2006) and (Sweet and Grace-
Martin, 2010). Therefore, in this study, this type of regression is used.
28
Chapter 2 Relevant Theory and Background
The multi-linear regression model is a widely employed statistical method due to its
effectiveness for creating functional relationships among variables (Jain, 1991). Its aim
is to analyse the relationship between several variables. One variable is considered to be
the dependent or response variable and the others are considered to be independent or
descriptive variables (Chatteijee and Hadi, 2006). In order for the regression model to
be valid and accurate predictor, there are some assumptions that need to be followed
(Jain, 1991) and (Chatteijee and Hadi, 2006). These are as follows: (i) dependent
variable and independent variables need to be linearly related, (ii) the independent
variables is non-stochastic and measured without error, and (iii) model errors are
independent and normally distributed.
The formula of multi linear regression model as shown in equation (2.9) defines the
relationship model between dependent variable ( y ) and independent variables
(xlt x 2, ...., xn) as follows (Chatteijee and Price, 2006) and (Jain, 1991):
Y=XB + e (2.11)
e = Y —XB (2.13)
29
Chapter 2 Relevant Theory and Background
Fuzzy logic was introduced by Lotfi Zadeh in 1965 (Zadeh, 1965) as a methodology for
computing words rather than numbers. The concept of fuzzy logic is based on natural
human communication language because it has similarities with human knowledge and
reasoning (Klir and Yuan, 1995). The robustness of fuzzy logic due to the direct
expression of input/output relationships without a physical derivation of the rules, and
its flexibility to cope with imprecise and uncertain information and then draw definite
conclusions makes it an excellent and powerful mechanism (Jantzen, 1998) (Khoukhi
and Cherkaoui, 2008) and (Muyeen and Al-Durra, 2013). Unlike Binary logic (i.e.
Classical logic) which has a sharp boundary between true and false states, fuzzy logic
implements a gradient of possible states between true and false as shown in Figure (2-
10) (Cirstea et al, 2002).
Fuzzy logic is applied to many applications in various domains such as control, decision
making, optimisation, and evaluating systems (Klir and Yuan, 1995) (Naoum-Sawaya
and Ghaddar, 2005) (Saraireh et al, 2008) and (Muyeen and Al-Durra, 2013).
Fuzzy Inference System (FIS) is built upon the theory of fuzzy logic. FIS includes four
main components: fuzzification, rules base, inference engine, and defuzzification as
show in Figure (2-11) (Jantzen, 1998).
Rule Base
Inputs ^ Fuzzification
I
Inference
Defuzzification ■ f Output
engine
i :
30
Chapter 2 Relevant Theory and Background
FIS is used to interpret (i.e fuzzify) the crisp values of inputs into linguistic terms, and
based on a set of predefined rules, it calculates linguistic output value which in turn is
converted (i.e. defuzzified) into real crisp output value (Naoum-Sawaya and Ghaddar,
2005) and (Saraireh et al, 2008). The following subsections outline each component of
FIS.
2.4.2.1.1 Fuzzification
This is a process of converting numerical input values into linguistic terms and
determining the degree of belonging to the appropriate fuzzy sets via membership
functions. In fuzzy sets, Cirstea et al (2002) reported that an element (jc,) in the universe
of discourse X is assigned a degree of membership p(;c,) obtained by a membership
function as shown in Figure (2-12). A membership function allows gradual transition
from full-belonging to a fuzzy set to not-belonging at all with intermediate values
presenting degrees of belonging (Al-Sbou et al, 2006). In fuzzification process, different
types of membership functions can be employed. These include Triangular,
Trapezoidal, Gaussian, Generalised, and bell Sigmoid (Mathworks, 2012(a)).
0 ll X
(X)
This component contains a set of IF-THEN rules represented in linguistic variables. The
set of IF-THEN rules is the bases of decision making process of FIS. The number of
rules depends on the number of inputs and outputs variables as well as the number of
membership functions associated with them (Jantzen, 1998). The common form of IF-
THEN rules as follows:
31
Chapter 2 Relevant Theory and Background
where the Antecedent relates the linguistic term to a fuzzy set, and the Consequent
represents the conclusion from IF term. Each rule could have one or more connectives
(i.e. fuzzy operators). The most common fuzzy operations applied on IF-THEN rules are
Intersection, Union, and Complement which respectively defined by fuzzy operators
AND, OR, and NOT (Klir and Yuan, 1995) (Ross, 1995) and (Mathworks, 2012(a)). For
example, given that px and pYare the degrees of membership functions for fuzzy sets X
and Y respectively, the application of fuzzy operators AND, OR, and NOT can be
defined as given in equation (2.14) (Ross, 1995):
Fuzzy inference engine uses fuzzified inputs along with the rules to perform inferencing
(i.e. the process of implication and then aggregation) (Jantzen, 1998). The fuzzified
inputs can be related to more than one rule to specify how adequately each rule
describes the current situation by computing the degree of truth for IF condition. More
than one rule might be triggered at the same time describing the same situation. Each of
these rules produces Consequent or Conclusion to be taken in the THEN condition. This
process is performed by implication method which is defined as the shaping of output
membership functions. The input for the implication is a single number given by the
Antecedent of the rule, and the output is a fuzzy set. The truncated output fuzzy sets
from the implication process which describes the firing strength of the rules is then
processed by an aggregation method. In the aggregation process, the truncated output
fuzzy sets from the implication process are unified to produce one output fuzzy set
(Ross, 1995).
2.4.2.1.4 Defuzzification
This is the process that converts the output linguistic value (i.e. the aggregate output
fuzzy set) into a real numeric value. The input for the defuzzification process is the
aggregate output fuzzy set and the output is a single number. However, the aggregate of
a fuzzy set covers a range of output values which in turn must be defuzzified to produce
a single output value from the set. There are several methods can be used in
32
Chapter 2 Relevant Theory ana Background
Figure 2-13 shows the information flows through the process of fuzzy inference system:
commencing from fuzzifying inputs, through the process of applying fuzzy operator,
implication method, aggregation method, and terminating by defuzzification process
(Math work,2012(a)).
ZApoty 5frp
-AW
Aafon
1. e u z t f y f i p v t
tnemoo
fuzzy ru le 1
TV r>v etoponrjoncY
fuzzy ru le 2-
A .
1 3 a a 4. Appty
aggregation
fuzzy ru le 3 rm tfiod
input 1 input 2
/. * 6 D efu zztfy th e
H esu it OT a g g re g a te o u tp u t
a g g re g a tion (c*n trold>-
R esu /t o f o u tp u t
d efuzzification
Fuzzy inference system has two methods: Mamdani and Sugeno inference methods. The
procedure of fuzzifying the inputs and applying the fuzzy operator during the fuzzy
inference process are similar in both methods. However, the main difference between
Mamdani and Sugeno is the manner the outputs are determined. Mamdani-type FIS is
based on defuzzification process to generate crisp output from output fuzzy set, while
Sugeno-type FIS uses weighted average to compute the crisp output. The advantage of
33
Chapter 2 Relevant Theory and Background
Mamdani FIS is that its outputs are expressed and interpreted. This feature is lost in the
Sugeno FIS since the consequents of the rules are not fuzzy (Arshdeep and Amrit, 2012).
The other difference is that Mamdani FIS has output membership functions whereas
Sugeno FIS has no output membership functions. Due to the interpretable and intuitive
nature of the rule base, Mamdani-type FIS is widely used particularly for decision
support application (Haman and Geogranas, 2008). Therefore, in this study, Mamdani-
type FIS will be used in this study.
Clustering techniques are used to partition data into correlated groups where different
data objects should belong to different clusters and similar data objects are assigned to
the same cluster. The aim of clustering is to reveal the underlying structures of data and
explore its nature (Rokach and Maimon, 2005). Clustering techniques can be classified
based on many criteria. For instance, clustering techniques can be divided into two main
groups: hierarchical and partitioning techniques (Farley and Raftery, 1998). Another
criterion divided clustering methods into: density-based methods, model-based
clustering, grid based methods, and soft computing methods (Han and Kamber, 2006).
In this study, the focus will be on Fuzzy C-means (FCM) as one o f the most widely
employed soft clustering methods. Unlike hard clustering methods where an object must
belong to only one particular cluster, in FCM, it can belong to several clusters with
different degrees of membership between 0 and 1 to indicate their partial membership
(Parker et al, 2012). Therefore, in this study, FCM was used to analyse network QoS
because the natural characteristics of network QoS patterns partly cover more than a
single cluster.
FCM clustering algorithms was originally introduced by Dunn in 1973 and improved by
Jim Bezdek in 1981 (Nascimento et al, 2000). This algorithm is one of the most widely
used clustering algorithms. FCM is defined as a mechanism to discover certain features
in a set of data and classify each element of data into a number of clusters with different
degrees of memberships (Wang, 2009) and (Chaabane et al, 2008). FCM can be applied
to partition a set of data with a form of matrix X of size n X AT as shown in equation
(2.15).
34
Chapter 2 Relevant t heory and .Background
number of features p. FCM operates on the matrix X and minimises the FCM objective
function given in Equation (2.16) in order to partition matrix X into C clusters (Lei, et al
, 2012).
m n2 (2.16)
] QC'.u.V) = Zf=1Z?=1G»y)m Dfj
The value m controls the degree of fuzziness for the membership of the cluster where m
e [1, oo]. As the value of m decreased, the membership of the cluster becomes closer to
the binary clustering. U the membership matrix includes n x C values and can be
expressed as in formula (2.17):
where each value of the matrixp ^ j = 1, ...,n and i = 1, ...,C indicates the degree of
membership between vector Xj and cluster Q and must meet the following criteria:
• 2 f= iJ«y = 1. Vy=
During the clustering process, the objective function J (U, V) is minimized with the
following iterative steps (Parker et al, 2012) (Chen et al, 2009) and (Timo et al, 2002):
{pij) XJ v .= 1 c (2.18)
1 1 .......
3. The distance Dfj which is the Euclidian distance between Xj to the centre of
2
(2.19)
i
(2 .20)
Equations (2.18) - (2.20) are repeated until the termination criteria of FCM are met. The
process of FCM can be terminated when the maximum number of iteration is reached or
the objective function improvement between two consecutive iterations is less than the
minimum amount of improvement (Timo et al, 2002).
ANN is an adaptive parallel processing system capable of achieving results through the
process of learning. It provides a mean to model the human brain in a simplified form.
An ANN consists of a number of highly interconnected neurons that learn by interaction
with each other as each interconnection has an associated weight (Abraham, 2005). In
an ANN, the function of each neuron is to receive information, process it, and produce
an output (Haykin, 1999). During the training process, the output is used to adjust the
weight values to optimise the neural network performance. The mechanism to determine
the amount of change in the weights is the neural network’s learning algorithm (Cirstea
et al, 2002) and (Zaknich, 2003). ANNs can be classified based on the manner of
learning into supervised and unsupervised neural network. In supervised learning, they
are provided with training examples from known classes together with their desired
outcomes. In unsupervised learning, the neural network requires only training examples
to be trained (Nogueira et al, 2006).
The advantage of ANN is the ability to derive meaning from imprecise values with
highly parallel computing structure. This capability gives ANN the strength to model
complex systems in efficient and effective manner and then achieve desired results. In
this study, a combination of supervised and unsupervised neural networks was
considered to evaluate the QoS parameters for multimedia applications. The transmitted
application’s QoS parameters were initially analysed by the unsupervised learning
Kohonen neural network. The analysed QoS parameters were then used as inputs to a
supervised learning Multi-Layer Perceptron (MLP) neural network in order to quantify
36
Chapter 2 Relevant Theory and Background
the overall QoS. An explanation of MLP and Kohonen neural network (i.e. Self-
Organising Maps SOM) are in the next sections.
In this study, a multi-layer perceptron (MLP) neural network was employed to assess
the overall QoS due to its suitability and effectiveness. As one of the most popular
supervised ANN, MLP needs to be provided with representative examples from each
class, together with their corresponding class category. The architecture of MLP neural
networks is shown in Figure 2-14 (a) (Abraham, 2005).
MLP composes of an input layer, one or two hidden layers and an output layer. At each
layer, there are a number of neurons (i.e. processing elements). The function of each
neuron is to receive information (i.e. the inputs with their associated weights), process
them, and produce an output (Zaknich, 2003).
The operation of an MLP is shown in Figure 2-14 (b). During the training phase, a
known pattern (x,-) is applied to the input layer of the MLP, and its target (i.e. desired
value (d)) is applied to the output layer of the MLP. The elements of input (x,-) are
multiplied with their associated connection weights (w,) and the resulting value (s) is
obtained using a summation function as in equation (2.21) (Cirstea et al, 2002):
(2.21)
e — d —y (2 .22)
The calculated error is then used by the learning algorithm to adjust the weights in order
to reduce the error in the next iteration.
37
Chapter 2 Relevant Theory and Background
The process of learning and adapting are achieved by the learning algorithm. The
function of the learning algorithm is to use the calculated error (e) and the input data (x,-)
to adjust the values of the connections' weights (w,) in such a way as to reduce the
magnitude of the error in the following training iteration (Haykin, 1999). There are
several methods to realise the required learning. These include gradient descent back-
propagation, gradient descent with adaptive learning rate back-propagation, gradient
descent with momentum back-propagation, and gradient descent with momentum and
adaptive learning rate back-propagation (Abraham, 2005),(Cirstea et al, 2002).
(a)
Inputs Outputs
(b) Output
Error
Learning algorithm
Figure 2-14. Multi-Layer Perceptron: (a) The architecture, (b) MLP operation.
The MLP training process is repeated until either the specified number of iterations is
reached or when there is insignificant error between the network output (y) and the
desired result (d) (Mathworks, 2012(b)). The termination criteria evaluate how effective
the MLP is trained.
After the completion of training phase, the trained MLP is examined during the test
phase. Unknown input patterns are fed to MLP input layer and outputs are produced by
the output layer.
38
Chapter 2 Relevant Theory and Background
In contrast to the multilayer perceptron, the Kohonen network (i.e Self Organising Maps
SOM) is one of the most popular unsupervised neural networks require only training
examples to learn (i.e. no desired output is required). Self-Organizing Map was
introduced by Teuvo Kohonen as a data visualization technique (Kohonen, 1982). SOM
visualizes data by reducing its dimensions to a map, and represents similar data objects
into correlated clusters. An aspect of this study is that QoS parameters of multimedia
applications were intelligently classified using SOM. In situations such as network QoS,
where the natural characteristics of traffic cover multiple clusters, SOM could be an
effective clustering technique to analyse network QoS patterns.
The Kohonen network has a single layer of neurons known as a Kohonen map as shown
in Figure (2-15). The Kohonen map can be arranged in various topologies such as
rectangular and hexagonal topology.
A neighbourhood neuron
Inputs
Xj
x2
Xi
As shown in Figure (2-15), the j th neuron in Kohonen map is connected to ith input
feature of a certain pattern (x) to the neural network. Each connection has an associated
weight(wij). The connections' weights are initially set to random values between 0 and
1. Then, the network learns by determining the Euclidean distance dj between the
features of normalised input pattern (xi) and the neuron’s weights. For each j th neuron
and N features for each example, the Euclidean distance is calculated using equation
(2.23).
39
Chapter 2 Relevant Theory and Background
The neuron that obtained the smallest Euclidean distance with the input patterns is
considered as the winning neuron. Its weights are adjusted using the learning algorithm
as expressed in equation (2.24).
where, Wij(n + 1) is the updated weight, tj(n) is the learning rate 0 < a < 1, and the
term (** (n) — Wj; (n )) represents the error. The learning rate is usually a value between
0 and 1 which in turn controls the adaptation speed. The learning algorithm ensures that
the winning neuron's weights become iteratively closer to the input pattern category.
This in turn allocates the winning neuron to become representative o f that specific
category.
The weights associated with a number of neurons around the winning neuron are also
updated to a lesser extent. This enables specific regions of the Kohonen map to be
associated with different pattern categories. The neurons around the winning neuron
which their weights are updated are known as neighbourhood neurons, and the area of
the Kohonen map covered by them is referred to as the neighbourhood region as shown
in Figure (2-15) (Haykin, 1999).
2.5 Summary
This chapter provided a theoretical background related to the QoS of multimedia
networks. This includes the definitions, QoS parameters, QoS requirements of
multimedia applications, and QoS components. It also discussed the QoS in wireless
and wired networks. The IEEE 802.l i e as an emerging WLANs standard to provide
QoS was explained, and packet scheduling mechanisms, the most commonly
mechanisms implemented in wired network to support QoS were reviewed. This chapter
also provided the relevant theoretical background to statistical and Artificial Intelligent
(AI) techniques which were used in this study. The fundamental principles of regression
analysis which is one of the most widely employed statistical modelling methods were
discussed. The basic concepts of fuzzy logic, fuzzy C-means clustering algorithm,
Multilayer Perceptron Neural Network, and Kohonen neural network were explained.
The next chapter reviews the literature relevant to the process of managing QoS for
multimedia computer networks
40
Chapter 3 Literature Review
3.1 Introduction
Managing Quality of Service (QoS) of multimedia applications is currently one of the
principle research topics in the field of computer networks. Two important factors make
the QoS management an issue of great importance, (i) Computer networks are
increasingly integrated (i.e. networks consist of heterogeneous networks: wired and
wireless), (ii) Computer networks carry diverse multimedia applications involving
video, audio, and data which require a large bandwidth and perceived quality
(Mohammed et al, 2001).
In this study, network QoS management refers to evaluation and improvement of QoS
provided by computer networks. Evaluation of QoS aims to analyse and quantify
network performance with respect to meeting the applications' transmission
requirements. The QoS improvement involves the ability to take actions to enhance
network QoS or change network performance toward a desired operation. However,
there are many issues related to the process of managing multimedia computer
networks. The main issues are: (i) Multimedia applications generate an extensive
amount of data in the form of information packets. The collection and processing of all
these packets in real time are not practically feasible, (ii) Gathered network information
which represents network performance in delivering diverse applications include a
multitude of parameters related to QoS. These parameters need evaluating in an
effective manner, (iii) Multimedia applications require a large bandwidth, that in turn
are considered to be as a critical issue because of the limitations of the physical
communication channels as well as the interfering noise, particularly in the wireless side
of the network. Accordingly, the QoS of the transmitted applications could be
unpredictable, (iv) The existing monitoring tools are unable to get directly the overall
network QoS. Network managers have to do an extra evaluation to assess the overall
network QoS. This process could be complicated, expensive, and time consuming.
In this chapter, the previous studies relevant to the aforementioned issues of managing
QoS of multimedia network are critically analysed and discussed. The aim of this
41
Chapter 3 Literature Review
chapter is to identify the potential gaps associated with these issues which in turn
require a further development and investigation.
This chapter is organised as follows: section 3.2 reviews the sampling techniques used
to gather information from network traffic. In section 3.3, the current QoS analysis and
assessment techniques used to evaluate network QoS are discussed. The relevant studies
considering QoS support in wireless and wired networks are evaluated in section 3.4.
Section 3.5 critically analyses the exiting monitoring tools used to assess the network
performance. Section 3.6 reviews the applications of statistical and artificial intelligent
techniques (used within this study) into the field of computer network management.
Finally, in section 3.7, a summary of this chapter is provided.
In non-adaptive sampling techniques, the packet selection method can be time interval
based or packet number based. In the former method, the selection is based on
predefined time intervals, whereas in the latter, the packet selection decision is based on
a packet count (Zseby, 2004) and (Claffy et al, 1993).
42
Chapter 3 Literature Review
1 d d d d
Systematic sampling V
oC3
b • • • 1 O
Random sampling
h 1• 1 I
Stratified sampling Sampling interval
(a) (b)
Figure 3-1. Classification of sampling techniques: (a) Non adaptive sampling, (b) The
concept of adaptive sampling.
Gathering traffic information from the network using conventional sampling techniques
poses a challenge as the process should reflect on the nature of ongoing network traffic
activity. Therefore, in order to reduce the biasness presented in conventional sampling
methods and to increase the effectiveness of measuring QoS parameters, traffic should
be sampled adaptively. This process is illustrated in Figure 3-1 (b).
43
Chapter 3 Literature Review
during the period of high activity, whereas a larger sample interval is required during
the period of low traffic activity (Giertl et al, 2006).
A number of studies have been conducted using adaptive sampling to gather network
information. For example, in (Hernandez et al, 2001), two adaptive sampling schemes
have been proposed based on linear prediction and fuzzy logic. The experimental results
showed that both approaches achieved better results compared with systematic
sampling.
An adaptive sampling based on fuzzy logic was reported in (Giertl et al, 2006). The
contribution of the study was to assess the utilisation of bandwidth for real TCP/IP
activity of network interface, connecting a local network to the Internet using adaptive,
random and stratified sampling techniques. The study showed that adaptive sampling
was more effective than the non-adaptive sampling techniques.
In (Ma et al, 2003, and 2004), an adaptive sampling technique was devised using the
time-sliding window (TSW) algorithm to estimate traffic rate. A key element of the
devised technique was to predict future behaviour based on observed behaviour. The
experimental results showed that measuring QoS parameters of voice traffic using the
devised adaptive sampling technique was more accurate than the conventional sampling
techniques. Gan et al (2009) proposed another method to sample packets in an adaptive
manner. Their adaptive sampling algorithm automatically adjusted the sampling
granularity according to the packet arrival interval. Their results showed that the
proposed adaptive sampling method is more accurate and economical than static
sampling methods.
Adaptive sampling methods are not only used for traffic measurement, but they also can
be used for a number of other applications (i.e. network security application). For
instance, Zhang et al (2007) proposed a small packet threshold adaptive sampling
algorithm to capture malicious packets, which speeded up the sampling process and
achieved accurate results. Patcha and Park (2006) in their paper proposed an adaptive
sampling algorithm based on weighted least squares prediction. The proposed sampling
algorithm was tailored to enhance the capability of network based IDS at detecting
denial of-Service (DoS) attacks. The algorithm was not only proposed to reduce data
analysed by IDS but also, it maintained the intrinsic self-similar characteristic of
network traffic. This feature can be used by IDS to detect DoS attacks by using the fact
44
Chapter 3 Literature Review
From the above discussion, most of the previous adaptive sampling schemes require
sophisticated computations as in the case o f (Hernandez et al, 2001), (Giertl et al, 2006),
and (Giertl et al, 2008). Other schemes consider one parameter as the reference-
coefficient of sampling granularity as the case in (Gan et al, 2009). None of the previous
studies considered the traffic’s statistics such as mean, median, and standard deviation
during the process of sampling multimedia traffic in an adaptive manner.
45
Chapter 3 Literature Review
Wilbur, 2001), and (Malan and Jahanian, 1998). The following subsections discuss the
current QoS analysis and measurement techniques used to evaluate network QoS.
Analysis of network QoS plays a crucial role in practicing effective management for
multimedia computer networks. The aim of QoS analysis is to obtain a comprehensive
view about the state of the network and simultaneously discover important details from
the transmitted traffic (Timo et al, 2002). The network traffic analysis has been
investigated by a number of studies using several mechanisms. The most explorative
techniques used to analyse the characteristics of network traffic are statistical analysis
techniques and AI techniques (Liu et al, 2012) and (Ting et al, 2010).
Statistical parameters such as mean, standard deviation, and mode that are used to
analyse network traffic may not be as effective as AI techniques. This is because
outliers of the analysed traffic using statistical parameters could affect the final
conclusion about traffic characteristics which in turn may give inaccurate results about
the analysed traffic (Jain, 1991). Therefore, statistical parameters will not be used in this
study to analyse QoS.
Fuzzy C-means (FCM) and Self-Organising Maps (SOM) (i.e. Kohonen neural
network) were previously used in a number of study to analyse network traffic..
FCM was used to cluster network traffic and produce application profiles which
contained significant information about the current status of the network in order to
manage network resources as reported in (Timo et al, 2002). A network administrator
assistance system was proposed based on FCM. The proposed system utilised a FCM
method to analyse users’ network behaviours and traffic-load patterns based on the
measured traffic data of an IP network. Analysed results were used to assist
administrators to determine efficient controlling and configuring parameters of the
network management (Chen et al, 2009). In wireless sensor networks, FCM algorithm
was used to create clusters which reduced the spatial distance between sensors nodes
46
Chapter 3 Literature Review
(Hoang et al, 2010). FCM was employed to detect routing attacks caused by abnormal
flows in a wireless sensor network. The study demonstrated that FCM can be a valuable
tool for intrusion detection (Wang et al, 2009).
SOM was also used by several studies to analyse network traffic. For example, Kemel-
SOM (KSOM) was proposed to introduce network traffic classification approach (Ting
et al, 2010). The experimental results showed that KSOM achieved high classification
accuracy and successfully categorised network traffic characteristics. In (Kiziloren and
Germen, 2007), network traffic types were analysed using SOM. The aim was to
distinguish between normal traffic and anomaly traffic, such as port scanning and
Denial of Service Attacks. The results demonstrated the usefulness of SOM to
distinguish three traffic types: port scanning, heavy-download, and other traffic.
However, none of the previous studies discussed in this section have utilised FCM or
SOM to analyse and classify QoS parameters (i.e. delay, jitter, and packet loss ratio). A
novel aspect of this study is that QoS parameters of multimedia applications are
intelligently classified using either FCM or SOM. In situations such as network QoS
parameters where the natural characteristics of traffic cover more than a single cluster,
FCM algorithm or SOM could be an effective QoS analysis technique.
A number of studies quantified QoS based on objective approaches. For instance, Sun
and Ifeachor (2002) examined the impact of packet loss and gender of talker on
perceived speech quality using ITU PESQ measurement algorithm. In the study, they
found that the packet loss and gender of talker have an impact on perceived speech
quality. The quality for female talkers was worse than the male talkers for the same
47
Chapter 3 Literature Review
network conditions. Palomar et al (2008) assessed the quality o f audio streaming over
WLAN using EAQUAL which is a software tool based on ITU-R for objective
assessment. Nevertheless, the experimental results showed that the EAQUAL was not a
good approach to assess the audio quality in cases like the effect of packet loss.
Subjective approaches on the other hand measure the overall multimedia quality based
on human subjects (ITU-T, 2008) and (Mohammed et al, 2001). Subjective approaches
are carried out by having n human subjects viewing the distorted multimedia signals and
then rate their quality based on predefined scale. The most common scalar used by
subjective approaches is Mean Opinion Score (MOS) (ITU-T, 1998) (Patel et al, 2003).
Several studies have used subjective approaches to evaluate network performance. For
instance, Brauer et al (2008) assessed Voice and Video over IP (VVoIP) quality in IP
networks. They used Mean Opinion Score (MOS) Absolute Category Rating (ACR)
scalar to obtain the quality of VVoIP under certain conditions. Teyeb et al (2006)
evaluated the performance of heterogeneous networks in subjective manner. In their
study, the QoS of web browsing and video streaming services was subjectively
evaluated through usability test. The aim was to find out the effect of network
parameters on users' perception of quality of web browsing and video streaming.
48
Chapter 3 Literature Review
each LSP. (Ma et al, 2003) and (Ma et al, 2004) used adaptive sampling and active
measurement to evaluate the performance of voice traffic in Multiprotocol Label
Switching MPLS-based IP networks.
There are several studies based on passive approaches to quantify QoS. For instance,
Cranley and Davis (2005) investigated the effect that the background traffic has on
video streaming traffic. Their approach non-intrusively measured and recorded the
bandwidth utilisation of video streaming traffic. Al-Sbou et al (2008) passively
evaluated network performance of multimedia applications in mobile ad hoc networks
(MANET). Their proposed system indicated that the measured QoS can be used as an
indication of the network conditions and resource availability.
However, there are some drawbacks with both passive and active measurements.
Performing active measurements that give representative results is not a trivial task,
because excessive probing generates a significant load that might disturb the operation
of the network, whereas infrequent probing might not reveal the performance
characteristics of the network. The disadvantage of passive measurement however is the
requirement of collecting and processing a large amount of recorded packets which is
not practically feasible in real time (Brekne et al, 2002).
The limitations of the above QoS measurement approaches highlight the need for
further investigation and development. The contribution of this study is to develop QoS
assessment techniques that overcome some drawbacks in subjective and objective
approaches. The techniques should evaluate QoS in a manner similar to human subjects
and quantify the QoS without the necessity for complex mathematical models taking
into the account the QoS requirements of each type of multimedia application. In
addition, the QoS assessment techniques should not add extra load to the network as the
case of active approaches, nor depend on the whole collected packets like passive
approaches. The proposed assessment technique will be based on the analysed traffic
generated from the proposed analysis techniques in order to overcome some drawbacks
of both active and passive measurement methods. An aspect o f this study is that a
49
Chapter 3 Literature Review
regression model is developed and Multi-Layer Perceptron (MLP) neural network was
trained in order to combine the QoS parameters (i.e. delay, jitter, and packet loss ratio)
for each QoS class (or cluster) identified by SOM or FCM to estimate the overall QoS.
This is because a single QoS parameter could not reflect an application’s transmission
requirements. For instant, delay, jitter, and packet loss ratio could all have significant
effects on VoIP quality (Al-Sbou, 2010).
Neural networks were also used in the area of network QoS. For instance, in (Nogueira
et al, 2006), a modelling approach based on neural network was proposed to predict the
media access delay in a Wireless Local Area Network (WLAN). MLP was used to
predict packet loss in a real-time video transmission. The results showed that MLP
could predict packet loss rate with accuracy of 96% (Lavington et al, 1999). Random
Neural Networks (RNNs) were devised to automatically quantify the quality o f video
flows. The achieved results correlated well with human perception methods (Mohamed,
2002). In (Radhakrishnan and Larijani, 2011), three non-intrusive RNNs models (simple
feed-forward model, Multilayer feed forward model, and recurrent architecture) were
employed to measure and monitor voice quality transmission. The results indicated that
the feed forward architecture produced the best results as compared with the other two
architectures.
50
Chapter 3 Literature Review
Networks
QoS improvement in this study involves the ability to take actions to improve QoS or
change network performance toward desired situation. Improving QoS has become
more pronounced in particular with the presence of multimedia applications and
integration of wired and wireless networks.
In many instances, users in WLAN exchange and access multimedia applications with
users in wired networks. Figure 3-2 shows how voice calls can take place between a
user connected to wireless network and user connected to wired network. Considering
that multimedia applications which are sensitive to transmission parameters, it is vital to
improve and sustain QoS in the integrated networks in order to enable the effective
delivery of multimedia services.
Voice
station
The integration of wired and wireless networks poses challenges to overcome QoS
issues in both network sides: wired and wireless. This is because the QoS issues are
different between wired and wireless networks. The QoS support in WLAN is enabled
at MAC layer whereas the most wired networks enable QoS at the IP layer (Senkindu
and Chan, 2008). Therefore, providing and improving network QoS for traffic being
exchanged between two different environments is considered to be a challenging task.
In terms of wireless networks, the demand for supporting QoS for various applications
has led to the development of a WLAN standard so called IEEE 802.1 le. This standard
provides two schemes to access the wireless channel: Enhanced Distributed Channel
Access (EDCA) and Hybrid Coordination Function (HCF) Controlled Channel Access
51
Chapter 3 Literature Review
In contrast with the earlier legacy IEEE 802.11 DCF access method, IEEE 802.l i e
EDCA differentiates traffic according to their QoS requirements as reported in (Liang et
al, 2006), (Rauf et al, 2009), (Alahmadi et al, 2008), and (Peng et al, 2010). For
instance, in (Liang et al, 2006), stations operating the IEEE 802.1 le EDCA were
compared with IEEE 802.1 lb for transmitting a number of applications. Results showed
that IEEE 802.1 le EDCA mechanism provided a better QoS as compared with IEEE
802.11b DCF. An evaluation reported in (Rauf et al, 2009) showed that IEEE 802.l i e
EDCA introduced an effective service differentiation mechanism and provided QoS
support under light network load. Peng et al (2010) presented an analytical model to
study the performance of EDCA service differentiation schemes. The general
effectiveness of EDCA service differentiation was proved through analytical and
simulation results.
52
Chapter 3 Literature Review
Most previous studies either supported QoS provided by IEEE 802.11 EDCA for high
priority traffic, which may starve other transmitted traffic as in (Politis et al, 2011), or
required modification of all wireless stations, which in turn complicates the WLAN
operation as in (Lin et al, 2009). The contribution of this study is the development of an
adaptive allocation algorithm at the wireless Access Point (AP) to further improve the
QoS of IEEE 802.1 le EDCA. The algorithm determines the Packet Arrival Rate (PAR)
of the uplink and downlink traffic for each Access Category (AC). The algorithm then
dynamically allocates the traffic from a lower priority AC to the next higher AC, when
there is no traffic for the higher AC. The algorithm enables lower priority traffic to have
access to a higher priority AC ensuring more efficient use of network resources.
Most current wired networks which are based on IEEE 802.3 have low bit error rates in
transmitting multimedia applications as compared with wireless networks. However,
one of the most challenging tasks that affects the QoS in wired is traffic congestion.
Congestion occurs when multiple stations transmit traffic simultaneously using the
same router. Congestion has a negative impact on the network performance as increases
the probability of packet loss ratio and collision.
Several mechanisms and approaches were used to overcome the QoS issues and
enhance network performance (Farrel, 2008). One of these approaches is over
provisioning which refers to enhancing the network capability by simply providing the
network with enough bandwidth in order for all traffic to meet their QoS requirements
(Fraleigh et al, 2003). Other approaches such as traffic class, resource reservation, and
queuing mechanism utilise available network resources according to the application’s
QoS (Farrel, 2008).
53
Chapter 3 Literature Review
Several studies have used packet scheduling mechanisms to improve network QoS. For
instance, Frantti and Jutila (2009) proposed Adaptive Weighted Fair Queuing (AWFQ)
to differentiate service for traffic according to QoS requirements. The study showed that
AWFQ achieved improved results than traditional WFQ. Epiphaniou et al (2010)
discussed the performance of three different mechanisms: FIFO, RED and DiffServ, and
their effects on real-time voice traffic. The experimentation results proved that under
burst traffic conditions up to a congestion level, DiffServ seems to perform better on all
three categories examined, by mainly providing a better queue management and
throughput, reduced packet drop rates based on data transmitted, and One Way Delay
(OWD) within the acceptable levels for an acceptable voice call. Miaji and Hassan
(2010) investigated the performance of three scheduling mechanisms: First Come-First-
Serve, Priority Queuing, and Weighted Fair Queuing. According the their simulation
results, WFQ provided a better enhancement for multimedia applications and hence a
higher QoS. Balogh and Medvecky (2011) carried out a comparison between Weighted
Fair Queuing (WFQ), Worst-case Fair Weighted Fair Queuing+ (WF2Q+) and
Weighted Round Robin (WRR) from the point of their usage in modem converged
telecommunication networks. Simulation results showed that WFQ and WF2Q+ provide
fair distribution of bandwidth for variable length packets due to the calculation of
packet size at cost of high computing and memory usage requirements which limited
their usages in high speed backbone network. On the other hand, WRR was a quick
algorithm with low computing requirements which allows its usage in high speed
backbone networks though WRR does not take into the account the length o f packet
size.
From the above discussion, the trade-off between different queuing mechanisms is their
complexity, control ability and level of fairness. An aspect of this study is an
appropriate utilisation of a suitable queuing scheduling mechanism. FIFO does not
support QoS because it treats traffic equally. PQ provides premium service to the high
priority traffic at the expense of the lower priority traffic, causing the latter traffic to
experience excessive delay. FQ is not designed to support traffic with different QoS
54
Chapter 3 Literature Review
requirements, as it allocates the same amount of bandwidth among multiple traffic. The
computational complexity of WFQ algorithm affects its scalability to support larger
traffic with different requirements at the edge of the network (Semeria, 2001),
(Epiphaniou et al, 2010), and (Balogh and Medvecky, 2011).
WRR addresses the limitations of FIFO, PQ, and FQ by classifying traffic based on
their QoS requirements, and ensuring that low priority traffic can access to buffer space
and output port bandwidth. The implementation of WRR is more popular and its
operation is less complex as compared with WFQ. Therefore, in this study, WRR was
considered to provide traffic prioritisation because of its practicality and low
complexity.
55
Chapter 3 Literature Review
However, the existing QoS monitoring tools have some limitations. For instance, they
are not determine directly the overall network QoS as in (Graham et al, 1998) and
(Malhotra et al, 2011). Network managers have to do a variety of operations to assess
the overall network QoS. Also, these tools were not developed to work as stand-alone
device as in (Zseby and Scheiner, 2004) and (Carvalho et al, 2009). From these
limitations, it can be concluded that the process of monitoring QoS can be complicated,
expensive, and time consuming. Therefore, developing a portable hand-held device that
accurately determines the overall network QoS for multimedia applications can be very
valuable. In this study, a mechanism that assesses QoS which in turn taking into the
account the QoS requirements of multimedia applications is implemented on a portable
microprocessor board to build QoS monitoring tool. The proposed tool could work on
its own to assess the QoS of multimedia applications based on their QoS requirements.
Artificial intelligence techniques on the other hand were used widely in the area of
computer networks. For instance, Khoukhi and Cherkaoui (2008) proposed a fuzzy
logic system called FuzzyMARS for call admission control and service differentiation
56
Chapter 3 Literature Review
for wireless ad hoc networks. Their results showed that FuzzyMARS achieved service
differentiation delivery and reduced delay transmission of multimedia applications.
Frantti and Jutila (2009) embedded fuzzy expert system for Adaptive Weighted Fair
Queuing (AWFQ) to differentiate service for traffic according to QoS requirements. The
simulation results showed that AWFQ reacted faster to differentiate traffic classes than
traditional WFQ. Fuzzy logic was used to assess QoS and adjust the minimum
contention window ( CWmin) in mobile ad hoc networks (MANET) (Seriareh et al,
2008) and (Al-Sbou, 2010). The devised approach effectively assessed and improved
the QoS of multimedia applications in MANET.
Several studies have used FCM clustering in network traffic domain. For instance, a
network administrator assistance system was proposed based on FCM (Chen et al,
2009). The proposed system utilised a FCM method to analyse users’ network
behaviours and traffic-load patterns based on the measured traffic data o f an IP network.
Analysis results can be used to assist administrators to determine efficient controlling
and configuring parameters of the network management. In wireless sensor networks, a
FCM algorithm was used in order to create clusters which reduced the spatial distance
between sensors nodes (Hoang et al, 2010). A FCM clustering algorithm was also
developed to detect routing attacks caused by abnormal flows in a wireless sensor
network. The study demonstrated that FCM can be a valuable tool for intrusion
detection (Wang et al, 2009).
Neural network has been also used in the area of computer networks. In (Nogueira et al,
2006), a modelling approach based on a neural network was proposed to predict the
media access delay in WLAN. The prediction of the model was accurate even when the
number of active nodes was changed significantly. Multilayer perceptron MLP was
used to predict packet loss of real-time video transmission. The results showed that
MLP was capable of predicting the number of lost packets with 96% accuracy
(Lavington et al, 1999). Self-organising map was used to cluster network traffic types
and produce application profiles, which contained significant information about the
current status of the network, in order to manage network resources (Timo et al, 2002)
and (Kiziloren, and Germen, 2007). A network traffic classification approach based on
Kemel-SOM (KSOM) was proposed in (Ting et al, 2010). The experimental results
showed that the Kohonen SOM achieved high classification accuracy and successfully
categorised network traffic characteristics.
57
Chapter 3 Literature Review
3.7 Summary
The main objective in this chapter was to provide an extensive literature review of
previous studies in the area of managing QoS of multimedia computer networks. The
main QoS management issues of multimedia networks that need further development
and investigations were discussed. Firstly, the state of the art of sampling techniques for
multimedia traffic was reviewed. Then, the current QoS analysis and assessment
techniques used to evaluate network QoS were reviewed. The relevant studies that
considered the QoS support in wireless and wired networks were discussed. The exiting
monitoring tools used to assess the network performance were critically analysed. The
applications of statistical and artificial intelligent techniques used within this study into
the field of computer network management were reviewed.
The next chapter focuses on a description of the experimental approaches that were
carried out to evaluate and validate the proposed techniques involved in the process of
managing QoS.
58
Chapter 4 Experimental Methodology
4.1 Introduction
An explanation of network evaluation methodologies, tools, and general experimental
procedure used throughout this study is provided in this chapter. A more detailed
discussion of the procedures which are specific to individual studies are discussed later
in the relevant chapters. Section 4.2 of this chapter covers an overview of network
evaluation approaches, network simulation tools, description of simulation environment
and protocols, and traffic type and characteristics. The measurement processes which
include a description of QoS metrics and requirements, and analysis procedure of
simulation output are included in section 4.3. The main issues are summarised in section
4.4.
59
Chapter 4 E x p e rim e n ta l M e tn o ao io g y
NS-2 is a discrete event simulator based on two object oriented languages: C++ with an
Object Tool Command Language (OTcl) as shown in Figure 4-1. NS-2 has a rich set of
protocols such as TCP and UPD and traffic source behaviour such as FTP, Telnet, Web,
CBR and VBR (Chung and Claypool, 2004). NS-2 is also capable of transmitting video
streaming such as MPEG-4 and H.264 using Evalvid framework tool-set (Ke et al,
2008). Full details about the general architecture of the network components in NS-2
can be found in the documentation supported by NS-2 group (Fall and Varadhan, 2011).
In the simulation process of NS-2 as shown in Figure 4-1, users generate Tool
Command Language (TCL) script files to specify network topology, traffic applications,
and all the required settings. The TCL files are then handled by the C++ libraries and
OTcl interpreter. After the termination of simulation process, NS-2 produces simulation
results in two output files: NAM file and trace file. The NAM file is used to support
graphical tool called Network Animator (NAM) to visualize simulation traces. The trace
file contains information about packets (i.e. transmitted packets, received packets,
60
Chapter 4 experimental ivietnoaoiogy
dropped packet, packet types, packet ID, etc) (NS, 2012). NS-2 Users can extract
relevant information from the output trace file using script languages such as Awk, Perl,
or Grep and plot them using other script languages such as Xgraph, Gunplot, or Matlab
(Kumar, 2008).
Summary
Trace
files
OTcl interpreter
TCL Script
Simulation
File t NAM
C++ library
File
Simulation
r NAM Network
NS-2
Results Animator
Figure 4-1. The simulation process of NS-2.
Evalvid framework tool set can be integrated within NS-2 simulator to facilitate the
transmission of real video applications over simulated networks. The formats of the
video applications supported by Evalvid framework which can be later transmitted
using NS-2 are YUV QCIF (176 x 144) or YUV CIF (352 x 288) (Zhou and Sik-Jang,
2008). Figure 4-2 shows the component of Evalvid framework and NS-2 simulator used
to transmit real video applications. The process of transmitting a video application using
Evalvid framework and NS-2 simulator is as follows (Ke et al, 2008) and (Abdel-Hady
and Ward 2007):
i. Video Encoder: the encoder is used to convert the source video file from YUV
format to a compressed H.264 or MPEG4 format at the sender side.
ii. Video Trace Generator: this component reads the compressed H.264 or MPEG4
file generated by video encoder and then fragments each video frame into
smaller segments of 1000 bytes for transmission, if the size of the video frame is
larger than the preset maximum packet size (Maximum packet length is 1028
bytes, including 20 bytes for IP header and 8bytes for UDP header). Video trace
generator produces a video trace file that contains information about every frame
in the real video file.
iii. During the simulation process of NS-2, there are three agents implemented
between Evalvid framework and NS-2. These are MyTrafficTrace, MyUDP, and
61
Chapter 4 experimental Metnoaoiogy
MyUDPSink agent. MyTrafficTrace agent reads the frame type and the frame
size from the video trace file, and sends these frame segments to the transport
layer according to the preset time settings specified in the simulation script file.
At the transport layer, MyUDP which is an extension of UDP agent allows users
to specify the output file name of the sender trace file and then records the
timestamp of each transmitted packet, the packet ID, and the packet size. At the
transport layer of the receiver side, MyUDPSink agent receives the fragmented
video frame packets sent by MyUDP and then records the timestamp for each
received packet, packet ID, and packet size in the receiver trace file specified by
user.
iv. Video Reconstruction: after simulation is terminated, the three trace files (i.e.
video trace file, sender trace file, and receiver trace file) are reconstructed by the
video reconstruction component to produce the received video file in a
compressed H.264 or MPEG4 format.
YUV M PEG-4
NS-2 Environment
Sender trace
Simulated
^^^ender^^" Network - ^ ^Receiv^^
MPEG-4
Reconstruction 2. Sender trace file
3. Received trace file
Figure 4-2. The process of transmitting video using Evalvid framework and NS-2.
62
^napier * e x p e r im e n ta l ivietnoaoiogy
Wireless-cum-wired network topologies with different sizes (i.e. small, medium, and
large network) were simulated using the Network Simulator-2 (NS2) as shown in Figure
4-3. As the number of stations plays an important role in network performance, the
number of stations in the simulated network was varied from 8 to 64 stations according
to selected scenario. The connections between stations in wireless-cum-wired network
topology were unidirectional. The number of these connections was varied from 4 to 32
according to the number of transmitted stations. In most simulation scenarios, half of
the connections transmitted traffic from wireless to wired network, whereas the other
half transmitted traffic from wired to wireless. Each station transmitted one type of
traffic to its corresponding destination. At the wired side of network, all links had 5
Mbps bandwidth and 2 ms propagation delay. The WLAN side o f the network was
based on IEEE 802.1 le, and it used the Enhanced Distributed Channel Access (EDCA)
scheme. The main parameters that modelled the wireless channel were the default
settings for IEEE 802.1 le as shown in Table 4-1.
The network topology covered an area of 500m X 500m and the stations were
positioned randomly within the specified area. This position was fixed during the
simulation time. Simulations were repeated 10 times. Each time a different initial seed
value was used to randomly position the stations and manage which node transmitted
first, as all nodes were requested to transmit at a given time. The randomness introduced
using different seeds avoided the bias of random number generation. The results o f the
10 simulations were then averaged. Simulation time was between 300 - 500 seconds.
These settings were considered appropriate to examine the long term behaviour o f the
IEEE 802.l i e protocol.
63
Chapter 4 Jkxpem iieiiiai iviem uuuiugj
At the wireless side of the network, the main physical layer parameter considered was
channel bit rate which includes basic rate for control frame transmission and data rate
for data transmission. Basic rate was set at 1 Mbps, while data rate was set to be 2 Mbps
or 11 Mbps for some selected simulation scenarios. The physical layer was modelled to
work as Lucent WaveLAN at a frequency of 914 MHz and DSSS radio interface card
(Fall and Varadhan, 2011). A summary of PHY layer parameters of this model is listed
in Table 4-1 (NS, 2012).
In this research study, the MAC layer was based on IEEE 802.1 le at the wireless side of
the network. This standard provides two schemes to access the wireless channel. These
are: Enhanced Distributed Channel Access (EDCA) and Hybrid Controlled Channel
Access (HCCA) (IEEE, 2005). This study focused on the IEEE 802.1 le EDCA due to
its simplicity as compared with HCCA. The main parameters that modelled EDCA were
the default settings for IEEE 802.1 le. A summary of MAC parameters of this model is
listed in Table 4-1 (NS, 2012).
Table 4-1. Simulation settings of MAC and PHY parameters in IEEE 802.1 le.
Parameter Value
Capture Threshold 10
Carrier Sense Threshold 1.559e-ll
Receiving Threshold 3.652e-ll
Power Transmission 0.28183815
Frequency Band 914e+6 i
Data Rate 2 .0 -1 1 .0 Mbps
Basic Rate 1.0 Mbps
I Modulation Technique DSSS
PHY Header 24 bytes
MAC Header 28 bytes I
SlotTime 20psecs
SIFS 1Op secs
Preamble Length 144 bits
PLCP Header Length 48 bits
RST Threshold 3000
ShortRetryLimit 7
LongRetryLimit 4
64
l^napier 4 niA peniiiem ai ivicuiuuuiugj
In this study, the transmitted traffic over IEEE 802.l i e EDCA were VoIP, video, best
effort traffic, and background traffic. These traffics were mapped into different Access
Categories (ACs) to represent different levels of priority as shown in Table 4-2 (IEEE,
2005).
^ ^ T y p e of traffic
Best effort Background
VoIP Video
traffic traffic
Parameters
AIFS 2 2 3 7
c w min 7 15 31 31
c w max 15 31 1023 1023
TXOP 3.008 6.016 0 0
Classifying the traffic into different ACs was based on their QoS requirements. Due to
the high sensitivity of VoIP to QoS parameters, it was assigned to AC with the smallest
values of AIFS, CWmin, CWmax and largest value of TXOP. In contrast, background
traffic was assigned to AC with the largest values of AIFS, CWmin, CWmax and
smallest value of TXOP because of its tolerance to some QoS parameters such as delay.
Accordingly, VoIP had the highest priority, whereas the background traffic had the
lowest priority.
65
s im p le r 4 l-/ApVl
In the network, when multiple packets are serviced through a congestion point such as a
router, queuing mechanisms are required to determine the bandwidth allocation among
transmitted packets and the manner in which to service various applications with
different QoS requirements. In this study, two queuing scheduling mechanisms were
employed. These were First-In-First-Out (FIFO), and Weighted Round Robin (WRR).
Although no preference was given to the transmitted traffic regarding to its QoS
requirement in FIFO queuing scheme as its basis was first packet come first packet
served, FIFO was employed in most simulation scenarios due to its management
simplicity and implementation popularity. The default queue size of FIFO used in this
study was 50 packets. In some experimental scenarios in Chapter 7, W RR queue
scheduling mechanism was implemented between the router and the Access Point (AP)
at the wired side of the network to improve its QoS. WRR addresses the limitations of
FIFO by classifying traffic based on their QoS requirements, and ensuring that different
traffic priorities can access the buffer space and output port bandwidth (Semeria, 2001).
The operation of WRR is explained under Section 2.3.2 in Chapter 2. In this study,
time-sensitive applications had a higher priority than time-insensitive applications. This
was because the former had larger weights than the latter as shown in Table 4-3.
Preset No. of
Queue 1 Queue 2 Queue 3 Queue 4
queues in WRR
Video Best effort Background
Application type VoIP
streaming traffic traffic
WRR weights 3 3 2 2
Queue length 25 25 25 25
In this study, different types of traffic were transmitted over the simulated networks.
These were: Voice over IP (VoIP), videoconferencing, video streaming, best effort
traffic, and background traffic. Constant Bit Rate (CBR) traffic was adapted to model
VoIP, videoconferencing, and best effort traffic. The VoIP packet size was 160 bytes
and its inter-packet interval was 20 ms, corresponding to G .711 voice encoding scheme
with 64 kbps Pulse Code Modulation (PCM) voice flows. The packet size of the video
66
Chapter 4 C A J J C l IIU C IIK U lT lC U I U U U lU g J
traffic was 512 bytes and the inter-packet interval was 10 ms. This generated a packet
transmission rate of 384 kbps (Markopoulou et al, 2003) and (Saraireh et al, 2007).
The video streaming sources were YUV QCIF (176 x 144) Foreman (400 frame) and
YUV QCIF (176 X 144) Highway (2000 frame) (YUV QCIF, 2012). Prior to the
transmission, each video frame was fragmented into packets, which in turn had a
maximum length of 1024 bytes. Video streaming applications were encoded using
MPEG-4 encoding scheme which defines three types of video frames: I (Intra-coded)
frame, P (Predictive-coded) frame, and B (Bidirectionally predictive-coded) frame as
shown in Table 4-4.
Table 4-4. The number of video frames and packets of the video streaming sources.
The I frame is encoded and decoded independent of previous or successive frames. The
encoding of P frame requires information from preceding I or P frame in the video
sequence. The predictions from previous and successive I or P frames are also required
to encode the B frame. According to MPEG-4 scheme, the I frame is the most important
frame among other types of frame. Comparing P and B frames, MPEG-4 scheme
specifies that the former frame is more important than the latter (Lin et al, 2009).
During the decoding process, the video frames can be decompressed into Group Of
Pictures (GOP), which its pattern is defined by two parameters G (M, N), where N is the
I-to-I frame distance and M is the I-to-P frame distance as shown in Figure 4-4 (Zhou
and Sik-Jang, 2008).
Q
B B B B
67
cnapier 4 UjAJIEI
packet size of 1000 bytes with 12.5 ms packet interval was to generate 500 kbps data
rate. File Transfer Protocol (FTP) application was also used as background traffic. FTP
was transmitted over TCP, whereas other traffics were transmitted using UDP transport
protocol.
Due to their importance in the process of QoS evaluation, the transmission requirements
of applications are considered during the analysis the simulation results. Throughout
this research, delay, jitter, and packet loss ratio are considered to be the main QoS
parameters. These parameters and the manner they were calculated are explained in the
following sections.
Table 4-5. QoS requirements for voice, video, and data as recommended by ITU group
(Zhai et al, 2005).
68
i^ n a p ie r * XifApCl Ilildliai lUCUIVUUlVgJ
This section explains the calculation method of QoS parameters (i.e. delay, jitter, and
packet loss ratio). After the TCL script file is simulated by NS-2, a detailed trace file is
generated which contains extensive information about transmitted traffic. This
information include: packet status (i.e. departed, arrived, and dropped), its timestamp,
packet ID, packet type, packet size, flow ID, sequence number, node ID, source and
destination addresses.
In this study, the QoS parameters were extracted from the data trace file using AWK
script language (Aho et al, 1988). Delay calculation process was associated with three
main fields: packet sent time, packet received time, and its unique ID. To calculate
delay, the sent time a packet was subtracted from the received time for the same packet.
Equation 4.1 illustrates how delay is calculated for particular packet:
D i= R i - Si (4.1)
Where Dt is the delay of the ith packet arrived, Rt and S* are the arrival and sending
Jitter was computed by calculating the absolute value of the difference between two
consecutive packets delays as shown in equation 4.2:
Where Ji is the absolute jitter of the ith packet, Dt is the delay of packet i, and is
the delay of the previous packet.
The percentage of packet loss ratio during certain time interval was calculated based on
the total number of received packets with respect to the total number o f transmitted
packets during that time interval as in equation 4.3:
Where PL\ is the loss ratio in percentage (%) during the ith interval, and £ Ri (t) and
2 Si (t) are the total number of received and transmitted packets with the ith interval
respectively.
When the values of QoS parameters were calculated, they were fed to MATLAB for
further analysis (MATLAB, 2012). This analysis included averaging the values of QoS
69
v^napier h UAJJCl 'BJ
parameter for every n consecutive packets. The averaged values were then normalised
and limited in order to ensure that all values had the same contribution in QoS
evaluation process.
4.4 Summary
This chapter described the experimental procedure used to evaluate and validate the
techniques proposed throughout this study. The use of NS-2 to simulate computer
network scenarios (wireless-cum-wired network) was discussed. The network settings
including routing protocols, queuing mechanisms, PHY, and MAC layer parameters
were also explained. The characteristics and the QoS requirements of applications
transmitted over the simulated network were clarified. The chapter concluded by
describing the calculation of QoS parameters using AWK script language and then
MATLAB for further qualitative analysis. Chapters 5, 6 , 7, and 8 rely on the outlined
experimental approach discussed in this chapter in order to test, validate, and evaluate
the proposed techniques to manage QoS of multimedia computer networks.
70
Chapter 5 Development and Evaluation of Adaptive
Statistical Sampling Techniques for Multimedia
Traffic
5.1 Introduction
The rapid growth of real-time applications transmitted over multimedia networks,
makes measurement of their traffic increasingly important. These measurements allow
Quality of Service (QoS) for the transmission of the applications to be assessed.
However, most real-time applications such as VoIP and video-conferencing generate an
extensive amount of traffic data. Analysing these data in real-time is computationally
intensive. Therefore, in order to reduce the amount of collected data and their
processing, appropriate sampling techniques are required.
In fixed rate sampling techniques, the number of data packets processed remains
unchanged even when traffic characteristics change. However, in adaptive sampling, the
number of packets sampled varies in accordance with traffic fluctuations. This ensures
appropriate amounts of data are processed.
In this chapter, statistical adaptive sampling techniques to adjust sampling rate based on
traffic's statistics were developed based on a linear adjustment approach, quarter
adjustment approach, and fuzzy inference system. A comparison of the devised methods
versus conventional sampling techniques (i.e. systematic sampling, stratified sampling,
and random sampling) was also carried out using a simulated computer network.
The organisation of this chapter is as follows: Section 5.2 reviews the state-of-the-art of
adaptive sampling approaches which are used to reduce network QoS parameters.
Section 5.3 includes a description of the proposed adaptive statistical sampling
approaches, implementation of conventional sampling techniques, calculation of QoS
parameters from sampled versions, methods of sampling analysis, and demonstrates the
simulation set up. The experimental results are discussed in section 5.4. The summary
of this chapter is provided in section 5.5.
71
i^napier d /\uitpuvc ouiusuuti omupimg icuiiuqu cs
Other adaptive sampling approaches were used for a variety of applications. These
applications were: a control of resources allocation for network performance reported
by (Graci'a et al, 2008), an estimate of traffic rate proposed by (Ma, et al., 2004), and
capturing Denial of Service (DoS) attack packet as in (Zhang, et al., 2007).
In our work, novel statistical adaptive sampling methods were developed based on
traffic's statistics (Dogman, et al., 2010a), (Dogman, et al., 2010b), and (Dogman, et al.,
2011). The methods adjusted the sampling interval by considering the traffic's statistics
between two consecutive sampled sections.
72
Chapter 5 /\uapuve oiauMicai sampling icum^ucs
• Pre- and post-sampling sections of the traffic: these traffic sections contained the
packets to be sampled. The number of packets in these sections was not changed
during the sampling operation.
• Inter-Sampling Section Interval (ISSI) of the traffic: the position of this traffic section
was between the pre- and post-sampling sections. The length of this section was
adaptively determined during the sampling process. A simple linear adjustment
mechanism, quarter adjustment mechanism, and fuzzy inference system (FIS) were
developed to adjust the length of ISSI. Section 5.3.2 demonstrates how the ISSI was
adjusted using these mechanisms.
• Threshold value p : this value was used to assess the variation between the statistics of
the pre- and post- sampling sections. The threshold value influenced when the ISSI
was increased or decreased.
A flow chart of the statistical sampling algorithm’s operation is provided in Figure 5-2.
The user is required to initialise the length of the pre- and post- sampling sections,
length of Inter-Sampling Section Interval (ISSI), and threshold value p . The lengths
pre- and post-sampling sections are the same and are not changed during the algorithm’s
73
Chapter 5 Adaptive statistical sampling lecumquca
operation. An increase in the length of pre- and post- sampling sections enlarges the
sample size. As the goal of the algorithm is to produce smallest sample size with the
highest accuracy, the user needs to select a suitable value for the length of the pre- and
post-sampling sections, through experimenting with a number of different values. The
initial length of ISSI was used as the current sample interval for the first iteration of the
sampling process. However, the initial length of ISSI was not critical during the first
iteration of sampling process because it will be adjusted in the following iterations.
The algorithm determined the statistics (i.e. mean, median and standard deviation) of
the QoS parameter (i.e. throughput) for the pre- and post- sampling sections and then
quantified their overall statistic using equation (5.1).
Where abs represents the absolute value, meanl and mean2 are the mean values o f pre-
and post- sampling sections respectively for the throughput of traffic being sampled,
medianl and median2 are the median values of pre- and post-sampling sections
respectively for the traffic throughput, and std l and std2 are the standard deviation
values of pre- and post-sampling sections respectively for the traffic throughput. The
overall statistic assessed the discrepancy of statistics between pre- and post-sampling
sections. The algorithm updates the new length of ISSI based on the comparison
between the quantified overall statistic and threshold value p using simple linear
The above operation represents one update of the ISSI length. During the next step, the
updated ISSI is the current sample interval, the current post-sampling section becomes
the next pre-sampling section, and the location of next post-sampling section is
determined based on the updated value of ISSI. This process is repeated until the traffic
is fully sampled.
74
V/iiapici j ounnutai odiupiuig lcuuiiqucs
CL Start algorithm
Quantify the overall statistic of the pre- and post-sampling sections using
Equation (5.1)
Quantify the new length of ISSI based on the comparison between the
quantified overall statistic and threshold value p using different adjustment
mechanisms:
1. Linear adjustment mechanism.
2. Quarter adjustment mechanism.
3. Fuzzy inference system.
No
Is traffic fully
sa m n led ?
Yes
CL End of aleorithm
Figure 5-2. The flow chart of the adaptive statistical sampling algorithm.
Three adjustment mechanisms were proposed and incorporated into statistical sampling
algorithm to adaptively adjust the length of Inter-Sampling Section Interval (ISSI).
75
Chapter 5 Aaapuve aiausucai sampling leumiqucs
This scheme examines the overall statistic along with predefined threshold value p and
then linearly increases or decreases the length of ISSI. If the overall statistic value is
less than the threshold, then the length of ISSI is updated using equation (5.2).
Updated ISSI = current sample interval + //, (5.2)
If the overall statistics value is more than or equal to the threshold value, then the ISSI
is updated using equation (5.3).
Updated ISSI = current sample interval - p 2 (5.3)
The terms//, and p 2in equations (5.2) and (5.3) control the update magnitude. The initial
value for both p xand ju2 was 1. During the sampling operation, a further increase to the
value of //, and p 2 is applied which in turn increases or decreases the length of ISSI
linearly. However, in case of the overall statistic value is equal or more than threshold
value p , the value of p 2 was less than the length of current sample interval, at least by
one.
In this approach, the calculated overall statistic was compared with the user specified
threshold value p . If the overall statistic value is less than the threshold, then the Inter-
p was determined though experimenting with different values. Small value o f p may
significantly change the length of ISSI, whereas as larger values o f p make finer
changes when ISSI length was updated. As the goal of the sampling algorithm is to
produce smallest sample size with the highest accuracy, the user needs to select a
76
A u a p u v c o u i u a u u t i o a iiip iiu g ic u u i^ u ra
study, the value of // was determined though experimenting with different values and a
value of 4 was chosen for this parameter. Therefore, the amount of ISSI change was a
quarter of its previous length.
In this approach, the Mamdani type of Fuzzy Inference System (FIS) was used to adjust
the length of ISSI. Two inputs were fed into the FIS: the length of current sample
interval and the overall statistic, which was used to measure the discrepancy of statistics
between pre- and post-sampling sections. The overall statistic was calculated using
equation (5.1).
Each fuzzy input variable was represented by five fuzzy sets to create input membership
functions. The amount of overlap and the range of each variable were determined by
experimenting with a number of suitable values and selecting the ones that gave best
outcomes. The locations, the degree of overlap between the generated membership
functions, and their corresponding fuzzy linguistics variables are shown in Figure 5-3
(a).
The fuzzy input variables were used to produce a single fuzzy output called Sample
Interval Difference (SID) which was then used to determine whether ISSI should be
increased or decreased or remain unchanged. The fuzzy output variable was also
partitioned into five membership functions as shown in Figure 5-3 (b).
The fuzzy inputs and the output were fuzzified using the Gaussian membership
function. This membership function is smooth and has concise notation. The
mathematical formula of Gaussian membership function is expressed in equation (5.6):
where ct and cr* are the mean and standard deviation of the ith fuzzy set A1, respectively
(Saraireh et al., 2007).
Tables (5.1) - (5.3) show respectively the values of membership function parameters for
fuzzy inputs (i.e. the length of current sample interval and the overall statistic) and
fuzzy output (i.e. Sample Interval Difference SID).
77
C hapter 5 A daptive sta tistica l sa m p lin g 1 ecm uques
0.005 0.01 0.015 0.02 0.025 0.03 0.035 0.04 0.045 0.05 350
Corrent sample interval
Threshold values
(b)
Figure 5-3. Graphs of FIS for adjusting ISSI: (a) Fuzzy inputs, (b) Fuzzy output.
Table 5-1. Mean and standard Table 5-2. Mean and standard deviation
deviation of overall statistic input fuzzy of current sample interval input fuzzy
_______ membership functions.________ membership functions.________
Current sample
Membership Overall statistics Membership
interval
functions functions
Mean St. dev. Mean St. dev.
Very low 0 0.005 Very small 0 53.09
Low 0.012 0.005 Small 125 53.09
Medium 0.025 0.005 Medium 250 53.09
High 0.037 0.005 Large 375 53.09
Very high 0.05 0.005 Very large 500 53.09
Table 5-3. Mean and standard deviation of sample interval difference output fuzzy
membershi p functions.
Sample interval difference
Membership functions
Mean St. dev.
Decrease low (DL) -100 21.24
Decrease High (DH) -50 21.24
No change (NC) 0 21.24
Increase low (IL) 50 21.24
Increase high (IH) 100 21.24
78
Uftapter 5 g-vuajjiive cjuimuLai aam p m ig k u u u iju m
The relationship between the inputs and the output was defined by a set of fuzzy rules.
The number of fuzzy rules was set according to the number of inputs and their
associated fuzzy sets. Examples of the rules are provides in Table 5-4.
Fuzzy reasoning (i.e. the process of implication and then aggregation) is based on
(minimum-maximum) inference method. Each rule is applied to the corresponding
membership function and the minimum is mapped into associated output membership
function. The output fuzzy set from the implication process for each rule is combined
together via aggregation process to produce one fuzzy set. In this study, the FIS output
was generated from aggregated fuzzy set (i.e. defuzzification) using the centriod
scheme. The centriod method returns the centre of area under the curve of the
aggregated output values using equation (Al-Sbou et al, 2006) (5.7).
r = I ly M J L n , ^
79
Chapter 5 AUitJJUVC I3UIUSI1UII o a i l l J J l l u g ic v iu m ju v j
where m is the number of fuzzy sets obtained after implication, y t is the centriod of
fuzzy region i, and p* is the output membership value.
The adaptive sampling algorithm used the generated fuzzy output (i.e. Sample Interval
Difference SID) along with the current sample interval to update the length of ISSI.
Equation (5.8) is used to calculate the new length of ISSI.
Updated ISSI = round ((SID/current sample interval)* 100+current sample interval)
(5.8)
In this study, conventional sampling techniques (i.e. systematic, stratified, and random
sampling) were implemented using the count based approach where the packet selection
decision was based on the packet count. This approach was chosen due to its simplicity
(Zseby, 2002). The implementations of conventional sampling techniques are as
follows:
i. Systematic sampling: in this approach, for every n packets, the nth packet is
selected. In implementation of systematic sampling a counter is set initially to n and
it is decreasing it by 1 on receiving each packet. When the counter is zero, the
packet is selected. This operation represents one packet selection. During the next
step, the counter is reset and the process is repeated. For several experimental runs,
the starting point of packet selection is chosen randomly to get different sets of
samples for the same size.
ii. Random sampling: in this approach, for a sample size of n taken from a population
of N, n random numbers need to be generated for a range 1 to N, and then the
packet selection is performed according to the position of the packets in the flow.
For each experiment, a new n random numbers should be generated in order to get
different sets of samples of the same size.
iii. Stratified sampling: is similar to the implementation of random sampling. For
every strata which has a size of N packets, random numbers n are generated in the
range [1, IV], and the packets are selected according to their position. For every run,
a new n random numbers are generated for the same sample size.
techniques. It also demonstrates the methods used to compare the sampled versions
versus original populations.
i. Throughput calculation: throughput was calculated by multiplying the number of
received packets (N) by the packet size (PS) and then dividing it by the difference
between receiving time of two successive sampled packets.
ii. Delay calculation: end-to-end delay was calculated by the difference of arrival and
sending times of packets which were selected during the sampling process.
iii. Jitter calculation: the difference between the delays of two consecutive sampled
packets for specific flow was used in order to calculate the jitter.
iv. Packet loss calculation: the percentage of loss ratio during the ith time interval was
calculated based on the total number of received and transmitted packets during that
th
time interval. The number of received packets during the i time interval was
obtained from the sampled version, whereas the number of sent packets was
computed from the difference between the sequence numbers of two successive
sampled packets.
After calculating the QoS parameters (i.e. throughput, delay, jitter, and packet loss ratio)
of the sampled versions, obtained using adaptive statistical sampling and non-adaptive
sampling approaches, a comparison between the QoS parameters of sampled versions
and the QoS parameters of original population was carried out. The aim of the
comparison was to determine which sampling approach could be used to produce
sampled version that effectively represented the whole population. The comparison was
carried out by calculating the mean, and standard deviation of the original population
and its sampled versions using adaptive and non-adaptive sampling techniques.
In order to compare the sampled version versus original populations, the mean and
standard deviation of the sampled version may not be sufficient to assess the accuracy
of sampled version in terms of representing the original population as they are affected
by the outliers (Brase, 2010). Therefore, additional criteria were used to assess the
discrepancy between the original population and its sampled version. These were:
i. Bias: the bias shows how far the mean of the sampled version lies from the mean of
the original traffic (Zseby, 2004). Bias is the averaged difference of all samples of
the same size. Equation (5.9) was used to calculate the bias:
81
^napier z> / i U i t p u v c o u i u d u u t i o a n i p i i i i g 1CU11U4UM
where N is the number of simulation runs, Mi and M are the means of the QoS
parameters for the sampled version and its original population respectively.
ii. Relative Standard Error (RSE): RSE measures the reliability of sampled version.
RSE is expressed as a percentage and can be defined as the standard error of the
sample (SE) divided by the sample size (n) as show in equation (5.10)
iii. Curve fitting (i.e. data trend): another criterion to examine the behaviour of
sampled version in terms of representing the original population is evaluating the
trend of sampled data versus its original counterpart by applying the curve fitting.
Curve fitting is a useful tool for representing a data set in a linear, quadratic or
polynomial fashion. Curve fitting could be based on two functions, polynomial
curve fitting function, and polynomial evaluation function which can quickly and
easily fit a polynomial to a set of data points. The general formula for a polynomial
is given in equation (5.11):
f ( x ) = a 0x N + + a 2x N~2 + ...... 4- a N- i x + aN (5.11)
The degree of a polynomial is equal to the largest value of the exponents (N), (x) is
a set of data, and (a) is a set of polynomial coefficients. Polynomial curve fitting
function computes a least squares polynomial for a given set o f data (x) and
generates the coefficients of the polynomial which can be used to simulate a curve
to fit the data according to the specified degree (N). Whereas, polynomial
evaluation function evaluates a polynomial for a given set of (jc) values and then
generates a curve to fit the data based on the coefficients found using curve fitting
function (Lindfield and Penny, 2012).
82
v^napier d ziuapuvc juiiiduuti oaiiipimg i^uuu\juvo
The traffic transmitted over the simulated network were: VoIP, video streaming, best
effort traffic, and background traffic. Constant Bit Rate (CBR) traffic was adapted to
model VoIP. VoIP was modelled as G.711 voice encoding scheme. The packet size of
VoIP was 160 bytes and the transmission rate was 64 kbps. The video streaming source
was YUV QCIF (176 X 144) Foreman (400 frame) (YUV QCIF, 2012). Prior to its
transmission, each video frame was fragmented into packets, which in turn had a
maximum length of 1024 bytes. The best-effort traffic had a fixed packet size of 1000
bytes and 125 kbps transmission rate. File Transfer Protocol (FTP) was used for the
background traffic. FTP was transmitted over TCP, whereas other traffics were
transmitted using the UDP transport protocol. The transmitted traffic over IEEE 802.1 le
EDCA (i.e. VoIP, video, best effort traffic, and background traffic) were mapped into
different access categories to represent different levels of priorities as shown in Table 4-
2. VoIP had the highest priority, whereas the background traffic had the lowest priority.
Each simulation lasted 500 seconds. Simulations were repeated 10 times for each
experiment. Each time, a different initial seed was used in order to randomly manage the
node’s transmission period as all the nodes were requested to transmit and stop at a given
time. The randomness introduced for different seeds avoided the bias of random number
generation.
5.4.1 Throughput
Figures 5-4 (a) - (g) show respectively the actual throughput with its sampled versions
using adaptive statistical sampling based on fuzzy inference system (FIS), adaptive
statistical sampling based on linear adjustment mechanism, adaptive statistical sampling
based on quarter adjustment mechanism, systematic sampling, stratified sampling, and
random sampling. The results obtained using adaptive statistical sampling approaches
were based on the initial parameters settings shown in Table 5-5. These initial
parameters settings were chosen experimentally, i.e. different values were tested to
monitor the response of the adaptive statistical sampling approaches and the most
suitable settings were chosen.
83
^ n a p ie r a n u ajiu y c k
3 uai.iai.ivai aaiiipnug i t m m iju w
I I 1 1 1 1 1 1 1
I I 1 1 1 1 1 1 1
1 1114 1 li ill 1 1 1 1 1 1 1 -L 1- 1 4
| I t U
'H I HI i | | in li 1 1 1 i i 1 T 1 1 T
y y k a | |
ipiii y pm
/ U T O lI " l i
f i i 1 1 1 1 1 1 1
.1___ ii___ ii___ i___ i___ 1i____ .1............
i----- 11----- . 1----- 11----------
_l---------- ----
0 50 100 ISO 200 250 300 350 400 450 500
Simulation Time (Sec)
Sampled throughput using adaptive linear sampling approach Sampled throughput using adaptive quarter sampling approach
“ Data trend Data trend
Sampled throughput using systematic sampling approach — Sampled throughput using stratified sampling approach 1
Data trend _________ Pd a trend _ _ _ _ _ _ J
1
. — 1--------- 1 -------- 1------------- i------------ --------- 1
l i l t l i
(e)
450
(g)
Figure 5-4. Comparison of throughput with its sampled versions using: (a) Actual, (b)
Adaptive sampling based on FIS, (c) Adaptive sampling linear adjustment, (d) Adaptive
sampling quarter adjustment, (e) Systematic, (f) Stratified, (g) Random sampling.
84
Chapter 5 /vuapuve oiauM icai aam pu ug i cuuuijura
In Figures 5-4 (b) - (g), the sample size for each sampled version was 240 packets (i.e.
sample fraction was 4.8 % of the actual traffic). The data trends shown in Figures 5-4
(a)-(g) were used to describe the behaviour of the observed sampled versions of
throughput and to illustrate the degree of discrepancy for each sampling method from
the original population (i.e. actual throughput).
It can be noticed from the trend of data in Figures 5-4 (a) - (g), how different sampling
approaches represent the actual throughput. As each sampling approach follows certain
procedure, the degree of discrepancy from the actual throughput is different for each
sampling method. However, it is observed that the sampled versions of throughput
obtained using adaptive statistical sampling approaches is closer to the actual
throughput as compared with the non-adaptive sampling approaches (i.e. systematic,
stratified, and random sampling). This is because the adaptive statistical sampling
approaches based on FIS, linear adjustment mechanism, and quarter adjustment
mechanism varied the length of Inter Sampling Section Interval (ISSI) according to the
statistical variations of throughput.
Figures 5-5 (a) - (c) show respectively the response of adaptive statistical sampling
approaches based on FIS, linear adjustment mechanism, and quarter adjustment
mechanism. It can be seen from Figures 5.5 (a) - (c) that during the sampling process of
adaptive statistical sampling approaches based on linear adjustment mechanism, quarter
adjustment mechanism, and FIS, whenever the calculated overall statistic of throughput
was less than pre-defined threshold values indicated in Table 5-5, the ISSI was
increased according to equation (5.2), (5.4), and (5.8) respectively. Otherwise, the ISSI
was decreased using equations (5.3), (5.5), and (5.8).
85
i^napier a ziuap uvc ou iusu i.ai oaiu pu ug i clihiiijucs
3fl 40 50
Iteration number
40 50 60 15 20 25
Iteration number Iteration number
(b) (c)
Figure 5-5. The length of ISSI for adaptive statistical sampling using: (a) Fuzzy
inference system, (b) Linear adjustment mechanism, (c) Quarter adjustment mechanism.
Tables 5-6 (a) - (f) summarise respectively the throughput measurement for different
sample fractions using adaptive statistical sampling based on (FIS), linear adjustment
mechanism, quarter adjustment mechanism, and non-adaptive sampling methods (i.e.
systematic, stratified, and random). It is established from Tables 5-6 (a) - (f) that the
variations of sampling versions using all sampling methods from actual mean and actual
standard deviation are increased as the sample size is decreased. The calculated absolute
error is also increased for all sampling methods as the sample fraction is decreased and
vice versa.
The summary statistics provided in Tables 5.6 (a) - (f) also indicate that sampled
versions of throughput for different sampling fractions obtained using the three adaptive
sampling approaches are closer to the original throughput as compared with the versions
obtained using non adaptive sampling approaches. This indicates that the proposed
adaptive sampling techniques outperform conventional sampling techniques.
86
v^napier 3 n u a p u v c o u tiu u ta i aaiiipiiug ic u im q u c a
Table 5-6. Throughput measurement results using different sampling methods: (a)
Adaptive sampling based on FIS, (b) Adaptive sampling based on linear adjustment, (c)
Adaptive sampling based on quarter adjustment, (d) Systematic, (e) Stratified, (f)
Random sampling.
__________________________________(a)__________________________________
Actual values Sample fraction (%)
Units: (kbps)
15.7 8.18 4.8 3.34
Mean throughput 68.35 68.37 68.45 67.57 69.33
Standard deviation 19.73 19.58 19.46 19.15 20.5
Absolute error 0.02 0.1 0.78 0.98
(b)
(c)
Units: (kbps) Actual values Sample fraction (%)
15.7 8.18 4.8 3.34
Mean throughput 68.35 67.74 67.59 69.37 67.15
Standard deviation 19.73 21.4 21.87 22.41 23.27
Absolute error 0.61 0.76 1.02 1.2
(d)
(e)
Units: (kbps) Actual values Sample fraction (%)
15.7 8.18 4.8 3.34
Mean throughput 68.35 69.1 67.5 67.04 66.17
Standard deviation 19.73 24.19 22.33 23.8 24.6
Absolute error 0.75 0.85 1.31 2.18
(f)
87
Chapter 5 AUajJUVC kJUtll»U(.ai OaiII|JlllIg Iiwuui|uvi}
The mean and standard deviation were not only the criteria used to access the accuracy
of throughput sampled version for representing the actual population. Bias and RSE
calculated using equations 5.9 and 5.10 respectively were also used to assess the
discrepancy between the original throughput and its sampled versions.
Figures 5-6 (a) - (c) illustrate respectively the comparison of the bias of sampled
throughput versions from the mean of actual throughput for different sample fractions
using conventional sampling approaches versus adaptive sampling based on FIS, linear
adjustment mechanism, and quarter adjustment mechanism. The results illustrate that
the bias was decreased and became closer to zero for all sampling approaches when the
sample size was increased. However, it can be observed from the figures that the
adaptive statistical sampling approaches have a lower bias as compared with systematic,
stratified, and random sampling techniques. For example, the bias of sampled
throughput for 8.18% sample fraction (i.e. sample size 409 packets) using adaptive
statistical sampling based on FIS was 0.1, whereas the bias values when using
systematic, stratified, and random sampling were 0.9, -0.85, and -0.97 respectively.
S flJ
(a)
1 "
* i 4 Original Throughput * 1 4 Original Throughput
i 1
+ •
• ■ Adaptive sampling using linear adjustment ■ Adaptive sampling using quarter adjustment
1
(b) (c)
Figure 5-6. Comparison of bias of sampled throughput obtained from non-adaptive
sampling with bias obtained using: (a) Adaptive sampling based on FIS, (b) Adaptive
sampling based on linear adjustment, (c) Adaptive sampling based on quarter
adjustment.
Chapter 5 A daptive sta tistica l sa m p lin g i eu im q u cs
RSE calculated from throughput sampled versions obtained from conventional sampling
approaches (i.e. systematic, stratified, and random) compared with the RSE obtained
using adaptive statistical approaches based on FIS, linear adjustment, and quarter
adjustment as shown in Figures 5-7 (a) - (c) respectively. It is established that the results
obtained from the three adaptive sampling approaches are more accurate as compared
with systematic, stratified, and random sampling. For example, the overall reductions of
RSE were respectively 13.46%, 16.67% and 16.67% when using adaptive sampling
based on FIS as compared with systematic, stratified, and random sampling.
1.2
A daptive sam pling using F IS
1.1 ISystematic sam pling
IStratified sam pling
IR andom sam pling
0.9
0.7
(a)
0.5
0.4
0.3
0.2
0.1
1 I I 1 1
(b) 1 I i 1 1
' I i 1 1
! i i 1 1
1 I 1 1
IIH i _ _ _ _ _ _ _ : ____
1 I 1 1
I I j
1
_L_ H I m l
4 6 8 10 12 14 16 18
Sam ple fraction (% )
1
l A daptive sam pling using q u a r te r ad ju stm e n t
I H System atic sam pling
i
H i Stratified sam pling
I
i H R andom sam pling
i i i
i i i
i i i
i i i
r I i
,i i i
(c) i i i
i i i
!i i i
i
nIII i
i
t i
i
i II 8 10 12
mi
Sam ple fraction (% )
Figure 5-7. RSE of sampled throughput using conventional sampling versus: (a)
Adaptive sampling based on FIS, (b) Adaptive sampling based on linear adjustment, (c)
Adaptive sampling based on quarter adjustment.
89
unapter s /lUUpUVC iJUlUaUMll Oailipmig aW1UU1|UVU
5.4.2 Delay
Delay was another QoS parameter measured from sampled versions obtained using
adaptive statistical sampling based on FIS, linear adjustment mechanism, and quarter
adjustment mechanism, along with non-adaptive sampling techniques (i.e. systematic,
stratified, and random sampling) as depicted in Figure 5-8 (b) - (g).
Figures 5-8 (a) - (g) show the comparison between the actual delay and its sampled
versions using the above mentioned sampling techniques respectively. The mean and
standard deviation of data trend for original delay were 33.5 msec and 7 msec
respectively, whereas the sampled versions o f delay obtained from the adaptive
sampling based on FIS, linear adjustment mechanism, and quarter adjustment
mechanism had the mean of 32.67 msec, 33.99 msec, and 32.33 msec and standard
deviation of 7.7 msec, 5.7 msec, and 7.5 msec respectively. However, the mean and
standard deviation of data trend for sampled delay using systematic, stratified, and
random sampling were (35.82 msec, 34.95 msec, and 35.85 msec) and (10.73 msec,
5.42 msec, and 5.3 msec) respectively. This indicates that sampled versions of delay
using adaptive statistical sampling approaches represented the original delay more
accurately and effectively.
Tables 5-7 (a) - (f) statically demonstrate how different sample fractions obtained using
adaptive and non-adaptive sampling approaches represent the actual delay. For all
sampling methods, as the sample size was increased, the variation of sampled mean,
standard deviation from the actual mean and standard deviation decreased accordingly.
This is because a large sample size includes more packets that in turn increase the
probability of obtaining more details from the actual delay. However, the delay sampled
versions obtained from the proposed adaptive sampling techniques represent the actual
delay more closely than the conventional sampling techniques. For example, the
absolute error of 8.18% sample fraction of systematic, stratified, and random sampling
was increased by 32.2%, 31.03%, and 33.33%, as compared with the absolute error
obtained using the adaptive sampling using linear adjustment mechanism.
90
cnapier 3 nua|Jiivc ju iiisu i.ai uaiii[iuiig iwun»|tavu
200 250 300 50 100 150 200 250 300 350 400 450 500
Simulation Time (set) Simulation time (set)
(b)
Sampled delay using adaptive linear sampling approach ~ Sampled delay using adaptive quarter sampling approach
"Data trend ■j 1--------- [--------- -j-. Data trend ______ ______ ______ _ _ _ _ _ _
— Sampled delay using systematic sampling approach Sampled delay using stratified sampling approach
_ _ 1 I____ Data trend Dati trend
o 80
400
(g)
Figure 5-8. Comparison of delay with its sampled versions using: (a) Actual, (b)
Adaptive sampling based on FIS, (c) Adaptive sampling linear adjustment, (d) Adaptive
sampling quarter adjustment, (e) Systematic, (f) Stratified, (g) Random sampling.
91
cnapter 5 Adaptive statistical sampling i ecnniques
Table 5-7. Delay measurement results using different sampling methods: (a) Adaptive
sampling based on FIS, (b) Adaptive sampling based on linear adjustment, (c) Adaptive
sampling based on quarter adjustment, (d) Systematic, (e) Stratified, (f) Random
sampling.
(a)
(b)
Units: (msec) Actual values Sample fraction (%)
15.7 8.18 4.8 3.34 !
Mean delay 33.5 33.98 34.3 34.48 32
Standard deviation 22.63 22.7 22.2 21.93 21.75
Absolute error 0.48 0.8 0.98 1.5
(c)
Units: (msec) Actual values Sample fraction (%)
15.7 8.18 4.8 3.34
Mean delay 33.5 32.85 34.45 34.94 35.26
Standard deviation 22.63 22.88 23.08 23.81 24.05
Absolute error 0.65 0.95 1.44 1.76
(d)
Units: (msec) Actual values Sample fraction (%)
15.7 8.18 4.8 3.34
Mean delay 33.5 32.64 34.68 35.24 31.57
Standard deviation 22.63 24.19 24.81 24.54 24.82
Absolute error 0.86 1.18 1.74 | 1.93
(e)
Units: (msec) Actual values Sample fraction (%)
15.7 8.18 4.8 3.34
Mean delay 33.5 34.46 34.66 35.03 31.73 |
Standard deviation 22.63 24.19 25.64 24.32 24.17
Absolute error 0.96 1.16 1.53 1.77
(f)
Units: (msec) Actual values Sample fraction (%)
15.7 8.18 4.8 3.34
Mean delay 33.5 32.52 34.7 35.85 35.88
Standard deviation 22.63 24.19 24.94 25.68 24.61 !
Absolute error 0.98 1.2 2.35 2.38
92
^napier s A daptive d ia u su ca i sa m p lin g i ecnm ques
Bias and RSE were also used to assess the accuracy of sampled delay. The bias for
adaptive sampling based on FIS, linear adjustment mechanism, quarter adjustment
mechanism were closer to zero as compared with the non-adaptive sampling approaches
as shown in Figures 5-9 (a) - (c) respectively. For example, the bias of 3.34% sample
fraction was reduced by 55.46%, 36.97%, and 26.05% in case of adaptive sampling
based on FIS, linear adjustment mechanism, quarter adjustment mechanism respectively
comparing with random sampling approach.
♦ Original Delay
-— 1 - - =-
•
Adaptive sampling using FIS
1
1
1
1
1
1
1
1
1
1
1
1
1
% Systematic sampling
1
1
1
1
j
^ Stratified sampling
Random sampling
4
-------- 1--------
____ 1 ____
-------- 1--------
1
im
1
L
1
1
1
1
1
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1
1
1
1
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i
i
i
i
i
i
i
r
i
i
i
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i i i i i
---------
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1 1 1 1 1
t i l l 1
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l ■ L lB i - — — 4 .
t i l l i
t i l l i
J L J L L
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Sample Traction (%)
(a)
------ 1 ■ |
• 1 f Original Delay 4 Original Ddai
i i
i i 1 Adaptive sampling using inear adjustment I Adaftii e sampling using quarter adjustnwnt
£ Systematic sampling i Systematic sampling
i i ★ Stratified sampling
* ♦ Stratified sampling
i i
i Randomsampling I Random sanpline ______
i i i
1
------------- _ _ « --------
----------- r
>1 i i
i i i
J 1 L
1 1 1
1 1 1
♦ it 7 i — ♦
i i i
i i i
i t r
i i i
i i i
i
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i i i
------- I - --------------
i i i
★ i i i
* i i i........ -
8
Sample fraction Irr)
18 8 II
Sample fraction IIt)
(b) (c)
Figure 5-9. Comparison of bias of sampled delay obtained from non-adaptive sampling
with bias obtained using: (a) Adaptive sampling based on FIS, (b) Adaptive sampling
based on linear adjustment, (c) Adaptive sampling based on quarter adjustment.
Relative Standard Error RSE as shown in Figures 5-10 (a) - (c) proves that adaptive
statistical approaches based on FIS, linear adjustment, and quarter adjustment
respectively outperformed conventional sampling approaches.
93
cnapier 3 A daptive a ia u su c a i sa m p lin g 1 ecnniques
Although the value of RSE decreased and become closer to zero for all sampling
methods when the sample size was increased, the RSE values obtained from the three
adaptive sampling approaches were less than RSE calculated from sampled delay using
systematic, stratified, and random sampling. The overall RSE was increased by 3.7%,
1.85% and 3.7% when using systematic, stratified, and random sampling respectively
compared with adaptive statistical sampling based on quarter adjustment.
1.2
A daptive sam pling using F IS
1.1 I System atic sam pling
0.8
0.7
S 0.6
(a) as
0.5
0.4
0.3
in
0.2
0.1
10 12
Sam ple fraction (% )
1 1
1 1 A daptive sam pling using lin e a r ad ju stm en t
1 H System atic sam pling
1
1
Hi Stratified sam pling
1 H R andom sam pling
r I n --------r
i i i i
- - -
I i i i
;1 i i i
L
(b) 1
1
i
i J
i i
i
;1 i i i
11 i i i
1 i i i
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r i n t
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i i i i
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10 12
Sam ple fractio n (% )
1 1
1 1 A daptive sam pling using q u a r te r ad ju stm en t
1 1 H S ystem atic sam pling
1 1 H S tratified sam pling
r i
i i H R andom sam pling
r r
i i i
i i i
(C) - - -
! i i 1
i 1
i i 1
i i 1
i i 1
i i 1
n i 1
i i 1
i
i 1
H i
i 1
i
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8 10 12
S am ple fractio n (% )
Figure 5-10. RSE of sampled delay using conventional sampling versus: (a) Adaptive
sampling based on FIS, (b) Adaptive sampling based on linear adjustment, (c) Adaptive
sampling based on quarter adjustment.
94
Chapter 5 Adaptive Statistical Sampling Techniques
5.4.3 Jitter
Jitter was another important QoS parameter to be measured from sampled versions
obtained using the adaptive and non-adaptive sampling approaches. The results shown
in Figures 5-11 (a) - (g) highlighted the potential of sampling the actual jitter using
different sampling approaches.
It can be perceived from Figures 5-11 (b) - (d) how adaptive statistical sampling based
on FIS, linear adjustment, and quarter adjustment represent the actual jitter shown in
Figure 5-11 (a). The statistic of data trend obtained from jitter sampled versions using
adaptive statistical sampling based on FIS, linear adjustment, and quarter adjustment
had respectively; 12.99 msec, 12.63 msec, and 12.34 msec for their means and 0.76
msec, 1.18 msec, and 1.57 msec for their standard deviations. These values were very
close to the means and standard deviations of the trend of actual jitter which were 12.92
msec, and 0.81 msec. However, the mean and standard deviation of data trend obtained
using the non-adaptive sampling techniques as shown in Figures 5-11 (e) - (g) were
respectively 13.19 msec, 2.14 msec for systematic, 13.27 msec, 3.17 msec for stratified,
and 13.7 msec, 1.81 msec for random sampling. This indicates that the three proposed
adaptive statistical sampling approaches provided results closer to the actual jitter as
compared with the non-adaptive sampling techniques. The statistic results of jitter for
different sample fractions using adaptive statistical sampling based on FIS, linear
adjustment, quarter adjustment approaches and conventional sampling techniques are
summarised in Tables 5-8 (a) - (f) respectively.
From the Tables 5-8 (a) - (f), the percentage of difference between the values of mean
and standard deviation for 3.34% sample fraction obtained from adaptive sampling
approaches and the actual mean and standard deviation were respectively 4.01%, 2.33%
for fuzzy adjustment approach, 7.45%, 5.5% for linear adjustment approach, and 3.37%,
10.53% for quarter adjustment approach. Whereas the percentage of difference between
the values of the actual mean and standard deviation, and the values of mean and
standard deviation obtained from conventional sampling techniques were respectively
8.24%, 16.56% for systematic sampling, 9.31%, 20.42%, for stratified sampling, and
11.57%, 29.01% for random sampling. This analysis verified that the proposed adaptive
statistical sampling with fuzzy adjustment, linear adjustment, and quarter adjustment
mechanisms generated sampled versions which are very close to the original jitter as
compared with non-adaptive sampling approaches.
95
Chapter 5 Adaptive statistical sampling 1 ecnmques
500
Simulation Time (sec) Simulation Time (see)
Sampled jitter using adaptive linear sampling approach Sampled jhter using adaptive quarter sampling approach
70-
500
(c) (d)
— bampted jitter using svstemanc sampling approacn Sampled jitter using stratified sampling approach
Data trend___________________________ — Data trend
50-
30-
500
Simulation time (set) Simulation time (set)
(e)
— Sampledjitter using random sampling approacb
Data trend _________________
(g)
Figure 5-11. Comparison of jitter with its sampled versions using: (a) Actual, (b)
Adaptive sampling based on FIS, (c) Adaptive sampling linear adjustment, (d) Adaptive
sampling quarter adjustment, (e) Systematic, (f) Stratified, (g) Random sampling.
96
Chapter s Adaptive statistical sam pling i ecmuqucs
Table 5-8. Jitter measurement results using different sampling methods: (a) Adaptive
sampling based on FIS, (b) Adaptive sampling based on linear adjustment, (c) Adaptive
sampling based on quarter adjustment, (d) Systematic, (e) Stratified, (f) Random
sampling.
__________________________________ (a)__________________________________
Units: (msc) Actual values Sample fraction (%)
15.7 8.18 4.8 3.34
Mean jitter 12.92 12.99 13.02 13.09 13.46
Standard deviation 7.56 7.57 7.47 7.67 7.74
Absolute error 0.07 0.1 0.17 0.54
(b)
Units: (msc) Actual values Sample fraction (%)
15.7 8.18 4.8 3.34
Mean jitter 12.92 13.1 13.18 13.42 13.96
Standard deviation 7.56 7.39 7.77 7.11 8
Absolute error 0.18 0.26 0.5 1.04
(c)
Units: (msc) Actual values Sample fraction (%)
15.7 8.18 4.8 3.34
Mean jitter 12.92 13.01 13.1 12.64 13.37
Standard deviation 7.56 7.9 8.01 8.23 8.45
Absolute error 0.09 0.18 0.28 0.45
(d)
(e)
Units: (msc) Actual values Sample fraction (%)
15.7 8.18 4.8 3.34
Mean jitter 12.92 12.7 13.23 12.22 11.82
Standard deviation 7.56 9.68 9.1 9.67 9.5
Absolute error 0.22 0.31 0.7 1.1
(f)
Units: (msc) Actual values Sample fraction (%)
15.7 8.18 4.8 3.34
Mean jitter 12.92 13.16 13.27 13.7 11.58
Standard deviation 7.56 10.01 8.72 10.04 10.65
Absolute error 0.24 0.35 0.78 1.34
97
Chapter 5 Adaptive statistical sampling i ecnniques
The bias values shown in Figures 5-12 (a) - (c) demonstrate that the bias obtained from
adaptive and non-adaptive sampling approaches decreased accordingly as the sample
size increased.
The bias values obtained from adaptive sampling based on fuzzy approach, linear
adjustment approach, and quarter adjustment approach were closer to zero as compared
with non-adaptive sampling approaches. For instance, the biases of sample size of 409
packets (i.e. 8.18% sample fraction) obtained using adaptive sampling were 0.1 for
fuzzy approach, 0.26 for linear adjustment approach, and 0.18 for quarter adjustment
approach whereas the biases obtained from the same sample size using conventional
sampling techniques were 0.29 for systematic sampling, 0.31 stratified sampling, and
0.35 for random sampling. This indicates that actual jitter could be represented
effectively using adaptive sampling approaches rather than conventional sampling
approaches.
ft Original Jitter
I Adaptive sampling using FIS
^ Systematic sampling
★ Stratified sampling
ft Random sampling_______
(a)
1 t Original Jitter
ui— ♦ Original JiUer
1 1 Adaptive sampling using linear adjustment I Adaptive sampling using quarter adjustment
* 1 ^ Systematic sampling ^ Systematic sampling
1 1 f Stratified sampling * Stratified sampling
1
ft Random sampling i Random sampling_ _ _ _ _ _ _ _ _ _
1 ft 1
1 1
------- - 4 --------------+ -------
1
t
ff
----------- +
- ~ . L - ♦ —
i
i * ! *
i I
i I
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i I
i I
“1
★ r I
i I
• i I
i i._
4 4 8 II 14 16 1
Sam ple fra c tio n I rt) Sample fraction (%)
(b) (c)
Figure 5-12. Comparison of bias of sampled jitter obtained from non-adaptive sampling
with bias obtained using: (a) Adaptive sampling based on FIS, (b) Adaptive sampling
based on linear adjustment, (c) Adaptive sampling based on quarter adjustment.
98
Chapter 5 Adaptive statistical sampling i ecnmques
Relative Standard Error RSE as shown in Figures 5-13 (a) - (c) also illustrated how the
sampling errors for all sample fractions obtained from jitter sampled versions using
adaptive statistical sampling approaches were closer to zero as compared with non-
adaptive sampling approaches.
0.5
A daptive sam pling using F IS
0.45 ! System atic sam pling
I Stratified sam pling
0.4 i R andom sam pling__________
0.35
0.3
| 0.25
(a)
0.2
0.15
0.1
0.05
1 J i Il
10 12
Sam ple fraction (% )
0.5 1
1 1
1 1 f c : A daptive sam pling using lin ear a d ju stm en t
0.45 - - - H S ystem atic sam pling
1 1
1 1 H S tratified sam pling
0.4 1 “1 R £ R andom sam pling
1 1 1 " i " ............. i
1 1 J 1 1
0.35 1 1 1 1
1
1 1 1 1 1
0.3
1 1 1 1 1
1 1 1 1 1
(b) gu s 1 1 1 1 1
. I_ 1 1 1 1
0.2 1 1 1 1 1
1 1 1 1 1
0.15
1 1 1 1 1
11
0.1 ------- - 11
1
11 l i j
0.05 1
1
_1_ II H l i b
10 12
Sam ple fraction (%)
0.5
Adaptive sam pling using q u a r te r ad ju stm en t
0.45 I S ystem atic sam pling
I S tratified sam pling
0.4 I R andom sam pling_________________________
0.35
0J ------- 4 ---
(c) u
£ 0 .2 5
0.2
0.15
0.1
0.05
0 10 12
Sam ple fraction (% )
Figure 5-13. RSE of sampled jitter using conventional sampling versus: (a) Adaptive
sampling based on FIS, (b) Adaptive sampling based on linear adjustment, (c) Adaptive
sampling based on quarter adjustment.
99
Chapter 5 A d a p tiv e S ta tistic a l S am p lin g T e c h n iq u e s
As an example of the accuracy of adaptive statistical sampling, the RSE value for 4.8%
sample fraction as shown Figure 5-13 (a) was reduced by 16.67%, 23.08%, and 25.93%
respectively when fuzzy adjustment approach was applied as compared with systematic,
stratified, and random sampling.
Figures 5-14 (a) - (g) illustrate the actual packet loss ratio in addition to the sampled
versions of loss ratio using adaptive and non-adaptive sampling approaches with a
sampling fraction of 3.34%. These figures show how adaptive and non-adaptive
sampling approaches tracked and sampled the actual traffic losses.
It can be observed from Figures 5-14 (a) - (g) that the accurate estimation of packet
losses were obtained using adaptive sampling techniques based on fuzzy adjustment
approach, linear adjustment approach, and quarter adjustment approach. The mean and
standard deviation of sampled packet losses obtained from adaptive sampling
techniques were 0.27, and 0.82 for fuzzy adjustment approach, 0.26, and 0.77 for linear
adjustment approach, and 0.38, 0.91 for quarter adjustment approach. These results
were very close and comparable with the mean and standard deviation of actual packet
loss ratio which were respectively 0.31, and 0.81. The data trends shown in Figures 5.14
(a) - (g) also demonstrate that sampled versions of packet loss ratio obtained using the
adaptive statistical sampling techniques with the three different approaches closely
represented the actual packet loss ratio as compared with non adaptive sampling
approaches.
Tables 5-9 (a) - (f) demonstrate how the actual packet loss ratio was represented by
different sample fractions obtained using adaptive and non adaptive sampling
approaches. For all sampling approaches, as the sample size was decreased, the
variation of sampled mean, standard deviation from the actual mean and actual standard
deviation increased accordingly. However, the packet loss ratio calculated from
sampled versions obtained using the proposed adaptive sampling techniques represented
the actual loss ratio better than the conventional sampling techniques. For example, the
absolute error of sample fraction of 4.8% was increased by 15%, 25%, and 60% in case
of systematic, stratified, and random sampling comparing with the absolute error
obtained using adaptive sampling based on quarter adjustment which was 0.2.
100
Chapter 5 Adaptive Statistical sampling 1 ecnniques
— Original packet loss ratio — Sampled packet loss ratio using adaptive fuzzy sampling approach
— Data trend — Data trend ___________________
200
Simulation rune (sec) Simulation Time (sec)
(a)
l — Sampled packet loss ratio using adaptive linear sampling approach — Sampled packet loss ratio using adaptive quarter sampling approach
I — Data trend t t t — Data trend_ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _
L _ _ J I I 1 ____ I I- ----------------- 5----- 1----- 1------ i----- 1------1------ i------ 1----- 1------ t------
_
I I I I I I I
I I I I I I I
I I I I I I I
150 200 250 300 350 400 450 500 0 50 100 150 200 250 300 350 400 450 500
Simulation Time (sec) Simulation Time (sec)
(c)
• I I I
Sampled packet loss ratio using systematic sampling approach 1 1 I Sampled packet loss ratio using stratified sampling approach
— Datatrend ____________________ 1 1 1 — Datatrend
J 1 ' L -t I
(g)
Figure 5-14. Packet loss ratio with its sampled versions using: (a) Actual, (b) Adaptive
sampling based on FIS, (c) Adaptive sampling linear adjustment, (d) Adaptive sampling
quarter adjustment, (e) Systematic, (f) Stratified, (g) Random sampling.
101
cnapter :> Adaptive siau su cai sam pling i ecnniques
Table 5-9. Packet loss ratio measurement results using different sampling methods: (a)
Adaptive sampling based on FIS, (b) Adaptive sampling based on linear adjustment, (c)
Adaptive sampling based on quarter adjustment, (d) Systematic, (e) Stratified, (f)
Random sampling.
(a)
Units: (%) Actual values Sample fraction (%)
15.7 8.18 4.8 3.34
Mean packet loss ratio 0.31 0.31 0.31 0.29 0.27
Standard deviation 0.81 0.8 0.82 0.8 0.82
Absolute error 0.005 0.01 0.07 0.11
(b)
Units: (%) Actual values Sample fraction (%)
15.7 8.18 4.8 3.34
Mean packet loss ratio 0.31 0.3 0.32 0.29 0.26
Standard deviation 0.81 0.8 0.83 0.76 0.77
Absolute error 0.02 0.03 0.07 0.16
(c)
Units: (%) Actual values Sample fraction (%)
15.7 8.18 4.8 3.34
Mean packet loss ratio 0.31 0.3 0.29 0.37 0.38
Standard deviation 0.81 0.81 0.76 0.9 0.91
Absolute error 0.015 0.05 0.2 0.21
(d)
Units: (%) Actual values Sample fraction (%)
15.7 8.18 4.8 3.34
Mean packet loss ratio 0.31 0.3 0.33 0.27 0.38
Standard deviation 0.81 0.8 0.83 0.75 0.95
Absolute error 0.018 0.07 0.23 0.23
(e)
Units: (%) Actual values Sample fraction (%)
15.7 8.18 4.8 3.34
Mean packet loss ratio 0.31 0.33 0.37 0.23 0.39
Standard deviation 0.81 0.83 0.82 0.63 0.93
Absolute error 0.08 0.18 0.25 0.26
(1D
Units: (%) Actual values Sample fraction (%)
15.7 8.18 4.8 3.34
Mean packet loss ratio 0.31 0.37 0.38 0.41 0.47
Standard deviation 0.81 0.88 0.9 0.96 0.99
Absolute error 0.21 0.23 0.32 0.54
102
Chapter 5 Adaptive statistical sampling i ecnniques
The accuracy of sampled packet loss ratio was also measured using bias as shown in
Figures 5-15. As shown in the Figures, the bias of sampled packet loss ratio using
adaptive sampling based on FIS, linear adjustment mechanism, quarter adjustment
mechanism were closer to zero for all sampling fractions as compared with the non-
adaptive sampling approaches. For example, the bias of 4.8% sample fraction was
reduced by 75%, 75%, and 25% in case of adaptive sampling based on FIS, linear
adjustment mechanism, quarter adjustment mechanism respectively comparing with
stratified sampling approach.
0 .2 5
* O rig in a l P a c k e t loss ratio
■ A d ap tiv e sam p lin g u sin g F IS
0.2 * Sy stem atic sam p lin g
* S tra tified sam p lin g
(a)
£ 0 .0 5
♦- - 1~
-0 .0 5
-0 . 1.
0 .2 5
* O rig in a l P a c k e t loss ratio
■ A d a p tiv e s a m p lin g u sin g lin e a r a d ju s tm e n t
0.2 * S y stem atic sa m p lin g
* S tra tified sa m p lin g
0 .1 5 * R a n d o m sa m p lin g _________ _______________
0.1
(b)
r
-0 .0 5
-0.1,
S am p le fra c tio n ( % )
0 .2 5
* O rig in a l P a c k e t loss ratio
■ A d ap tiv e s a m p lin g u sin g q u a rte r a d ju s tm e n t
0.2
* System atic sa m p lin g
* S tra tified sa m p lin g
0 .1 5 * R an d o m sam p lin g __________________________
0.1
(c)
a 0 .0 5
-
t
-0 .0 5
- 0 . 1.
Figure 5-15. Comparison of bias of sampled packet losses obtained from non-adaptive
sampling with bias obtained from adaptive sampling based on: (a) FIS, (b) linear
adjustment approach, (c) quarter adjustment approach.
103
Chapter 5 Adaptive Statistical sampling l ecnniques
The comparisons between RSE of sampled packet losses obtained using conventional
sampling approaches, and the RSE obtained using adaptive statistical approaches based
on FIS, linear and quarter adjustments are shown in Figure 5-16 (a) - (c) respectively.
The results obtained from the three adaptive sampling approaches were more accurate
as compared with systematic, stratified, and random sampling. The overall reductions of
RSE were respectively 10.5%, 15% and 22.7% when using adaptive sampling based on
FIS compared with systematic, stratified, and random sampling.
0.05 1 1
1 1 ■ j A daptive sam pling using F IS
0.045 H System atic sam pling
1 1
H I S tratified sam pling
1 1
0.04 1 1 ■ R andom sam pling
1 1
0.035
1 1
1 1
0.03 1 1
1 1
0.025 - - - 1-
(a) DC 1 1
L 1
0.02 i 1
l 1
0.015
I 1
L
0.01
0.005
:i
1
l; r
1
1 1
III
Ih III 1
0.
4 6 8 10 12 14 16 18
Sam ple fractio n (% )
0.05
i A daptive sam pling using lin e a r ad ju stm e n t
0.045 — u --------------- H System atic sam pling
I
I H i S tratified sam pling
0.04 I ■ R andom sam pling
I l I
0.035
i i i
i i i
0.03 i i I
i I I
(b) 2060-025 i I i
i I I
0.02 I i
I I
1
0.015
1
i J
0.01 i I
i i
|
0.005
! 1 1 II I m i
0.
10 12
Sam ple fractio n (% )
0.05
A daptive sam pling using q u a rte r ad ju stm en t
0.045 I S ystem atic sam pling
I S tratified sam pling
0.04 I R andom sam pling_________________________
0.035
0.03
0.025
(C) DC
0.02
0.015
0.01
0.005
0 llll
10 12
Sam ple fraction (%)
Figure 5-16. RSE of sampled packet loss ratio using conventional sampling versus: (a)
Adaptive sampling based on FIS, (b) Adaptive sampling based on linear adjustment, (c)
Adaptive sampling based on quarter adjustment.
104
tn ap ier a Adaptive diausucai sampling lecnmques
5.5 Summary
In this chapter, three adaptive statistical sampling techniques to adjust the sampling rate
of multimedia traffic were developed and evaluated. A novel aspect of the proposed
techniques was an adjustment of sampling rate based on traffic's statistics. The sampling
rate of the three devised sampling techniques was controlled using three different
mechanisms: a simple linear adjustment mechanism, a quarter adjustment mechanism,
and a fuzzy inference system. The proposed techniques decreased the sampling rate
when the statistics of the traffic did not significantly change and increased the sampling
rate when the statistics of the traffic significantly changed.
Conversely, the sampling rate of conventional sampling techniques was either constant
as in systematic sampling or changed randomly as in stratified and random sampling.
The fixed and random sampling rates may result in a significant discrepancy between
the actual data and its sampled version.
Advantages of the proposed adaptive statistical sampling techniques were the ease of
implementation and the quick response to traffic changes.
105
Chapter 6 Techniques to Evaluate Network Quality
of Service Using Statistical and Artificial Intelligence
6.1 Introduction
Evaluation of QoS is an important task in managing computer networks. This is
currently carried out by analysis or measurement techniques. Analysis techniques are
used to examine the characteristics of the traffic, whereas measurement techniques are
applied to determine how well the network treats an ongoing traffic. The contribution of
this study is to propose mechanisms that combine analysis and measurement techniques
to evaluate QoS in multimedia applications in an effective manner. In this chapter, two
innovative %QoS evaluation approaches are proposed. The first approach combined
Fuzzy C-Means (FCM) and regression model to analyse and assess QoS o f multimedia
applications in a simulated network, whereas the other analysed and assessed QoS in
multimedia applications using a combination of supervised and unsupervised neural
networks. The transmitted application’s QoS parameters were initially analysed either
by FCM clustering algorithm or by the unsupervised learning Kohonen neural network
(i.e. Self-Organising Maps (SOM)). The analysed QoS parameters were then used as
inputs to a regression model or supervised learning Multi-Layer Perceptron (MLP)
neural network in order to quantify the overall QoS. The proposed QoS evaluation
system provided information about the network’s QoS patterns and based on this
information, the overall network’s QoS was successfully quantified.
Section 6.2 of this chapter presents a discussion of the related studies. The proposed
FCM clustering algorithm, regression model, Kohonen neural network, and Multi-Layer
Perceptron (MLP) neural network in addition to the simulation of experiments and
traffic models are discussed in section 6.3. The findings of these studies are presented in
section 6.4. Finally, the summary of this chapter is provided in section 6.5.
106
c n a p te r o E v a lu a tio n 01 iNetworK Q u a lity 01 s e rv ic e
network traffic as reported in (Chen et al, 2009), (Timo et al, 2002), (Hoang et al, 2010),
(Wang et al, 2009), (Ting et al, 2010), and (Kiziloren and Germen, 2007) or by
measuring the network performance to determine how well the network treats the
ongoing traffic as reported in (Palomar et al, 2008), (Brauer et al, 2008), (Mishra and
Sharma, 2003), (Al-Sbou et al, 2008), (Brekne et al, 2002),. and (Mohammed et al,
2001). A novel aspect of this study is to propose an evaluation system that combines
analysis and measurement techniques to effectively evaluate network QoS. The first
contribution of this study is to analyse and classify network QoS parameters (i.e. delay,
jitter, and packet loss ratio) using, either Fuzzy C-means (FCM) or Self Organizing
Maps (SOM). Due to the ability of FCM and SOM to derive meaning from imprecise
values as reported in (Cirstea et al, 2002) and because of the natural characteristics of
network QoS parameters, where a single cluster could not be clearly identified, FCM
algorithm and SOM are suitable for QoS analysis.
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Chapter 6 Evaluation of INetworK quality 01 service
inputs to the FCM clustering algorithm or to the Kohonen network to be analysed and
organised into correlated groups. The regression model and MLP relied on the identified
groups of QoS parameters generated by FCM or Kohonen network to assess the overall
QoS. The following subsections explain how the aforementioned mechanisms were
developed to perform the QoS evaluation process.
< s>
<£>
Classes of QoS
parameters:
Class 1
Class 2
Class i
Calculate the overall QoS assessment for class 1, class2 ,.. and class i using
1.Regression Model
2. Multi-Layer Perceptron (MLP)
Outcomes:
Assessed QoS for class 1, class2 ... and class i
108
Chapter 6 Evaluation of Network Quality oi service
FCM was developed to partition the QoS parameters (i.e. delay, jitter, and packet loss
ratio) of critical real-time applications (i.e. VoIP and video applications) into clusters.
This classification provided an informative view about multimedia traffic behaviour and
consequently discovered valuable information from ongoing traffic.
The values of delay, jitter, and packet loss ratio were measured respectively using
equations (2.2), (2.4), and (2.6) which can be found in section 2.2.2, chapter 2. The
measured QoS parameters values were then represented by a matrix ( Q o S _ P ) to be used
as inputs to FCM algorithm as shown in equation (6.1)
D 1 J i P L R -l
D 2 J2 P L R 2
QoSP = (6 . 1)
P n Jn P L R n .
where Q o S _ P is the matrix of QoS parameters, DjJj, and P L R j , j = 1,2, ...,n are the
measured delay, jitter, and packet loss ratio respectively. In this study, FCM was
employed at predefined time interval. FCM operated on the matrix ( Q o S _ P ) and
minimised the FCM objective function given in equation (6.2) in order to partition
(Q o S _ P ) matrix into ( C ) clusters, generate membership matrix (U), and produce clusters
centres (V ).
c n
The exponent value for partition (i.e. m) controls the degree of fuzziness for the
membership of the clusters (Chen et al, 2009). This is commonly set to 2 as other values
either introduce insufficient or too much fuzziness.
During the clustering process as explained in section 2.4.3 (chapter 2), the elements of
U were updated using Equation (2.17), the clusters centres V = {v lf v 2, —, v c) were
calculated according to Equation (2.18), and the Euclidian distance Dfj between
D j,J j, and P L R j to the centre v t of cluster i were calculated using equation (2.19). The
clustering process was terminated when the maximum number of iteration was
performed or the objective function improvement between two consecutive iterations
was less than the minimum set amount of improvement.
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Chapter 6 Evaluation of Network Quality of Service
In this study, the maximum number of iterations was 200, and the minimum set amount
of improvement was 10'5. These parameters were chosen experimentally, i.e. different
values were experimented to monitor the FCM clustering response and the best values
were selected.
matrix U expressed as
The number of clusters C was chosen based on the Xie-Beni index cluster validity
method (Xie and Beni, 1991). The function of Xie-Beni index method is defined by
nmintjWvi-Vjf
In this study, the optimal number of clusters for VoIP and video traffic was three,
associated with small Xie-Beni index S=0.0002 for VoIP, and S=0.0004 for video
traffic. The FCM algorithm produced three clusters classifying the QoS parameters into
three categories: Low, Medium, and High. These clusters were represented by clusters
centres matrix V as expressed as
D ei J c i P L R ci
where Dci,Jci, PLRci , i = 1,2,3 are the cluster centres of delay, jitter, and packet loss
ratio respectively.
Kohonen neural network (i.e. Self Organising Map SOM) with size of 100 neurons (i.e.
a 10 by 10 neuron grid) as shown in Figure 6-2 was trained to partition the QoS
parameters of multimedia applications (i.e. VoIP and video) into their correlated groups.
The aim of partition using SOM was to provide information about multimedia traffics
behaviour and their inherent groups of QoS.
110
C h a p te r 6 ii valuation 01 ivetworK quality 01 service
OOOOOOOOOO X Dj Jj PLRj
OOOOOOOOOO
OOOOOOOOOO D 2 J2 P L R 2
OOOOOOOOOO
OOOOOOOOOO D i h P L R i
OOOOOOOOOO
Kohonen map A neighbourhood region
Prior to the training process, the values of QoS parameters for VoIP or video application
were pre-processed and arranged to form examples to train the Kohonen neural
network. The values of QoS parameters (delay, jitter, and packet loss ratio) were
measured using equations (2.2), (2.4), and (2.6). The training examples considered the
classification of QoS parameters of VoIP and video applications listed in Table 2-1 (see
chapter 2). The QoS parameters of VoIP (i.e. delay, jitter, and packet loss ratio) were
classified into: Low (i.e. delay < 1 5 0 ms, jitter < 1 ms, and packet loss ratio < 2%),
Medium (i.e. 150 < delay < 400 ms, 1 < jitter < 3 ms, and 2% < packet loss ratio < 4%),
and High (i.e. delay > 400 ms, jitter > 3 ms, and packet loss ratio > 4%), whereas the
QoS parameters of video application were classified into: Low (i.e. delay < 1 5 0 ms,
jitter < 10 ms, and packet loss ratio < 1%), Medium (i.e. 150 < delay < 400 ms, 10 <
jitter < 20 ms, and 1% < packet loss ratio < 2%), and High (i.e. delay > 400 ms, jitter >
20 ms, and packet loss ratio > 2%). After considering the classification o f QoS
parameters and during the arrangement phase, the QoS parameters (delay, jitter, and
packet loss ratio) were labelled to provide visual differentiation between the generated
QoS classes. Label (L) indicated Low range for delay, jitter, and packet loss ratio. Label
(M) indicated Medium range for delay, jitter, and packet loss ratio, whereas label (H)
indicated High range for delay, jitter, or packet loss ratio. The values of QoS parameters
and their labels were represented in a matrix notation as shown equation (6.6) in order
to train the Kohonen network.
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Chapter 6 E v a lu a tio n o t INetworK q u a l ity 01 s e rv ic e
D1 J-l PLRi Li
^ 2 J2 P L R 2 L>2
QoS p a r a m e t r e s = (6.6)
f i n Jn PLR n Ln .
where Di,Jit PLRi Lit i = 1,2,..., n were respectively the measured delay, jitter, packet
loss ratio, and their labels (i.e. L,M, or H).
During the training process, the ith input feature of delay, jitter, or packet loss ratio was
connected to j th neuron in Kohonen map. Each connection was associated with weight
Wij. The network learned by determining the Euclidean distance dj between input
patterns with N elements and the connection weights vtfy for j th neuron as explained in
Equation (2.23) (see chapter 2). The neuron where its associated weights provided the
smallest Euclidean distance to the input pattern was considered as the winning neuron.
The weights associated with the winning neuron were then updated according to the
Kohonen learning rule represented by Equation (2.24). This ensured that the winning
neuron's weights iteratively moved closer to the specific input pattern categoiy.
However, the weights associated with a number of neurons around the winning neuron
were also updated to a lesser extent. This allowed improved training.
In this study, the maximum number of training iterations was 1000. According to
(Vesanto et al, 1999), the number of training steps should be at least 10 times of the
number of neurons. The initial learning rate of learning Kohonen algorithm rj was set to
(0.5). This value was decreased inversely proportional to the number of iterations
(Haykin, 1999). This allowed the training to be course to start with and then it became
gradually finer. The analysis process of Kohonen network produced three distinct groups
that correlated with low, medium, and high QoS.
During the test phase, the trained Kohonen neural network was fed with other values of
QoS parameters in order to classify the QoS parameters into their correlated groups. The
groups produced by the Kohonen network provided information about the relationships
between different QoS parameters of transmitted VoIP or video traffics and subsequently
discovered relevant information about network operation.
Due to the relevance between the quality of multimedia applications and their QoS
parameters (delay, jitter, and packet loss ratio) as shown in Table 2-1 (see Chapter 2),
regression model was devised in this study to combine the QoS parameters of VoIP, and
C h a p te r 6 E v a lu a tio n o r INetworK q u a l ity or s e rv ic e
The regression model was developed by using the theory of regression analysis
discussed in Section 2.4.1, Chapter 2. The values of the independent variables
(xl t x2, x 3) in the regression model were represented by delay, jitter, and packet loss
ratio respectively, whereas the values of dependent variable (y) were represented by the
overall QoS. The regression expression was determined by considering the QoS
requirements listed in Table 2-1 in order to provide the outputs that reflect the overall
QoS. The QoS parameters shown in Table 2-1 were categorised as: Low, Medium, and
High. The overall QoS on the other hand was classified as Good, Average, and Poor
quality corresponding to the categories of QoS parameters. In this study, the overall
QoS spanned between (0-100%). QoS=0% represented the worst case of network
performance, whereas the best network performance was for QoS=100%.
The regression formula for VoIP application was calculated by arranging the values of
independent variables (i.e. delay, jitter, and packet loss ratio) and the values of
dependent variable (i.e. overall QoS) into matrices as follows: Low QoS parameters (i.e.
delay < 1 5 0 ms, jitter < 1 ms, and packet loss ratio < 2%) corresponded to good
overall QoS which ranged between (67-100%), medium QoS parameters (i.e. 150 <
delay < 400 ms, 1 < jitter < 3 ms, and 2% < packet loss ratio < 4%) corresponded to
average QoS (i.e.33% < QoS < 67%), whereas high QoS parameters (i.e. delay > 400
ms, jitter > 3 ms, and packet loss ratio > 4%) corresponded to poor QoS (i.e. QoS <
33%).
The regression formula for video application was calculated using the same mapping
used in VoIP application. However, good overall QoS which ranged between (67-
100%) corresponded to low value of delay < 150 ms, jitter < 10 ms, and packet loss
ratio < 1%), average QoS (i.e.33% < QoS < 67%) corresponded to medium QoS
parameters (i.e. 150 < delay < 400 ms, 10 < jitter < 20 ms, and 1% < packet loss ratio
< 2%), whereas high QoS parameters (i.e. delay > 400 ms, jitter > 20 ms, and packet
loss ratio > 2%) corresponded to poor QoS (i.e. QoS < 33%).
After the mapping process of QoS parameters to the overall QoS, the matrices of QoS
parameters and the overall QoS were then organised to form the regression formula as
expressed by
113
Chapter 6 e v a lu a tio n 01 JNeiworK q u a l ity 01 service
Q0 S 1 1 D± A PLR± rfcoi ei
QoS2 ~ 1 D2 J2 PLR2 bi e2
+ (6.7)
b2
jQoSn. Dn Jn P L R nm -b^. ? n.
where DiJi,PLRi,QoSi , i = 1,2, ...,n are delay, jitter, packet loss ratio, and overall
QoS of VoIP or video applications. The regression coefficients b0, b 1, b 2, b 3 were
determined from the recorded data using equation (2.12). The vector o f residual (i.e.
error terms) was then calculated using equation (2.13). In this study, the calculated
errors produced from regression formula for VoIP and video traffics were normally
distributed. This indicated that the mean of error terms ei f i = 1,2, ...,n was zero. This
implied that the estimated regression model determined was:
where Q0 S1 ,Di,Ji,PLRi t i = 1 ,2 ,...,n are the overall QoS, delay, jitter, packet loss
ratio for ithpacket respectively.
A Multi-Layer Perceptron (MLP) neural network was chosen to assess the overall QoS
due to its suitability and effectiveness (Abraham, 2005). The proposed MLP neural
network model in this study composed of an input layer with three neurons, a hidden
layer with three neurons, and an output layer with one neuron as shown in Figure 6-3.
The inputs A , and P L R i fed into the input layer of MLP were delay, jitter, and packet
loss ratio respectively, whereas the values of desired output (QoSi) were represented by
the overall QoS.
QoS]
Q 0 S2
Q o S i
w37 w67
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Chapter 6 Evaluation of Network Quality of Service
The inputs (i.e. delay, jitter, and packet loss ratio) and their desired output(i.e. QoS) were
arranged in a matrix by considering the QoS requirements listed in Table 2-1 in order to
provide outputs that reflected the overall QoS. The QoS parameters of VoIP and video
application shown in Table 2-1 were categorised into three classes: Low, Medium, and
High. The overall QoS on the other hand was classified as Good, Average, and Poor
quality corresponding to the categories of QoS parameters. The overall QoS spanned
between (0-100%).
Low QoS parameters of VoIP application (i.e. delay < 1 5 0 ms, jitter < 1 ms, and packet
loss ratio < 2%) corresponded to good overall QoS which ranged between (67-100%),
medium QoS parameters (i.e. 150 < delay < 400 ms, 1 < jitter < 3 ms, and 2% <
packet loss ratio < 4%) corresponded to average QoS (i.e.33% < QoS < 67%), and
high QoS parameters (i.e. delay > 400 ms, jitter > 3 ms, and packet loss ratio > 4%)
corresponded to poor QoS (i.e. QoS < 33%). Whereas in video application, good
overall QoS which ranged between (67%-100%) corresponded to low value of delay <
150 ms, jitter < 10 ms, and packet loss ratio < 1%), average QoS (i.e.33% < QoS <
67%) corresponded to medium QoS parameters (i.e. 150 < delay < 400 ms, 10 < jitter
< 20 ms, and 1% < packet loss ratio < 2%), and high QoS parameters (i.e. delay > 400
ms, jitter > 20 ms, and packet loss ratio > 2%) corresponded to poor QoS (i.e. QoS <
33%).
The matrix expressed in equation (6.9) which included training examples was fed into
MLP for its training. The matrix represented the QoS parameters and the overall QoS
where D iJ ifPLRi,QoSi , i = 1,2, ...,n were delay, jitter, packet loss ratio, and the
overall QoS respectively:
D± Ji PLRi QoSi
^2 J2 PLR2 Q0 S 2
Training examples = (6.9)
&n Jn PLRn QoSn.
During the training phase of MLP, the inputs (Di,Jit PLRi) were multiplied with their
associated weights (Wi) and the resulting values are summed by the summation function
using equation (2.21) (see Chapter 2). The output from summation function (5 ) was then
processed by the activation function to produce the output (y). In this study, the
hyperbolic tangent activation function was used as it gives continuous output between -
1 and +1 and thus it gave a larger range than sigmoid activation function which provides
115
Chapter 6 Evaluation of Network Quality of Service
a range between 0 and 1 (Karlik and Olgac, 2010). Equation (6.10) formulates the
hyperbolic tangent function:
(6. 10)
The calculated output (p(s) (i.e. y) was subtracted from the desired output (QoSi) as *n
equation (2.22) (See Chapter 2) to produce an error (e) which in turn was used by the
learning algorithm in order to reduce the magnitude of the error in the next iteration.
In this study, the commonly used a gradient descent with momentum learning algorithm
was employed. The function of this algorithm was to use the calculated error (e) and the
input data (DiJit PLRi) to the processing neuron to adjust the values of the connections'
weights (W/) which in turn reduced the magnitude of the error in the following training
iteration. Gradient descent with momentum learning algorithm updated the weights using
equation (6.11) (Eberhart and Dobbins, 1990):
where Wnew is the new updated weight, W0id is the previous weight, the term rj is the
learning rate parameter, (e) is the calculated error, x is the input to the processing
element, the momentum term ( a ) ensures that the learning algorithm does not get stuck
in a local minimum and finds the desired global minimum, and the term A W0id
represents the amount of change in the weights from the previous iteration.
The learning rate parameter rj is a positive constant limited to the range 0 < i} < 1
whereas the momentum factor a can take values between 0 and 1. In this study, the
learning rate parameter rj and the momentum factor a were set to their default values
which were 0.01 and 0.9 respectively. These values provided the most reliable results.
The training process of MLP was terminated when the maximum number of iterations
reached (in this study was 1000) or when there was insignificant error (i.e. 0.001)
between network output (y) and desired result (QoSi). During the test phase, the trained
MLP was fed with other values of QoS parameters (D*,/*, PLRi) in order to assess the
overall QoS. The actual output values used to train MLP and the MLP output values
during test phase were then correlated to ensure that MLP had been trained effectively.
The prediction accuracy of proposed QoS assessment methods was measured using a
correlation coefficient. The correlation coefficient (R) is widely used to evaluate the
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C h a p te r 6 E v a lu a tio n o t INetworK q u a lity 01 s e rv ic e
prediction accuracy (Chatteijee and Hadi, 2006). The magnitude of R is between 0 and
1. The magnitude closest to 1 indicates a perfect correlation, whereas a correlation less
than 0.5 would be described as weak correlation. The correlation between the actual
values (yA) and predicted values (y P) is calculated using equation (6.12).
n _ Z C y A i-W Cy pi -y p)
y A ' yp V S (y /ii-5M)2 Z ( y p i- y F )2
where y^is the actual value, Ya ls the mean of the actual values, y Pis the predicted
value, and y k is the mean of the predicted values.
The types of traffic transmitted over the simulated network were: VoIP, video streaming,
best effort and background. VoIP was modelled as G.711 voice encoding scheme by
adapting Constant Bit Rate (CBR) traffic. The packet size for VoIP was 160 bytes and
the transmission rate was 64 kbps. The video streaming source was YUV QCIF (176 X
144) Highway (2000 frame) (YUV QCIF, 2012). Prior to its transmission, each video
frame was fragmented into packets that in turn had a maximum length of 1024 bytes.
The best-effort traffic had a fixed packet size of 1000 bytes and 125 kbps transmission
rate. File Transfer Protocol (FTP) application was used for the background traffic. FTP
was transmitted over TCP, whereas other traffics were transmitted using UDP transport
protocol. The transmitted traffic over IEEE 802.l i e EDCA (i.e. VoIP, video, best effort
traffic, and background traffic) were mapped into different access categories to represent
different priority levels, as shown in Table 4-2. VoIP had highest priority, whereas the
priority of the background traffic was lowest.
The simulation time was 500 seconds. During the first third of the simulation, two VoIP
applications, and two best effort traffics were transmitted. During the period (170s-
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Chapter 6 n,valuation 01 i\eiw orK i^uamy 01 service
340s), the number of transmitted applications increased to three VoIP, two video, two
best effort traffics, and background traffic making the network load heavy and
approximately 100% of channel capacity was used. However, after the transmission of
video traffic was completed, the network load became light during the last third of
simulation time.
This section is divided into four subsections: two sections provide QoS analysis results
for Fuzzy C-Means and Kohonen Neural Network. The remaining two sections provide
QoS assessment results using Regression Model and Multi-Layer Perceptron.
In this study, FCM clustering algorithm was applied at predefined regular time intervals
to analyse VoIP and video traffic. FCM analysis results for VoIP traffic during the
simulation time interval 350 - 400 seconds are shown in Figures 6-4. The values of QoS
parameters (i.e. delay, jitter, and packet loss ratio) of VoIP were grouped into three
clusters, representing Low, Medium, and High values. Each cluster was represented by
its own centre. Figure 6-4 shows that the packet loss ratio of VoIP is zero. This is
because VoIP was mapped to the access category that had the highest priority.
However, due to the heavy network load, a number of VoIP packets experienced
fluctuated delay that in turn resulted in high jitter values.
_□ _ _
5
i*
-
ts 3 Centers of clustefc
2 G Clusterl
n Cluster2
0 Cluster3
1 © Centerl
□ Centerl
© © Q Centerl
°10 15 20 25 30 35 40 45 50 55
Delay
Figure 6-4. Clustering QoS parameters of VoIP application at predefined time interval.
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cnapier o evaluation 01 i>eiworK ^mamy 01 service
FCM analysis result for video traffic during the simulation time interval 200 - 250s is
presented is presented in Figure 6-5. Due to the mapping of video application to the
Access Category (AC) that had a lower priority as compared with the (AC) assigned to
VoIP, and the heavy load on the network, the QoS parameters (i.e. delay, jitter, and
packet loss ratio) varied extensively. These variations resulted in three clusters
representing Low, Medium, and High QoS parameters. During this interval, the video
quality was affected by the high values of QoS parameters in cluster 3. The centre of
cluster 3 characterised high delay, high jitter, and medium packet loss ratio which were
respectively 540.6 ms, 24.4 ms, and 1.2%.
2.5 O Cluster!
D Cluster2
0 Cluster3
, Centers of dusters 0 Centerl
1 Center 2
^ Center 3
-<r
-O ' '
600
500
400
-o- 300
200
100
Figure 6-5. Clustering QoS parameters of video application at predefined time interval.
The fuzzy partition matrix (produced by FCM algorithm) indicated the degree of
membership of each QoS parameter to each cluster. Figures 6-6 (a) - (b) show
respectively the degree of membership for a sample of delay for VoIP, video
application, and their clusters’ centres. As indicated in the Figures, each value had
different degree of membership to the three clusters between 0 and 1 with the total
being 1. In this study, the fuzzy partition feature of FCM made it a valuable clustering
tool because the characteristics of QoS parameters did not allow crisp (binary) partition.
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Chapter h evaluation 01 iveiworK i^uaiuy ui service
H C l u s te r l
E S I C lu s te r2
I i C lu s te r3
S a m p le d v a lu e s o f d e la y
(a)
WMC l u s te r l
H W C la s te r 2
1 IC lu s te r 3
S a m p le d v a lu e s o f d e la y
(b)
Figure 6-6. The degree of membership between a sample of delay and the cluster's
center for: (a) VoIP, and (b) Video application.
During the clustering process of VoIP and video application, the objective function
indicated the progress of FCM clustering algorithm over the number of iterations
performed. This is shown inFigures 6-7 (a) - (b). The clustering process of FCM
stopped either when it reached the maximum number of iterations orwhen the objective
function improvement between two consecutive iterations was less than the predefined
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L^napier o u vaiuiiuu ii ui n ciw u iK v^uamy ui cjci vice
minimum amount of improvement. From Figures 6-7 (a) - (b), it can be noticed that the
clustering process of VoIP and video application terminated when the objective function
improvement between two consecutive iterations was less than le-5 (predefined
minimum improvement) although the maximum number of iterations was set as 200.
,x i ol
c
1s
£
Ic
Iteration Count
(a)
Iteration Count
(b)
Figure 6-7. The progress of objective functions during FCM analysis of: (a) VoIP, (b)
Video application.
Figure 6-8 illustrates the cluster centres for 50 seconds time intervals during
transmission of VoIP. The FCM algorithm classified the QoS parameters of VoIP into
three levels at each time interval. The Figure shows that VoIP started its transmission
approximately at 170s of the simulation time. During that time, there were two other
VoIP applications using the channel since their transmission started during the first third
of the simulation (i.e. Os-170s). Therefore, during the time interval 170s-185s, the
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v^iiapicr o LvYdiuciuuii u i n c m u i i\ y u a m j u i j c i v it c
values of delay, jitter, and packet loss ratio of VoIP were at medium range of QoS
parameters. As the number of transmitted applications increased to three VoIP, two
video applications, two best effort traffics, and background traffic during the time
interval 215s - 275s, the load became heavy. This in turn made the network incapable of
meeting minimum QoS requirements for a VoIP application. The values of QoS
parameters for cluster 3 were in the high range, 500 ms for delay, 5 ms for jitter, and 6%
for packet loss ratio, indicating a poor quality VoIP. However, once the two video
applications were transmitted during 240s - 260s, the network load became light. This
allowed the network to meet the QoS requirements of the VoIP application during the
time interval 275s - 500s. The values of delay, jitter, and packet loss ratio were in the
low range of QoS parameters.
The FCM algorithm classified the QoS parameters of video application into three levels
for each time interval (i.e. 5 seconds) as shown in Figure 6-9. Video application
transmission started approximately at 170s and terminated at around 240s. Figure 6-9
shows that the majority of QoS parameters values for video application were in the high
range, particularly in cluster 3, around 550 ms for delay, 30 ms for jitter, and 3% for
packet loss ratio during the simulation time 170s - 230s. The reasons for this were: (i)
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c iid p ie r u £ iV iiiu aiiu ii u i i i c m u i i v y u a m j u i i j t i y iv t
the allocation of the video application to an access category which had a lower priority
than the access category that VoIP was assigned (ii) the network load was heavy due to
the transmission of multiple applications during the same simulation time interval.
Consequently, the network was incapable of meeting the QoS requirements for the
video application.
600
500
o
100
170
175
180
185
190
195
200
205
210 Lc3
215 Jc3
D c3
220 Lc2
225 Jc2
230 Jcl
Lcl D e l
Simulation Time (Sec) D el
Centers of Clusters
In this section, the developed regression model was used to combine the QoS
parameters (i.e. delay, jitter, and packet loss ratio) for each centre of the generated
clusters by FCM in order to estimate the overall QoS. Figure 6-10 shows the results
from the devised regression model. The results show the predicted QoS of VoIP for the
centres of generated clusters at each 50 seconds time interval. Figure 6-10 shows that
the QoS values reflected the corresponding QoS parameters indicated in Figure 6-8. In
other words, as the values of QoS parameters decreased, the values of overall QoS
increased accordingly. When VoIP started its transmission at 170s, its overall quality
was poor (i.e. below 30%). This is because of its contention to access the channel which
carried other VoIP applications with the same priority. As the number of transmitted
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L^napier o j ^ v a i u a u u u u i i i c i w u i iv v £ u a ii i j u i o c i t iv c
applications increased to three VoIP, two videos, two best effort traffics, and
background traffic during the time interval 215s - 335s, the VoIP quality degraded by
overall 51.83% as compared with its quality of 55.97% during the period 1 8 5 -2 1 5 . The
heavy load on the network made its performance incapable of meeting minimum QoS
requirements for VoIP application. However, during the last third of simulation time,
the network load became light due to the termination of two video applications. The
VoIP quality increased sharply at certain time interval. For example, VoIP quality
reached 87.9% at time interval 365s - 395s.
H Q oS of center 1
I IQoS of center 2
1 IQoS of center 3
The developed regression model was also used to assess the QoS of video application
for each centre of the FCM generated clusters shown in Figure 6-9 by combining the
QoS parameters: delay, jitter, and packet loss ratio. Figure 6-11 shows the predicted
QoS of the video application for the centres of generated clusters at each 5 seconds time
interval. The Figure shows that the video quality was good at 170s. This is because the
transmission of video started prior to other traffic such as VoIP, other video application,
and background traffic during the second third of simulation time (170s - 340s).
However, due to the low priority of video as compared with VoIP and the load of the
network became heavy, the quality of video degraded sharply and its QoS range
remained between poor and average level (i.e. 3.04% - 73.05%).
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L.napter o CiVillUilllUH u i l i c m u i i v y u a m j u i u v i > iw
f 50
170 175 180 185 190 195 200 205 220 225 230
Simulation Time (Sec)
The results obtained from the devised regression model were compared using other QoS
assessment methods. These were: fuzzy inference system mechanism, and distance
measurement evaluation system reported in (Al-Sbou, 2010(a)) and (Al-Sbou, 2006(b)).
Tables 6-1 and 6-2 show respectively the comparison between the aforementioned
assessment methods used to quantify the overall QoS of VoIP, and video application.
From the sampled values of evaluated QoS provided in Tables 6-1 and 6-2, it is
indicated that the three assessment methods provided results which were comparable.
The values of the correlation coefficients between the QoS determined using regression
model, and the other QoS assessment methods (i.e. fuzzy inference system mechanism,
and distance measurement evaluation system) were respectively: 0.95 and, 0.97 for
VoIP application, and 0.89, and 0.93 for video application. Although some outputs were
slightly different, they were still in the same region (i.e. poor, average, or good). The
discrepancies were due to the fact that each method followed a different operation.
However, the values of QoS obtained from devised regression model spanned between
(0%-100%), whereas the range of QoS values produced by FIS was between (10%-
90%). This indicates that the devised regression model provides more accurate results.
125
cnapier o H r V itiu a u u ii u i m i n u i i\ y u n u i y u i o c i * n -c
Table 6-1. QoS parameters of VoIP and expected QoS using: fuzzy inference system,
distance measurement, and regression model.
QoS parameters Assessment of overall QoS using:
Fuzzy
Delay Jitter Packet loss Distance Regression
inference
(ms) (ms) ratio (%) measurement model
system
36.69 4.62 0.00 9.64 29.69 29.78
108.84 5.00 0.00 9.28 24.13 21.92
599.94 5.00 6.00 9.28 0.01 4.31
24.21 2.87 0.00 71.50 55.79 64.14
Table 6-2. QoS parameters of video and expected QoS using: fuzzy inference system,
distance measurement, and regression model.
QoS parameters Assessment of overall QoS using:
Fuzzy
Delay Jitter Packet loss Distance Regression
inference
(ms) (ms) ratio (%) measurement model
system
57.15 13.03 0.00 80.42 92.69 82.29
85.37 24.52 2.61 13.81 24.90 40.72
289.30 25.64 1.38 12.22 32.19 33.59
534.18 24.00 1.06 10.11 20.25 15.69
600.00 29.55 1.16 9.29 8.30 3.04
126
Chapter o n .v u iu a ii u n u i ix c i w u i k y u a u i j ' u i a c i v itc
Kohonen neural network (i.e. Self-Organising Map SOM) processed the QoS
parameters (i.e. delay, jitter, and packet loss ratio) of VoIP, and video applications into
correlated groups. The values of delay, jitter, and packet loss ratio of transmitted VoIP
and video were grouped into three classes representing Low, Medium, and High values
as shown in Figures 6-12 and 6-13. Low values of delay, jitter, and packet loss ratio are
represented in blue, Medium values are represented in green, and High values are
represented in red.
Figures 6-12 (a), (c) show that low delay, and packet loss ratio activate most neurons in
SOM. This was due to VoIP having the highest priority to access the channel amongst
other transmitted traffic such as video, and best effort traffic. However, the heavy load
on the network and the contention with other VoIP applications having the same priority
caused high fluctuation to the delay. This in turn generated high values of jitter as
shown in Figure 6-12 (b). The high values of jitter would have a negative effect on the
overall VoIP quality.
The labels of the QoS parameters in Figure 6-12 (d) provided a differentiation between
the three generated QoS classes. Label L identified the active neurons which represent
Low values of QoS parameters; label M covered the region where Medium values of
QoS parameters were located; whereas High QoS parameters were represented by other
neurons with label H. The groupings produced by the Kohonen network provided
information about the relationships between the different QoS parameters of transmitted
VoIP traffic and subsequently discovered how VoIP was treated as high priority traffic
transmitted over heavy loaded network.
Figure 6-12 (e) shows the response of a Kohonen neural network to the QoS parameters
of VoIP. SOM differentiated delay, jitter, and packet loss ratio into almost three
clusters. The Figure shows that Low and Medium QoS parameters have well defined
regions. The winning neurons for these regions were characterised by small number of
QoS values. In contrast, the neighbourhood region of high QoS parameters was large.
This was because most jitter values for VoIP were high .This in turn had an impact on
VoIP quality, although other values such as packet loss ratio were 0% during most of
VoIP transmission period. The winning neuron for high QoS parameters region was
activated by 69 values.
127
VyliapiVl A VI T1VV
306 2 .7 3
1 8 .6 0 .4 5 6
(a) (b)
Packet loss ratio (%)
5 .8 1
2 .9
(c) (d)
Medium QoS parameters
m m f* ° o
0 0 0 0 0 3
0 0 0 0 0 0 0
o 0 0 0
5}(7) ® (§) <2 0 (g)
(e)
Figure 6-12. Classifying QoS parameters of VoIP using kohonen network: (a) Delay,
(b) Jitter, (c) Packet loss ratio, (d) QoS parameters labels, (e) SOM sample hits.
The analysis of SOM for QoS parameters of video application is shown in Figure 6-13.
During its transmission, the video application experienced high delay, jitter, and packet
loss ratio as indicated in Figures 6-13 (a) - (c). The network was incapable of meeting
the QoS requirements for video application. This was because of the heavy load on the
network as multiple applications were being transmitted at the same simulation time
interval. Moreover, the video application was allocated to a lower priority access
category compared with VoIP allocation.
Although some individual values of delay, jitter, and packet loss ratio were within an
acceptable level of quality, high values of one the aforementioned parameters would
have a negative effect on the overall quality of video. The labels of QoS parameters and
the response of SOM to the QoS parameters of video application shown in Figures 6-13
128
(d) and (e) respectively provided a differentiation between the generated QoS classes.
Both figures indicated that Low QoS parameters identified by label L covered a very
small region of SOM. Two neurons were activated by low QoS values and the winning
neuron contained 16 values. The region of Medium values of QoS parameters which
had label M was large as compared with Low QoS parameters region. The
neighbourhood region of high QoS parameters represented by label H covered most of
the Kohonen map. Figure 6-13 (e) shows that high QoS parameters had four sub-
regions. The number of values activated the winning neurons for these regions were
relatively large (i.e. 121, 56, 19, and 8). It can be concluded from Figure 6-13 that high
value of QoS parameters of video application: delay, jitter, or packet loss ratio could
have negative impact on its overall quality. The analysis of SOM demonstrated how the
simulated network could manage and treat different applications with different priorities
during light and heavy loads.
8 5 .1
2 .9 3
(c) (d)
L o w Q oS p aram eters
M ed iu m ^ ^
Q oS ^ m
p aram eters W
0 O’
(e) @0
® o
0 CD
ZlSD@®
H ig h Q oS p aram eters
Figure 6-13. Classifying QoS parameters of video application using kohonen network:
(a) Delay, (b) Jitter, (c) Packet loss ratio, (d) QoS parameters labels, (e) SOM sample
hits.
t^napier o j&vaiuauuii ui i^ eiw o m v^uauiy ui o erv icc
After the QoS parameters of VoIP and video application were processed by Kohonen
neural network, the average of QoS parameters (i.e. delay, jitter, and packet loss ratio)
activated a winning neuron and a number of neurons in the neighbourhood region as
shown in Figures 6-12 and 6-13 were combined by MLP to assess the overall QoS. The
MLP was required to be trained correctly in order to assess the QoS in an effective
manner. Figures 6-14 (a) and (b) show respectively the training process of the MLP to
assess the QoS of VoIP and video application. The training process of MLP was
terminated either when the maximum number of iterations was reached or when there
was insignificant error (i.e. 0.001) between MLP outputs and desired results. In this
study, the training process of MLP in case of QoS assessment of VoIP or video
application was terminated when the maximum number of training iterations (i.e. 1000)
was reached. The value of maximum number of training iterations was chosen
experimentally, i.e. different values were tested and the MLP response was monitored.
It is also indicated from the Figures that the Mean Squared Error (MSE) values for MLP
when the training process terminated to assess the QoS of VoIP and video were
respectively 0.003 and 0.005. These values indicated an insignificant error between
MLP outputs and desired results.
Train I
— Best |
•gio'
£i<f
0 100 200 300 400 500 (00 700 800 900 1000
1000 Epochs
— Train
Best
ElO
0 100 200 300 400 500 600 700 800 900 1000
1000 Epochs
Figure 6-14. The progress of training MLP to assess the QoS for: (a) VoIP, (b) Video.
130
^ iia p ier o r.vaiuauoii ui n eiw orK v£u a ,,lJ U1 vice
Figures 6-15 (a) and (b) show respectively the comparison between normalised actual
QoS and calculated outputs from MLP during its training to assess QoS of VoIP and
video. It can be perceived from the Figures that the actual output values used to train
MLP and the output values following the termination of its training were strongly
correlated. The correlation coefficients were 0.997 when MLP trained to assess QoS of
VoIP and 0.996 when MLP trained to assess QoS of video application. The correlation
coefficient values indicated that the MLP had been trained effectively.
Training: R=0.99673
O.
O.
o.
« o.
-o.
-o.
-o.
Target
(a)
Training: R=0.99535
Target
(b)
Figure 6-15. Comparison between normalised actual QoS and calculated outputs from
MLP in case of: (a) VoIP QoS assessment, and (b) Video QoS assessment.
131
so ap ier o
The results in Figures 6-16 (a) and (b) show respectively the predicted QoS of VoIP and
video applications obtained from MLP during the test phase. From the Figures, it can be
observed that the range of QoS values reflected the QoS parameters regions shown in
Figures 6-12 (e) and 6-13 (e). In other words, Good QoS reflects the region of Low QoS
parameters, whereas High QoS parameters region produced Poor QoS. Due to its
priority over video application, the average estimated QoS of VoIP for Low, Medium,
and High QoS parameters regions was elevated by 56.01% as compared with the
average estimated QoS of video application. However, the heavy load on the network at
certain time intervals made its performance incapable of meeting the minimum QoS
requirements for VoIP and video applications.
90
• Good QoS
■ Average QoS
80 ♦ Poor QoS
70
60
50
40
30
20
lO
8 lO 12 14 16 18 20
No. of Neuron
100
• P oor Q oS
■ A v era g e Q oS
90 ♦ G ood Q oS
80
70
60
50
40
30
20 10 11 12 14
N o . o f N eu ro n
Figure 6-16. The QoS assessment of: (a) VoIP, and (b) Video application using Multi-
Layer Perceptron.
The results of expected QoS for VoIP and video obtained using MLP were compared
with other QoS assessment methods such as the developed fuzzy inference system FIS
(Al-Sbou, 2 0 1 0 ( a ) ) , Euclidean distance measurement system (Al-Sbou, 2 0 0 6 ( b ) ) , and
132
^napier n jj>yaiuauuu ui n c in u m ^ u a n tj v i uvi »iw
regression model (Dogman et al, 2012)) as shown in Tables 6-3 and 6-4 respectively.
The values of correlation coefficients between the QoS of VoIP determined using multi
layer perceptron neural networks, and the other QoS assessment methods were: 0.90 for
FIS, 0.99 for distance measure, and 0.98 for the regression model. Whereas correlation
coefficients between the QoS of video determined using MLP and other QoS
assessment techniques were: 0.97 for FIS, 0.97 for distance measure, and 0.99 for
regression model. From the values of correlation coefficient and the values of evaluated
QoS provided in Table 6-3 and 6-4, it was concluded that the aforementioned QoS
assessment techniques for VoIP and video application provided results which were
closely comparable. Although some outputs were slightly different, they were in the
same QoS region. The discrepancies were due to the fact that each method followed a
different process. However, the values of QoS obtained from the MLP architecture
ranged from (1% - 100%), whereas the range of QoS values produced by the FIS was
between (10%-90%). This indicates that the MLP was more effective.
Table 6-3. QoS parameters of VoIP and expected QoS using: Fuzzy Inference System
(FIS), Distance Measurement (DM), Regression Model (RM), Multi-Layer Perceptron
Packet Loss
Delay (ms) Jitter (ms) FIS DM RM MLP
ratio (%)
55.44 4.59 0.00 9.73 29.98 29.29 27.41
| 35.36 4.58 0.00 9.76 30.26 30.63 29.16 j
180.85 5.00 1.25 9.28 23.94 21.90 26.12
35.23 2.26 0.00 81.98 67.77 69.95 72.98
37.96 3.18 0.00 60.30 50.42 54.20 51.68
20.28 2.41 0.00 80.39 65.44 68.27 66.29
71.67 2.99 0.00 67.51 52.66 55.46 52.89
24.24 2.76 0.00 74.18 57.69 62.11 56.44
133
i ^ v a i u a u u u u i n c m u i iv y u a i u j u i ljv i ▼i w
Table 6-4. QoS parameters of video and expected QoS using: Fuzzy Inference System
(FIS), Distance Measurement (DM), Regression Model (RM), Multi-Layer Perceptron
(MLP)
QoS parameters Assessment of overall QoS using:
Packet loss
Delay (ms) Jitter (ms) FIS DM RM MLP
ratio (%)
This chapter introduced two novel Quality of Service (QoS) assessment systems. The
first system included a combination of fuzzy C-means (FCM) and regression model to
analyse and measure the QoS of VoIP and video traffic transmitted over a simulated
network. Whereas the other system used a combination of supervised and unsupervised
neural networks (i.e. Kohonen network and multi-layer perceptron (MLP)) to evaluate
the QoS of VoIP and video applications.
The QoS parameters of VoIP and video traffic were analysed by FCM and Kohonen
network. The capability and robustness of these techniques to cope with imprecise QoS
patterns made them effective clustering mechanisms for QoS analysis. FCM and
Kohonen network classified the values of QoS parameters of transmitted VoIP and
video into clusters representing Low, Medium, and High values of QoS. The regression
134
V /U c tJ J tC l U i^Ydiuaiiuu u i i i c i n u i *v y u a u i j u i a c t t i v t
model and MLP in turn combined the QoS parameters (i.e. delay, jitter, and packet loss
ratio) for each centre of generated clusters and produced a single value that represented
the overall QoS. The overall QoS was a good indication of network performance. The
overall QoS can be used to monitor the network and to avoid congestion. The values of
assessed QoS were strongly correlated to a number of previously studied QoS
assessment methods.
135
Chapter 7 Improvements in Quality of Service in
Computer Networks
7.1 Introduction
In this chapter, a new Quality of Service (QoS) enhancement scheme for WLAN-wired
networks is developed and its performance is evaluated. The proposed scheme consists
of an adaptive Access Category (AC) traffic allocation algorithm that is incorporated
into the network’s wireless side to improve the performance o f IEEE 802.1 le Enhanced
Distributed Channel Access (EDCA) protocol, and a Weighted Round Robin (WRR)
queuing scheduling mechanism that is incorporated into the wired side of the network.
The adaptive traffic allocation algorithm determines the Packet Arrival Rate (PAR) of
up-link and down-link traffic for each AC. It then dynamically allocates the traffic of
the lower priority AC to the next higher AC, when the higher AC is not receiving traffic
at the time. On the wired side of the network, the aim of WRR is to share the network
resources based on the traffic’s quality o f service (QoS) requirements. The performance
of the proposed scheme was compared with the standard IEEE 802.1 le EDCA and
FIFO queuing mechanisms (i.e. WLAN-wired network legacy scheme). The
incorporation of the scheme improved the performance of the WLAN-wired network,
thus enhancing the QoS for transmitted applications. The scheme allowed an end-to-end
QoS to be set up which in turn provided improved delivery of a variety of applications
in the context of wired-cum-wireless networks.
This chapter is organised as follows: the relevant studies are discussed in section 7.2.
Section 7.3 introduces a detailed description of the proposed QoS enhancement scheme.
The simulated network and traffic models are presented in section 7.4. The results are
presented and discussed in section 7.5. The conclusions are provided in section 7.6.
136
Lnapter / xmprovemenu* 111 ^uaui}' ui aervite in ^uinpuier i^eiworiu»
communication parameters such as packet delay, jitter and throughput. On the other
hand, some time-insensitive applications, such as File Transfer, are more affected by
packet loss and reliability (this includes bits error rate), but are unaffected by jitter
(Kurose and Ross, 2005). The interconnection between wired and wireless networks also
requires that QoS of traffic being exchanged to be appropriately realised.
Most previous studies explored QoS support either in wireless local area networks
(WLANs) or in wired networks as discussed in section 3.4 of chapter 3. However, there
were only few studies to enable end-to-end QoS for wired and wireless networks. For
instance, Skyrianoglou et al. (2002) proposed a Wireless Adaption Layer (WAL) to
provide an integrated QoS between WLAN and IP infrastructure. The WAL was located
between the MAC layer and the IP layer. The function of WAL was to provide service
differentiation to the IP traffic transmitted between WLAN and a fixed IP network to
improve the performance in a wireless IP networks. An architecture to map IP layer
Differentiated Services Code Point (DSCP) QoS to MAC layer EDCA Access categories
(ACs) was proposed by Park et al. (2003). The integrated scheme used DSCP of the IP
packets to allocate packets to an appropriate AC. Another mapping scheme of Enhanced
Distributed Channel Access (EDCA) access categories to IP traffic class in wired
network was proposed by Senkindu and Chan (2008). Their mapping aim was to ensure
that end-to-end service guarantees were provided for multimedia applications.
However, most previous studies either required modifications of wireless stations, which
in turn complicated the WLAN operation, or supported QoS of high priority traffic
which in turn starved other transmitted traffic. The limitation of WAL proposed by
Skyrianoglou et al. (2002) is that an intermediate layer was introduced between the MAC
and IP layers in wireless stations. This in turn resulted in more complexity to WLAN
management. Also, the modification of wireless stations to support DiffServ
functionality was a disadvantage of the method introduced by Park et al. (2003).
The scheme by Senkindu and Chan in (2008) was simple to implement, but the low
priority traffic’s packet drop rate was high due to link congestions and QoS prioritisation.
Therefore, the challenge in end-to-end QoS is that both wired and wireless parts of the
network provide suitable treatment for each class of traffic and to efficiently use network
resources.
The main contribution of this study is the development of a new QoS enhancement
scheme that improves delivery of a variety of applications in both wired and wireless
sides of the network. The scheme provides an integrated MAC layer QoS for the
137
L im p ier / uiijji uvciiiciiio in \ £ u a n i y ui oci vice 111 cu iu p u in licinuiivd
wireless side of the network and network layer QoS controls in the wired side. A novel
aspect of the approach is that an adaptive AC traffic allocation algorithm was devised
and incorporated into wireless access point (AP) in order to improve QoS in the IEEE
802.l i e EDCA protocol. Also, Weighted Round Robin (WRR) queuing scheduling
mechanism was implemented into the congestion point (i.e. router) in wired networks to
support a fair distribution of bandwidth among different traffic types.
The adaptive traffic allocation algorithm was located in the access point in the wireless
side of the network. The algorithm is outlined in the flow chart shown in Figure 7-1.
The algorithm monitors the Packet Arrival Rate (PAR) of the uplink and downlink
traffic for each AC, in order to allocate traffic to an appropriate AC. The PAR
parameter was chosen because it signify how well real-time and non-real-time
applications are delivered. Also, it can be easily computed in real-time. This in turn
facilitates quick allocation of the arrived traffic to the most appropriate AC.
PAR is defined as the number of successfully received packets in a given time interval
(Wang et al., 2000). Equation (7.1) was used to calculate PAR:
where, PARi (t ) is packet arrival rate during ith time interval. £ Pi (t) is the total
number of packet received during the ith interval, and tj is the time duration of the ith
interval.
138
Chapter 7 improvements in quality 01 service in com puter i^eiworics
Incoming packets
Yes No
During transmission, the algorithm calculates at each time interval, the PAR for each
AC (i.e., ACo to AC3). AC0 receives the highest priority traffic, while AC3 is for the
lowest. When the PAR at a higher AC is zero, the algorithm allocates the incoming
traffic to a lower priority AC to the next higher priority AC. This operation increases
the lower priority traffic’s ability to access the channel and thus it transmits its packets
at a faster rate. This also ensures the network resources do not remain idle.
The algorithm waits for a pre-set time interval and re-calculates the PAR associated
with traffic for each AC. If there is an application for transmission at a higher priority
AC, the lower priority traffic previously diverted to the higher AC, would be moved
139
L.napter / improvements in viiiamy 01 service 111 ^umpuier neiwui ks
back to its original default AC. However, if higher priority traffic started its
transmission during the pre-set time interval, it would be directed to its original AC and
the lower priority traffic would be moved back to its original AC after the current
interval is terminated. Therefore, the pre-set time interval must be chosen carefully to
accommodate the process. This is because in case of a large time interval, the higher
priority traffic would be negatively affected, whereas in case of short time interval, the
high computational load would be experienced (that is an issue for real-time
operations).
In the wired side of the network, WRR scheduling mechanism was integrated between
the router and the AP in order for the transmitted traffic to share the network resources,
based on QoS requirements. The WRR takes the incoming packets from a number of
traffic types and schedules them according to their allocated respective weights. The
allocation of weights considers the traffic priority, application's packet size, and its
transmission rate. In this study, four types of traffic were considered: VoIP and video as
time-sensitive applications and best effort traffic and background traffic as time-
insensitive applications. Time-sensitive applications were high priority and were
assigned 60% of the bandwidth, whereas best effort and background traffic were treated
as low priority and were allocated 40% of the bandwidth.
The respective weights for the aforementioned applications are shown in Table 7-1.
However, when one type of application was not transmitted at a particular time interval,
WRR divided the bandwidth allocated to that application to other applications being
transmitted according to their respective weights (Semeria, 2001).
Weights 3 3 2 2
140
unapter / improvements in quality 01 service in computer i^eiworics
The traffic transmitted over the simulated networks was: VoIP, video streaming, best
effort traffic, and background traffic. Constant Bit Rate (CBR) traffic was adapted to
model VoIP. VoIP was modelled with the G.711 voice encoding scheme. The video
streaming source was YUV QCIF (176 x 144) Foreman with 400 frames (YUV QCIF,
2012). Prior to transmission, each video frame was fragmented into packets that in turn
had a maximum length of 1024 bytes. The best-effort traffic was modelled using CBR
with different packet sizes and generation rates that corresponded to non- VoIP usage.
File Transfer Protocol (FTP) application was used for the background traffic. FTP was
transmitted over TCP (to ensure reliability), whereas other traffics were transmitted
using UDP (to ensure high transmission rate) transport protocol. Table 7-1 shows the
traffic characteristics of the aforementioned applications.
Two different scenarios were considered for each simulated network. The first scenario
excluded the presence of QoS enhancement scheme. The WLAN was based on IEEE
802.l i e EDCA, The main parameters that modelled the wireless channel were the
141
Chapter 7 improvements in quality ot service in computer i>eiworKs
default settings for IEEE 802.lie . These parameters are shown in Table 4-1. The
transmitted traffic (i.e. VoIP, video streaming, best effort traffic, and background
traffic) were mapped to the ACs to represent different levels of priority as shown in
Table 4-2. VoIP had highest priority, while the priority of the background traffic was
lowest. At the wired side of the network, First-in-First out (FIFO) queue scheduling
mechanism (queue size=50) was implemented between the router and the access point.
FIFO was chosen due to its simplicity that facilitates real-time operations.
However, in the second scenario where the QoS enhancement scheme was included, the
adaptive traffic allocation algorithm was operated at the wireless side of the AP. The
main parameter of adaptive allocation algorithm was the pre-set time interval which was
set up to be 2.5s. This value was chosen experimentally to provide best results. In the
wired side of the network, the queue scheduling mechanism implemented between the
router and the access point was Weighted Round Robin (WRR). WRR classified traffic
based on their QoS requirements. Time-sensitive applications had a higher priority than
time-insensitive applications. The main parameters of WRR are shown in Table 7-3.
The network topology covered an area of 500m x 500m and the stations were
positioned randomly within this area. This setup remained unchanged during the
simulations. Simulations were repeated 10 times. Each time a different initial seed value
was used to randomly position the stations and to control which node transmitted first.
The randomness introduced using the different seeds avoided the bias of random
number generation. The results of the 10 simulations were then averaged. Simulation
duration was 300 seconds. These periods were considered sufficient to examine the
behaviour of IEEE 802.1 le protocol.
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^napier / xinpruveiiieiiui 111 yuam j ui service 111 v^uinpuier nciwuiivs
7.2.1 Delay
In this study, the measurement of delay includes various types of delays such as queuing
delay, transmission delay, processing delay, and propagation delay. Figure 7-2 (a) - (d)
show respectively the average values of delay for VoIP, video streaming, best effort
traffic, and background traffic under different network loads. It is indicated from the
Figures that the average values of delay using legacy and QoS enhancement scheme
were increased as the number of connections increased. This is because increased
probability of packets collisions, which in turn caused the nodes to retransmit the
collided packets. In this case, the delay between the consecutive packets would be
further increased.
In addition, Figures 7-2 (a) - (d) indicate that both schemes treated the transmitted
traffic based on their QoS requirements. VoIP gained the lowest delay because it was
assigned by default to the highest priority access category (i.e. ACo), whereas the
background traffic experienced the highest delay among other traffics as it was assigned
by default to the lowest priority access category (i.e. AC 3).
The VoIP that was assigned to ACo had a small value for AIFS (i.e. equivalent to 2ps)
allowing ACo to start its backoff procedure after detecting the channel was idle for an
AIFS period. Also, small values for CWmin, and CWmax which were respectively 7 slots,
and 15 slots for ACo, made VoIP traffic to have a small waiting duration before
accessing the channel. In addition, large value of TXOP (i.e. equivalent to 3.008 ms)
allowed ACo to transmit multiple data frames continuously during that particular time
interval defined by TXOP. The values of aforementioned EDCA parameters gave VoIP
traffic the highest priority to access the channel. Therefore, the expected packet delay
would be lower than other transmitted traffic.
143
unapier / improvements 111 v£uamy ui oti vite 111 v^uiupuicr iicmuina
In contrast, the background traffic that was assigned to AC 3 experienced highest delay
among other transmitted traffics. For example, the delay values of the background
traffic at medium network loads (i.e. 16 connections) using legacy and QoS
enhancement scheme were receptively 186.6 ms, and 130.5 ms. These values were
much higher as compared with delay values of other traffic under the same network
conditions. The delay values of video streaming for instance when using legacy and
QoS enhancement scheme were respectively 17.1 ms, and 48.5 ms. This is because of
the values of EDCA parameters assigned to AC 3 . Large values of AIFS, CWmin, and
CWmax which were respectively 7 ps, 31 slots, and 1023 slots made the traffic assigned
to AC 3 to wait for long period of time before accessing the channel and then to start its
backoff procedure after detecting the channel was idle for an AIFS period.
However, Figures 7-2 (a) - (d) also indicate that the traffic transmitted using the
proposed QoS enhancement scheme experienced a lower delay for all ACs as compared
with the legacy network scheme. The average delay for video streaming, best effort
traffic, and background traffic were decreased by 57.6%, 63.6%, and 31.5%
respectively when the QoS enhancement scheme was introduced. The reductions in
delay values were due to the allocation of lower priority traffic to the next higher AC in
cases of Packet Arrival Rate (PAR) for the higher priority traffic was zero. For instance,
the packets of video streaming were transmitted by default using ACi which its EDCA
parameters value were 2 ps, 15 slots, and 31 slots for AIFS, CWmin, and CWmax
respectively. However, when QoS enhancement scheme was applied, the packets of
video streaming were transmitted for particular time interval using ACo when VoIP
traffic did not transmit at that particular time interval. The allocation of video streaming
traffic's to ACo (which its EDCA parameters value were 2 ps, 7 slots, and 15 slots for
AIFS, CWmin, and CWmax respectively) increased its ability to access the channel and
thus reduced the average value of delay.
Also, when QoS enhancement scheme was applied, the fairness in treating traffic with
different QoS requirements introduced by WRR at the wired side of the network had
been a factor in improving network performance. For example, the reduction o f 66.2%
in delay for VoIP was due to the implementation of WRR, as the adaptive allocation
algorithm did not shift the traffic transmitted by the highest priority AC.
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Chapter 7 im provem en ts in v^uaiuy ui a er v itc m v.uiupuici n c m u in a
(a)
No. of connections
(b)
No. o f connections
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□ Q o S enhancem ent scheme
i
i
1
i
L
i
i
i
i
i
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i
1 i
' m 1
i i
70
i t
I i
60
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i
i
(c) ^ 50
.2
i
i
« 40
30
20
i 1 B
10
1 No. of connections
32
(d)
No. of connections
Figure 7-2. Bar chart representation of average delay with and without QoS
enhancement scheme for: (a) VoIP, (b) Video, (c) Best effort traffic, (d) Background
traffic.
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C h a p te r / improvements in yuitiuj' ui oei vice in v^uni|mici ncmuina
7.2.2 Jitter
The average values of jitter for VoIP, video streaming, best effort traffic, and
background traffic under different network loads are shown respectively in Figures 7-3
(a) - (d). Each Figure provides a comparison between the jitter's values obtained using
the proposed QoS enhancement scheme and the legacy scheme under a light network
load (i.e. 8 connections), a medium network load (i.e. 16-24 connections), and a heavy
network load (i.e. 32 connections) respectively.
It is shown from the Figures that the values of jitter for all transmitted traffic gradually
increased as the network became more congested for both legacy and QoS enhancement
schemes. For example, the values of jitter for VoIP traffic under a light network load
were respectively 1.7 5ms, and 3.3 ms using QoS enhancement scheme and the legacy
scheme. Whereas, the jitter's values for VoIP traffic using QoS enhancement scheme
and the legacy scheme under a heavy network load were respectively 4.9 ms, and 5.7
ms.
In other words, as the number of active stations increased; the probability of collisions
increased accordingly due to increased competition between the stations. This forced the
MAC protocol to retransmit the collided packets. When the collided packets were
successfully received; the time between any two consecutive packets that were
successfully received at the destinations were increased, and subsequently increasing
the jitter.
However, Figures 7-3 (a) - (d), also indicates that the implementation of QoS
enhancement scheme reduced the jitter values of all transmitted traffic as compared
when the legacy scheme was applied. The average jitter for VoIP, video streaming, best
effort traffic, and background traffic were reduced by 36.3%, 36%, 20.5%, and 35.1%
respectively.
The reduction of jitter for traffic transmitting by lower priority access category (i.e. ACi
- AC3) was due to the adaptive allocation algorithm at the wireless network, and WRR
implemented at wired side of the network.
The jitter's reduction of best effort traffic for instance was due to the dynamic allocation
for CBR packets from the lower priority AC (i.e. AC 2) to the next higher AC (i.e. ACi)
when the ACi was not transmitting video traffic at that time. In other words, best effort
traffic were transmitted by default using AC 2 which its EDCA parameters were 3 ps, 31
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liu p iu v u iiu iid in y u r n iij u i a c t v i t c 111 V / U i u p u i c i iic m u ix v a
slots, and 1023 slots for AIFS, CWmin, and CWmax respectively. However, when QoS
enhancement scheme was applied, the CBR packets were transmitted for a particular
time interval using ACi when video traffic did not transmit at that time interval. The
allocation of CBR traffic to ACi which its EDCA parameters were 2 ps, 15 slots, and 31
slots for AIFS, CWmin, and CWmax respectively increased its ability to access the
channel and thus reduced the average jitter.
Also, the fairness of WRR to share network resources among transmitted traffic based
on their QoS requirements decreased the values of jitter. WRR allocated 20% o f the
bandwidth to CBR traffic making the packet collision to decrease accordingly. This is
unlike the FIFO mechanism in legacy scheme where high and low priority traffic were
treated in the same manner. The arrived packets dropped regardless of their priorities
when the queue of FIFO became full. As a result, the implementation of QoS
enhancement scheme reduced the possibility of CBR packets to collide and
subsequently reduced the variations in delays between any two consecutive CBR
packets.
The jitter for VoIP traffic which was reduced by 36.3%, when QoS enhancement
scheme was implemented was due to the implementation of WRR only. This is because
the adaptive allocation algorithm does not shift the traffic transmitted by default using
the highest priority AC (i.e. ACo).
As the highest priority traffic, 30% of the bandwidth was allocated to VoIP when WRR
was implemented. This is unlike FIFO mechanism in legacy scheme which made VoIP
as high priority traffic to be treated in the same manner as low priority traffic.
Subsequently, the arrived packets were dropped regardless of their priorities when the
buffer at the router became full.
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Chapter 7 improvements in yuaiiiy 01 service in L.ompuier iveiworKs
(a)
No. of connections
(b)
No. of connections
(c)
16 32
No. of connections
(d) rao
No. of connections
Figure 7-3. Bar chart representation of average jitter with and without QoS
enhancement scheme for: (a) VoIP, (b) Video, (c) Best effort traffic, (d) Background
traffic.
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unapier / i i u p i U Y C i u c i i U ) 111 v ^ u a i u j u i o e i v i c e i n ^ u m p u ic i n c iw u in s
Figures 7-4 (a) - (d) depict comparison of packet loss ratio for VoIP, video streaming,
best effort traffic, and background traffic obtained using QoS enhancement scheme and
the legacy scheme.
The results indicate, the increase in packet loss ratio was based on the number of
connections, and the priority of the traffic for both schemes. For example, packet loss
ratio for video streaming increased as the number of connections increased as shown in
Figure 7-4 (b). Packet loss ratios for video streaming under a light load (i.e. 8
connections) were respectively 0.001%, and 0.1% for QoS enhancement scheme and the
legacy scheme. Whereas, packet loss ratios for video traffic using QoS enhancement
scheme and the legacy scheme under a heavy network load (i.e. 32 connections) were
respectively 1.9%, and 3.5%. This was caused by a high degree of competition between
the transmitting stations, which in turn increased the probability o f packets colliding.
The priority of the traffic also influenced packet loss. For example, Figure 7-4 (a) shows
the packet loss ratios for VoIP under a medium network load (i.e. 32 connections)
which were respectively 0.1%, and 0.25% using the QoS enhancement scheme and the
legacy scheme. Whereas, the packet loss ratios for the video application under the same
load were respectively 0.51%, and 1.71% using the QoS enhancement scheme and the
legacy scheme as shown in Figure 7-4 (b). This is because; VoIP traffic was assigned to
the highest priority AC (i.e. ACo) with small values of AIFS, CWmin, and CWmax.
Whereas video traffic was assigned to the lower priority AC (i.e. ACi) with large values
of AIFS, CWmin, and CWmax as compared with ACo. Small values of AIFS, CWmin,
and CWmax increased the traffic’s ability to access the channel and thus the AC
transmitted its packets faster.
Figure 7-4 (d) shows that the packet drop rate for FTP which was transmitted using the
lowest priority AC (i.e. AC 3) was lower than the packet loss ratio for CBR shown in
Figure 7-4 (c). For example, the packet loss ratios for FTP under a medium load (i.e. 16
connections) were respectively 0.21%, and 1.68% using QoS enhancement scheme and
the legacy scheme. Whereas, the packet loss ratios for CBR traffic under the same
network load were respectively 0.7%, and 2.05% using QoS enhancement scheme and
the legacy scheme.
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Chapter 7 improvements in viuauty 01 service m t^umpuier ncivruiM
Although the CBR traffic was located at a higher priority AC (i.e. AC 2), FTP was
transmitted using Transmission Control Protocol (TCP) whereas CBR traffic
transmitted using User Datagram Protocol (UDP). TCP as a connection-oriented
protocol provides a reliable, ordered, and error-checked delivery of packets.
Figures 7-4 (a) - (d) also indicate the proposed QoS enhancement scheme reduced
packet loss ratio for all ACs as compared with legacy scheme. The overall packet loss
ratios for the VoIP, Video streaming, CBR traffic, and FTP application decreased by
72.2%, 6 8 %, 26.1%, and 81.9% respectively.
The reductions in packet loss ratios were due to two factors: the adaptive traffic
allocation algorithm at the wireless side of the AP, and the WRR queuing scheduling
mechanism which was implemented between the AP and the router at the wired side of
the network.
The former factor affected the video, CBR, and FTP traffic as it shifted traffic from a
lower AC to the next higher AC, whereas the latter factor reduced the packet loss for all
transmitted traffic due to its fairness distribution o f network resources. As an example,
the packet loss ratio for video streaming under a heavy network load (i.e. 32
connections) was 3.5% using the legacy scheme, whereas its packet loss ratio under the
same condition using QoS enhancement scheme was 1.9%. This reduction was due to
the dynamic allocation for the video packets from a lower priority AC (i.e. ACi) to the
highest AC (i.e. ACo) when the ACo was not transmitting VoIP traffic at that time. The
values of EDCA parameters for ACi were 2 ps, 15 slots, and 31 slots for AIFS, CWmin,
and CWmax respectively. Whereas, the values of AIFS, CWmin, and CWmax for ACo were
2ps, 7 slots, and 15 slots respectively. This operation increased video traffic’s ability to
access the wireless channel and transmit its packets faster.
The 20% of the bandwidth allocated by WRR to video traffic made the probability of
packet collision to be decreased. WRR implemented in QoS enhancement scheme
treated traffic based on their QoS requirements. This is unlike FIFO mechanism in the
legacy scheme where high and low priority traffic were treated in the same manner.
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Chapter 7 Improvements in quality 01 service in computer iveiwurKs
0.7
]Q oS enhancem ent scheme
M Legacy network_________
0.6
0.5
(a)
-0 .3
0.2
0.1
_ J
No. o f connections
(b)
16 32
No. of connections
(c)
(d) iu
K No. of connections
Figure 7-4. Bar chart representation of average packet loss with and without QoS
enhancement scheme for: (a) VoIP, (b) Video, (c) Best effort traffic, (d) Background
traffic.
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Chapter 7 improvements in i^uamy 01 service 111 computer neiwui
Figures 7-5 (a) - (d) show the overall QoS for VoIP, video streaming, best effort traffic,
and background traffic respectively when legacy and QoS enhancement schemes were
implemented.
The overall QoS was assessed using a supervised learning Multi-Layer Perceptron
(MLP) neural network (Dogman et al, 2012). In this technique, the QoS parameters (i.e.
delay, jitter, and packet loss ratio) for the transmitted applications were used as inputs to
the trained MLP. The MLP then quantified the overall QoS based on the values of
delay, jitter, and packet loss ratio taking into the account the QoS requirements for each
application. The overall QoS spanned between (0%-100%). A poor QoS did not exceed
33%, whereas a Good QoS could not be below 67%. More details about QoS
assessment using MLP can be found in section 6.3.4 (see chapter 6 ).
From Figures 7-5 (a) - (d), the overall QoS for all transmitted traffic were decreased as
long as the number of connections in the network was increased for both the legacy and
QoS enhancement schemes. For example, the values of overall QoS for the video
streaming under a light network load (i.e. 8 connections) were respectively 82.18%, and
42.46% using the QoS enhancement scheme and the legacy scheme. Whereas, the
overall QoS values for the video application using QoS enhancement scheme and the
legacy scheme under a heavy network load (i.e. 64 connections) were respectively
22.1%, and 15.01%. This was due to the gradual increase of delay, jitter, and packet loss
ratio for all transmitted traffic when the load on the network was increased. In both
schemes, as the number of active stations increased; the probability o f collisions
increased due to a high degree of competition between stations. This is in turn forced
the MAC protocol to retransmit the collided packets, and subsequently increased the
values of delay, jitter, and packet loss.
Also indicated from Figures 7-5 (a) - (d), is that the priority o f transmitted applications
could affect their overall QoS. For instance, the values of overall QoS for the VoIP
application under a medium network load (i.e. 32 connections) were respectively
75.46%, and 46.62% using the QoS enhancement scheme and the legacy scheme.
Whereas, the overall QoS values for CBR traffic using the QoS enhancement scheme
and the legacy scheme under the same network load were respectively 35.86%, and
21.55%. This is because VoIP was transmitted using the highest priority AC (i.e. ACo)
while the best effort traffic was transmitted by default using the lower priority AC (i.e.
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i,napier / i m p i u v c m c u i s 111 y u m u j ' u i o c i y iw c 111 V / u m j i u i c i n c in u iiu
AC2). The values of EDCA parameters for ACo facilitated its traffic to access the
channel and transmitted at faster rate. The small values of AIFS, CWmin, and CWmax
which were respectively 2 ps, 7 slots, and 15 slots enabled VoIP transmitted by ACo to
have a small waiting period before accessing the medium. Also, the large value of
TXOP, which was equivalent to 3.01 ms, allowed ACo to transmit multiple data frames
continuously during that particular time interval defined by TXOP.
However, the transmission mechanism, and the QoS requirements for a transmitted
application could have an impact on its overall QoS rather than the traffic priority.
Although the background traffic was transmitted by default using the lowest priority AC
(i.e. AC3), its overall QoS was outperformed on the overall QoS of the video traffic
which was transmitted using A Q . For instance, the values of the overall QoS for
background traffic under heavy network load (i.e. 64 connections) were respectively
49.18%, and 33.48% using the QoS enhancement scheme and the legacy scheme.
Whereas, the overall QoS values for the video traffic using the QoS enhancement
scheme and the legacy scheme under the same network load were respectively 2 2 . 1 %,
and 15.01%. This is because the background traffic was transmitted using TCP, whereas
the video traffic was transmitted using UDP. TCP is a connection-oriented protocol that
provides a reliable, ordered, and error-checked delivery of packets transmitted between
stations, while UDP is a connectionless Internet protocol with no acknowledgment of
packet delivery. Also, the background traffic is tolerant of the QoS parameters such as
delay, and jitter as compared with the video traffic. The acceptable QoS requirements
for video application in terms of delay, jitter, and packet loss ratio should not exceed
150 ms, 10 ms, and 3% respectively.
Figures 7-5 (a) - (d) indicate that the overall QoS of traffic transmitted for all ACs using
the proposed QoS enhancement scheme was higher as compared with the legacy
network. For example, the average QoS for the VoIP, video, best effort traffic, and the
background traffic, when the network load was light (i.e. 8 connections), were improved
by 80.9%, 93.5%, 7.8%, and 29.4% respectively. The improvements in QoS were due to
allocating the lower priority traffic to the next higher AC, in cases of zero PAR for the
higher priority AC and the fairness introduced by WRR in treating the traffic based on
their QoS requirements.
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Chapter 7 im provem en ts in v£uaiuy ui se rv ic e m \^oui|jui.ci n c m m iv s
No. of connections
■
No. of connections
No. of connections
No. of connections
Figure 7-5. Bar chart representation of overall assessed QoS with and without QoS
enhancement scheme for: (a) VoIP, (b) Video, (c) Best effort traffic, (d) Background
traffic.
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Figure 7-6 shows a visual comparison of sample images from the Foreman video
transmitted over a medium network load (i.e. 24 connections) using the proposed QoS
enhancement scheme. It was observed that the image quality with the QoS enhancement
scheme was higher than the image quality without it. The number of received I, P, and
B frames for the transmitted Foreman video was greater when using the QoS
enhancement scheme as shown in Table 7-4. This improvement was due to the shift of
the video packets from ACi to ACo in the absence of VoIP and the fairness of
distributing network resources by WRR.
(a) (b)
Figure 7-6. Visual comparison of reconstructed Foreman video using: (a) QoS
enhancement scheme, (b) legacy network scheme.
Table 7-4. The Amount of sent, received, and lost Foreman video frames.
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7.3 Summary
A new QoS enhancement scheme that consisted of a combination of an adaptive access
category (AC) traffic allocation algorithm implemented on the wireless side o f a
simulated network, and the Weighted Round Robin (WRR) scheduling implemented on
its wired side was devised and its performance was evaluated. The traffic allocation
algorithm assigned traffic of a lower priority AC to the next higher priority AC in the
absence of any higher priority traffic to further improve the performance of IEEE
802.l i e EDCA standard, whereas WRR managed to fairly allocate network resources
among transmitted traffic types based on their QoS requirements. The performance of
the proposed scheme was compared with the standard IEEE 802.1 le EDCA and FIFO
queuing mechanisms (i.e. WLAN-wired network legacy scheme). The proposed scheme
significantly improved the QoS for transmitted applications. The average QoS for VoIP,
video, best effort traffic, and background traffic increased from their original values by
72.5%, 70.3%, 44.5%, and 45.2% respectively. The QoS proposed scheme allowed an
end-to-end QoS to be set up which in turn provided an improved delivery for a variety
of applications in the context of wired-cum-wireless networks.
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Chapter 8 Microcontroller Board Implementation
8.1 Introduction
Network QoS assessment plays an important role in managing network resources and
ensuring that various applications receive an appropriate priority or sufficient resources.
Therefore, developing a hand-held system that accurately assesses QoS for different
applications is very valuable.
Following the measurement of the QoS parameters: delay, jitter and packet loss ratio of
multimedia applications, they are fed into the microcontroller board. The board then
analysed the parameters based on their transmission requirements and produced the
corresponding overall QoS. The performance of the system device is compared with
other QoS assessment methods (e.g. QoS assessment using Fuzzy Inference System
introduced by (Al-Sbou et al, 2006), and Neural Network QoS monitoring approach
proposed by (Dogman et al, 2012b). The results indicated that the developed system is
capable of accurately assessing QoS.
This chapter is organised as follows: the relevant studies about QoS monitoring tools
are discussed in section 8.2. Section 8.3 explains the MCB2300 KEIL ARM
microcontroller board. Section 8.4 outlines how the QoS assessment technique using
regression modelling was devised, and implemented on the MCB2300 KEIL ARM
microcontroller board. The experimental procedures are described in section 8.5. The
results are discussed in section 8 .6 . The conclusion is presented in Section 8.7.
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V /iiit p ie r o 1T1 IL1 uvuiili u n ti n u p itiii^ iiia iiu ii ui y vl7 njjvajiiivu^ ujo^vau
There are a number of QoS monitoring tools that have been proposed to monitor
network performance as in (Graham et al, 1998), (Zseby and Scheiner 2004), and
(Carvalho et al, 2009). A critical analysis of these tools can be found in Section 3.5,
Chapter 3. The existing network QoS monitoring tools have a number of shortcomings.
Some methods cannot determine directly the overall network QoS as in (Graham et al,
1998). Network managers have to do a variety of operations to assess the overall
network QoS. Other QoS monitoring tools are not stand-alone devices as in (Zseby and
Scheiner, 2004) and (Carvalho et al, 2009). From these limitations, the process of
monitoring QoS can be complicated, expensive, and time consuming.
Therefore, in this study, a hand-held system that accurately determines the overall
network QoS for multimedia applications was designed. The proposed device assessed
network QoS for multimedia applications taking into the account the QoS requirements
of these applications. A novel aspect of this study is that a microcontroller board with
integrated QoS monitoring tool is used.
The use of embedded systems in recent years has increased in such a way that it is used
in various mobile and multimedia applications. An efficient environment for
microcontroller applications is KEIL Microcontroller Development Kit (MDK-ARM)
(MDK-ARM, 2013). The features of KEIL MDK-ARM are its ease o f use, and the
manner it can be redesigned based on application's requirements without incurring
major Non-Recurring Engineering (NRE) costs.
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manufacturing applications.
The algorithm which was proposed by (Thangaraj et al, 2006) enhanced MCB 2300
KEIL microcontroller performance. Most applications that run on embedded platform
were constrained by limited built in memory available for storing the application
programs. Therefore, the proposed algorithm reduced the KEIL ARM memory
requirement needed by an application program.
The contribution of this study is the implementation of the regression modelling used to
assess QoS which was reported by (Dogman et al, 2012a) on the KEIL ARM MCB2300
microcontroller board. The proposed system could work independently to assess the
QoS of multimedia applications based on their transmission requirements (Dogman et
al, 2012d), and (Dogman et al, 2013).
The hardware components and interfaces of the Keil MCB2300 evaluation board are
shown in Figure 8-1 (MDK-ARM, 2013). The interfaces on the Keil MCB2300
evaluation board provide an easy access to the on-chip peripherals.
• ARM C/C++ Compiler: this component allows the user to write ARM
applications in C or C++ and then compiles C/C++ source files to have the
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Chapter 8 Microcontroller uoara implementation 01 y o a assessm eiu oysiem
• CAN Driver: MDK tool kit includes a CAN interface layer which provides an
easy and quick approach to implement a CAN network. It also provides a
standard programming API for all supported microcontrollers.
• Flash File System: the Flash File System allows the embedded applications to
create, save, read, and modify files in standard storage devices such as ROM,
RAM, Flash ROM, and SD/MMC/SDHC Memory Cards.
• USB Host and Device Interface: MDK-Professional provides USB Host and
USB Device support for embedded systems. The USB Host library is an
embedded USB stack supporting USB MSC (Mass Storage Class) and HID
(Human Interface Device) classes. The USB Device Interface uses standard
device driver classes that are available with all Windows PCs.
• TCP/IP Networking Suite: the full TCP/IP Networking Suite is designed for
ARM processor-based microcontrollers. It is highly optimized, has a small code
footprint, and gives an excellent performance.
• Graphic User Interface Library (GUI): the GUI Library is a fully featured
graphics suite that makes it possible to add graphical user interfaces to
embedded applications. It supports a large number of displays such as
monochrome, grayscale and colour LCDs.
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cnapier a Microcontroller r>oaru implementation ui yuo nsscMmcm opicm
USB Device
(Power)
Ethernet
CAN 2
CA N1
f - \ ( \
r \
/ \ f \
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8-3. This used the regression model (explained in Section 6.3.3, Chapter 6) on a KEIL
MCB2300 microcontroller board.
The QoS parameters of transmitted multimedia applications (i.e. delay, jitter, and packet
loss ratio) were obtained from the generated trace files of the simulated network. These
were used as inputs to KEIL MCB2300 microcontroller in order to quantify the overall
QoS. The following subsections explain the QoS assessment technique using regression
model and its implementation on the MCB2300 KEIL ARM board microcontroller.
Assessed QoS
In Section 6.3.3, Chapter 3, the QoS assessment technique using regression model was
proposed and evaluated.
The regression expression was developed based on QoS requirements listed in Table 2-
1 in order to provide the outputs that reflected the overall QoS. The QoS parameters and
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unapter » M icrocontroller u u a ru iiiipieiiieiiuuiuu ui yuo amcsmuciu ojaicui
the overall QoS were then arranged in matrices in order to feed them to the regression
model as follows:
Q o S i 1 D ± A PLR{ \b o] ei
Q o S 2 1 ^2 J2 PL R 2 h e2
= +
b2
Q o S n . .1 D n Jn P LRn_ -b 3- .e n .
where Dp /*, PLRi, QoSit i = 1,2, ...,n are delay, jitter, packet loss ratio, and overall
QoS respectively. The regression coefficients b0, blt b2, b3 were determined from the
recorded data using equation (2.12). The vector of residual (i.e. error terms) was then
calculated using equation (2.13). In this study, the means of vector o f residual produced
from regression formula for VoIP and video traffics were zeros (i.e. mean o f error terms
,i = 1,2, ...,n was zero). This implied that the estimated regression model
determined was as expressed in equation ( 8 . 1 ):
Q o S i = b0 + hi * Dj + b2 * Ji + b3 * P L R t (8.1)
where Q o S i , D it J if P L R i i = 1 ,2 ,...,n are the overall QoS delay, jitter, packet loss
ratio for ith packet respectively.
In this section, the QoS assessment technique using regression model is implemented on
Keil MCB2300 microcontroller. The code for developed QoS assessment technique
using regression model was written in C language in order to be implemented on
MCB2300 board.
The KEIL pVision was used to create, compile, download, debug, and run the C
program on the MCB2300 board. Once the C program operation was successfully
verified, the KEIL pVision downloaded the C code on the MCB2300 microcontroller.
The KEIL ARM MCB2300 microcontroller then was able to work as a standalone QoS
monitoring device.
Figure 8-4 shows the pseudo code of the C program implemented on the MCB2300
microcontroller. First, the QoS parameters of multimedia applications (i.e. delay, jitter,
and packet loss ratio) were fed into the QoS monitoring device (i.e. programmed
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Chapter 8 Microcontroller isoara implementation 01 assessment. opicm
MCB2300 board) through the SD connector. The Liquid Crystal Display (LCD) of the
QoS monitoring device indicated when the SD card was inserted and then the data
processing was initiated. The device then processed the QoS parameters file, read the
values of delay, jitter, and packet loss ratio, and assessed the overall QoS using equation
( 8 . 1).
Afterward, the device created an output file and recorded the values of delay, jitter, and
packet loss ratio, and their corresponding QoS value.
As soon as the QoS monitoring device completed the data processing, the overall QoS
was displayed on the LCD of the device.
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lv iiu u tu m iu u u D U d iu x i i i|i i c i u e i i u t u u i i u i ^ u o /iM C d d iu c iu
The queue management mechanism was Drop-Tail and the queue size was 50 packets.
A number of traffic types were transmitted over the simulated network. These were;
VoIP, video-conferencing, and best effort traffic. Constant Bit Rate (CBR) traffic was
adapted to model VoIP, videoconferencing, and data. VoIP modelled as G.711 voice
encoding scheme with 160 packet size and 64 kbps generation rate. The packet size of
the video traffic was 512 bytes and the inter-packet interval was 10 ms. This generated a
packet transmission rate of 384 kbps. The best effort traffic was modelled using
different packet sizes and the generation rates that corresponded to non
videoconferencing or VoIP usage. All traffic were transmitted using UDP.
The simulation time was 500 seconds and was repeated 10 times for each experiment.
Each time a different initial seed was used in order to randomly manage which node
transmitted first as all the nodes were requested to transmit at a given time. The
randomness introduced using the different seeds avoided the bias of random number
generation.
The transmitted traffic (VoIP, video-conferencing, and best effort traffic) were mapped
into three access categories (ACs) to represent the three priority levels as shown in
Table 8-1. VoIP had the highest priority, whereas best effort traffic had the lowest
priority (IEEE Computer Society, 2005).
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Chapter 8 Microcontroller isoara implementation 01 vjos Assessm ent system
^ < 5^ <S*
Figure 8-5. The Simulated Network.
This section explains the setup for the MCB2300 microcontroller board. This includes
the details about connecting and configuring procedures for the MCB2300 evaluation
board.
The following components were needed in this experiment in order to use the
MCB2300 Evaluation Kit:
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Chapter 8 Microcontroller board implementation 01 y o s Assessment system
The Keil ULINK-ME USB-JTAG Adapter which is shown in Figure 8-6 was first
connected to the MCB2300 board via its JTAG plug. Then the ULINK-ME USB-JTAG
Adapter was connected to the PC s USB port using a standard USB cable. This allows
the user to power to board, and download programs on the M CB2300 evaluation board.
Figure 8-7 shows the connection between the PC and the MCB2300 evaluation broad
using the Keil ULINK-ME USB-JTAG Adapter.
Figure 8-7. The connection between the PC and the MCB2300 using ULINK-ME
JTAG Adapter.
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Chapter 8 Microcontroller isoara implementation 01 Assessment system
The MCB2300 Evaluation Kit uses the pVision IDE software tool. In this experiment,
the MCB2300 Evaluation Board was connected directly to the KEIL pVision IDE
software which was installed on the PC via the KEIL ULINK-ME USB-JTAG Adapter.
The Keil pVision IDE software was the front-end used with ULINK-ME adapter to
create, download, and test the embedded application on the MCB2300 microcontroller
board. Therefore, no additional software was required to run the board. Figure 8-8
shows the snapshot of pVision IDE software tool (MDK-ARM, 2013).
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The project development cycle of the Keil pVision IDE is similar to any other software
development project. The development cycle started with creating a project for an
embedded application until downloading the application on target hardware. In this
experiment, the following stages were implemented to create an embedded application
with pVision IDE software:
1. Create a project, select the target device from the Device Data base, and configure
the tool settings: in this study, a new project file called "SD-File" was created from
168
Chapter 8 M ic ro c o n tro lle r Jtsoara im p ie m e n ia u o n u i y o a A b sessm cu i a j s ic u i
"New Project" option in "Project" Menu. pVision then created a new, empty project
file with the specified name. After a file name for the project was selected, a target
microcontroller was chosen. pVision had an option to select a target microcontroller.
The "Select Device for Target option" from "Project" Menu lists all the devices from
the pVision Device Database. The target device was MCB2300 microcontroller. This
step is very important, since pVision customizes the tool settings, peripherals, and
dialogs for a chosen device.
2. Create source files in C/C++ or Assembly: the "New" option from "File" Menu was
used to create a source code written in C language. The source code was saved in
"main.c" file. This file contained the pseudo code of QoS assessment shown in Figure
8-4. After a source file was created and saved, it was added to the project. The "SD-File"
project had a single source file called "main.c". This file contained a C function which
was used to read the QoS parameters of multimedia applications and assess the overall
QoS using regression model for QoS assessment.
3. Building the "SD-File" project with the Project Manager: There are several
commands were used from the "Project" Menu to compile and link the files in the
project. These were: "Translate File" command to compile the selected file in the
Project Workspace, "Build Target" command to compile files that have changed since
the last build and link them, and "Rebuild All Target Files" command to compile and
link all files in the project.
4. Downloading the "SD-File" program to MCB2300 board and test the linked
application: the "Configure Flash Tools" command from "Flash" Menu was used to
configure the target driver for flash programming. After the pVision IDE was
configured, the "Download" command from "Flash" Menu used the specified adapter
for Flash programming in order to flash the "SD-File" application program to the target
hardware (i.e. MCB2300 board). Blinking LED indicated successful download of the
program into the target hardware.
5. Debug the "SD-File" program application: the aim of debugging was to verify and
optimize the "SD-File" program application. The Options for Target - Debug dialog
was used to verify the configuration settings for the Debugger. The "Start/Stop Debug
Session" command from "Debug" Menu was used to debug "SD-File" program
application.
169
cnapier a lviicrucumruiier ouaiu xiiipiemciiuiuuii ui v uo /xascasiiiciu ajBicm
Due to the high sensitivity of multimedia applications to the QoS parameters as shown
in Table 2-1 (See Chapter 2), in this chapter, the QoS for VoIP, and video traffic were
measured and evaluated. The following subsections respectively evaluate the assessed
QoS for the transmitted traffic (i.e. VoIP, and video traffic).
After the traffic was configured and the network topology was simulated, the QoS
parameters (i.e. delay, jitter, and packet loss ratio) of VoIP were extracted from the
generated trace files of the simulated network. The QoS parameters were then used as
inputs to the KEIL MCB2300 microcontroller in order to quantify the QoS for VoIP.
Table 8-2 shows the QoS parameters of VoIP, and the evaluated QoS obtained from the
QoS Monitoring Device (QoS_MD), Fuzzy Inference System technique (FIS), and
Multi-Layer Perceptron neural network (MLP). The QoS assessment techniques as well
as the QoS monitoring device quantified the overall QoS based on the values of delay,
jitter, and Packet Loss Ratio (PLR) taking into the account the QoS requirements for
VoIP application. The overall QoS spanned between (0%-100%). Poor QoS did not
exceed 33%, whereas Good QoS could not be below 67%. More details about the
aforementioned QoS assessment techniques can be found in (Al-Sbou et al, 2006),
(Dogman et al, 2012b), (Dogman et al, 2012d), and (Dogman et al, 2013).
From Table 8-2, it can be observed that the QoS values reflected the corresponding QoS
parameters. In other words, as the values of QoS parameters increased, the values of
overall QoS decreased accordingly. For example, when the values of delay, jitter, and
packet loss ratio were 16.9ms, 0.8ms, and 0% respectively, the values of QoS for VoIP
were in Good region, i.e. 92.5% for QoS monitoring system, 89.6% for FIS based
technique and 96.9% for MLP neural network based technique.
170
unapter a M icrocontroller n o a r a iiiip ieiiieiiia iio ii 01 A ssessm ent o p ic in
In contrast, when the values of QoS parameters were high (i.e. 422.3ms for delay, 5.0
ms for jitter, and 6% for packet loss ratio), the corresponding QoS were at the Poor
level, i.e. 1.8% for QoS monitoring system, 9.3% for FIS based technique, and 4.2%
using MLP neural network. This was due to the gradual increase o f delay, jitter, and
packet loss ratio for VoIP traffic when the load on the network was increased. As the
number of active stations increased; the probability of collisions increased due to the
high degree of competition between stations. This is in turn forced the MAC protocol to
retransmit the collided packets, and subsequently increased the values of delay, jitter,
and packet loss.
Table 8-2. QoS Parameters, and the evaluated QoS of VoIP application using QoS
Monitoring Device (QoS_MD), Fuzzy Inference System (FIS), and Multi-Layer
Perceptron (MLP).
As shown in Table 8-2, the results obtained from QoS monitoring device were
compared with other QoS assessment methods (i.e. QoS assessment using FIS technique
introduced by (Al-Sbou et al, 2006), and MLP neural network QoS monitoring
approach proposed by (Dogman et al, 2012b)). The correlation coefficient (R ) was used
to evaluate the accuracy of the proposed device compared with the other QoS
assessment techniques. When using the correlation coefficient, the magnitude of R is
between 0 and 1. The magnitude closest to 1 indicates a perfect correlation, whereas a
171
Chapter 8 Microcontroller noara impiemeniauon 0 1 yoa Assessm ent aysiem
correlation less than 0.5 would be described as weak correlation. More details about
measuring accuracy using correlation coefficient can be found in Section 6.3.5 (see
Chapter 6)
In this study, the values of the correlation coefficients between QoS determined using
the monitoring tool, and the other QoS assessment methods were: 0.97 for FIS
technique, and 0.99 for MLP neural network.
From the values of evaluated QoS provided in Table 8-2, it can be concluded that all
approaches provided results which were closely comparable. Although some outputs
were slightly different, they were in the same QoS region (i.e. Poor, Average, or Good).
The discrepancies were due to the fact that each method followed a different scheme to
determine QoS. However, the values of QoS obtained from the proposed QoS
monitoring device spanned between 1-100%, whereas the range o f QoS values produced
by FIS was between 10-90%. These indicate that the QoS monitoring system was
accurately in its operation.
The QoS parameters of video traffic were also fed to the QoS monitoring device (i.e.
programed KEIL MCB2300 microcontroller board) to assess the overall QoS. The
extracted delay, jitter, and packet loss ratio o f video traffic were combined by the QoS
monitoring device which in turn produced its overall QoS.
As shown in Table 8-3, the QoS parameters of video application were processed using
QoS monitoring system, Fuzzy Inference System technique (FIS), and Multi-Layer
Perceptron neural network (MLP). The QoS assessment processes for these techniques
including the QoS monitoring system were based on the QoS requirements for video
application. A good overall QoS which ranged between 67-100%) corresponded to low
value of QoS parameters (i.e. delay < 1 5 0 ms, jitter < 1 0 ms, and packet loss ratio <
1%), An average QoS (i.e.33% < QoS < 67%) corresponded to medium QoS
parameters (i.e. 150 < delay < 400 ms, 10 < jitter < 20 ms, and 1% < packet loss ratio
< 2%), and high QoS parameters (i.e. delay > 400 ms, jitter > 20 ms, and packet loss
ratio > 2%) corresponded to a poor QoS (i.e. QoS < 33%).
Therefore, it can be observed from Table 8-3 that the values of evaluated QoS are
reflecting the corresponding QoS parameters. Low values of QoS parameters produce
172
cnapter 8 M icrocontroller n o a ra im plem en tation 01 v jo s A ssessm ent sy stem
high values of overall QoS and vice versa. For example, when the values of delay, jitter,
and packet loss ratio were 55.2ms, 11.32ms, and 0% respectively, the values of
expected QoS for video application were 77.14% when QoS monitoring device was
used, 80.69% when FIS technique was used, and 81.73% when using MLP neural
network. The expected QoS values were all in Good QoS region. In contrast, when the
values of QoS parameters were high (i.e. 533.4ms for delay, 24.5ms for jitter, and 3%
for packet loss ratio), the corresponding QoS were in poor region. The values of QoS
were 7.81% using QoS monitoring device, 9.81% using FIS technique, and 5.27% using
MLP neural network.
Table 8-3. QoS Parameters, and the evaluated QoS of video application using QoS
Monitoring Device (QoS_MD), Fuzzy Inference System (FIS), and Multi-Layer
Perceptron (MLP).
From Table 8-3, the results obtained from the QoS monitoring device, QoS assessment
using FIS technique, and MLP neural network QoS monitoring approach were
comparable. The correlation coefficient (R ) was used to evaluate the accuracy of the
proposed QoS device as compared with the other QoS assessment techniques.
In this study, the values of the correlation coefficients between QoS determined using
the monitoring system, and the other QoS assessment methods were: 0.98 for FIS
technique, and 0.99 for MLP neural network.
173
<^napier o iviicrucumruiicr dusiu liiipiemeiiutuuii ui y u j rtsstssmciu oyaicm
From the values of evaluated QoS obtained from the QoS monitoring system, QoS
assessment using FIS technique, and MLP neural network QoS monitoring approach as
provided in Table 8-3, it can be concluded that all approaches provided results which
were closely comparable. Nevertheless, some outputs were slightly different. These
divergences were because each method followed a different scheme to determine QoS.
However, the values of QoS obtained from the proposed QoS monitoring system
spanned between 1-100%, whereas the range of QoS values produced by FIS was
between 10-90%. This indicates that the QoS monitoring device could be more accurate.
8.7 Summary
In this study, a portable hand-held system to assess the QoS for multimedia applications
was designed and evaluated. Our developed QoS assessment technique which was
based on the regression model was implemented on the MCB2300 KEIL ARM
microcontroller board. The proposed system analysed the QoS parameters for
multimedia applications to measure the overall QoS. The QoS parameters (i.e. delay,
jitter, and packet loss ratio) were fed into the proposed device which in turn produced a
single value that represented the overall QoS.
The QoS assessment results were highly correlated with results obtined from a number
of previously developed QoS assessment methods. This indicated the correctness of the
developed system in monitoring QoS.
Further discussion about the results obtained from the developed approaches in this
thesis will be presented in the next chapter. Chapter 9 will conclude the thesis and
provide recommendations for future work.
174
Chapter 9 Conclusions, and Future Work
9.1 Conclusions
In this research, network QoS management referred to evaluation and improvement of
QoS provided by wired and wireless computer networks. Therefore, the main focus was
on development of techniques to evaluate QoS in multimedia networks and the use of
this information as part of network management to improve its performance.
The main QoS parameters of VoIP traffic (i.e. throughput, delay, jitter, and packet loss
ratio) with their sampled versions using adaptive and non-adaptive sampling techniques
were discussed in chapter 5. It was concluded from the findings in chapter 5 that the
sampled versions of throughput, delay, jitter, and packet loss ratio obtained using the
adaptive statistical sampling approaches were closer to the actual population than the
non-adaptive sampling approaches (i.e. systematic, stratified, and random sampling).
For instance, the bias values of sampled jitter versions obtained from adaptive sampling
based on fuzzy approach, linear adjustment approach, and quarter adjustment approach
were closer to zero as compared with non-adaptive sampling approaches as shown in
Figure 5-12 (See chapter 5).
This concludes that the developed adaptive statistical sampling methods were more
effective than conventional sampling methods in representing the traffic. This is because
the sample interval was adjusted during the sampling process in case of adaptive
175
Chapter 9 conclusions, ana ruiure vvo i k
Chapter 6 of this research is about development of techniques to analyse and assess QoS
parameters of multimedia networks accurately in real time. The multimedia QoS
information collected by adaptive statistical sampling techniques was considered.
Two innovative QoS evaluation approaches that combine analysis and measurement
techniques were developed. The first approach combined FCM and regression model to
analyse and assess QoS of multimedia applications in a simulated network. The second
approach analysed and assessed QoS in multimedia applications using a combination of
supervised and unsupervised neural networks.
The contribution of chapter 6 was to analyse and classify network QoS parameters (i.e.
delay, jitter, and packet loss ratio) using either Fuzzy C-means (FCM) or Self
Organizing Maps (SOM). Another contribution was to propose QoS assessment
techniques. The proposed techniques assess QoS in a manner similar to human subjects
and quantified the QoS without the necessity for complex mathematical models. Also,
the proposed QoS assessment techniques did not add significant extra load to the
network. The proposed assessment techniques were based on the traffic generated from
the proposed analysis techniques. A regression model was developed and a multi-layer
perceptron (MLP) was trained to combine the QoS parameters (i.e. delay, jitter, and
packet loss ratio) for each QoS class identified by SOM or FCM to estimate the overall
QoS. Both regression model and MLP were capable of combining QoS parameters (i.e.
delay, jitter, and packet loss ratio) to provide overall QoS.
The accuracy of QoS evaluation approaches was examined using typical values of QoS
parameters for VoIP, and a video. The findings of chapter 6 showed that FCM and
Kohonen network classified the values of QoS parameters of transmitted VoIP and
video into clusters representing Low, Medium, and High values of QoS as illustrated in
Figures 6-8, 6-9, 6-12, and 6-13. The regression model and MLP in turn combined the
176
uiajJier y VsUHClllMUIla, illlU CU1UIC VVU11V
QoS parameters (i.e. delay, jitter, and packet loss ratio) for each centre of generated
clusters and produced a single value that represented the overall QoS. The overall QoS
was an accurate indication of network performance as indicated in Tables 6-3, and 6-4.
Chapter 7 of this thesis considers the use of QoS information as part of multimedia
network management to improve its performance. Therefore, another object of this
study was about deployment network QoS enhancement which can be an effective
solution for multimedia applications to be shared under finite network resources.
A new QoS enhancement scheme for WLAN-wired networks was proposed. The
devised enhancement scheme consisted of an adaptive Access Category (AC) traffic
allocation algorithm which was incorporated into the network's wireless side to improve
the performance of IEEE 802.1 le Enhanced Distributed Channel Access (EDCA)
protocol, and a Weighted Round Robin (WRR) queuing scheduling mechanism that was
incorporated into the wired side of the network.
The algorithm considered the Packet Arrival Rate (PAR) of the uplink and downlink
traffic for each access category (AC), in order to allocate traffic to an appropriate AC.
PAR was chosen because it is an important QoS parameter that affects most real-time
and non-real-time applications. In addition, PAR can be easily computed while
applications are being transmitted. This in turn facilitated quick allocation o f the arrived
traffic to the most appropriate AC. Once PAR values for the up-link and down-link
traffic for each AC were determined, the algorithm dynamically allocated the traffic of a
lower priority AC to the next higher AC, when the higher AC was not receiving traffic
at the time. On the wired side of the network, a weighted round robin (WRR) shared the
network resources, based on the traffic’s quality o f service requirements.
The performance of the proposed scheme was compared with the standard IEEE
802.1 le EDCA and FIFO queuing mechanisms (i.e. WLAN-wired network legacy
scheme). The incorporation of the scheme enhanced the performance o f the WLAN-
wired network and significantly improved the QoS for transmitted applications. The
average QoS for VoIP, video, best effort traffic, and background traffic were increased
from their original values by 72.5%, 70.3%, 44.5%, and 45.2% respectively.
The QoS scheme proposed allowed an end-to-end QoS to be set up. This in turn
provided an improved delivery of a variety of applications in the context of wired-cum-
wireless networks.
177
Chapter 9 i^unciusiuus, aim r u iu re v t u i iv
In chapter 8, a network QoS monitoring device was designed and evaluated. The
existing QoS assessment devices do not determine directly the overall network QoS.
Network managers have to carry out a variety of tasks to assess the overall network
QoS. This makes the process complicated, expensive, and time consuming. Therefore,
developing a hand-held system that accurately assesses QoS for multimedia applications
is very valuable. The proposed QoS monitoring device used the QoS assessment
approach which was based on regression model which is described in section 6.3.3,
Chapter 6. The QoS assessment approach was implemented on the MCB2300 KEIL
ARM microcontroller board to design a hand-held device that assessed QoS of
multimedia applications.
The QoS parameters (delay, jitter and packet loss ratio) for multimedia applications
were fed into the proposed device to determine their overall QoS. The QoS monitoring
device analysed to QoS parameters of multimedia applications based on their QoS
requirements, and then produced an output that reflected their overall QoS. The
performance of the proposed device was compared with other QoS assessment
methods, it was observed from the findings that the overall QoS values obtained from
the proposed device were highly correlated with the QoS values obtained from QoS
assessment using Fuzzy Inference System introduced by (Al-Sbou et al, 2006), and
Neural Network QoS monitoring approach proposed by (Dogman et al, 2012b). The
obtained results indicated the effectiveness of the developed device in monitoring
multimedia QoS accurately.
178
u ia p ie r y cunciusiuiis, aim r u i u r e vvorK
CAC is to determine whether a new traffic should be admitted into the network. This
operation depends on many factors. The factors could include the overall QoS, QoS
requirements for new traffic, and the state of the network. Consequently, the QoS
requirements of admitted traffic will be satisfied.
• Proposed Adaptive Sampling Techniques and QoS Evaluation Methods over
other Packet Networks: Although the focus of this thesis was on wired-wireless
networks, the sampling approaches and the QoS evaluation methods can be applied
to other networks such as Mobile Ad hoc Networks (MANETs). This is because
adaptive sampling and QoS evaluation approaches are based on the traffic QoS
requirements, and measured QoS parameters.
179
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