WiMAX AND WLAN NETWORKS FOR VOICE OVER IP APPLICATION

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PERFORMANCE EVALUATION OF
INTEGRATED WiMAX AND WLAN
NETWORKS FOR VOICE OVER IP
APPLICATION

Abbreviations
AAA Authentication, Authorization and Accounting
ADPCM Adaptive Differential Pulse Coded Modulation
AES-CCM Advanced Encryption Standard Counter with
CBC MAC
AMC Adaptive Modulation and Coding
AP Access point
ARQ Automatic Repeat Request
ASN Access Service Network
AWGN Adaptive White Gaussian Noise
BE Best Effort Service
BPSK Binary Phase Shift Keying
BSS Base Service Set
BWA Broadband Wireless access
CBR Constant Bit Rate
CID Connection Identifier
CS Convergence Sub-layer
CS-ACELP Conjugate Structure Algebraic-Code Excited
Linear Prediction
CSMA/CA Carrier Sense Multiple Access/ Collision
Avoidance
CSN Connectivity Service Network
CCA Clear Channel Assessment
DBPSK Differential Binary Phase Shift Keying
DCF Distributed Coordination Function
DCME Digital Circuit Multiplication Equipment
DHCP Dynamic Host Control Protocol
DL Downlink
DLC Data Link Control Layer

DL-MAP Downlink Map
DOCSIS Data over cable service interface specification
DQPSK Differential Quadrature Phase Shift Keying
DSL Digital Subscriber Line
DSSS Direct Sequence Spread Spectrum
EAP Extensible Authentication Protocol
ertPS extended real time Polling Service
ESS Extended Service Set
FDD Frequency Division Duplexing
FHSS Frequency Hop Spread Spectrum
FTP File Transfer Protocol
GFSK Gaussian Frequency Shift Keying
GRE Generic Routing Encapsulation
IETF-EAP Internet Engineering Task Force-Extensible
Authentication Protocol
IEEE Institute of Electrical and Electronic Engineers
IP Internet Protocol
IR Infra Red
ISI Inter Symbol Interference
ISM Industrial, Scientific and Medical
ITU-T Telecommunication Standardization Sector of the
International Telecommunications Union
LAN Local Area Network
LD-CELP Low-Delay Code Excited Linear Prediction
LLC Logical Link Control
LOS Line Of Sight
MAC Medium Access Control
MAC CPS MAC Common Part Sub-layer
MN Mobile Node

MOS Mean Opinion Score
MS Mobile Station
nrtPS non-real time Polling Service
NSP Network Service Provider
NWG Network Working Group
OFDMA Orthogonal Frequency Division Multiple Access
PC Point coordinator
PCF Point Coordination Function
PHY Physical layer
PLCP Physical Layer Convergence Protocol
PMD Physical Medium Dependent
PMKv2 Privacy and Key Management Protocol version 2
PPP Point to Point Protocol
PSTN Public Switched Telephone Network
PTM Point To Multipoint
PTP Point To Point
QAM Quadrature Amplitude Modulation
QoS Quality of Service
QPSK Quadrature Phase Shift Keying
RLC Radio Link Control
rtPS real time Polling Service
RTS/CTS Request-To-Send/ Clear-To-Send
SDU Service Data Units
SIP Session Initiation Protocol
SISO Single Input Single Output
SONET Synchronous Optical Network
SS Subscriber Station
TDD Time Division Duplexing

TDM Time Division Multiplexed
TDMA Time Division Multiple Access
TEK Traffic Encryption Key
UGS Unsolicited Grant Service
UL Uplink
UL-MAP Uplink Map
VBR Variable Bit Rate
VoIP Voice over IP
Wi-Fi Wireless Fidelity
WiMAX Worldwide Interoperability for Microwave
Access
Wireless MAN Wireless Metropolitan Area Network
Wireless HUMAN Wireless High-Speed Unlicensed Metropolitan
Area Networks
WLAN Wireless Local Area Network
WWAN Wireless Wide Area Network

Contents
Chapter 1: Introduction……………………………………………..1
1.1 Introduction………….…………………..……………………….2
1.2 Voice over IP (VoIP).……………………………………………4
1.3 Problem Statement….……………………………………………9
1.4 Related Work……….……………………..……………………10
1.5 Thesis outline……….…………………..………………………10
Chapter 2: Effect of Terrain and Pathloss on Wireless Networks.12
2.1 Introduction…………………………………………………..…13
2.2 Background………………………………………………..……13
2.2.1 Free Space Propagation Model……………………………….…...13
2.2.2 Erceg‟s Suburban Fixed Model………………………………........14
2.2.3 Outdoor-to-Indoor and pedestrian pathloss Model…………..……16
2.2.4. Vehicular environment…………………………………….…..….16
2.3. Simulation Setup…………………………………………….....16
2.4. Simulation Results and Discussion……………………….……18
2.4.1 Pathloss……………………………………………………..……..18
2.4.2 Average Throughput………………………………………………19
2.4.3 Average Packet end to end delay……………………………...….20
2.5 Conclusion…………………………………………………..….21
Chapter 3: VoIP over WiMAX Network…………………………..22
3.1 Introduction……………………………………………………..23
3.2 Background of IEEE 802.16 and WiMAX …………………….23

3.3 Overview of 802.16 MAC Layer………………….……………25
3.4 Overview of 802.16 PHY Layer………………………………..28
3.5 WiMAX Network Architecture…………………...……………29
3.6 Experimental Setup……………………………………………..31
3.6.1 Scenario 1: Simulation Setup and Results………………………....31
3.6.1.1 Average Jitter…………………………………………....32
3.6.1.2 Average Packet End to End Delay……………………...34
3.6.1.3 Average MOS…………………………………………..35
3.6.2 Scenario 2: Simulation Setup and Results…………………………35
3.6.2.1 Average Jitter and packet end to end delay……………..37
3.7 Conclusion…………………………………………...………….38
Chapter 4: Voice over WLAN……………………………...…...….39
4.1 Introduction……………………………………………...……...40
4.2 Background of IEEE 802.11 and WLAN……………….………40
4.3 Protocol Architecture…………………………………………...41
4.4 Overview of IEEE 802.11 MAC Layer…………………………43
4.4.1 IEEE 802.11 MAC: DCF and PCF………………………..………43
4.5 Overview of IEEE 802.11 PHY layer……………………...…...44
4.5.1 Frequency Hop Spread Spectrum (FHSS)…………………………44
4.5.2 Dynamic Sequence Spread Spectrum (DSSS)……………….……44
4.5.3 Infrared……………………………………………………….……45
4.6 WLAN System Architecture………………..……………….….45
4.6.1 Infrastructure mode……………………………………45
4.6.2 Adhoc mode…………………………………………...46

4.7.1.1 Average Jitter……………………………………………48
4.7.1.3 Average MOS…………………………………………..50
4.7.2.1 Average Jitter……………………………………………52
4.8 Conclusion………………………………………………………53
Chapter 5: Voice over WiMAX-WLAN Integrated Network……54
5.1 Introduction……………………………………………………..55
5.2 WLAN vs. WiMAX…………………………………………….56
5.3 Types of Integration…………………………………………….58
5.3.1 Loose Coupling……………………………………………………58
5.3.2 Tight Coupling………………………………………………….…58
5.4.1.1 Average Jitter……………………………………………60
5.4.1.2 Average MOS …………………………………………..61
5.4.1.3 Average Packet End-to-End Delay……………………...62
5.4.2 Scenario 2: Simulation Setup and results………………………….63
5.4.2.1 Average jitter………………………………...………….63
5.4.2.2 Average Packet End-to-End Delay……………………...64
5.6 Conclusion………………………………………………………65
Chapter 6: Development of Scenarios in OPNET ……..…………66
6.1 Why OPNET?..............................................................................67

6.2 Development of WiMAX in OPNET…………………………...67
6.2.1 Static Scenario……………………………………………………..68
6.2.2 Mobile Scenario……………………………………………………76
6.3 Development of WLAN in OPNET…………………………….82
6.3.1 Static Scenario……………………………………………………..82
6.3.2 Mobile Scenario……………………………………………………87
6.4 Development of WLAN-WiMAX Integrated Scenario in
OPNET……………………………………………………………...88
6.4.1 Static Scenario……………………………………………………..88
6.4.2 Mobile Scenario……………………………………………………90
Chapter 7: Comparative Analysis….………………………………92
7.1 Introduction……………………………………………………..93
7.2 Comparative Results……………………………………………93
7.2.1 Stationary network…………………………………………………93
7.2.2 Mobile network……………………………………………………97
7.3 Conclusion………………………………………………………99
Chapter 8: Conclusion and Future Work………………………..100
Bibliography

Chapter 1
Introduction
Chapter Outline:-
 Introduction

 Voice over IP (VoIP)

 Problem Statement

 Related Work

 Thesis Outline
1

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t.com
Introduction
1.1 Introduction
Wireless networking has become an essential part in the modern
telecommunication system. The demand of high speed data transfer with quality
has led to the evolution of technologies like WiMAX and WLAN and is still
increasing. Hence, new ways to enhance quality and speed of connectivity are
being searched for. With step towards the fourth generation communication
networks, integrated networks are coming into operation. Also voice over IP is
expected to be a low cost communication medium. The voice codecs are big
constraints which affect the quality of the voice in a network. Hence, before real
time deployment of VoIP over a network it is necessary to evaluate the voice
performance over varying networks for various codecs.
WLANs [1] are mostly designed for private wired LANs and have been
enormously successful for data traffic but voice traffic differs fundamentally
from data traffic in its sensitivity to delay and loss [2]. Voice over WLAN is
popular, but maintaining the speech quality is still one of many technical
challenges of the VoIP system. VoIP is spreading rapidly and there is need to
support multiple concurrent VoIP communications but WLAN support handful
number of users [3] [4].
The IEEE 802.11 MAC specifies two different mechanisms, namely the
contention-based Distributed Coordination Function (DCF) [1] and the polling-based
Point Coordination Function (PCF) [1]. The DCF uses a carrier sense multiple access
with collision avoidance (CSMA/CA) scheme for medium access and the optional
four way handshaking request-to-send/clear-to-send [1] mechanism (RTS/CTS).
Incapability of providing differentiation and prioritization based upon traffic type
results in providing satisfactory performance for best-effort traffic only, but inferior
support for QoS requirements posed by real time traffic. These requirements make
the DCF scheme a less feasible option to support QoS for VoIP traffic. The PCF
mode enables the polled stations to transmit data without contending for the channel.
Studies on VoIP over WLAN in PCF mode [5] shows that the polling overhead is
high with increased number of stations in a basic service set (BSS). This results in
excessive delay and poor performance of VoIP
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under heavy load conditions. Thus, both DCF and PCF have limited support for
real-time applications. Supporting VoIP over WLAN using DCF mode poses
significant challenges, because the performance characteristics of their physical
and MAC layers are much worse than their wired counterparts and hence
considered in our system.
WiMAX (Worldwide Interoperability for Microwave Access) [6][16] on the
other hand is designed to deliver a metro area broadband wireless access (BWA)
service. So, while wireless LAN supports transmission range of up to few hundred
meters, WiMAX system ranges up to 30 miles [6]. Unlike a typical IEEE 802.11
WLAN with 11Mbps bandwidth which supports very limited VoIP connections [4],
an IEEE 802.16 WiMAX with 70Mbps bandwidth [7] can support huge number of
users. These motivations led to study and comparison of the VoIP quality of service
in IEEE 802.11b WLAN and IEEE 802.16 WiMAX network.
IEEE 802.16 support 5 types of service classes, namely UGS (Unsolicited
Grant Service), rtPS (real time Polling Service), nrtPS (non-real time Polling
Service), BE (Best Effort Service), ertPS (extended rtPS service) [8]. UGS
supports fixed-size data packets at a constant bit rate (CBR). It supports real time
applications like VoIP or streaming applications but wastes bandwidth during the
off periods. rtPS supports variable bit rate(VBR) real-time service such as VoIP.
Delay-tolerant data streams such as an FTP is designed to be supported by the
nrtPS. This requires variable-size data grants at a minimum guaranteed rate. The
nrtPS is similar to the rtPS but allows contention based polling. Data streams,
such as Web browsing, that do not require a minimum service-level guarantee is
supported by BE service. BE connections are never polled but receive resources
through contention. ertPS was introduced to support VBR real-time services such
as VoIP and video streaming. It has an advantage over UGS and rtPS for VoIP
applications as it carries lower overhead than UGS and rtPS [9] and hence is
modeled in the system.
Since, WiMAX is expected to create the opportunity to successfully penetrate
the commercial barrier by providing higher bandwidth, establishing wireless
commons becomes an important factor. Also, bandwidth crunch and network
integration are some of the major technical and social challenges regarding the
future of the community-based Wi-Fi networks [10]. According to [10], the
foundation of the WiMAX PTP commons is the process of hot-spot
3

interconnection and integration. Instead of global Internet connectivity, many
current applications and businesses are expected to be better utilized by using the
localized Wi-Fi constellation.
With a step towards the next generation, it is expected that an integrated
network as shown in Figure 1.1, comprising of both the WiMAX and WLAN
network and using mobile nodes with dual stack is expected to provide a better
performance than a similar WiMAX or WLAN network.
FIGURE 1.1. Wi-Fi integration using WiMAX [10].
VoIP has been widely accepted for its cost effectiveness and easy
implementation. A VoIP system consists of three indispensable components,
namely 1) codec, 2) packetizer, and 3) playout buffer. Analog voice signals are
compressed, and encoded into digital voice streams by the codecs. The output
digital voice streams are then packed into constant-bit-rate (CBR) voice packets
by the packetizer. A two way conversation is very sensitive to packet delay jitter
but can tolerate certain degree of packet loss. Hence a playout buffer is used at
the receiver end to smooth the speech by removing the delay jitter.
Quality of noise sensitive VoIP is usually measured in terms of jitter,
MOS and packet end-to-end delay. Perceived voice with zero jitter, high MOS
and low packet end-to-end delay is considered to be the best. With the two
competing wireless networks namely WLAN and WiMAX, this paper analyses
the perceived voice quality as measured using OPNET simulation environment.
1.2 Voice over IP (VoIP)
Voice is analog and is converted to digital format before transmitting over
Internet. This process is called encoding and the converse is called decoding and
both are performed by voice codecs [11]. With bandwidth utilization becoming a
huge concern, voice compression techniques are used [11] to reduce bandwidth
consumption. Voice compression by a codec adds an additional overhead of
4

algorithmic delay. Thus, a codec is expected to provide good voice quality even
after compression, with minimum delay.
Table 1.1 shows the bandwidth requirements of some common codecs. G.711 is
the international standard for encoding telephone audio. It has a fixed bit rate of
64kbps. G.723 and G.729 are low bit rate codecs at the expense of high codec
complexity. G.723 is one of the most efficient codecs with the highest compression
ratio and is used in video conferencing applications. G.729 is an industry standard
with high bandwidth utilization for toll-quality voice calls [12]. G.726 uses ADPCM
speech codec standard, and transmits at rates of 16, 24, 32, and 40 kbps. G.728
officially codes speech at 16 Kbit/s using low-delay code excited linear prediction
[13]. For example, during a call using G.711 as codec, the amount of data transfer for
both uplink and downlink will be 87.2 x 2 = 174.4Kbps = 0.1703 Mbps = 10.21 MB
per minute. So, G.711 uses 10.21 Mb/min per VoIP call where as G.729 uses
0.5MB/min per voice call in the same way.
TABLE 1.1. Bandwidth Requirement of Some Common Codecs [11][14]
Codecs Algorithm Bandwidth Ethernet
(Kbps) Bandwidth
Usage (Kbps)
G.711 PCM(Pulse Code Modulation) 64 87.2
G.729 CS-ACELP (Conjugate Structure 8 31.2
Algebraic-Code Excited Linear
Prediction)
G.723.1 Multi Rate Coder 6.3 21.9
G.723.1 Multi Rate Coder 5.3 20.8
G.726 ADPCM(Adaptive Differential 32 55.2
Pulse Code Modulation)
G.726 ADPCM(Adaptive Differential 24 47.2
Pulse Code Modulation)
G.728 LD-CELP (Low-Delay Code 16 31.5
Excited Linear Prediction)
5

Figure 1.2. Packet Creation based on voice activity.
Moreover, recently voice codecs are developed to detect talk-spurt [15]
and silence lengths [15] within a conversation. Silence in a communication
period leads to packetization of the background noise and sending it over the
network. This causes bandwidth wastage. Usually, during a conversation we talk
35% of the time and remain quiet rest of the time [15]. With silence suppression
during the silence period, the codec does not send data as shown in Figure 1.2.
This decreases channel utilisation and thereby saves bandwidth.
Voice communication is noise sensitive. Noise causes the signal to reach
the destination with a lead or lag in the time period. This deviation is called jitter.
Lead causes negative jitter and lag causes positive jitter and both degrade the
voice quality. The time taken by voice to be transmitted from the mouth of the
sender to the ear of the receiver is called packet end-to-end delay. The packet
end-to-end delay should be very less for voice communication. Perceived voice
quality is typically estimated by the subjective mean opinion score (MOS), an
arithmetic average of opinion score. MOS of a particular codec is the average
mark given by a panel of auditors listening to several recorded samples. It ranges
from 1(unacceptable) to 5 (excellent). It depends on delay and packet dropped by
the network. The E-model, an analytical model defined in ITU-T
recommendation, provides a framework for an objective on-line quality
estimation based on network performance measurements like delay and loss and
application level factors like low bit rate codecs. The result of the E-model is the
calculation of the R-factor (best case 100 worst case 0) [5].
R = R0 – Is – Id – Ie + A. (1.1) Where R0 groups the
effects of noise, Is includes the effects of the other impairments related to the
quantisation of the voice signal, Id represents the impairment caused due to delay,
Ie covers the impairments caused by the low bit
6

rate codecs and packet losses. The advantage factor A compensates for the above
impairments under various user conditions. A is 10 for mobile telephony but 0 for
VoIP [5]. R0 is considered to be 94.77 and Is is considered to be 1.43 in OPNET
14.5.A. The relation between MOS and R-factor:
MOS = 1 + 0.035R + 7.10-6R(R – 60) (100 – R) (1.2) Packet networks
operate on packet switching principles; hence voice in an IP or WiMAX network
would be transmitted to the destination as a collection of
packets where each one might follow different routes, thus arrive at the
destination with different delays. Typically originating and terminating parts
would respectively be the in-house portions of an enterprise network. The part
lying in Internet or some public WAN (i.e. shared medium) is termed the core
network. The packet flow is assumed to be from left to right. For flows in the
opposite direction, the terminology of originating and terminating is
interchanged. Unless stated otherwise, in view of delay calculations, originating
and termination parts are regarded to be identical.
On the originating side, the analog voice signal is digitized into pulse
code modulation (PCM) signals by the voice codecs. Then the PCM samples are
compressed and converted into packet format, thus ready to be sent across the
net. For some network configurations, the edge router may also perform codec
and compression functions. In the subsections below, we attempt to describe
various kinds of delays together with their associated formulation in units of
milliseconds (ms).
o Look Ahead Delay: The codecs conforming to different standards may use
voice blocks of different sizes. The block size used by the G.729 A codec is 10
ms whereas; codecs of G.723.1 have block sizes of 30 ms [16]-[18]. The
present packet is checked while the previous one is being compressed. Note
that such an operation is not possible at the very beginning of transmission.
Thus, a fixed delay of 5 ms, called look-ahead delay, occurs in this section of
packet voice transmission [16]-[17].
o Encoding and Compression Delay: Generally the G.729 A type codec is used
in VoIP having a mean opinion score (MOS) value of 4.2 which almost
provides toll quality [19]. This type of codec has an 8 kbps modulation rate,
hence producing a 10 ms delay for encoding the packet [17]-[24].
7

o Packetization Delay: The time spent for packetization of encoded and
compressed data packets is called packetization delay. To support a good
quality voice call, the packetization delay should be less than 30 ms [17].
The packetization process commences after storing the packets into a
buffer. As long as the buffering time remains below 10 ms, the compression
and buffering periods will overlap and there will be no additional delays
introduced at the buffer [17]. Otherwise, the buffering time would exceed
the compression time, and the remaining period would add to the total delay
created during the buffering operation.
o Serialization delay (SD): Serialization of the data/voice packets to the core
network is going to be implemented at the ingress point of the originating
network. This creates an extra delay, which may be calculated as [19]
SD= (payload + 48) x 8 / Link Speed (1.3) The term “payload”
corresponds to the average number of bytes stored in payload of the packet
and 48 indicates the number of bytes added by real time protocol, user
datagram protocol and internet protocol (RTP/UDP/IP) headers including
some optimizations [17], [19].
o Network Delays: An IP packet suffers several other delays while traversing
the whole network. The sources of these delays are explained below:
o Switching Delay: The packets wait for 10 ms at each switch in the
originating network and 1 ms in the core network [17]. The total
switching delay becomes
Switching delay = NRo x 10 + NRc x 1 (1.4) Where NRo
represents the number of routers in the core network and the NRc
represents the number of routers in the core network. In the
simulation, the originating/terminating hop counts are both taken
as 1.
o Propagation Delay (PD): Assuming that a signal propagates with a
velocity of two thirds of the speed of light, each km of propagation
will contribute an extra 5 μs delay. The propagation delay may be
written as
PD = Distance (in km) x 5 x 0.001 (1.5)
o Data Queuing Delay (DQD): The packets are serial, so they are
served sequentially. The waiting time of each packet will be
8

different, depending on the packet size. Also, these queuing delays
will also vary depending on the transmitted IP packet size.
Because of these complications, data queuing delay is not easy to
formulate. In the simulation, IP is assumed to carry only data
packets. Therefore, the voice packet will have the highest priority
as there will be no video packets. In this way the data packets will
therefore jump ahead of the other packets and wait only for the one
that is currently being processed. Even with this assumption an
accurate prediction of queuing delay is still impossible, but may be
simplified to [19].
o Voice/Data Contention Delay (VDCD): Voice data contention
delay is expressed as
VDCD = (MTU + 48) / Link Speed (1.6)
The term MTU (maximum transferable unit) appears, because in
worst case assumption, the maximum payload size of the data
packet will be equal to MTU.
o Jitter Buffer Delay: Each packet serialized to the network will follow
a different route according to the status of network and routers at that
moment, so each packet will arrive to destination with a different
delay which is proportional to the maximum jitter size.
o Voice Decoding Delay: The decoding time for each packet at the receiver is
given by [19].
Voice Decoding Delay = Encoding Delay (1.7)
Since a G.729 codec imposes an encoding delay of 10 ms [17]-[23] our
decoding delay will simply be 10 ms for each packet.
The purpose of this modeling is to compare the performance parameters for
the voice codecs considering both with and without silence suppression in WiMAX
802.16d, WLAN 802.11b and their integrated network and thereby show that the
integration provides optimal network capacity and quality of service.
1.3 Problem Statement
With voice over IP coming into existence, the maximum utilisation of the spectrum
has also become a concern. To accommodate maximum number of users with
9

considerably good quality of voice has become a concern of the researchers.
Innovative methods and techniques are coming up each day to suffice the needs.
With the next generation communication networks coming into being our target
is to analyse of the performance of voice over IP over a WLAN-WiMAX
integrated 4G network. WLAN is chosen because it is a widely used low cost
technology and WiMAX is expected to outshine the conventional DSL cables for
providing broadband wireless access. Hence, it is expected that an integrated
WLAN-WiMAX
1.4 Related Work
[1], [2], [4], [5], [7] and [13] discusses the performance and capacity of the
WLAN networks while [8], [9], [16] discusses the capacity and performance of
WiMAX network. The idea of the voice codecs has been discussed in [15]. It also
discusses the concept of silence suppression in it. In [10] it is proposed that an
integrated architecture of WiMAX and WLAN is expected to perform better than
the conventional WiMAX or WLAN network. Among them [2], [7] and [13]
discusses the capacity improvement of WLAN network for voice traffic while
quality of the voice traffic has been taken care of in [4] and [5]. Papers [6] [8] [9]
have discussed the performance of the WiMAX network for voice over IP and
multimedia traffic. To the best of our knowledge, the performance of WLAN-
WiMAX integrated network has not been evaluated yet. In this thesis we provide
a comparative discussion of the performance of a WLAN-WiMAX integrated
network with respect to the upcoming application voice over IP.
1.5 Thesis Outline
The thesis is organised as follows:
 Chapter 2 discusses the performance of WiMAX network for various
pathloss models.

 Chapter 3 discusses the performance of a WiMAX network in both static
and mobile conditions for the application voice over IP

 Chapter 4 discusses the performance of a conventional WLAN network in
both static and mobile conditions for the application voice over IP.
10

 Chapter 5 discusses the performance of the WLAN-WiMAX integrated
network in both static and mobile conditions for the voice over IP
application.

 Chapter 6 discusses the simulation setup in OPNET to perform the
simulations in the aforesaid chapters

 Chapter 7 shows the comparative study of the network performances for
VoIP in both static and moving conditions

 Chapter 8 discusses the future work that can be done as an extension to
this thesis and finally concludes the thesis.
11

Chapter 2
Effect of Terrain and Pathloss on
Wireless Networks
Chapter Outline:-
 Introduction

 Background

Free Space Propagation Model
Erceg’s Suburban Fixed Model

Outdoor-to-Indoor and pedestrian pathloss Model
Vehicular environment

 Simulation Setup

 Simulation Results and Discussion
Pathloss
Average Throughput

Average Packet end to end delay

 Conclusion
12

Effect of Terrain and Pathloss on
Wireless Networks
2.1. Introduction
With various wireless technologies coming up, performance of the
wireless networks has become a concern. Mobile WiMAX is expected to be the
wireless technology of the next generation as it supports vehicular mobility with
broad coverage area. Among various other parameters which affect network
performance, the terrain feature on which the network is deployed and the path
loss model affects the network performance substantially. Air being the
communicating medium in wireless networks, any hindrance on the
communication path like buildings, trees, etc., affects the wireless signal
propagation. Hence, prior network deployment, analysis of the terrain is essential.
Voice over IP as discussed in Chapter 1 is expected to be the communication
medium of the next generation and File Transfer Protocol is one of the most
popular data transfer protocol. Hence, in this chapter the behaviour of various
propagation models in WiMAX network with respect to path loss, throughput and
delay for various terrain models with voice and FTP applications is studied.
2.2. Background
The common propagation models namely Free Space path loss model,
Suburban Fixed (Erceg), Outdoor to Indoor and Pedestrian Environment and
Vehicular Environment are discussed in this paper. The Suburban Fixed path loss
model is further subdivided in three categories based on the building and tree
density namely terrain type A, B and C. The models are briefly discussed in the
next sections.
2.2.1. Free Space Propagation model
The free space propagation model is mathematically given by:
13

(2.1)
Where, Prx is received power in watts and is a function of distance between
transmitter and receiver, Ptx is the transmitted power in watts, Grx and Gtx are the
gain of the receiving and transmitting antennas respectively, L is the system-loss
factor and is not related to propagation. It is usually greater than 1 and λ is the
wavelength in meters [25].
2.2.2. Erceg’s Suburban Fixed Model
The Erceg model is based on extensive experimental data collected at
1.9GHz in 95 macro cells of suburban areas across the United States [26]. This
model is a slope intercept model given by [26]:
(2.2) Where, PL is the
instantaneous attenuation, H is the intercept and is given by free space path loss
at the desired frequency over a distance of d0 = 100 meters:
(2.3)
Where, λ is the wavelength. The parameter is a Gaussian random variable over
the population of macro cells within each terrain category. It can be written as
[26] [27].
(2.4)
where hb is the height of the base station antenna in meters, σγ is the standard
deviation of , x is a zero-mean Gaussian variable of unity standard deviation N[0,
1], and e, g, k and σγ are all data-derived constants for each terrain category. The
shadow fading components varies randomly from one terminal location to
another within any given macro-cell. It is a zero-mean Gaussian variable and can
be expressed as [26] [27].
(2.5)
14

where y and z are the zero-mean Gaussian variables of unit standard deviation
N[0, 1], σ is the standard deviation of s, µσ is the mean of σ, and σσ is the
standard deviation of σ. µσ and σσ are both data-derived constants for each terrain
category. The numerical values of the above parameters are given in Table 2.1.
The correlation factors of the model for the operating frequency and for
the MS antenna height are given in [27] as:
for Terrain type A and
for Terrain type C
Where, f is the frequency in MHz and hr is the height is the MS antenna above
ground in meters.
The Terrain Type A is a hilly terrain with moderate to heavy tree density,
representing rural environments and has highest path loss. Terrain Type B is
characterized by either a mostly flat terrain with moderate to heavy tree density or a
hilly terrain with light tree density. Terrain Type C is a flat terrain with light tree
density and is associated with the lowest path loss for rural environments [27].
TABLE 2.1. NUMERICAL VALUES CONSIDERED FOR THE PARAMETERS [26]
Parameters Terrain Terrain Terrain
Type A Type B Type C
e 4.6 4.0 3.6
g m
1
0.0075 0.0065 0.005
k m 12.6 17.1 20.0
0.57 0.75 0.59
10.6 9.6 8.2
2.3 3.0 1.6
15

2.2.3. Outdoor-to-Indoor and Pedestrian Pathloss Model
This environment is characterized by small cells and low transmission
power. Base stations with low antenna heights are located outdoors; pedestrian
users are located on streets and inside buildings and residences.
(2.9) The above equation
describes the path loss in dB where R is the distance
between the base station and the mobile station in kilometres and „f‟ is the carrier
frequency of 2000 MHz for IMT-2000 band application [28].
2.2.4. Vehicular environment
This environment is characterized by larger cells and higher transmits power. The
path loss in vehicular environment in dB is given by:
hb) log10 R – 18 log10 hb – 21 log10 f + 80dB (2.1
where, R is the distance between the base station and the mobile station, and f is
the carrier frequency of 2000 MHz and Δhb is the base station antenna height in
meters measured from the average roof top level.
2.3. Simulation Setup
The Wireless Deployment Wizard of OPNET is used to deploy a 7 celled
WiMAX network, with multiple subscriber stations in the range of a base station as
shown in Figure 2.1. The base stations are connected to the core network by an IP
backbone. There is a server backbone containing the voice server which is
configured as the SIP server. The IP backbone is connected to the server backbone
via an ASN gateway. This node, configured as the ASN gateway supports the
mobility in the WiMAX network. The IP backbone, server backbone and the ASN
gateway together represent the service provider company network. The green
bidirectional dotted lines represent the generic routing encapsulation (GRE) tunnels.
The cell radius is set to 30 kilometres. There are 10 nodes under the base station 2. 5
of these nodes are communicating with 5 mobile nodes under base station 7 and other
5 are communicating with 5 mobile nodes under base station 3. Voice call of PSTN
quality is configured between these mobile nodes. The nodes participating in the
same session are connected by the blue bidirectional dotted lines. The mobile nodes
under base station 2 are configured to move at a speed of 50 km/hr in the path as
shown by the white lines in Figure 2.1. The remaining key
16
P
L
=
4
0
(
1
–
4
X
1
0-
3

network configuration parameters in OPNET are summarized as in Table 2.2 and
the attributes of the network components are shown in Table 2.3.
FIGURE 2.1. Network Model for WiMAX.
TABLE 2.2. NETWORK CONFIGURATION DETAILS
Network
Cell Radius
No. of Base Stations
No. of Subscriber Stations per BS
No. of Mobile nodes in the
network Speed of the mobile
nodes Simulation time
Base Station Model
Subscriber Station Model
ASN Gateway Model
IP Backbone Model
Voice Server Model
Link Model (BS-Backbone)
Link Model (ASN - Backbone)
7 celled WiMAX
network 30km
7
10
10
50 km/hr
600 sec
wimax_bs_ethernet4_slip4_router
wimax_ss_wkstn
ethernet4_slip8_gtwy ip32_cloud
ppp_server
PPP_DS3
PPP_SONET_OC12
17

TABLE 2.3. ATTRIBUTES OF THE NETWORK COMPONENTS
Attributes Value
Physical Layer Model OFDMA 20Mhz
MAC Protocol IEEE 802.16e
Multipath Channel Model ITU Vehicular A
No. of Transmitter per BS SISO
Traffic Type of Service Interactive Voice and Data
Scheduling Type ertPS, nrtPS
Application Voice (PSTN quality), FTP
Voice Codec G 711
FTP Load High
2.4. Simulation Results and Discussion
The performance metric used to analyse the network performance is path
loss, average throughput of the WiMAX network and packet end to end delay.
Using OPNET, the simulation is conducted for the common path loss models and
the three types of terrains: Terrain Types A, B, C of erceg path loss model. We
are interested to investigate the combined effect of terrain (buildings, trees, etc.)
and vehicular mobility on the Mobile WiMAX performance in OPNET. To
investigate the combined effect of terrain and vehicular mobility, the terrains are
simulated by choosing the terrain type in OPNET which is thereby selected by
the simulator, based on the actual location and surrounding terrain of the
transmitter-receiver pair.
2.4.1. Pathloss
Path loss models are broadly classified into four groups namely free
space, Suburban fixed, Outdoor to indoor and pedestrian, vehicular environment.
The path loss due to these models in a WiMAX network with mobile nodes
moving at a speed of 50 km/hr is modeled in OPNET modeler and the result is
shown in Figure 2.2.
18

FIGURE 2.2. Path loss due to various pathloss models in decibel.
From the graph it is observed that the path loss for outdoor to indoor and
pedestrian is the highest and the same for free space is the lowest. This is due to
the fact that the path loss value varies with the amount of reflection in the
communicating path. As a mobile node moves from outdoor to indoor the
number of reflections in the communicating path increases rapidly thereby
causing huge path loss. Also, we see that the path loss for free space is the least.
This is concluded in [27] that the path loss increases with the number of
obstructions in the communication path. Free Space implies a terrain with no or
very less obstruction. Hence, number of obstruction decreases and pathloss
thereby is minimum.
2.4.2. Average Throughput
This metric measure the amount of voice traffic and FTP traffic received
in bits per second on an average for each connection. From Figure 2.3, it is
observed that the throughput of the network with free space path loss model is the
highest and the same for outdoor to indoor and pedestrian is the lowest for both
voice and FTP traffic. This is due to the fact that as the density of obstacles
increases, the Line of Sight (LOS) gets affected. Thus the number of times the
signal gets obstructed and reflected is more. This results in increasing attenuation
and diffraction due to the building structures, trees or mountains. As the Line of
Sight (LOS) between the transmitting and receiving nodes decreases, it causes
delay. This results in packet loss thereby causing fall in the average throughput.
19

FIGURE 2.3. WiMAX Network Throughput in Bits per second
2.4.3. Average Packet end to end delay
This metric measures the time taken by each voice packet to travel from
the mouth of the transmitter to the ear of the receiver and each FTP packet from
the application layer of the sender to the application layer of the receiver, on
average for each connection. From Figure 2.4, it is observed that the packet end
to end delay of the network with free space as the pathloss model is the lowest
and the same for outdoor to indoor and pedestrian is the highest irrespective of
the application. This is due to the fact that as the density of the building
structures, trees or mountains increases, the number of times the signal gets
obstructed and reflected is more thereby increasing attenuation and diffraction.
Hence, the LOS between the transmitting and receiving nodes decreases. This
causes the mobile node to get disconnected from the base stations and thereby
perform re-registration [1] to get re-connected to a base station.
The registration procedure is time consuming and hence adds up to the
packet end to end delay. With flat terrain model and very less or nil obstructions,
the mobile node remains connected to some or the other base stations and thus
the network scanning and registration process is avoided thereby causing less
delay as shown in Figure. 2.4.
20

FIGURE 2.4. Packet end to end delay in seconds
2.5. Conclusion
The target of this work was to check the variation of the network
performance due to varying pathloss and terrain models. This work shows the
variation in the WiMAX network performance with varying path loss models
deployed over suburban areas of terrain types varying from hilly terrain with
moderate to heavy tree density to flat terrain with light tree density for Voice over IP
and File Transfer Protocol application with mobile nodes moving at a speed of 50
km/hr. The aim was to evaluate the effect of various path loss models over varying
suburban terrains, on the basis of average throughput and packet end to end delay
and path loss of Mobile WiMAX. The reduction in Line-Of-Sight due to the terrain
model of any area influences the network throughput directly by increasing the
attenuation and diffraction losses, and indirectly affects the packet end to end delay
by causing nodes to initiate network re-registration more frequently under fluctuating
cell coverage. This is already mentioned in [26] [27] [29]. We conclude that our
results tally with the already taken measurements of the various path loss models in
[26] [27] [29] and thus, free space pathloss model is chosen for the future works as it
is a basic pathloss models with all other parameters related to terrain and building
density set as constant.
21

Chapter 3
VoIP over WiMAX Network
Chapter Outline:-
 Introduction

 Background of IEEE 802.16 and WiMAX

 Overview of 802.16 MAC Layer

 Overview of 802.16 PHY Layer

 WiMAX Network Architecture

 Experimental Setup

Scenario 1: Simulation Setup and Results
Average Jitter

Average Packet End to End Delay
Average MOS

Average Jitter


 Conclusion
22

VoIP over WiMAX Network
3.1. Introduction
In recent years, internet access has moved to a new dimension. It is now
not restricted to Web browsing and emailing. Multimedia services including
Voice-over-IP (VOIP) and media streaming have become the expectation of the
next generation. To provide users with such connectivity, Broadband Wireless
Access (BWA) comes into the picture. It promises users to be provided with
megabit internet access seamlessly. One of the many technologies under BWA is
WiMAX (Worldwide Interoperability for Microwave Access). Based on IEEE
802.16 it has been designed to provide metro area broadband wireless access.
With 70 Mbps speed and over 50 miles of coverage area [30], WiMAX supports
mobility up to 70-80 miles/hr and is expected to be the replacement of cable and
Digital Subscriber Line (DSL) [31]. Hence, with an overview of the IEEE 802.16
BWA network, a discussion of the quality of voice achieved by using a WiMAX
network supporting both in static and moving nodes is done in this chapter.
3.2. Background of IEEE 802.16 and WiMAX
The IEEE 802.16 group was formed in 1998 to develop an air interface
standard for wireless broadband. The group‟s initial focus was the development
of a LOS based point-to-multipoint wireless broadband system for operation in
the 10-66 GHz millimeter waveband. The resulting standard – the original IEEE
802.16 standard completed in December 2001 – was based on a single carrier
physical (PHY) layer with a burst time division multiplexed (TDM) MAC layer.
Many of the concepts related to the MAC layer were adapted for wireless from
the popular cable modem DOCSIS (data over cable service interface
specification) standard.
The IEEE 802.16 group subsequently produced 802.16a, an amendment to
the standard, to include NLOS applications in the 2 GHz – 11 GHz band using
orthogonal frequency division multiple access (OFDMA), were also included.
Further revisions resulted in a new standard in 2004, called IEEE 802.16d – 2004
which replaced all prior versions and formed the basis for the first WiMAX
23

solution. These early WiMAX solutions based on IEEE 802.16d – 2004 targeted
fixed applications, and are referred to as Fixed WiMAX [32]. In December 2005, the
IEEE Group completed and approved IEEE 802.16e – 2005, an amendment to the
IEEE 802.16d – 2004 standard that added mobility support. The IEEE 802.16e
– 2005 forms the basis for the WiMAX solution for nomadic and mobile
applications and is often referred to as mobile WiMAX [33]. The basic
characteristics of various IEEE 802.16 standards are summarized in Table 3.1.
TABLE 3.1. Basic Characteristics of various IEEE 802.16 standards.
802.16 802.16d – 2004 802.16e – 2005
Completed
Completed June
Completed
Status December December
2004
2001 2005
10GHz-66GHz for
Frequency
fixed;
10GHz-66GHz 2GHz-11GHz 2GHz-11GHz for
Band
mobile
applications
Application Fixed LOS Fixed NLOS
Fixed and Mobile
NLOS
MAC
Point-to-
Point-to-Multipoint, Point-to-
Multipoint,
Architecture mesh Multipoint, mesh
mesh
Single carrier, 256
Single Carrier, 256 OFDM OR
Transmission Single Carrier OFDM or 2048 Scalable OFDM
Scheme only
OFDM
with 128, 512,
1024, or 2048
subcarriers
Modulation
QPSK,16QAM, QPSK, 16QAM, 64 QPSK, 16QAM,
64QAM QAM 64 QAM
Gross Data 32Mbps-
1Mbps – 75Mbps 1Mbps – 75Mbps
Rate 134.4Mbps
24

Multiplexing
Burst Burst TDM/ TDMA/ Burst TDM/
TDM/TDMA OFDMA TDMA/ OFDMA
Duplexing TDD and FDD TDD and FDD TDD and FDD
1.75 MHz, 3.5 MHz,7 1.75 MHz, 3.5
20MHz, MHz, 14 MHz, 1.25 MHz,7 MHz, 14
Channel 25MHz, MHz, 5 MHz, 10 MHz, 1.25 MHz,
Bandwidths 28MHz MHz, 15 MHz, 8.75 5 MHz, 10 MHz,
MHz
15 MHz, 8.75
MHz
WirelessMAN – SCa
WirelessMAN –
SCaWirelessMAN –
–
Air-interface WirelessMAN OFDM
WirelessMAN
OFDM
designation -SC WirelessMAN–
WirelessMAN –
OFDMS
OFDMA
WirelessHUMAN
WirelessHUMAN
WiMAX
None
256 – OFDM as Fixed Scalable OFDMA
implementation WiMAX as Mobile
WiMAX
3.3. Overview of 802.16 MAC Layer
IEEE 802.16 Medium Access Control (MAC) generally follows point to
multi-point (PMP) network topology with support for mesh topology.
From the reference model as shown in Figure 3.1 there are three sub
layers in the MAC:
Service Specific Convergence Sub-Layer (CS): providing any
transformation or mapping of external network data through CS SAP (CS
Service Access Point)
MAC Common Part Sub-layer (MAC CPS): classifying external network
service data units (SDUs) and associating these SDUs to proper MAC
service flow and connection identifier (CID). Multiple CS specifications
are provided to interface with various protocols.
Privacy (or Security) Sub-layer: supporting authentication, secure key
25

exchange, and encryption.
FIGURE 3.1. Reference Model of IEEE 802.16
Unlike the random access techniques of typical MACs of IEEE 802, IEEE
802.16 MAC is connection oriented and similar to time division multiple access
(TDMA). As a subscriber station (SS) enters the network, it creates one or more
connections with the base station (BS). It also performs link adaptation and
automatic repeat request (ARQ) functions to maintain the target bit error rate.
Further to support multimedia services, the IEEE 802.16 MAC may have to use
radio resources and provide quality of service (QoS) differentiation in services
which are not considered typical MAC functions. The five different types of
services to support QoS differentiation for different applications provided by
IEEE 802.16 are as follows [35]:
o Unsolicited Grant Services (UGS): UGS is designed to support fixed-size
data packets at a constant bit rate (CBR). It supports real time applications
like T1/E1 emulations but wastes bandwidth during the off periods or
silence periods. Hence, it cannot be used for Voice over IP without silence
suppression.
o Real-Time Polling Services (rtPS): It is designed to support variable
sized data packets, periodically like MPEG video and Voice over IP with
silence suppression.
o Non-Real-Time Polling Services (nrtPS): nrtPS is designed to support
non real time services that require data bursts of variable size on a regular
basis like FTP. It is similar to rtPS except for the contention based polling
scheme.
26

o Best Effort (BE): Data streams that do not require a minimum service-level
guarantee is supported by BE service. It counts typical data traffic like
Internet web browsing and FTP file transfer.
o Extended Real-Time Polling Service (ertPS): It is designed to support
variable rate real-time services such as Voice over IP and video streaming. It
has an advantageous over UGS and rtPS for VoIP applications because it
carries lower overhead than UGS and rtPS [9] and hence is considered in
our system.
To support OFDMA PHY, the MAC layer is responsible for assigning frames
to the proper zones and exchanging this structure information with the SSs in the
DL and UL maps. IEEE 802.16 MAC is connection oriented. As BS controls the
access to the medium, bandwidth is granted to the SSs on demand. At the
beginning of each frame, the BS schedules the uplink and downlink grants to
meet the negotiated QoS requirements. Each SS learns the boundaries of its
allocation under current uplink and sub-frame via the UL-MAP message. The
DL-MAP delivers the timetable of downlink grants in the downlink sub frame.
There are different bandwidth request schemes in WiMAX. For UGS, a
fixed amount of bandwidth is requested periodically at the set-up phase of uplink
and no bandwidth is requested explicitly after that. The unicast poll allocates
necessary bandwidth for a polled uplink connection. The broadcast polls are
issued by the BS to all uplink connections, while a truncated exponential back-off
algorithm is employed to resolve possible collisions in polling. Based on the
bandwidth requested and granted, the BS uplink scheduler estimates the
residential backlog at each uplink connection, and allocates future grants. An SS
scheduler must be implemented with each SS MAC, in order to re-distribute the
granted capacity to all its connections. However, note that IEEE 802.16 does not
specify scheduling algorithms that are left to manufacturers.
Similar to the concept of cellular layer - 2/3, IEEE 802.16 MAC has the
radio link control (RLC) to control PHY transition from one burst profile to
another, in addition to traditional power control and ranging.
Another important sub-layer in the IEEE 802.16 MAC is the security sub-
layer, and an improved version has been developed for the IEEE 802.16e. The
Privacy and Key Management Protocol version 2 (PMKv2) is the basis of Mobile
WiMAX security. Device and user authentication adopts IETF EAP protocol. The
27

traffic encryption follows the IEEE 802.11i using AES-CCM to protect traffic
data. The keys used to derive the cipher text are generated from the EAP
authentication. To avoid further attacks and hostile analysis, a periodic key
(TEK) refreshing mechanism enables improved protection. A three-way
handshake scheme in Mobile WiMAX optimizes the re-authentication
mechanism for fast handover by preventing man-in-the-middle-attacks.
To deal with mobility in Mobile WiMAX, IEEE 802.16e the MAC
specifies MAC layer handover procedure, while the exact handover decision
algorithm is not specifically defined.
Further details regarding the IEEE 802.16 MAC are discussed in [30] [32]
[33] [35] [9] [36] [37].
3.4. Overview of 802.16 PHY Layer
The Figure 3.2 shows the schematic representation of WiMAX
PHY layer. The diagram has basically three blocks which are
o Transmitter,
o Receiver and
o Channel.
FIGURE 3.2. Schematic Model of WiMAX PHY Layer
The transmitter consists of channel encoding, digital modulations, serial to
parallel conversion block followed by an OFDM modulation block and a parallel
to serial conversion block has been shown that has the specific purpose of
serializing the data bits.
The channel encoding block has “block encoder”, “convolutional encoder”
and “puncture vector block” associated with it. The redundancy added to the data
sequence is the sole purpose of the encoder block. In WiMAX scenario, we can
28

only opt for BPSK, QPSK, 16-QAM and 64-QAM modulation. WiMAX PHY
uses Adaptive Modulation and Coding (AMC) which takes into account the
channel SNR to dynamically select the proper modulation technique appropriate
for that channel condition to deliver the maximum throughput. Serial to parallel
conversion of the incoming bit becomes the critical set of operation that
determines the parallel transmission of data bits. OFDM modulation being the
key multiplexing technique in WiMAX helps the transmission of data bits at a
very high rate at a negligible amount of Inter Symbol Interference (ISI) with
minimal amount of packet loss and bit error.
The data transmitted through wireless channel reaches the receiver. The
channel which might be AWGN, Rayleigh etc. determines the effective channel
impairment introduced in the receiver. The receiver section does exactly the
opposite to the transmitter.
3.5. WiMAX Network Architecture
The IEEE 802.16e – 2005 standard provides the air interface for WiMAX
but does not define the full end-to-end WiMAX network. The WiMAX Forum's
Network Working Group (NWG) is responsible for developing the end-to-end
network requirements, architecture, and protocols for WiMAX, using IEEE
802.16e – 2005 as the air interface. The WiMAX NWG has developed a network
reference model to serve as an architecture framework for WiMAX deployments
and to ensure interoperability among various WiMAX equipment and operators.
The network reference model envisions unified network architecture for
supporting fixed, nomadic, and mobile deployments and is based on an IP service
model. Below is simplified illustration of IP-based WiMAX network
architecture. The overall network may be logically divided into three parts:
o Mobile Stations (MS) used by the end user to access the network.
o The Access Service Network (ASN), which comprises one or more base
stations and one or more ASN gateways that form the radio access
network at the edge. The ASN gateway typically acts as a layer 2 traffic
aggregation points within an ASN. Additional functions that may be part
of the ASN gateway include intra-ASN location management and paging,
radio resource management and admission control, caching of subscriber
profiles and encryption keys, AAA client functionality, establishment and
29

management of mobility tunnel with base stations, QoS and policy
enforcement, and foreign agent functionality for mobile IP, and routing to
the selected CSN.
o Connectivity Service Network (CSN), which provides connectivity to the
Internet, ASP, other public networks, and corporate networks. The CSN is
owned by the NSP and includes AAA servers that support authentication for
the devices, users, and specific services. The CSN also provides per user
policy management of QoS and security. The CSN is also responsible for IP
address management, support for roaming between different NSPs, location
management between ASNs, and mobility and roaming between
ASNs.
The network reference model developed by the WiMAX Forum NWG
defines a number of functional entities and interfaces between those entities.
Figure 3.3 below shows some of the more important functional entities.
o Base station (BS): The BS is responsible for providing the air interface to
the MS. Additional functions that may be part of the BS are micro
mobility management functions, such as handoff triggering and tunnel
establishment, radio resource management, QoS policy enforcement,
traffic classification, DHCP (Dynamic Host Control Protocol) proxy, key
management, session management, and multicast group management.
FIGURE 3.3. WiMAX Network Architecture [39]
30

3.6. Experimental Setup
To analyze VoIP in a network, it is necessary to study real life scenarios.
Hence, OPNET 14.5.A is chosen as the simulation tool as it will reflect the actual
deployment of the WiMAX network.
3.6.1. Scenario 1 – Static Nodes
Figure 3.4 shows the simulation setup used for WiMAX network. Using
the Wireless Deployment Wizard of OPNET a 7 celled WiMAX network, with
multiple subscriber stations in the range of a base station is deployed. The base
station is connected to the core network by a server backbone via an IP backbone.
The server backbone is further connected to the voice server which is configured
as the SIP server. The base station, IP Backbone, Server Backbone and the Voice
Server together represent the service provider company network. The cell radius
is set to be 30 km. The Base Station transmission power is set to 10W and the
same for subscriber station is set to 0.5W based on [35] as shown in Figures 3.5
and 3.6 respectively. The number of subscribers in cells 2 and 3 are 10 and Voice
over IP calls are setup between them in mesh using SIP.
FIGURE 3.4. Network Model for WiMAX in OPNET
31

FIGURE 3.5. Base Station Parameters
FIGURE 3.6. Subscriber Station Parameters
With the simulation setup as mentioned before, the voice codecs being
used for the Voice over IP calls are varied and the corresponding variation in
voice jitter, MOS (Mean Optimal Score) and Packet End-to-end delay are noted.
3.6.1.1. Average Jitter
Figure 3.7 shows the variation of jitter for the WiMAX network
without using silence suppression for various codecs. Perceived voice quality is best
if the jitter is zero. As shown in the figure, average voice jitter is almost 0 for the
voice codecs G 723.1 with both 5.3Kbps and 6.3Kbps and G 726 with 32Kbps
implying very good quality of voice whereas all other codecs shows some deviation.
A positive jitter of 0.000000925926 seconds is shown by G 711 while all others
show negative jitter of about 0.000001841621 seconds. Since, the bit rate of G 723.1
is 6.3 or 5.3 Kbps; it results in generation of small packets. As stated in [15] modem
and fax signals cannot be carried by G 723.1 and hence it can be used only for
narrow band communications. Like G 711, G 726 has its roots in the PSTN network.
Hence, it is expected to provide users with good quality of voice. It
32

is primarily used for international trunks to save bandwidth. Unlike G 711, G 726
uses 32Kbps to provide nearly the same quality of voice. This is because 32 Kbps
is its de facto standard [34].
To increase the number of users supported count, silence suppression
technique is important. As mentioned in Chapter 1, human speech during one
way conversation has 35% talk spurt length and 65% silence length. Silence
suppression prevents the packetization of the silence length and thereby save
bandwidth. But usage of silence suppression technique increases the positive
jitter considerably compared to the other voice codecs. This is shown in Figure
3.8. Hence, G 726 cannot be used in cases where silence suppression technique is
used irrespective of its performance in cases where silence suppression has not
been used.
Mean Jitter without silence suppression (sec)
0.0000015
0.000001
Time(sec)
0.0000005
0
G729 G711 G723_1_5_3 G723_1_6_3 G726_16 G726_24 G726_32 G726_40 G728_12_8 G728_16
-0.0000005
-0.000001
-0.0000015
-0.000002
Voice Codecs
FIGURE 3.7. Average voice jitter without Silence Suppression
Mean Jitter with Silence Suppression (sec)
0.03
0.025
Time(sec)
0.02
0.015
0.01
0.005
0
-0.005
G729 G711 G723_1_5_3 G723_1_6_3 G726_16 G726_24 G726_32 G726_40 G728_12_8 G728_16
Voice Codecs
FIGURE 3.8. Average voice jitter using Silence Suppression
33

3.6.1.2. Average Packet End to End Delay
As stated in Chapter 1, packet networks operate on packet switching
principles; hence voice in an IP or WiMAX network would be transmitted to the
destination as a collection of packets where each one might follow different routes,
thus arrive at the destination with different delays. Factors affecting packet end to
end delay include Look Ahead Delay, Packetization Delay, Serialization Delay,
Network Delay, etc. (Details covered in Chapter 1). Figure 3.9 and 3.10 shows, the
Packet End-to-End delay for the voice codec G 723.1 is the highest irrespective of
silence suppression. This is because; G 723.1 uses coding rate of 5.3 Kbps or 6.3
Kbps which results in the formation of packets of smaller size and larger count. Now
as the number of packets increases in the network, the congestion in the network
increases. Congestion directly affects the network packet delay and thus results in
increased packet end to end delay.
Mean Packet-End-to-End Delay without Silence Suppression
0.14
0.12
(sec)
0.1
0.08
Time
0.06
0.04
0.02
0
G729 G711 G723_1_5_3 G723_1_6_3 G726_16 G726_24 G726_32 G726_40 G728_12_8 G728_16
Voice Codecs
FIGURE 3.9. Average packet end to end delay without Silence Suppression
Mean Packet End-to-End Delay with Silence Suppression
0.14
0.12
(sec)
0.1
0.08
Time
0.06
0.04
0.02
0
G729 G711 G723_1_5_3 G723_1_6_3 G726_16 G726_24 G726_32 G726_40 G728_12_8 G728_16
Voice Codecs
FIGURE 3.10. Average packet end to end delay using Silence Suppression
Also the Look Ahead delay of G 723.1 voice codec is 7.5 msec [35] while the
same for G 729A is 5 msec and other considered voice codecs is 0 msec. On the
other hand the serialization delay of G 726 32Kbps is quite high. As mentioned in
Chapter 1 the serialization delay is directly proportional to the payload size and is
inversely proportional to the link speed. Since the test network has same link
34

configuration, the link speed is same for all codecs. Hence, depending on the
payload size, the variation in the serialization delay affects the total packet end to
end delay significantly.
3.6.1.3. Average MOS
Mean MOS without Silence Suppression
4
3.5
3
MOS
2.5
2
1.5
1
0.5
0
G729 G711 G723_1_5_3 G723_1_6_3 G726_16 G726_24 G726_32 G726_40 G728_12_8 G728_16
Voice Codecs
FIGURE 3.11. Average MOS without Silence Suppression
Average MOS with Silence Suppression
4
3.5
3
MOS
2.5
2
1.5
1
0.5
0
G729 G711 G723_1_5_3 G723_1_6_3 G726_16 G726_24 G726_32 G726_40 G728_12_8 G728_16
Voice Codecs
FIGURE 3.12. Average MOS with Silence Suppression
The Mean Optimal Score (MOS) as shown in Figures 3.11 and 3.12 is
independent of Silence Suppression. As mentioned in Chapter 1, MOS depends
on number of packets dropped. G 723.1 is a low bit rate codec which generates
packets of size 5.3 Kbps or 6.3 Kbps. This results in network congestion and
thereby packet drop. Hence, the MOS value for the voice codec G 723.1 is quite
low. Voice having MOS of 3 can be considered to be of considerable quality.
Hence, all other codecs considerable with respect to their MOS value.
3.6.2. Scenario 2 – Mobile Nodes
Figure 3.13 shows the simulation setup used for WiMAX network.
Using the Wireless Deployment Wizard of OPNET a 7 celled WiMAX network,
with multiple subscriber stations in the range of a base station is deployed. The
35

base station is connected to the core network by a server backbone via an IP
backbone and an ASN Gateway which controls the mobility of the mobile nodes. The
server backbone is further connected to the voice server which is configured as the
SIP server. The base station, IP Backbone, Server Backbone, ASN Gateway and the
Voice Server together represent the service provider company network.
Generic Routing Encapsulation (GRE) tunnel is setup between the ASN
gateway and the base stations to control the mobility of the mobile nodes (for
further details refer to Appendix A). The cell radius is set to be 30 km. The
mobile nodes are configured to move at a rate of 50 km/hr, 100 km/hr and 150
km/hr. The Base Station transmission power is set to 10W and the same for
subscriber station is set to 0.5W based on [35]. The number of subscribers in cells
2 is 10 and the same in cells 3 and 7 are 5 each. Voice over IP calls are setup
between them using SIP and are shown by the blue lines in the Figure 3.13.
With the simulation setup as mentioned before, the voice codecs being
voice jitter and Packet End-to-end delay are noted.
Due to the limitation of OPNET, Mean Optimal Score (MOS) could not be
estimated. Also only some of the voice codecs could be modeled. G 723.1 with 5.3
kbps, G 726 with 32 Kbps and G 728 with 16 Kbps could only be modeled.
FIGURE 3.13. Network Model for WiMAX
36

3.6.2.1. Average Jitter
Mean Jitter(sec)
0.02
0.018
0.016
0.014
J
i
t
t
e
r
(
s
e
c
)
0.012 150km/hr0. 008
50km/hr
0.01 100km/hr
0.006
0.004
0.002
0
G711 G723.1 G726 G728 G729
Voice Codec
FIGURE 3.14. Mean Jitter without Silence Suppression
Average Packet end to end delay
0.09
0.08
D
e
l
a
y
(
s
e
c
)
0.07
0.06
0.05
50km/hr
100km/hr
0.04
P
a
c
k
e
t
150km/hr
0.03
0.02
0.01
0
G711 G723.1 G726 G728 G729
Voice Codecs
FIGURE 3.15 Mean Packet end to end delay without Silence Suppression
G 711 codec has packet rate of 64 Kbps which is quite large compared to
that of G 723.1 which is about 5.3 Kbps. Thus, G711 has fewer number of
packets compared to G 723.1 for fixed amount of voice. For larger packets the
overhead due to header is small [38]. As a result, for scenario 1, the delay for G
711 is lesser compared to G 723.1. Moreover, in scenario 1, the nodes are fixed
hence the packets follow almost the same route and arrive in order more or less
but wireless networks, have a inherent property of discarding any packets
containing one or more erroneous bits. So the probability of discarding G723.1
packets are less and as a result the jitter is also minimum for this codec.
However in the mobile scenario, scenario 2, the probability of packet loss is
more and any packet loss in the physical layer is interpreted as delay by the upper
layers due to the hiding property of the MAC or link layer protocols [38]. Since,
37

the packet size of G 711 is larger; it has higher probability of suffering from
packet loss than G 723.1 thereby having higher delay than G 723.1. This is shown
in Figure 3.15. As the subscriber stations become mobile, their supporting base
stations changes frequently due to handover and the mobile nodes get distributed
between different base stations depending upon their trajectory. Hence, the
packets may follow different path and arrive out-of-order. Since, the number of
packets for G 723.1 is much more than G 711, this problem is more severe than G
711. As a result it suffers from maximum jitter. This is shown in Figure 3.14.
3.7. Conclusion
The target of this work is to observe the variation of jitter and packet end
to end delay in a WiMAX network under various mobility scenarios. This work
shows the variation of the mentioned parameters with respect to a stationary
network having all stationary nodes and with respect to a mobile network with
varying speed of 50km/hr, 100km/hr and 150km/hr. Average packet end to end
delay is more in case of stationary nodes than mobile nodes. Figures 3.7, 3.8 and
3.14 show the variation of jitter in the above mentioned networks. As Figure 3.7
and 3.8 shows, the jitter in a stationary network increases considerably when
silence suppression is activated in the network and it increases further when
mobility is introduced in the network. This is because as a node becomes mobile
it undergoes handover. This creates inconsistency in the order of the packets
delivered thereby jitter. Figures 3.9, 3.10 and 3.15 show the variation of packet
end to end delay in the above mentioned WiMAX network. From Figures 3.9 and
3.10, it is observed that the packet end to end delay decreases in the overall
network when silence suppression is activated. This is due to the reduction in the
queuing delay. As the number of packets decreases with silence suppression, the
number of packets queued in the intermediate nodes decreases. This reduces the
queuing delay. The packet end to end delay decreases considerably when
mobility is introduced in the network. With mobility, the mobile nodes get
distributed under multiple base stations and this removes the bottle neck of one
base station in case of stationary nodes and thereby decreases the delay.
With a step towards the 4G mobile networks, this study is used in the
following chapters to compare and evaluate the performance of the loosely
coupled WiMAX WLAN integrated network.
38

Chapter 4
VoIP over WLAN Network
Chapter Outline:-
 Introduction

 Background of IEEE 802.11 and WLAN

 Protocol Architecture

 Overview of IEEE 802.11 MAC Layer

IEEE 802.11 MAC: DCF and PCF

 Overview of IEEE 802.11 PHY layer

Frequency Hop Spread Spectrum (FHSS)

Dynamic Sequence Spread Spectrum (DSSS)
Infrared

 WLAN System Architecture


Average Jitter

Average MOS

Average Jitter


 Conclusion
39

VoIP over WLAN Network
4.1 Introduction
WLANs [1] are mostly designed for private wired LANs and have been
enormously successful for data traffic like (email, media downloads etc).
WLANs are being studied as an alternative to the high installation and
maintenance costs incurred by traditional addition, removal, and changes
experienced in wired LAN infrastructures. It is an economic way to provide users
with ubiquitous connectivity - anywhere, anytime - to the Internet and to private
and corporate networks. It offers typical gross data rate of 10-50 Mbps as
opposed to 10-100 kbps offered by cellular technology and compared to the peak
70 Mbps offered by WiMAX. Voice traffic differs from data traffic mainly in its
sensitivity to delay and loss [2]. Although existing WLAN applications are
mainly data centric, there is a growing demand for real-time voice services over
WLAN. Driven by these two popular technologies, VoIP over WLAN has been
emerging as an infrastructure to provide low-cost wireless voice services.
4.2 Background of IEEE 802.11 and WLAN
The first generation of WLANs operated in the 900 MHz ISM band, with
symbol rates of around 500 Kbps, but they were exclusively proprietary, non-
standard systems, developed to provide wireless connectivity for specific niche
markets. The second-generation systems came around 1997. They operated in the
2.4 GHz range and provided symbol rates of around 2Mbps. The standards of
WLAN are as follows:
o IEEE 802.11b (Wi-Fi 2.4 GHz): The goal of the Task Group b was to
increase the maximum bit rate in the 2.4 GHz frequency range while
maintaining interoperability with the original standard. The MAC layer
was kept and the PHY was redefine to only work with DSSS (Direct
Sequence Spread Spectrum), thus increasing the spectral efficiency with
bit rates of up to 11 Mbps.
40

o IEEE 802.11a (Wi-Fi 5.2 GHz): The goal of this group was to provide
higher data rates and to port IEEE 802.11 to the newly available U-NII at
5.2 GHz. The original MAC layer was kept and the PHY was reworked to
provide rates up to 54 Mbps. Since the available band at U-NII is about
300 MHz, eight non-overlapping bands were defined. The spread spectrum
technology used in this case was OFDM (Orthogonal Frequency Division
Multiplexing), as DSSS was not efficient at working with these high bit
rates.
o IEEE 802.11g: Task group g is working on an extension to IEEE 802.11b
at 2.4 GHz, enabling transmission at symbol rates of 54 Mbps.
There are also other IEEE 802.11 Task Groups that focus on different aspects
such as:
o 802.11i - wireless security at the Mac layer.
o 802.11f - roaming between access points
o 802.11e - the Quality of Service
The IEEE 802.11 WLAN standard that operate in the 2.4 GHz or 5 GHz
unlicensed radio bands; the four are summarized in Table 4.1.
TABLE 4.1. Characteristic feature of Wireless LAN
Standard Maximum Fallback Channels Frequency Radio
Bit Rate Rates Provided Band Technique
802.11b 11 Mbps 5.5 Mbps, 2 Mbps 3 2.4 GHz DSSS
1 Mbps
802.11a 54 Mbps 48 Mbps, 36 Mbps, 12 5 GHz OFDM
24 Mbps, 18 Mbps,
12 Mbps, 9 Mbps,
6 Mbps
802.11g 54 Mbps Same as 802.11a 3 2.4 GHz OFDM
4.3 Protocol Architecture
Figure 4.1 shows the protocol Stack of IEEE 802.11. Part of the IEEE
802.11 protocol stack is shown in the figure 4.2. In this figure, only the relevant
parts of the protocol stack are depicted, as the upper layers are the same than the
41

other LANs and only the two layers represented (Physical Layer and Data Link
Control Layer) differ.
FIGURE 4.1. Protocol Stack
FIGURE 4.2. Part of the Protocol Stack
In the Physical Layer (PHY), several sub layers appear:
o Physical Medium Dependent (PMD). This sub layer‟s functions are the
modulation and codification of the signal.
o Physical Layer Convergence Protocol (PLCP): This level is in charge of
adapting the PHY service to the PMD sub layer.
o Physical Management. This level selects the channel to transmit with.
Just above the Physical Layer, the Data Link Control Layer (DLC) is formed by
the following sub layers:
o Medium Access Control (MAC). This sub layer is responsible for the
access, fragmentation and ciphering of the signal.
o Logical Link Control (LLC). This level makes the assignment of the
logical channels to the physical channels.
o MAC Management. This level is responsible of the synchronization,
roaming, consume management, etc.
o Finally, the Station Management is responsible of the coordination of the
management functions in the two layers.
42

4.4 Overview of the IEEE 802.11 MAC
802.11 support two modes of operation. The first, called Distributed
Coordination Function (DCF), does not use any kind of central control and the
second, called Point Coordination Function (PCF), and uses the base station to
control all activity in its cell. So, DCF can be used by all implementations of
WLANs and PCF is optional. These two modes will be explained in detail.
4.4.1 DCF and PCF
The original IEEE 802.11 standard [3] specifies two channel access
mechanisms: a mandatory contention-based distributed coordination function
(DCF) and an optional polling based point coordination function (PCF). The DCF
uses a carrier sense multiple access with collision avoidance (CSMA/CA) scheme
for medium access and the optional four way handshaking request-to-send/clear-
to-send [4] mechanism (RTS/CTS). DCF provides a best effort service and is not
capable of providing differentiation and prioritization based upon traffic type.
While DCF may provide satisfactory performance in delivering best-effort traffic,
it lacks the support for QoS requirements posed by real time traffic, and
especially VoIP which has stringent delay requirements. These requirements
make the DCF scheme a less feasible option to support QoS for VoIP traffic.
PCF mode, with a centralized controller, represented another promising
alternative to providing QoS in WLAN [5]. The centralized controller called the
point coordinator (PC) has its functionality embedded in the access point (AP) of
a BSS. The PC enables polling, enabling the polled stations to transmit data
without contending for the channel. Nevertheless, studies on carrying VoIP over
WLAN in PCF mode in [5] found that when the number of stations in a basic
service set (BSS) is large, the polling overhead is high. This results in excessive
end-to-end delay and VoIP gets poor performance under heavy load conditions.
Although the PCF mode is designed for real-time traffic, it is not widely
deployed due to its inefficient polling schemes, limited quality of service (QoS)
provisioning, and implementation complexity. Thus, both DCF and PCF have
limited support for real-time applications, the IEEE 802.11e has been proposed to
enhance the current 802.11 MAC to support applications with stringent QoS
requirements [7], but the wide deployment of the standard is yet to be done.
43

On the other hand, supporting voice traffic over WLANs using the DCF
mode poses significant challenges, because the performance characteristics of
their physical and MAC layers are much worse than their wired counterparts and
hence considered in our system.
4.5 Overview of IEEE 802.11 PHY layer
IEEE 802.11 supports three types of physical layer – Infrared, Frequency
Hop Spread Spectrum (FHSS), and Direct Sequence Spread Spectrum (DSSS).
The later two are based on radio transmission. All of them include the provision
of the clear channel assessment signal (CCA). The CCA is used for medium
access and indicates if the medium is idle. The transmission technology
determines the technique to obtain this signal. The PHY layer offers a service
access point (SAP) with 1 or 2 Mbps data rate to the MAC layer.
4.5.1 Frequency Hop Spread Spectrum (FHSS)
FHSS is a spread spectrum technique which allows for the coexistence of
multiple networks in the same area by separating different network using
different hopping sequence.
The standard specifies Gaussian shaped FSK (frequency shift keying),
GFSK, as modulation for FHSS PHY. For 1Mbps a two level GFSK is used (i.e.
1 bit mapped to 1 frequency) and for 2 Mbps a four level GFSK is used (i.e. 2
bits are mapped to 1 frequency). Sending and receiving in 1Mbps is mandatory
while sending and receiving in 2 Mbps is optional. This facilitated the production
of low cost devices for the lower rate only and more powerful devices for both
transmission rates in the early days of 802.11.
A FHSS frame consists of two parts, namely the PLCP part (preamble and
header) and the payload part. PLCP part is always transmitted at 1Mbps while the
payload can use 1 or 2 Mbps. Also the MAC data is scrambled using the
polynomial s(z) = z
7
+ z
4
+ 1 for DC blocking and whitening of the spectrum.
4.5.2 Dynamic Sequence Spread Spectrum (DSSS)
Direct sequence spread spectrum is the alternative spread spectrum method
separating by code and not by frequency. In case of IEEE 802.11 DSSS, spreading is
achieved by using the 11-chip Barker sequence (+1,-1,+1,+1,-1,+1,+1,+1,-1,-1,-1).
The key characteristics of this method are its robustness against interference
44

and its insensitivity to multipath propagation (time delay spread). However, the
implementation is more complex compared to FHSS.
IEEE 802.11 DSSS PHY also uses the 2.4 GHz ISM band and offers both
1 and 2 Mbps data rates. The system uses differential binary phase shift keying
(DBPSK) for 1 Mbps and differential quadrature phase shift keying (DQPSK) for
2 Mbps as modulation schemes. The symbol rate is 1MHz resulting in a chipping
rate of 11 MHz all bits transmitted by the DSSS PHY are scrambled with the
polynomial s(z) = z
7
+ z
4
+ 1 for DC blocking and whitening of the spectrum.
4.5.3. Infrared
The PHY layer which is based on infra red (IR) transmission, uses near
visible light at 850-950nm. Infra red light is not regulated apart from safety
restrictions. The standard does not require a line of sight (LOS) between sender
and receiver, but should also work with diffuse light. This allows for point to
multipoint (PMP) communication. The maximum range is about 10m if no
sunlight or heat sources interfere with the transmission. Typically, such a
network will only work in buildings, e.g. classrooms, meeting rooms, etc.
frequency reuse is very simple. A simple wall is more than sufficient to shield
one IR based IEEE 802.11 network from another. Today, no products are
available that offer infra red communication based on 802.11. Proprietary
products offer, e.g. up to 4 Mbps using diffused infra red light. Alternatively,
directed infra red communication based on IrDA can be used.
4.6 WLAN System Architecture
The architecture of a WLAN system is described according to the IEEE
802.11 standard. The main building block of such architecture is the Base Service
Set (BSS). There are two modes of configuration that can be used within the
standard:
o The “Infrastructure” and
o The “Ad Hoc” mode.
4.6.1 Infrastructure mode:
In the “Infrastructure” mode, the different BSS are interconnected with each
other via a component called the Distribution System (DS). Each BSS has one
Access Point (AP), through which the Mobile Nodes (MNs) access to the DS.
45

These interconnected components form the Extended Service Set (ESS). The ESS is
a large coverage area where MNs can get handed over from one BSS to another
without changes or notification to higher layers in the protocol stack. Finally, a
“Portal” is required to integrate the WLAN architecture into the wired network (e.g.,
Ethernet) and may be integrated with an AP in a single device attached directly to the
DS. Fig 4.3 shows the WLAN topology in infrastructure mode.
FIGURE 4.3. Infrastructure based architecture
4.6.2 Ad-Hoc mode:
In the “Ad-Hoc” mode, each MN can directly reach any MN within the
BSS without going through an intermediate node (i.e., AP). There is no backbone
network or distributed system associated with the mobile nodes and the BSS are
not connected to the wired network. The Adhoc mode of connection usually
covers small area and the nodes are allowed to move only within their BSS.
There is no concept of handover to support mobility across a large area. Fig 4.4
shows the topology of WLAN in Adhoc Mode.
FIGURE 4.4. Adhoc Architecture
46

4.7 Experimental Setup
To analyze VoIP in a network, it is necessary to study real life scenarios.
Hence, OPNET 14.5.A is chosen as the simulation tool as it will reflect the actual
deployment of the WiMAX network.
4.7.1 Scenario 1 – Static Nodes:
Figure 4.5 shows the simulation setup used for WLAN (IEEE 802.11b)
network. Using the Wireless Deployment Wizard of OPNET a 7 celled WLAN
network, with multiple subscriber stations in the range of an Access Point (AP) is
deployed. The AP is connected to the core network by a server backbone via an
IP backbone. The server backbone is further connected to the voice server which
is configured as the SIP server. The base station, IP Backbone, Server Backbone
and the Voice Server together represent the service provider company network.
The cell radius is set to be 100 m. The numbers of subscribers in cells are 10.
Voice over IP calls are setup between the subscriber stations of cells 2 and 3 in
mesh using SIP. The parameters set in the Access Points and the Subscriber
Stations are shown in figures 4.6 and 4.7 respectively.
FIGURE 4.5. Network Model for WLAN in OPNET
FIGURE 4.6. Access Point Parameters
47

With the simulation setup as mentioned, the common voice codecs being
voice jitter, MOS (Mean Opinion Score) and Packet End-to-end delay are noted.
4.7.1.1 Average Jitter
Figure 4.8 shows that all voice codecs except G 723.1 has non zero voice
jitter with positive variation. Voice quality is considered to be best if the jitter
value is zero. Thus, it can be said from this graph that G 723.1 provides best
performance with zero value and G 729 on the other hand produce worst
performance with highest jitter of 0.001 seconds. But as stated in [5], since G
723.1 is a low bit rate codec, it generates small packets of 5.3 Kbps and 6.3 Kbps
and hence can only be used for narrow band communications. Other than G
723.1, G 711 shows lower jitter compared to others at the cost of bandwidth
consumption. G 711 is a high bit rate codec and uses 64Kbps bit rate for data
transfer. It provides PSTN quality of voice. As already mentioned in chapter 1
voice communication has 35% voice and 65% silence in it. To accommodate
more users, silence present in the voice communication needs to be suppressed.
This is expected to save bandwidth. Results pertaining to silence suppression are
shown in figure 4.9. Figure 4.9 shows that the order of voice jitter decreases to an
order of 10
-5
from 10
-3
when silence is suppressed. This is quite less. After
silence suppression G 711 shows highest jitter among all the codecs considered.
G 726 (32Kbps) shows average position in both the graphs indicating optimal
performance. According to [7], G 726 has its roots in the PSTN network and
provides considerably good quality of voice. Since, it shows good performance in
both the cases (figures 4.7 and 4.8) it can be used in trunks and thereby can
accommodate more users by saving bandwidth.
48

Average Jitter without using silence suppression
0.0012
Second(sec)
0.001
0.0008
0.0006
0.0004
0.0002
0
G729 G711 G723.1 G723.1 G726 G726 G726 G726 G728 G728
5.3K 6.3K 16K 24K 32K 40K 12.8K 16K
Voice Codecs
FIGURE 4.8. Average voice jitters without Silence Suppression
Average Jitter using silence suppression
0.000016
0.000014
0.000012
Seconds(sec)
0.00001
0.000008
0.000006
0.000004
0.000002
0
-0.000002 G729 G711 G723.1 G723.1 G726 G726 G726 G726 G728 G728
5.3K 6.3K 16K 24K 32K 40K 12.8K 16K
Voice Codecs
FIGURE 4.9. Average voice jitters using Silence Suppression
4.7.1.2 Average Packet End to End Delay
The average packet end to end delay in WLAN network for voice codecs
without silence suppression is shown in figure 4.10 and is in the order of seconds.
This is because WLAN 802.11b provides a bandwidth of 11 Mbps and the Access
Points 2 and 3 [refer to figure 4.5] are overloaded and congested with the voice
traffic. This results in a bottleneck situation in the Access Points which results in
packet drop due to queue overflow in the access points. Any packet loss in the
physical and MAC layers is interpreted as delay by the higher layers. Hence, the
delay. Except G 723.1, all other voice codecs shows an average delay of more than 2
seconds if silence period is not suppressed. G 723.1 gives a packet end to end delay
of less than 0.25 seconds which is acceptable but can only be used for narrow band
communication. With silence suppression the packet end to end delay is considerably
less and in the order of 10
-2
seconds. The result is shown in figure 4.11. G 723.1
though giving a good performance without silence suppression shows terrible result
when silence is suppressed. The packet end to end delay for G 723.1 increases up to
0.1 seconds while the same for other is about 0.06
49

approximately which is quite less comparatively. G 723.1 creates small packets
of larger count. Though silence suppression decreases its number of packets but
still its packet count remains quite high compared to the other codecs. This
results in network congestion and thereby increased delay. Also, the look ahead
delay of G 723.1 is 7.5msec [8] while the same for G 729 is 5sec and others are
zero [8]. This adds up to the delay of G 723.1.
Average Packet End to End Delay without using silence suppression
5
4
Delay(sec)
3
2
1
0
G729 G711 G723.1 G723.1 G726 16K G726 24K G726 32K G726 40K G728 G728 16K
5.3K 6.3K 12.8K
Voice Codecs
FIGURE 4.10. Average packet end to end delay without Silence Suppression
Average Packet End to End Delay using Silence Suppression
0.12
0.1
(sec)
0.08
0.06
Delay
0.04
0.02
0
G729 G711 G723.1 G723.1 G726 G726 G726 G726 G728 G728
5.3K 6.3K 16K 24K 32K 40K 12.8K 16K
Voice codecs
FIGURE 4.11. Average packet end to end delay using Silence Suppression
4.7.1.3 Average MOS
Figure 4.12 shows the Mean Opinion Score (MOS) obtained by the voice
codecs in a WLAN network without using silence suppression. It shows that the
voice quality produced by all voice codecs except G 723.1 is unacceptable as they
have scored a MOS value of 1 which implies unacceptable quality of voice [5].
Hence, silence suppression has to be introduced and the improved results due to
silence suppression are shown in figure 4.13. With introduction of silence
suppression, the MOS value shows a considerable rise for all voice codecs except G
723.1. For most of the voice codecs the MOS value has exceeded 3.5 indicating very
good quality of voice. Hence to finally select a voice codec in a static WLAN
50

network accommodating large number of users, the voice codec G 726 is
expected to perform the best.
Average MOS without using silence suppression
3
MOS
2.5
2
Average
1.5
1
0.5
0
G729 G711 G723.1 G723.1 G726 G726 G726 G726 G728 G728
5.3K 6.3K 16K 24K 32K 40K 12.8K 16K
Voice Codecs
FIGURE 4.11. Average MOS without Silence Suppression
Average MOS using silence suppression
4
MOS
3.5
3
2.5
Average
2
1.5
1
0.5
0
G729 G711 G723.1 G723.1 G726 G726 G726 G726 G728 G728
5.3K 6.3K 16K 24K 32K 40K 12.8K 16K
Voice codecs
FIGURE 4.12. Average MOS using Silence Suppression
4.7.2 Scenario 2 – Mobile Nodes:
Figure 4.13 shows the simulation setup used for roaming enabled WLAN
network. Using the Wireless Deployment Wizard of OPNET a 7 celled WLAN
network, with multiple subscriber stations in the range of a base station is
deployed. The access points are connected to the core network by a server
backbone via an IP backbone. The server backbone is further connected to the
voice server which is configured as the SIP server. The base station, IP
Backbone, Server Backbone and the Voice Server together represent the service
provider company network. The cell radius is set to be 100 m. The mobile nodes
are configured to move at a rate of 50 km/hr, 100 km/hr and 150 km/hr along the
path shown in white lines. The number of subscribers in all cells is 10. Voice
over IP calls are setup between nodes of cells 2, 3 and 7 using SIP and are shown
by the blue lines in the figure 4.13.
51

FIGURE 4.13. Network Model for WLAN
Figure 4.14 shows the variation of jitter across the mobility supported
WLAN network with respect to the various voice codecs considered. From the
figure it is observed that the average jitter irrespective of speed decreases with
the decrease in data rate. G 711 has the highest data rate of 64Kbps and has
highest jitter. G 726 has a data rate of 32Kbps as considered and shows jitter less
than G 711. G 728 and G 729 has data rates of 16Kbps and 8Kbps and thereby
shows the decrease in the jitter and finally G 723.1 which has a minimum data
rate of 5.3Kbps and has minimum jitter. As the data rate decreases the load of the
network decreases. As the load decreases, the router processing happens in time
and the packets arrive in time. This thereby decreases the jitter.
Average Jitter
0.35
0.3
Second
s
0.25
50km/hr
0.2 150km/hr
100km/hr
0.15
0.1
0.05
0
G 711 G 723 G 726 G 728 G729
Voice Codecs
FIGURE 4.14. Average Jitter
4.7.2.2 Average Packet End to End Delay
As mobility is introduced in the network, the network performance
degrades to an unacceptable level. As the mobile nodes start moving the packet
end-to-end delay increases in the order of 10 seconds. This shows that due to low
bandwidth a WLAN network is unable to support voice communication at
vehicular speed. Overall we can say that such huge delay makes WLAN
incapable of supporting voice communication.
52

180
160
140
Sec
ond
s
120 50km/hr60 150km / hr
100
100km/hr
80
40
20
0
G 711 G 723 G 726 G 728 G729
Voice Codecs
Fig 4.15 Average Packet end to end delay
4.8 Conclusion
The target of this work is to observe the variation of jitter and packet end-to-
end delay in a WLAN network under various mobility scenarios. This work shows
the variation of the mentioned parameters with respect to a stationary network having
all stationary nodes and with respect to a mobile network with varying speed of
50km/hr, 100km/hr and 150km/hr. Unlike the WiMAX network average packet end
to end delay is less in case of stationary nodes than mobile nodes. Figures 4.8, 4.9
and 4.14 show the variation of jitter in the above mentioned networks. As figure 4.8
and 4.9 shows, the jitter in a stationary network decreases considerably when silence
suppression is activated in the network and it increases quite a lot when mobility is
introduced in the network. This is because as a node becomes mobile it undergoes
handover. This creates inconsistency in the order of the packets delivered thereby
jitter. Figures 4.10, 4.11 and 4.15 show the variation of packet end to end delay in the
above mentioned WLAN network. From figures 4.10 and 4.11, it is observed that the
packet end to end delay decreases in the overall network when silence suppression is
activated. This is due to the reduction in the queuing delay. As the number of packets
decreases with silence suppression, the number of packets queued in the intermediate
nodes decreases. This reduces the queuing delay. The packet end to end delay
increases largely when mobility is introduced in the network. With mobility, the
mobile nodes move at a very high speed over the network and while getting handed
over from one AP to another it moves to a third AP. This results in the fact that the
speed of handover is less compared to the speed of the mobile node. Hence, the
network fails and the effect is visible to us in terms of huge delay.
With a step towards the 4G mobile networks, this study is used in the
following chapters to compare and evaluate the performance of the loosely
coupled WiMAX WLAN integrated network.
53

Chapter 5
VoIP over WLAN-WiMAX Integrated
Network
Chapter Outline:-
 Introduction

 WLAN vs. WiMAX

 Types of Integration
Loose Coupling
Tight Coupling

 Working of the Integrated Network
Scenario 1: Stationary Nodes
Scenario 2: Mobile Nodes

Average Jitter

Average Packet End-to-End Delay
Average MOS

Scenario 2: Simulation Setup and results
Average jitter

Average Packet End-to-End Delay

 Conclusion
54

VoIP over WLAN-WiMAX Integrated
Network
5.1 Introduction
With growing demand for anytime anywhere connectivity, wireless networks
are gaining popularity and have become an integral part of human life. Researchers
of academia and industry are searching for new innovative methods to cater to the
requirement of the variety of roaming users. Nowadays the requirement is not only
restricted to connectivity but quality of service is also added to the requirement. With
voice over IP coming into existence this demand has gained more importance. By
distributing high-speed internet access from cable, digital subscriber line (DSL), and
other fixed broadband connections within wireless hotspots, WLAN has dramatically
increased productivity and convenience. The integration of WLAN into laptops has
accelerated the adoption of WLAN and an increased number of handhelds and
Consumer Electronics (CE) devices are thereby adding WLAN capabilities.
Moreover as shown in Chapter 4, the quality of service for voice application is of
acceptable quality in WLAN network.
WiMAX on the other hand has taken wireless Internet access to the next
level, and over time, could achieve similar attach rates to devices as WLAN.
WiMAX can deliver Internet access miles from the nearest WLAN hotspot and
blanket large areas -Wide Area Networks (WANs), be they metropolitan, suburban,
or rural - with multi-megabit per second mobile broadband Internet access. In the last
few years, WiMAX has established its relevance as an alternative to wired DSL and
cable, providing a competitive broadband service offering that can be rapidly and
cost effectively deployed. Now, Mobile WiMAX, as defined in the Institute of
Electrical and Electronic Engineers (IEEE) 802.16e-2005 standard, adds broadband
connectivity on the move. Mobile WiMAX, based on scalable Orthogonal Frequency
Division Multiple Access (OFDMA) technology, is capable of simultaneously
supporting fixed, portable, and mobile usage models.
Together, WiMAX and WLAN are ideal partners for service providers to
deliver convenient, affordable mobile broadband Internet services in more places.
55

Both are open IEEE wireless standards built from the ground up for Internet
Protocol (IP)-based applications and services. By using WiMAX backbone and
WLAN hotspot, service providers can deliver high-speed Internet connectivity
that subscriber‟s desire in more places. And WiMAX and WLAN technology
synergies enable seamless integration into laptops, CE devices, and generate a
new category of devices called “mobile Internet devices”. These devices are
expected to have dual stack for both WLAN and WiMAX.
5.2 WLAN vs. WiMAX
Although both WiMAX and WLAN provide wireless broadband
connectivity, they have been optimized for different usage models: WLAN for
very high-speed WLAN connectivity and WiMAX for high-speed Wireless
WAN (WWAN) connectivity. By combining WiMAX and WLAN technologies,
service providers can offer their subscribers a more complete suite of broadband
services in more places. Table 5.1 illustrates how WiMAX and WLAN
complement each other from an implementation and deployment perspective and
also shows the impact of the WLAN WiMAX integration synergy.
TABLE 5.1: WLAN and WiMAX Comparison [42] [43]
WiMAX 802.16e WLAN Synergy Impact
802.11b
Primary Deployed in wide Deployed in “Best-connected”
Application coverage areas, local coverage model: users connect
including areas, such as to WiMAX or WLAN
metropolitan areas public depending on their
for mobile hotspots, location, coverage,
broadband homes and and Quality of Service
wireless as well as businesses. (QoS) requirements.
rural or remote
areas for last-mile
connectivity and
portable service
Frequency Licensed/ 2.4 GHz ISM Service providers can
Band Unlicensed 2-11 Band leverage both types of
56

GHz spectrum; for
example, license
exempt for best effort
local area traffic and
licensed for wide area
and QoS sensitive
traffic.
Coverage Metropolitan area Short range Full range of services
mobile coverage of with up to 300 in the home and
up to several meters for a office, as well as on
kilometers for a single access the road. Economical
single base station. point. coverage of large
Longer range (up areas; for example,
to several miles) WLAN hotspots in
for fixed & lower- cafés, hotels, and
density airports, and WiMAX
deployments for blanket coverage
outside of hotspots.
Voice over VoIP is supported Voice over Both specifications
IP by the extended Internet support VoIP;
real-time polling Protocol however, operations in
class of service. (VoIP) is license-exempt
supported with spectrum limit QoS
enhancements assurance.
Mobility Provides fixed and Provides fixed Full range of services
portable solutions and portable in the home and
over a wide area of solutions office, as well as on
1-3 miles. within the the road.
coverage of the
hot spot.
Deployment Customer Premise Embedded in Integration into
Equipment (CPE) 97% of laptops devices is expected to
and PC cards will and many hand reduce device
57

be embedded in held and CE subsidies and lower
laptops and hand devices. Cost Per Gross Add
held devices (CPGA)
5.3 Types of integration
Mobility across WLAN and WiMAX can be integrated according to two
models namely loose coupling and tight coupling. Since, both the networks are IP
based; integration does not require translations in protocols when bridging the
infrastructure.
5.3.1 Loose Coupling
The lack of network management features in WLAN requires additional
components by the WLAN operator to obtain a higher degree of network
intelligence when performing vertical handover. Network intelligence can be
used to facilitate context transfer of AAA and services prior to handover. This
will reduce the latency of the handover and make the transition seamless. Loose
coupling enables flexibility for operators as roaming agreements can be made to
extend WiMAX data coverage without expanding or integrating WLAN in a
WiMAX network. Crucial to global roaming between WLAN hot spots and
WiMAX networks is better service tracking in WLAN where accounting and
billing is supported as WLAN currently lacks this facility.
5.3.2 Tight Coupling
In this type of integration, components of WiMAX network are already
present to support user mobility and AAA and integration proxies can be inserted in
the core network to support seamless connection transfers while avoiding dual traffic
flows. Care needs to be taken with such a solution so that end-to-end semantics are
preserved. Proxies also add a point of failure which requires some redundancy, but
managing at a higher level in the core infrastructure could mean better scalability.
Location management would also be made easier with the knowledge of network
infrastructure when moving between networks. With location management, mobile
devices can refrain from searching for other network topologies on other interfaces,
thus saving power. A WLAN attached to a WiMAX
58

network could use the CSN to supply users with IP-connectivity and IP-address
allocation.
5.4 Experimental Setup
Loosely coupled system has been developed and its performance with
respect to voice over IP is checked for varying voice codecs. The simulation
setup is as shown in the following sections.
5.4.1 Scenario 1: Stationary nodes
An integrated scenario comprising of WLAN hot spots and WiMAX
backbone is developed. The WiMAX backbone comprises of two base stations which
are connected to the SIP server via the routers. A special type of node called
SS_WiMAX_WLAN_AP having dual stack and behaving both as WLAN AP and
WiMAX subscriber station is placed under the coverage of the Base Station. This
node at the same time behaves both as the Subscriber Station of the WiMAX network
and Access Point of the WLAN hot spot and acts like a bridge. There are 10 WLAN
workstations under each such Access Points and VoIP calls are setup between the
workstations of the two APs. The parameters of the WiMAX Base Station and the
WiMAX interface of SS_WiMAX_WLAN_AP are same as that of WiMAX BS and
WiMAX SS set in Chapter 3 and the parameters of the WLAN workstations and
WLAN interface of the SS_WiMAX_WLAN_AP are same as shown in Chapter 4.
The scenario is as shown in Figure 5.1 and the protocol stack of
SS_WiMAX_WLAN_AP is shown in Figure 5.2. The entire process of developing
the scenario is discussed in Chapter 6. With the simulation setup as mentioned, the
common voice codecs being used for the Voice over IP calls are varied and the
corresponding variation in voice jitter, MOS (Mean Opinion Score) and Packet End-
to-end delay are noted.
FIGURE 5.1 Network Model for WiMAX-WLAN Static Integrated Network
59

FIGURE 5.2. Protocol Stack of SS_WiMAX_WLAN_AP
Figure 5.3 shows the variation of jitter for all voice codecs. It is seen that
the voice codec G 723.1 exhibits almost zero jitter which implies very good
quality of voice while others exhibit jitter varying from 0.0004 to 0.0005 seconds
and about 0.00055 for G729. The result when silence period is suppressed which
is about 65% of th entire communication period and is shown in Figure 5.4.
Average Jitter
0.0006
0.0005
Seconds
0.0004
0.0003
Integrated
0.0002
0.0001
0
-0.0001 9 1 K K K K K K K K
2 1 .3 .3 6 4 2 0 .8 6
7 7
.1
5
.1
6
6
1
6
2 3
6
4 2
8
1
G G 26 8 1
3 3 2
72 2 2
7 7 7
2 7
2 2
G G G G G7 7 7
G G Voice Codecs G
FIGURE 5.3 Average voice Jitter without Silence Suppression
Figure 5.4 shows the variation of jitter in the voice over IP for varying voice
codecs with voice activity detection enabled. It detects the silence period and
prevents paketisation of the background noise during the silence period. It is
observed from the graph that average voice jitter decrease to the order of 10
-5
to 10
-4
which is very nominal. Interestingly, G 723.1 shows unchanged performance. Voice
Activity Detection and silence suppression does not affect the performance of G
723.1 with respect to jitter. This is due to the fact that, G 723.1 is a low bit rate codec
and hence generates small packets. This thereby has much decreased load even
without silence suppression on the network unlike other codecs. Low load results in
low processing time and increased network performance. Also
60

among the two data rates of G 723.1, it is seen in figure 5.4 that the lower data
rate generates lower jitter than the higher data rate codec. G 711 having the
highest data rate displays maximum jitter.
Average Jitter
1.20E-04
1.00E-04
8.00E-05
Second
s
6.00E-05
4.00E-05
2.00E-05
Integrated
0.00E+00
-2.00E-05
9 1 K K
16K
K K
40K
K K
7 2 7 1 5 6
2 4 3 2 . 8 16
G G
. 3 . 3
12.1 .1 26 26 6 26 8 8
-4.00E-05 23 23 7 7 7 7 2 7
7 7 G G G G 7 G
G G G
-6.00E-05
Voice Codecs
FIGURE 5.4 Average voice Jitter with Silence Suppression
5.4.1.2 Average MOS
With MOS coming into the picture, figure 5.5 shows the mean optimal
score (MOS) scored by the various voice codecs in the WLAN-WiMAX
integrated network. As shown in the figure the highest MOS obtained by the
voice codecs with background noise packetised during the silence period i.e.
without silence suppression reaches a maximum value of 2.5 and for G 723.1
while for others it ranges from 1 to 2. From the MOS value we can conclude that
the network is incapable of supporting VoIP traffic.
A totally contradicting picture is observed in the figure 5.6 which shows
the performance of voice codecs with silence being suppressed in the
communication period. The MOS value scored jumps from the range of 1-2 to 3-
3.5 implying the network can provide acceptable quality of voice. Here comes the
importance of silence suppression. Again like average jitter the average MOS for
the voice codec G 723.1 remains unchanged irrespective of silence suppression
for the same reason as stated for jitter.
Average MOS
3
2.5
MOS
2
1.5
1
0.5
0
9 1 K K K 4K K K .8K K
2 1 .3 .3 6
32 40 6
7 7
.1
5
.1
6 1 2
6 12 8
1
G G 26 726 26 8
3 3 2 2
7 7 7
72
7
2 2
G G G G G7 7
G G G
Voice Codec
FIGURE 5.5 Average voice MOS without Silence Suppression
61

Average MOS with Silence Suppression
4
3.5
3
MOS
2.5
2
1.5
1
0.5
0
9 1 K K 6K K
32K
K K K
2 1 .3 .3 4 0 .8 6
7 7
.1
5
.1
6
6
1
6
2 4 2
8
1
G G
726 726 81
3 3 2 2 2
7 7
2 7
2 2
G G G G G7 7 7
G G G
Voice Codecs
FIGURE 5.6 Average voice MOS with Silence Suppression
5.4.1.3 Average Packet end-to-end delay
The packet end-to-end delay for the WLAN-WiMAX integrated network is
shown in figure 5.7. As shown in the figure, the packet end-to-end delay for all
voice codecs except G 723.1 is in the order of seconds varying from 1.5 sec for G
711 to above 3 seconds for G 728 and G 729. G 723.1 exhibits packet end-to-end
delay of less than 0.5 seconds. This is because G723.1 is a low bit rate codec and
generates small packets at the rate of 5.3Kbps or 6.3 Kbps compared to others.
Packet End-to-End Delay
4
3.5
3
Seconds
2.5
2
1.5
1
0.5
0
G729 G711 G723.1 G723.1 G726 G726 G726 G726 G728 G728
5.3K 6.3K 16K 24K 32K 40K 12.8K 16K
Voice Codecs
FIGURE 5.7 Average Packet end-to-end delay without Silence Suppression
Packet End-to-End Delay with Silence Suppression
0.14
0.12
0.1
Seconds
0.08
0.06
0.04
0.02
0
G729 G711 G723.1 G723.1 G726 G726 G726 G726 G728 G728
5.3K 6.3K 16K 24K 32K 40K 12.8K 16K
Voice Codecs
FIGURE 5.8 Average Packet end-to-end delay with Silence Suppression
The overall delay decreases to the order of 10
-2
seconds when silence
suppression is introduced and is shown in figure 5.8. Decreased delay is due to the
decreased load of the network. Silence suppression basically identifies the silence
62

period in the communication period by sensing the amplitude and stops
packetising the voice when silence is encountered. This decreases the number of
packets in the network thereby releasing the network load.
5.4.2 Scenario 2: Mobile Nodes
A scenario having two WLAN hot spots and a large span of area between
the hot spots covered by WiMAX is setup. The scenario is shown of Figure 5.3.
The WiMAX part of the network is deployed by the Wireless deployment wizard
of OPNET and WLAN Access Points are taken from the object palette. 10 special
nodes are configured to move from one Access point to the other access point i.e.
from one hot spot to another hot spot while communicating with 5 nodes of cell
number 3 of WiMAX and 5 nodes of cell number 7. Results show that the mobile
nodes are supported and traffic is send to them even when they are under the
jurisdiction of WiMAX network. The steps to develop the scenario are discussed
in Chapter 6.
FIGURE 5.9 Network Model for WiMAX-WLAN Integrated Mobile Network
The average voice jitter is found to be in the order of 10
-2
which is quite
high than static scenario. This is because as the mobility is introduced the routing
path varies. This results in a non serial way of packet arrival. Some packets reach
before time while others reach with a time lag resulting in increased jitter. Also it
63

is observed that jitter increases with speed. As the speed of the mobile node
increases, the mobile node gets handed over faster from the jurisdiction of one
BS/AP to another. This fast change of routing path causes frequent routing
update which in turn results in unordered packet and hence increased jitter.
Average Jitter
0.03
0.025
Second
s 0.02 50km/hr
150km/hr
0.015 100km/hr
0.01
0.005
0
G711 G723 G726 G728 G729
Voice Codecs
FIGURE 5.10 Average Voice Jitter without silence suppression
5.4.2.2 Average Packet end-to-end delay
The mobile scenario shows a decreased packet end-to-end delay. This is
shown in figure 5.11. As the mobile node starts moving, the load on the WLAN APs
which creates a bottleneck scenario is released. The entire load gets distributed
among the WiMAX base stations. This results in a fall in the packet end-to-end
delay. Also as the load of the packets decreases, the processing time of the BS and
the APs decreases considerably thereby saving time. Delay for G 711 is observed to
be quite high compared to the other voice codecs. This is because; G 711 is a high bit
rate codec and the rate of packets in about 64 Kbps. This results in a huge processing
time by the router hence increased delay. The delay increases with speed. As the
speed increases the mobile node faces increased number of hand over and hence the
routing path has to be recalculated which is time consuming. Also the packet
reassembly takes time and adds up to the delay.
0.3
0.25
Second
s
0.2 50km/hr
0.1 150km/hr
0.15 100km/hr
0.05
0
G711 G723 G726 G728 G729
Voice Codecs
FIGURE 5.11 Average Packet end-to-end delay without silence suppression
64

5.5 Conclusion
Overall it is concluded that, silence suppression improves the performance
of the network with respect to jitter, MOS and Packet end to end delay due to the
decrease in the network load. With mobility being introduced in the network the
average jitter increases considerably but the packet end-to-end delay decreases
remarkably. Due to the limitation of OPNET, the MOS value for the mobile
scenario could not be obtained but is since MOS is dependent on the delay of the
network, it is expected that the MOS value will increase in the mobile scenario.
Also, the network performance degrades with increase in speed.
65

Chapter 6
Development of Scenarios in OPNET
Chapter Outline:-
 Why OPNET?

 Development of WiMAX in OPNET
Static Scenario
Mobile Scenario

 Development of WLAN in OPNET
Static Scenario
Mobile Scenario

 Development of the WLAN-WiMAX Integrated Scenario
Static Scenario
Mobile Scenario
66

Development of Scenarios in OPNET
6.1 Why OPNET?
OPNET (Optimized Network Engineering Tool) [44] provides
comprehensive development environment supporting the modeling of
communication networks and distributed systems. Both behavior and
performance of the modeled systems can be analyzed by discrete event
simulation. Tool for all phases of our study including model design, simulation,
data collection and data analysis are incorporated in OPNET environment.
Various constructs pertaining to communication and information processing are
provided by OPNET. Thus it provides high leverage for modeling and distributed
systems. Graphical specifications of a model are provided by OPNET most of the
times. It provides a graphical editor to enter the network and model details. These
editors provide an intuitive mapping from the modeled system to the OPNET
model specification. OPNET provides 4 such type of editors namely the network
editor, the node editor, the process editor and parameterized editor organized in a
hierarchical way. It supports model level reuse i.e. models developed at one layer
can be used by another model at a higher layer. All OPNET simulations
automatically include support for analysis by a sophisticated interactive
debugger. Technology developers leverage advanced simulation capabilities and
rich protocol model suites to design and optimize proprietary wireless protocols.
In this work we have taken the advantages of OPNET Wireless modeler
suites (OPNET 14.5.A) particularly the WiMAX and WLAN platform for our
performance study.
6.2 Development of WiMAX in OPNET
To deploy a large 7 cell WiMAX network and examine the performance of voice
applications in static and mobile conditions the following steps are to be
followed: Methodology
1. Create a 7 cell WiMAX network and connect it to an IP backbone
2. Deploy voice applications between WiMAX nodes and configure the
voice server as the SIP Server.
67

3. Deploy voice QoS for WiMAX by choosing and tailoring a service class.
Map voice applications to WiMAX node by creating classifiers and
service flows on WiMAX nodes
4. Examine the performance of voice application in static conditions
5. Deploy trajectories on WiMAX SS nodes to make them mobile. Use ASN
anchored mobility for handover.
6. Deploy Application Demands.
7. The impact of mobility on application performance is then observed.
6.2.1 Static Scenario:
Steps to develop the static scenario are as follows:
1. Start OPNET Modeler by double-clicking on the OPNET Modeler 14.5
icon.
2. Select File > New.
3. A new window appears. Select Project from the menu and click OK
4. Another window appears and asks for project name and scenario name.
Enter project name as WiMAX_OPNET and scenario name as Static.
Uncheck the check box to not use the startup wizard and click OK.
5. Select Topology > Deploy Wireless Network. This launches the Wireless
Network Deployment Wizard.
FIGURE 6.1. Wireless Network Deployment Wizard
6. Click Continue and next until the Technology tab is reached. Select the
technology as WiMAX and set the Subscriber node transmission power as
0.5 W and Base Station transmission power as 10 W. Click next and move
to the next Tab of Topology.
68

FIGURE 6.2. Technology Window
7. In the topology tab enter the cell radius to be 30 km and from the drop
down menu select the node placement to be circular. This is because
circular nodes are located equidistantly from the base stations. The
settings are shown in the following figure.
FIGURE 6.3. Topology Window, First Tab
8. On clicking next the second tab of topology appears. Select the Subscriber
Station as wimax_ss_wkstn_adv from the dropdown list. This node allows
SIP configuration. Click Next.
9. The node mobility tab appears. Since we are deploying a static network
click on the row and click on delete row. Click Next.
10. The configuration summary appears as follows.
69

FIGURE 6.4. Configuration Summary
11. Click Finish. The deployed scenario consists of 7 celled WiMAX network
with the 7 BSs connected to a backbone.
12. To select Free Space as the pathloss model, Right-click on any of the
mobile nodes, and then choose Select Similar Nodes. Right-click on any
node again, and select Edit Attributes on any selected node in the
scenario space.
13. Check Apply changes to selected objects.
14. Expand WiMAX Parameters/SS Parameters and set Pathloss
Parameters to Free Space.
15. Click OK needed to apply changes to all selected nodes.
16. Click Yes to dismiss the compound attribute warning. You should see the
message “35 objects changed” in the status bar.
70

17. Click on the palette icon ( ), scroll up to the ppp_server and drag and
drop it on the work space. Similarly, drag an ip32_cloud and drop it on
the workspace. Rename the ppp_server as the Voice_Server.
18. Now scroll up to the Links_PPP and drag a PPP_SONET_OC1 link to
connect the ppp_server and the IP cloud. Similarly select a
PPP_SONET_OC_12 link and connect the IP cloud to the node called
backbone. The scenario now looks as follows. Right-click in the project
workspace after connecting the link to deactivate the handler.
FIGURE 6.6. Deployed WiMAX Scenario
19. Close the Object Palette.
20. From the object palette bring a Application Config and a Profile Config
node to configure the application.
21. Right click on the Application Config and select Edit Attributes
(advanced).
22. Under Application Definition add a row.
23. Enter Application Name as voice.
24. Expand the Description menu select the value of voice menu to edit.
25. Select the encoding scheme.
26. The voice codecs with (silence) implies silence suppression. By default
OPNET takes 65% of the communication period as silence and 35% as
the talk spurt.
27. The Profile Config node is configured as shown in the following figure.
71

FIGURE 6.7. Profile Setting
28. Now with the application and profile configured, we need to deploy the
application.
29. Select Protocols > Applications > Deploy Defined Applications
30. On the right-hand pane, expand Profile:”voice_profile” > Source and
click on Source. On the left-hand pane, while pressing Ctrl, select all
nodes of Cell 2. Once all of these nodes are selected, click on the
assignment arrow (>>) between the two panes.
31. Similarly, on the right-hand pane, expand Application:” voice” > Tier:”
Voice Destination” and click on “Voice Destination.” On the left-hand
pane, select all nodes of cell 3, and click on the assignment arrow
between the two panes. At this point, your right-hand pane should look
like the following figure. Click OK.
FIGURE 6.8. Application Deployment
72

32. The deployed voice application will be given strong QoS guarantees in
WiMAX network. For this reason, we will choose a service class using
ertPS scheduling.
33. Go to View/Show Network Browser; in the network browser pane, right-
click to Edit Attributes on the WiMAX_Config node. This node
contains the global parameters of the WiMAX network and is deployed
automatically when a WiMAX network is deployed. Edit the MAC
Service Class Definitions attribute by double-clicking in the
corresponding value field.
34. In the first row, modify the Gold service class by setting:
1. Scheduling Type to ertPS
2. Maximum Sustained Traffic Rate (bps) to 96000
3. Minimum Reserved Traffic Rate (bps) to 96000
4. Maximum Latency(milliseconds) to 10
35. Click OK once to accept changes.
36. On the WiMAX_Config node, set the Efficiency Mode to Physical Layer
Enabled.
37. Then click OK to accept the changes.
38. Right-click on any Mobile node in the network browser pane. The
wireless deployment wizard always deploys the WiMAX_Config node in
the top subnet, which explains its position outside the current subnet.
39. Deploy service flows and classifiers on the WiMAX mobile nodes, as
follows:
1. For each node of Cell 2, on which the voice application has been
deployed, deploy ertPS connections for both uplink and downlink
directions:
2. Select these nodes in the network browser (by holding down the Ctrl
key while selecting)
3. Right-click on one of these selected nodes in the scenario space, choose
Edit Attributes and checkmark Apply changes to selected objects.
4. Expand WiMAX Parameters/SS Parameters and click on the Value
field of Downlink Service Flows; add one row in the Downlink
Service Flows table and set:
1. Service Class Name to Gold
73

2. Modulation and Coding to QPSK 1/2; click OK once to accept
changes.
5. Similarly, add one row to the Uplink Service Flows table and set the
first row as follows:
2. Modulation and Coding to QPSK 1/2; click OK once to accept
changes.
6. While still editing the attributes of the mobile node, add a classifier
definition that directs voice traffic into the newly configured ertPS
connections for uplink traffic.
1. Click on the Value field of WiMAX Parameters/Classifier
Definitions and add one row to the Classifier Definitions table; in
the new row set:
1. Type of SAP to IP
2. Traffic Characteristics to IP ToS, Equals, Interactive Voice
(6)
4. Click OK once to accept classifier changes.
7. Click OK one final time to apply these changes to the selected objects.
8. Click Yes to ignore compound attribute warning.
9. Deploy classifier on the Base Stations for the downlink traffic.
1. In the Network Browser, select all base stations; right-click and
select Edit Attributes on any selected node in the scenario space.
Check Apply changes to selected objects.
2. Edit WiMAX Parameters/Classifier Definitions and add one
more row, setting it as in step 3.b.i.
3. Click OK as needed to apply changes to all selected nodes (BS
nodes).
40. OPNET provides a rich set of statistics to examine traffic performance from
the application as well as WiMAX perspective. In this scenario, we collect
global voice application and node level WiMAX connection statistics.
1. Right-click in the scenario space and Choose Individual DES
Statistics.
2. Expand Global Statistics group, and select Voice statistics.
74

3. Expand Node Statistics group, and select WiMAX Connection
statistic.
4. Click OK to accept selections
41. Click on the Configure/Run Discrete Event Simulation button in the
Project Editor. The Configure/Run DES dialog box appears. Notice the
simulation is configured to run for 600 seconds. Click Run to launch the
simulation. When the execution completes, click Close.
42. To view the results Select DES > Results > View Results.
43. To setup VoIP using Session Initiation Protocol (SIP), right click on the
voice server, expand the Edit Attributes (advanced) > SIP > SIP Proxy
Server Parameters and set Proxy Service to Enabled and then Enter the
Server Address as Proxy and Click OK.
FIGURE 6.9. SIP Server Attributes
44. Now select the mobile nodes among which the application is deployed.
Expand Edit Attributes > SIP > SIP UAC Parameters and set SIP
UAC Service to Enabled. Expand Proxy Server Specification and Select
the Proxy Server Name as Proxy. This is the name assigned to the Proxy
Server set in the Voice Server. Apply changes to the selected nodes and
Click OK.
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FIGURE 6.10. SIP UAC Parameter
6.2.2 Mobile Scenario
With mobility coming into consideration, a new node has to be included
to perform the functionality of the ASN. An ethernet4_slip8_gtwy node is taken
from the object palette and is placed between the backbone and is connected to
the backbone and IP cloud by point to point links. Generic Routing Encapsulation
Tunnels are also to be setup between the WiMAX Base Stations and the ASN
gateway. Generic Routing Encapsulation (GRE) is a tunneling protocol that
can encapsulate wide variety of network layer protocol packet types inside IP
tunnels, creating a virtual point to point links to routers at remote points over an
IP internetwork.
GRE tunnels are designed to be completely stateless. This means that each
tunnel end-point does not keep any information about the state or availability of
the remote tunnel end-point. A consequence of this is that the local tunnel end-
point router does not have the ability to bring the line protocol of the GRE tunnel
interface down if the remote end-point is unreachable. The ability to mark an
interface as down when the remote end of the link is not available is used in order
to remove any routes (specifically static routes) in the routing table that use that
interface as the outbound interface. Specifically, if the line protocol for an
interface is changed to down, then any static routes that point out that interface
are removed from the routing table. This allows for the installation of an alternate
(floating) static route or for policy based routing (PBR) to select an alternate
next-hop or interface.
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Normally, a GRE tunnel interface comes up as soon as it is configured
and it stays up as long as there is a valid tunnel source address or interface which
is up. The tunnel destination IP address must also be routable. This is true even if
the other side of the tunnel has not been configured. This means that a static route
or PBR forwarding of packets via the GRE tunnel interface remains in effect
even though the GRE tunnel packets do not reach the other end of the tunnel.
The GRE tunnel‟s keep alive mechanism is slightly different than for
ethernet or serial interfaces. It gives the ability for one side to originate and
receive keep alive packets to and from a remote router even if the remote router
does not support GRE keep alives. Since GRE is a packet tunneling mechanism
for tunneling IP-in-IP, a GRE IP tunnel packet can be built inside another GRE IP
tunnel packet. For GRE keep alives, the sender pre-builds the keep alive‟s
response packet inside the original keep alive request packet so that the remote
end only needs to do standard GRE decapsulation of the outer GRE IP header and
then forward the inner IP GRE packet. This mechanism causes the keep alive‟s
response to forward out the physical interface rather than the tunnel interface.
This means that the GRE keep alives response packet is not affected by any
output features on the tunnel interface.
Another attribute of GRE tunnel keep alives is that the keep alive timers
on each side are independent and do not have to match. The problem with the
configuration of keep alives only on one side of the tunnel is that only the router
that has keep alives configured marks its tunnel interface as down if the keep
alive timer expires. The GRE tunnel interface on the other side, where keep
alives are not configured, remains up even if the other side of the tunnel is down.
The tunnel can become a black-hole for packets directed into the tunnel from the
side that did not have keep alives configured. In a large hub-and-spoke GRE
tunnel network, it might be appropriate to only configure GRE keep alives on the
spoke side and not on the hub side. This is because it is often more important for
the spoke to discover that the hub is unreachable and therefore switch to a backup
path (Dial Backup for example).
Before GRE keep alives were implemented, there were only three reasons for a
GRE tunnel to shut down:
o There is no route to the tunnel destination address. o
The interface that anchors the tunnel source is down.
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o The route to the tunnel destination address is through the tunnel itself.
These three rules (missing route, interface down and mis-routed tunnel
destination) are problems local to the router at the tunnel endpoints and do not
cover problems in the intervening network. For example, these rules do not cover
the case in which the GRE tunneled packets are successfully forwarded, but are
lost before they reach the other end of the tunnel. This causes data packets that go
through the GRE tunnel to be "black holed", even though an alternate route that
uses PBR or a floating static route via another interface is potentially available.
Keep alives on the GRE tunnel interface are used in order to solve this issue in
the same way as keep alives are used on physical interfaces.
Steps to be done to a deployed static WiMAX network to add mobility to it:
1. Change the efficiency setting to use mobility in WiMAX scenario. Go to
View > Show Network Browser. In the network browser pane, right-
click to Edit Attributes on the WiMAX_Config node. Set Efficiency
Mode to Mobility and Ranging Enabled. Click OK.
2. To define the trajectory, go to Topology > Define Trajectory. A window
pops up where the trajectory name is to be entered. Check Coordinates
are relative to the objects and click define path.
3. A new window appears and allows entering the speed of the mobile node.
Click on the starting point of the trajectory and the ending point of the
trajectory the right click and select complete trajectory definition.
4. Select the nodes which will be configured to move. Right Click and select
Edit Attributes (advanced). From the drop down menu of the attribute
trajectory select the trajectory name that was defined.
5. A white line indicating the path of the motion of the mobile node appears.
6. The ASN gateway and the Base Stations are configured to setup the GRE
Tunnel. The configuration of the ASN gateway is as follows:
a. Right click on the ASN gateway -> Edit Attributes (advanced) ->
IP -> IP Routing Parameters -> Interface Information -> IF9 (any
other interface can also be chosen) -> set address field as
192.0.3.1. -> Click OK to save the settings.
b. Right click on the ASN gateway -> Edit Attributes (advanced) ->
IP -> IP Routing Parameters -> Tunnel interface -> set Number of
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rows with the number of tunnels to be established. Since we are
trying to develop 7 tunnels, we set the Number of Rows as 7 ->
Click OK to save the settings.
Tunnel Configuration between ASN Gateway and Base Station 1 is as
follows:
1. Right click on the ASN gateway -> Edit Attributes(advanced) -
> IP -> IP Routing Parameters -> Tunnel interface ->
Tunnel_BS1 -> set Address as 192.1.2.2 -> click OK to save
the settings.
2. Right click on the ASN gateway -> edit Attributes(advanced) -
> IP -> IP Routing Parameters -> Tunnel interface ->
Tunnel_BS1 -> Tunnel information -> set Tunnel source as
192.0.4.1 and set Tunnel Destination as 192.0.1.1.-> click OK
to save the settings.
> IP -> IP Routing Parameters -> Tunnel interface -> Tunnel0 -
> set Address as 192.1.2.1 and Subnet Mask as Class
C(natural)-> click OK to save the settings.
> IP -> IP Routing Parameters -> Tunnel interface -> Tunnel0 -
> Tunnel Information -> set Tunnel Source as 192.0.1.1 and
Tunnel Destination as 192.0.4.1 -> click OK to save the settings.
The ASN gateway is now configured for the GRE tunnel between
Base Station 1 and the ASN_gateway. Now the Base Station 1 is to
be configured as follows:-
1. Right Click on the Base Station 1 -> Edit
Attributes(advanced) -> IP -> IP Routing Parameters ->
Interface Information -> IF4(any other interfaces may also
be chosen) -> set Address field as 192.0.1.1 and subnet
mask as 255.255.255.0 -> click OK to save the settings.
2. Right Click on the Base Station 1 -> Edit Attributes
(advanced) -> IP -> IP Routing Parameters -> Tunnel
interface -> set Number of rows with the number of tunnels
to be established. Since we are trying to develop 1 tunnels,
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we set the Number of Rows as 1-> click OK to save the
settings
7. The Base Station 1 is now configured for the GRE Tunnel between Base
Station 1 and the ASN_gateway. To view the GRE Tunnel that has been
developed click View -> Visualize Protocol Configuration -> IP Tunnel
Configuration. Two unidirectional green colored lines will appear
between the ASN gateway and the Base Station1.These are the GRE
Tunnels. To configure the other base stations and the ASN gateway
following values are mapped with the example and set in the same way.
The detailed interface listing can be viewed from Edit -> Edit Objects
Using Template -> Click on the + symbol beside IP and select Tunnel
Interface Listing -> Click Generate. A table giving the details will appear.
TABLE 6.1. Tunnel Interface Listing
Node Tunnel Tunnel Tunnel Source IP Destination
name Address Subnet IP
Mask
ASN_GW Tunnel_BS1 192.1.2.2 Auto 192.0.1.1 192.0.2.1
Assigned
Tunnel_BS2 192.1.3.2 Auto 192.0.1.1 192.0.3.1
Assigned
Tunnel_BS3 192.1.4.2 Auto 192.0.1.1 192.0.4.1
Assigned
Tunnel_BS4 192.1.5.2 Auto 192.0.1.1 192.0.5.1
Assigned
Tunnel_BS5 192.1.6.2 Auto 192.0.1.1 192.0.6.1
Assigned
Tunnel_BS6 192.1.7.2 Auto 192.0.1.1 192.0.7.1
Assigned
Tunnel_BS7 192.1.8.2 Auto 192.0.1.1 192.0.8.1
Assigned
Base Tunnel0 192.1.2.1 Class C 192.0.2.1 192.0.1.1
Station_1
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Station_2
Station_3
Station_4
Station_5
Station_6
Station_7
8. After the GRE tunnels are setup, the application demand has to be setup.
9. From the object palette scroll to demands and after expanding demands select
IP_<Voice codec>_Voice. Click on the source node and further click on the
destination node. This creates a unidirectional flow of voice traffic. Similarly,
create the voice traffic from the destination to the source.
10. After the GRE Tunnels are setup and the Application Demands are setup
and mobility is assigned to the nodes the network appears as follows. The
green lines depicts the GRE tunnels that has been setup, the white lines
indicate the path of motion of the mobile nodes and the blue dotted
bidirectional arrows implies the Application Demand traffic configured.
FIGURE 6.11. Deployed Mobile WiMAX Network
11. Select the Demand Statistics from the DES which needs to be plotted and
the scenario is run for 600 sec.
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6.3 Development of Wireless LAN in OPNET
To deploy a large 7 cell WLAN network and examine the performance of
voice applications in static and mobile conditions the following steps are to be
followed:
Methodology
1. Create a 7 cell WLAN network and connect it to an IP backbone
2. Deploy voice applications between WLAN nodes and configure the voice
server as SIP Server.
3. Examine the performance of voice application in static conditions
4. Deploy trajectories on WiMAX SS nodes to make them mobile.
5. The impact of mobility on application performance is then observed.
6.3.1 Static Scenario:
Steps to develop the static scenario are as follows:
1. Start OPNET Modeler by double-clicking on the OPNET Modeler 14.5
icon.
2. Select File > New.
3. A new window appears. Select Project from the menu and click OK
4. Another window appears and asks for project name and scenario name.
Enter project name as WLAN_OPNET and scenario name as Static.
Uncheck the check box to not use the startup wizard and click OK.
5. Select Topology > Deploy Wireless Network. This launches the Wireless
Network Deployment Wizard.
6. Click Continue and next to move to the Next Tab of technology as shown
in the flowing figure. Select the technology as WLAN Infrastructure and
set the Subscriber node transmission power as 0.005 W, Operational
Mode as 802.11b and Data Rate as 11 Mbps. Click next and move to the
next Tab of Topology.
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FIGURE 6.12. Technology Selection Window
7. In the topology tab enter the cell radius to be 0.1 km and from the drop
down menu select the node placement to be circular. This is because
circular nodes are located equidistantly from the base stations. The
settings are shown in the following figure.
FIGURE 6.13. Topology Selection Window
8. On clicking next the second tab of topology appears as follows. Select the
Subscriber Station as wlan_wkstn_adv from the drop down menu. This
node allows SIP configuration. Click Next.
9. The node mobility tab appears. Since we are deploying a static network
click on the row and click on delete row. Click Next.
10. The configuration summary appears as follows.
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FIGURE 6.14. Configuration Summary Window
11. Click Finish. The deployed scenario consists of a 7 celled WLAN
network with the 7 access points connected to the backbone.
12. Expand the buffer size by, Right-clicking on any of the mobile nodes, and
then choosing Select Similar Nodes. Right-click on any node again, and
select Edit Attributes on any selected node in the scenario space.
13. Check Apply changes to selected objects.
14. Click on the palette icon ( ), scroll up to the ppp_server and drag and
drop it on the work space. Similarly, drag an ip32_cloud and drop it on
the workspace. Rename the ppp_server as the Voice_Server.
15. Now scroll up to the Links_PPP and drag a PPP_SONET_OC1 link to
connect the ppp_server and the IP cloud. Similarly select a
PPP_SONET_OC_12 link and connect the IP cloud to the node called
backbone. The scenario now looks as follows. Right-click in the project
workspace after connecting the link to deactivate the handler.
FIGURE 6.15. Deployed WLAN Network
84

16. Close the Object Palette.
17. From the object palette bring a Application Config and a Profile Config
node to configure the application.
18. Right click on the Application Config and select Edit Attributes
(advanced).
19. Under Application Definition add a row.
20. Enter Application Name as voice.
21. Expand the Description menu select the value of voice menu to edit.
22. Select the encoding scheme.
23. The voice codecs with (silence) implies silence suppression. By default
OPNET takes 65% of the communication period as silence and 35% as
the talk spurt.
24. The Profile Config node is configured as shown in the following figure.
FIGURE 6.16. WLAN Profile Configuration
25. Now with the application and profile configured, we need to deploy the
application.
26. Select Protocols > Applications > Deploy Defined Applications
27. On the right-hand pane, expand Profile:”voice_profile” > Source and
click on Source. On the left-hand pane, while pressing Ctrl, select all
nodes of Cell 2. Once all of these nodes are selected, click on the
assignment arrow (>>) between the two panes.
28. Similarly, on the right-hand pane, expand Application:” voice” > Tier:”
Voice Destination” and click on “Voice Destination.” On the left-hand
85

pane, select all nodes of cell 3, and click on the assignment arrow
between the two panes. At this point, your right-hand pane should look
like the following figure. Click OK.
FIGURE 6.17. WLAN Application Deployment
1. Right-click in the scenario space and Choose Individual DES
Statistics.
2. Expand Global Statistics group, and select Voice statistics.
3. Expand Node Statistics group, and select WiMAX Connection
statistic.
4. Click OK to accept selections
32. To setup VoIP using Session Initiation Protocol (SIP), configure the voice
server as done for WiMAX network.
33. Now select the mobile nodes among which the application is deployed
and configure the SIP UAC parameters like those in WIMAX.
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No new node needs to be added to create a mobile scenario of WLAN. Only
configuring mobility in the mobile nodes and adding Application Demands is all
that is to be done.
1. To define the trajectory, go to Topology > Define Trajectory. A window
pops up where the trajectory name is to be entered. Check Coordinates
are relative to the objects and click define path.
2. A new window appears and allows entering the speed of the mobile node.
Click on the starting point of the trajectory and the ending point of the
trajectory the right click and select complete trajectory definition.
3. Select the nodes which will be configured to move. Right Click and
select Edit Attributes (advanced). From the drop down menu of the
attribute trajectory select the trajectory name that was defined.
4. A white line indicating the path of the motion of the mobile node appears.
5. From the object palette scroll to demands and after expanding demands
select IP_<Voice codec>_Voice. Click on the source node and further click
on the destination node. This creates a unidirectional flow of voice traffic.
Similarly, create the voice traffic from the destination to the source.
6. The WLAN mobile scenario configuration is completed and looks as
follows. The white lines indicate the trajectory and the blue dotted lines
indicate the Application Demands.
FIGURE 6.18. Mobile WLAN Network
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6.3 Development of WLAN-WiMAX integrated network in OPNET
6.3.1 Static network
Steps to develop the WLAN-WiMAX integrated network is as follows:-
1. The integrated static scenario of WLAN-WiMAX integrated network
comprises of two WiMAX Base Stations (wimax_bs_router) connected
via three routers (ethernet4_slip8_gtwy) by PPP_DS3 link. The nodes
are dragged and dropped from the object palette and the links are also
taken from the subhead Links_PPP located in the object palette.
2. A ppp_server is configured as the SIP server as already discussed before
and is connected to the router located in the middle.
3. Two wimax_ss_wlan_router are dragged and dropped one near each
base station. These nodes have the dual stack and behave as a WiMAX
subscriber station and WLAN AP at the same time.
4. IP Auto addressing happens automatically for the WiMAX nodes based
on the association between SS nodes and their servicing BS nodes. In this
case, to associate the wimax_ss_wlan_router to the base station node, a
WiMAX MAC address has to be assigned to the Base Station Node. To
do this right click > Edit Attributes (advanced) > WiMAX Parameters
> BS Parameters > MAC address and then the same MAC address has
to be specified at the wimax_ss_wlan_router node. This can be done as
follows. Right Click > WiMAX Parameters > SS Parameters > BS
MAC Address.
5. The BSS ID of the WLAN also has to be configured manually on the
wimax_ss_wlan_router and the WLAN workstations which are dragged
and dropped into the workspace near the wimax_ss_wlan_router from the
object palette. It is configured on the wimax_ss_wlan_router as follows.
Right click > Edit Attributes (advanced) > Wireless LAN > Wireless
LAN Parameters (IF1 P0) > BSS Identifier. The same has to be put in
the WLAN workstations as follows. Right click > Edit Attributed
(advanced) > Wireless LAN > Wireless LAN Parameters > BSS
Identifier.
6. The Application Config and Profile Config nodes are to be configured
as they were configured in the previous cases.
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7. The Application then has to be deployed from Protocols > Applications >
Deploy Defined Applications.
8. The above configured scenario appears to be as follows.
FIGURE 6.19. Static WLAN-WiMAX Integrated Network
9. On the WiMAX_Config node, set the Efficiency Mode to Physical
Layer Enabled. Then click OK to accept the changes.
10. Add a classifier definition to the wimax_ss_wlan_router node that directs
voice traffic into the newly configured ertPS connections for uplink traffic.
a. Click on the Value field of WiMAX Parameters/Classifier
Definitions and add one row to the Classifier Definitions table; in
the new row set:
i. Type of SAP to IP
ii. Traffic Characteristics to IP ToS, Equals, Interactive
Voice (6)
iii. Service Class Name to Gold
iv. Click OK once to accept classifier changes.
b. Click OK one final time to apply these changes to the selected
objects.
c. Click Yes to ignore compound attribute warning.
d. Deploy classifier on the Base Stations for the downlink traffic.
i. In the Network Browser, select all base stations; right-click
and select Edit Attributes on any selected node in the
scenario space. Check Apply changes to selected objects.
ii. Edit WiMAX Parameters/Classifier Definitions and add
one more row and configure it as 10a.
iii. Click OK as needed to apply changes to all selected nodes
89

(BS nodes).
11. Configure the SIP Server as the Voice Server is configured in the
WiMAX and WLAN networks and also the SIP UAC parameters are to be
configured like the mobile nodes in the WiMAX and WLAN network.
a. Right-click in the scenario space and Choose Individual DES
Statistics.
b. Expand Global Statistics group, and select Voice statistics.
c. Expand Node Statistics group, and select WiMAX Connection
statistic.
d. Click OK to accept selections
1. Deploy a 7-celled WiMAX network and develop a WiMAX network the
way specified in 6.2.1.
2. Drag and drop two WLAN APs (wlan_ethernet_slip4_router) and place it
on either side of the deployed WiMAX network. Connect them to the
Server Backbone using PPP links. These two WLAN APs basically form
the hotspots.
3. 10 wimax_ss_wlan_router behave as handsets here are configured to
move from the coverage of one such hotspot to another moving through
the coverage of WiMAX network.
4. Their trajectory is configured the way already mentioned and Application
demands are configured between these nodes and the WiMAX mobile
nodes under the coverage of Cells 3 and 7.
5. The WiMAX efficiency mode is set to Mobility and Ranging Enabled.
Set up GRE Tunnels between the Base Stations and the ASN Gateway as
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shown in 6.2.2. This enables the Base Stations to send information about
the mobile node to the ASN Gateway and the ASN Gateway thereby
sends the traffic destined for the mobile node to be sent through the GRE
Tunnels.
6. The Routing information protocol (RIP) is configured throughout the
network and it is by this protocol that the mobile nodes get connectivity
throughout the network.
7. A lower RIP metric is assigned to the WLAN interface while a higher RIP
metric is assigned to the WiMAX interface. This will allow the mobile node
to use the WLAN network whenever both the networks are available.
8. Firstly, both the interfaces are given separate IP. This is done as follows.
Right Click on the mobile node > Edit Attributes (advanced) >IP > IP
Routing Parameters > Interface Information. 2 rows are added here
one for WLAN and the other for WiMAX and configured as follows. The
IF0 implies WiMAX Network and the IF1 implies the WLAN network.
TABLE 6.2. Interface Information
Name Address Subnet Routing Protocol MTU (bytes)
Mask
IF0 128.1.1.11 Class C RIP 1500
IF1 192.0.14.10 Class C RIP WLAN
9. To set the RIP metric the following steps are followed. Right click on the
mobile node > Edit Attributes (advanced) > IP Routing Protocols >
RIP Parameters > Interface Information. Set the cost parameter as 3
for IF0 and 1 for IF1.
10. Set up Application demands as mentioned in 6.2.2 between the mobile
nodes under the access points and the WiMAX Mobile Nodes located
under the coverage of the Base Stations 3 and 7.
11. After the Simulation Setup the scenario appears as follows.
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FIGURE 6.20. Deployed WLAN-WiMAX Integrated Mobile Network
12. Configure the server as the SIP Server and the mobile nodes as SIP UAC
Clients.
92

Chapter 7
Comparative Analysis
Chapter Outline:-
 Introduction

 Comparative Results

Stationary network
Mobile network

 Conclusion
93

Comparative Analysis
7.1 Introduction
Voice over IP is the expected to be in practice in the next generation
communication networks. The target of this paper is to send voice over IP
platform in the next generation networks and analyse the voice quality. There are
various types of competing networks like WiMAX, Wifi, etc. WiMAX having
higher bandwidth accommodates more users than WLAN but with degraded
performance as shown in this paper. Hence, an integrated network using WiMAX
backbone and WLAN hotspots have been developed with multiple competing
traffic using SIP signalling and analysed to support VoIP. To provide a real life
simulation environment, OPNET 14.5.A modeler is chosen as the simulation tool.
The quality of the service is analysed in all three networks with parameters like
average jitter, average MOS and average packet end to end delay for various
voice codecs. The network performance is also observed with respect to vehicular
mobility of 50, 100, and 150km/hr. It is concluded that the WiMAX-WLAN
integrated network accommodates users more than a similar WLAN but less than
a similar WiMAX network. It provides voice quality better than WiMAX and
almost like WLAN which is good and in the mobile scenario the integrated
network performs almost like WiMAX which is much better than WLAN as
WLAN cannot support vehicular mobility.
7.2 Comparative Results
The results obtained in the Chapters 3, 4 and 5 are compared in this section of
the thesis.
7.2.1 Stationary Network
The variation of jitter for the three networks with variation of the voice codecs is
shown in Figure 7.1. Perceived voice quality is considered to be good if the jitter is
zero. As, shown in the figure, the average voice jitter is almost 0 for WiMAX
implying very good quality of voice in comparison to WLAN which has a positive
jitter varying from about 0.0007 to 0.001 seconds. The integrated network shows
jitter variation from about 0.0004 to 0.0006 seconds. For voice codecs G723.1, the
94

average jitter is almost 0 irrespective of the network indicating a good
performance. This is because the bit rate of G723.1 is 6.3 or 5.3 Kbps which
results in generation of small packets. But modem and fax signals cannot be
carried by G723.1 [4]. It can be used only for narrow band communications.
Mean voice jitter(sec) for various codecs
Integrated Network WiMAX WLAN
(sec)
0.0012
0.001
jitter
0.0008
0.0006
voice
0.0004
0.0002Mean
0
-0.0002 G729 G711 G723.1 G723.1 G726 G726 G726 G726 G728 G728
5.3K 6.3K 16K 24K 32K 40K 12.8K 16K
Voice Codecs
Figure. 7.1 Average voice jitter
With silence suppression, the result is different. Like G711, G726 has its roots
in the PSTN network. It is primarily used for international trunks to save
bandwidth. Unlike G711, G726 uses 32Kbps to provide nearly the same quality
of voice. This is because 32 Kbps is the de facto standard.
As shown in Figure 7.2, the average voice jitter is almost 0 for all codecs in
both WLAN and WLAN-WiMAX integrated network where as the WiMAX
network shows a slight deviation for the voice codec G 726. The deviation is
highest for G726 with 32kbps and is about 0.03sec. Though G726 supports data
rate of 16 , 24, and 40Kbps also, the 24 and 16 Kbps channels are used for voice
in Digital Circuit Multiplication Equipment (DCME) and the 40 Kbps is for data
modem signals (especially modems doing 4800 Kbps or higher) in DCME. But
G726 with 32 Kbps follows the de facto standard and hence the deviation in jitter
is within the consideration limit.
Mean voice jitter (sec)
using silence suppression
Integrated Network WiMAX WLAN
jitter
3.00E-02
2.00E-02
voice
(sec)
1.00E-02
Mean
0.00E+00
-1.00E-02
G729 G711 G723.1 G723.1 G726 G726 G726 G726 G728 G728
5.3K 6.3K 16K 24K 32K 40K 12.8K 16K
Voice Codecs
Figure. 7.2. Average voice jitter with Silence Suppression
95

Voice Packet end to end delay (sec)
toend
5
delay(sec
)
4
Pack
etend
3
Integrated Network
WiMAX
2
WLAN
Voice
1
0
G729 G711 G723.1 G723.1 G726 G726 G726 G726 G728 G728
5.3K 6.3K 16K 24K 32K 40K 12.8K 16K
Voice Codecs
Figure. 7.3. Average packet end to end delay
As shown in Figure 7.3, the packet end to end delay for voice without silence
suppression is less than 0.5 seconds for WiMAX where as for WLAN high bit
rate codecs it is very high. This is because the silence periods is also packetised
and send thereby creating huge bandwidth requirement and congestion in the
WLAN network. The integrated network shows a packet end to end delay less
than the WLAN network as the capacity of the network has got enhanced due to
the integration thereby resulting in less congestion. On the other hand, as shown
in figure 7.4, the packet end-to-end delay with silence suppression shows that
WiMAX has a packet end-to-end delay more than WLAN and the integrated
network has the same almost as that of WiMAX network. This is because with
silence suppression the number of packets to be send decreases thereby releasing
the congestion in the WLAN network and also the distance to be traversed by the
packet for WiMAX is much higher than the distance to be covered by the packet
in case of WLAN and even for the integrated network the distance traversed by
the packet is almost is as huge as that of WiMAX. For WLAN the packet end-to-
end delay is about 0.06 seconds except for G 723.1 for which it is about 0.1
seconds where as for WiMAX and WLAN-WiMAX integrated network, the
delay varies from 0.08 seconds for G 729 and G 711 to 0.13 seconds for G723.1.
Voice Packet end to end delay(sec)
using silence suppression
to
0.14
VoicePack
etend
enddelay(
sec)
0.12
0.1 Integrated Network
0.08
WiMAX
0.06
WLAN0.04
0.02
0
G729 G711 G723.1 G723.1 G726 G726 G726 G726 G728 G728
5.3K 6.3K 16K 24K 32K 40K 12.8K 16K
Voice Codecs
Figure. 7.4. Average packet end to end delay with Silence Suppression
96

Figure 7.5 shows the variation of MOS for both WiMAX and WLAN network
with variation of the voice codecs.
M ea n V o ic e M O S v a lu e v a ry in g
c o d ec s
MOS
In te g ra te d W iM A X W L A N
4
3 . 5
Voice
Value
3
2 . 5
2
1 . 5
Mean
1
0 . 5
0
G 7 2 9 G 7 1 1 G 7 2 3 . 1 G 7 2 3 . 1 G 7 2 6 G 7 2 6 G 7 2 6 G 7 2 6 G 7 2 8 G 7 2 8
5 . 3 K 6 . 3 K 1 6 K 2 4 K 3 2 K 4 0 K 1 2 . 8 K 1 6 K
Figure. 7.5. Average voice MOS in WiMAX and WLAN and Integrated network
As shown in the figure, the MOS value obtained for WiMAX is above 3, for
WLAN is almost 1 and for the integrated network is about 1.5 except for G 723.1
for which it is about 2.5. This is because G723.1 is a low bit rate codec. In case of
WiMAX it has packet end to end delay higher than the other codecs hence lower
MOS. On the other hand it has low bandwidth requirement and hence less
packets are dropped for this codec in WLAN network. Hence higher MOS
compared to the other codecs.
Mean Voice MOS value varying
codecs with silence suppression
Integrated WiMAX WL AN
Value
4
3.5
3
eMOS
2.5
2
Voic
1.5
Mean
1
0.5
0
G 729 G 711 G 723.1 G 723.1 G 726 G 726 G 726 G726 G 728 G 728
5.3K 6.3K 16K 24K 32K 40K 12.8K 16K
Voice Codecs
Figure. 7.6. Average voice MOS with Silence Suppression
The result with silence suppression is totally different. As shown in Figure 7.6,
WLAN shows better performance among the three networks with a MOS value of
above 3.5 except for G711 for which it is about 3 and for G723.1 for which it is
about 2.5. This is because with the silence suppression the number of packets in
97

the network decreases thereby releasing the congestion in the WLAN network.
This decreases the amount of packet dropped and thereby increases the MOS
value. The codec G723.1 is a low bit rate codec having bit rate of 5.3 kbps and
6.3kbps and the packet end to end delay is larger for G 723.1 in comparison to
the other codecs. Hence the MOS decreases.
7.2.2 Mobile Network
Figure 7.7 shows the variation of jitter for all three networks in the mobile
scenario. The mobile nodes are configured to move at a speed of 50, 100,150
km/hr. As shown in the figure, the average jitter for WLAN network is very high
compared to the WiMAX network. The WLAN-WiMAX integrated network
shows variation of jitter almost like the WiMAX which is very low compared to
WLAN implying very good performance. The average performance of the
networks with respect to speed is almost same. Among the integrated network
and the WiMAX network, the WiMAX network performs slightly better than the
integrated network. The same with silence suppression could not be noted due to
the limitation of OPNET.
Average Jitter
G711 G723.1 G726 G728 G729
0.35
0.3
0.25
Sec
0.2
0.15
0.1
0.05
0
50km/hr
100km/
hr
150km/
hr
50km/hr
100km/hr
150km/hr
5
0
k
m
/
h
r
100km/
hr
150km/
hr
WiMAX WLAN Intergrated
Network
Figure 7.7 Average Jitter in mobile network
As shown in figure 7.8 the packet end-to-end delay for WiMAX and
WLAN-WiMAX is negligible compared to WLAN network. The delay exhibited
by the WLAN is of the order of minutes which make the network practically
unusable. Figure 7.9 shows the variation of delay between the WiMAX network
and the integrated network which is almost zero with respect to WLAN. Among
98

the two networks again WLAN is found to perform best and the integrated
network performs in a slightly degraded way. Also it is observed that delay
increases with the speed and G 711 has highest delay among other codecs for
integrated network. This is because G 711 is the highest bit rate codec and has a
data rate of 64Kbps. As the mobile nodes move, they get handed over both
horizontally and vertically. With high data rate, packets get dropped more during
the handover. Any packet dropped by the lower layer is perceived as delay by the
upper layers. Hence, the delay of G 711 is exhibited in figure 7.9.
G711 G723.1 G726 G728 G729
180
160
140
120
Sec
100
80
60
40
20
0
50km/hr
100k
m/hr
150km
/hr
50km/hr
100k
m/hr
150km/hr
5
0
k
m
/
h
r
100km
/hr
150km/
hr
WiMAX WLAN Integrated
Network
Figure 7.8 Average Packet End-to-End Delay for three networks
G711 G723.1 G726 G728 G729
0.3
0.25
0.2
sec
0.15
0.1
0.05
0
50km/hr 100km/hr 150km/hr 50km/hr 100km/hr 150km/hr
WiMAX Integrated
Network
Figure 7.9 Average Packet End-to-End Delay for two networks
99

7.3 Conclusion
In this chapter the simulated results of a VoIP application over a WiMAX
based BWA network, a WLAN 802.11b network and a WLAN-WiMAX
integrated network in OPNET 14.5.A are presented. Multiple competing traffic
sources using SIP signalling over the networks are simulated and the traffic traces
and measurements obtained from the simulation are presented.
Overall it can be concluded that to support increasing number of users,
silence suppression has to be used. With silence suppression introduced in the
network, WLAN is observed to perform best in stationary condition but WLAN
cannot support vehicular mobility. To support vehicular mobility WiMAX has to
be introduced. It is observed from the mobile scenario that WiMAX performs
much better than WLAN. The integrated network in the static scenario performs
better than WiMAX and almost like WLAN and in the mobile scenario; the
integrated network performs almost like WiMAX which is much better than
WLAN. Thus, we see that the integrated network gives the best performance in
both static and mobile scenario.
The mobile scenario with silence suppression could not be simulated due
to the limitations of OPNET. [45] states that the capacity of mobile WiMAX is
doubled when silence suppression is used. Hence, as presented in this thesis, if an
integrated network is developed comprising of WiMAX backbone and WLAN
hotspot, it is expected to be competent enough to provide optimal voice quality
with optimal network capacity.
100

Chapter 8
Conclusion and Future Work
Chapter Outline:-
 Conclusion

 Future Work
101

Conclusion and Future Work
8.1 Conclusion
The variation in performance of a WiMAX network deployed over
suburban areas of terrain types varying from hilly terrain with moderate to heavy
tree density to flat terrain with light tree density for VoIP application with mobile
nodes moving at a speed of 50 km/hr is investigated. The investigation evaluated
the effect of varying suburban terrain; on the throughput and packet end to end
delay performance of Mobile WiMAX. The reduction in Line-Of-Sight due to the
terrain impacts the throughput directly by increasing the attenuation and
diffraction losses, and indirectly affects the packet end to end delay by causing
nodes to initiate network registration more frequently under fluctuating cell
coverage. Thus, it is concluded that any experiment related to wireless network
should take into consideration the terrain on which the network is deployed
because this affects the network performance substantially.
In this project extensive simulation based performance analysis for VoIP
application is done over a WiMAX based BWA network, a WLAN (IEEE
802.11b), and also on a WLAN-WiMAX integrated network. For this OPNET
14.5.A and OPNET 15.0.A simulation platform have been used. Multiple
competing traffic sources using SIP signalling over the networks are generated
and the trace of traffic and measurements for different performance parameters
discussed in this paper are obtained.
Close observation of the results of static networks reveal that
WiMAX network performs better on the basis of jitter, MOS and packet end-to-
end delay than conventional WLAN 802.11b in case of voice applications when
no silence suppression is considered, i.e. with no bar on the bandwidth usage, but
when more users are to be accommodated, bandwidth becomes a constraint and
hence silence suppression has to be used. With silence suppression, WLAN
provides a better voice quality than WiMAX. Though, the WiMAX network
provides high capacity, degradation of the voice quality is observed i.e. reflected
from the MOS value. WLAN on the other hand has less capacity but provides a
better voice quality. Hence, the integrated deployment of WiMAX and WLAN is
102

expected to be competent enough to provide optimal voice quality with optimal
network capacity.
To support increasing number of users, silence suppression has to be used.
With silence suppression introduced in the network, WLAN is observed to perform
best in stationary condition but WLAN cannot support vehicular mobility. To support
vehicular mobility WiMAX has to be introduced. It is observed from the mobile
scenario that WiMAX performs much better than WLAN. The integrated network in
the static scenario performs better than WiMAX and almost like WLAN and in the
mobile scenario; the integrated network performs almost like WiMAX which is
much better than WLAN. Thus, we see that the integrated network gives the best
performance in both static and mobile scenario.
The mobile scenario with silence suppression could not be simulated due
to the limitations of OPNET. [45] states that the capacity of mobile WiMAX is
doubled when silence suppression is used. Hence, as presented in this thesis, if an
integrated network is developed comprising of WiMAX backbone and WLAN
hotspot, it is expected to be competent enough to provide optimal voice quality
with optimal network capacity.
8.2 Future Work
This thesis has worked on the performance of WLAN 802.11b and
WiMAX 802.16 networks for voice application. The work done in this thesis can
further be extended to observe quite a number of variations. Video traffic when
configured in the WLAN 802.11b network is observed to support hardly 2 to 3
users per access points. Hence, increasing network capacity can be a future step
for video applications. The performance of the networks for varying users can
also be a work in the future scope. Also the network performance for other
applications like file transfer and web browsing can be an extension of the work.
The integrated network used in this work has been configured using
WiMAX (IEEE 802.16) and WLAN (IEEE 802.11b). This part of the work can
be further extended by adding 3G network to the existing infrastructure. OPNET
15.0.A presently does not support the coexistence of WiMAX or WLAN and 3G.
It may be supported in Qualnet or in future releases of OPNET and thereby its
performance can be observed.
103

Voice Activity Detection is a technique required for silence suppression in
VoIP and is not currently supported in the application demands of OPNET. Also
the plot for Mean Optimal Score (MOS) calculation is not presently supported by
OPNET for WLAN-WiMAX integrated mobile scenario. This is a limitation of
OPNET 15.0.A. which may be overcome by modifying some codes of OPNET
modeler and can be in the scope of future work.
For representing the WLAN technology IEEE 802.11b has been considered
as it is low cost and widely deployed and accepted by the public, but IEEE 802.11b
supports highest data rate of 11 Mbps. On the other hand IEEE 802.11a supports high
data rate of 54 Mbps but is still in the development stage and not widely used. Hence,
performance of the WiMAX, WLAN and the WiMAX-WLAN integrated network
can be observed for IEEE 802.11a or the most recently developed IEEE 802.11n with
respect to this work as a future work.
104

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