Voip on Wimax

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Performance Enhancement of Voice over Internet Protocol
(VoIP) over WiMAX Networks Using IPV4 and IPV6
SYNOPSIS
MASTER OF TECHNOLOGY
IN
ELECTRONICS AND COMMUNICATION ENGINEERING
Under Guidance of: Submitted By:
Mrs.
(Asst. Professor) Roll No:
DEPARTMENT OF ELECTRONICS AND COMMUNICATION
ENGINEERING

ABSTRACT
The WiMAX system effectively supports wide variety of broadband wireless access
technologies, which including high speed internet and multimedia access with high Quality of
service (QoS) requirements. Real time services such as Voice over Internet Protocol (VoIP) are
becoming popular and are major revenue earners for network service providers. However, there
are still many challenges that need to be addressed to provide a steady and good quality voice
connection over the best effort WiMAX network .To support flexibility, efficiency and various
requirements of QoS over a range of different applications and environments several
provisioning and mechanisms are provided .
This research work investigates and improves the performance of Voice over Internet Protocol
(VoIP) traffic using IPV4 and IPV6 over WiMAX networks and the impact of various voice
codec schemes and statistical distribution for Voice over Internet Protocol (VoIP) over WiMAX
has been investigated in detail. Through various simulation experiments under realistic
networking scenarios, this study provides an insight into the Voice over Internet Protocol (VoIP)
performance in the WIMAX networks. The simulations results indicate that better choice of
voice codec's and statistical distribution have significant impact on Voice over Internet Protocol
(VoIP) performance in the WiMAX networks and Performance of selected parameters will be
done using the network simulator OPNET Modeler.
Keywords: VoIP, QoS, OPNET, WiMAX,

CHAPTER 1
INTRODUCTION
1.1 Introduction to Voice over Internet Protocol (VoIP):
Voice over Internet Protocol ( (VoIP), also called IP Telephony, is rapidly becoming a familiar
term and technology that is invading enterprise, education and government organizations. Voice
over Internet Protocol (VoIP) is designed to replace the legacy TDM technologies and networks
with an IP-based data network. Digitized voice will be carried in IP data packets over a LAN
and/or WAN network. Installing and testing the Voice over Internet Protocol (VoIP) network of
IP phones, gateways and servers requires new tools and expanded knowledge.
The legacy telephone network has provided reliable and high-quality voice communications for
many years. It delivers voice and speech over a standardized 64 Kbps channel. The 64 Kbps
bandwidth is guaranteed for each call and the speech path carries the voice as a continuous
digital stream. Digital voice is NOT carried in packets. Enterprise and residential callers use
DTMF (Touch Tone), TI channel and ISDN D channel signalling to set up and manage the call.
In a Voice over Internet Protocol (VoIP) network, there is a signalling protocol and a speech
transmission protocol. Both protocols require all information be carried in IP packets. Several
standards-based choices are available for signalling protocols, including H.323, SIP, MGCP and
H.248.
RTP is the standard speech transmission protocol used with VoIP networks. The speech is
digitized, placed in packets, and transmitted through the IP network. Multiple packets are
required to carry a single spoken word. The voice is digitized using one of the G.7xx standards.
1.2 Problem Statement:
The demand for multimedia applications in WiMAX networks is growing at a rapid pace. A
method for guaranteeing Quality of Service (QoS) for different classes of traffic is therefore
gaining importance. Hence designing and analyzing multimedia traffic and QoS parameters has

become central to this problem. In this study, firstly we have investigated the data and voice
support in the WiMAX network using IPV4 and IPV6 and to examine the capability of a
WiMAX network to deliver adequate QoS to voice and data applications. And secondly
improves the performance of Voice over Internet Protocol (VoIP) traffic over WiMAX networks.
1.3 Aims and objectives:
The objective of study is to guarantee QoS for multiple service class traffic in a multiple
connection environment and to examine a case of QoS deployment over a cellular WiMAX
network. In particular, the thesis compares the performance obtained using two different QoS
configurations differing from the delivery service class ,to guarantee QoS for multiple service
class traffic in a multiple connection environment in WiMAX network the Adaptive modulation
and coding scheme is used at the physical layer that adapts to the scheduled traffic to stabilize
the QoS requirements of different traffic classes, Configure QoS mechanisms to guarantee low
delay for multimedia application without drastically affecting data traffic. We analyze the
distribution impact on traffic arrival time to quality of Voice over Internet Protocol (VoIP) in
WiMAX network; analysis the quality of service (QoS) with Voice over Internet Protocol (VoIP)
over WiMAX will be performed. Performance of selected parameters will be done using the
network simulator, OPNET Modeler
1.4 Methodology:
In our case, we have used OPNET Modeler v16.0 and the study in presents a simulation model
to analyzes the performance of an IEEE 802.16 system by focusing on the MAC layer scheduling
and evaluate Voice over Internet Protocol (VoIP) traffic by using G.729 ,G .711 and G .723
codecs.
1.5 Why use IP for voice?
One or more of the following may justify the move to Voice over Internet Protocol (VoIP):
• Reducing long distance charges, especially international long distance

• Reducing staff by combining voice-network and data-network management and eliminating
redundant functions
• Adding expanded applications that are not offered by TDM-based systems
• Having one common network for different forms of communication
Fig. 1.1 Voice over Internet Protocol (VoIP)
There are two forms of a Voice over Internet Protocol (VoIP) call. You can set up a PC-to-PC
call without working with a call server. This is typically how the early users of Voice over
Internet Protocol (VoIP) made calls. However, the prevalent enterprise VoIP solution requires a
call server (the standards community calls this a “gatekeeper”) to be part of the network
configuration. Although it is called a “server”, the server does not operate like a traditional
server.

 Call Server: In Voice over Internet Protocol (VoIP), the call server controls all the
services offered, provides control over the call, supports the telephone features,
authenticates and authorizes the caller and implements security. The call server is NOT
the telephone switch. Once the call server sets up a phone (peer-to-peer) call, the server
becomes in standby during the speech transmission unless the phones contact the server
to indicate a change in status or the call server wants to change the call configuration,
such as indicating there is a call waiting. The server is there to process the signalling , but
does not switch the speech. The speech packets are passed directly from phone to phone.
 IP Phone: There are two major categories of IP phone implementations, hard phone
and soft phone. The hard phone contains all the hardware and software to implement
Voice over Internet Protocol (VoIP). It is not a PC, but is specifically designed as a
phone. Hard phones can be simple in their functions, but can also have colour displays
with touch sensitive screens and may even support web browsing. There is no typical
hard phone on the market. The second category, the soft phone, is a headset connected to
a PC with all the telephone features implemented by the sound card and software resident
in the PC.
 Access Gateway/ Trunk Gateway: The gateway is usually part of the Voice over
Internet Protocol (VoIP) network. Most organizations will have legacy phones, fax
machines, modems, connections to the PSTN, and other devices that originally connected
to the organization telephone switch, called a PBX. When migrating to Voice over
Internet Protocol (VoIP), these devices and interfaces will have to be connected to a
conversion system that supports the legacy devices and interfaces on one side and
connects to the IP network on the other. The legacy devices will be connected to an
access/gateway and the PSTN interface connection will be terminated on a trunk
gateway.
1.6 Standards for Voice over Internet Protocol (VoIP):

“Standards are great, I have so many to choose from” is a state that amply describes Voice over
Internet Protocol (VoIP). There are multiple signalling standards.
 H.323: Is the ITU-T standard for packet based multimedia communication, though
originally developed for multimedia conferencing over LAN's it was later modified for
Voice over Internet Protocol (VoIP) as well H.323 was published in 1995, started the
development of Voice over Internet Protocol (VoIP) products and services. There are
four versions available. V1 is obsolete and has been discontinued in virtually all products.
Versions 2, 3 and 4 are used in today‟s products. These three versions are similar in
design and are upwardly compatible. This is the dominant installed signalling protocol
for use with hard and soft phones. With versions coming out in 1996 and 1998, the
standard has faced stiff competition from the other protocol SIP that was specifically
designed for Voice over Internet Protocol (VoIP), but is more used because of its wide
existence in the already installed networks. The standard is interoperable and has both
point-to-point and multipoint capabilities. H.323 uses a number of other sub protocols
for the various functions.
 H.245: Terminal Capability Exchange, Media Description, Control of Logical Channel
Also H.323 offers specifications for call control, channel setup, codecs for the
transmission of Real time video and voice over the networks where the QoS and
guaranteed services are not available. For the transport RTP is used for real time audio
and video streaming.
 SIP: The session Initiation Protocol (SIP) is the IETF standard for Voice over Internet
Protocol (VoIP) signalling. It is based on the existing protocols like SMTP and HTTP,
and uses a text based syntax that is comparable to HTTP uses in web addresses. A web
address is comparable to a telephone number in a SIP network, also the PSTN phone
numbers are also compatible in a SIP network ensuring interfacing with PSTN systems.
SIP also provides a mobility function to the users. SIP also supports multiple media

sessions during a single call hence users can share a game, use instant message and talk at
the same time. SIP works with most protocols like RTP, Session Description Protocol
(SDP), Session Announcement Protocol (SAP).
1.7 Fundamentals of WiMAX:
WiMAX technology is a telecommunications technology that offers transmission of wireless
data via a number of transmission methods; such as portable or fully mobile Internet access via
point to multipoint’s links. WiMAX offers around 72 Mega Bits per second without any need for
the cable infrastructure and is based on IEEE 802.16, it usually also called as Broadband
Wireless Access. WiMAX technology is actually based on the standards that making the
possibility to delivery last mile broadband access as a substitute to conventional cable and DSL
lines.
1.8 Types of WiMAX:
The WiMAX family (802.16) concentrate on two types of usage models: a fixed WiMAX and
mobile WiMAX. The basic element that differentiates these systems is the ground speed at
which the systems are designed to manage. Based on mobility, wireless access systems are
designed to operate on the move without any disruption of service; wireless access can be
divided into three classes; stationary, pedestrian and vehicular. A Mobile WiMAX network
access system is one that can address the vehicular class, whereas the fixed WiMAX serves the
stationary and pedestrian classes. This raises a question about the nomadic wireless access
system, which is referred to as a system that works as a fixed WiMAX network access system
but can change its location.
 Fixed WiMAX: Broadband service and consumer usage of fixed WiMAX access
is expected to reflect that of fixed wire-line service, with many of the standards-based
requirements being confined to the air interface. Because communications takes place via
wireless links from WiMAX Customer Premise Equipment (WiMAX CPE) to a remote
Non Line-of-sight (NLOS) WiMAX base station, requirements for link security are
greater than those needed for a wireless service. The security mechanisms within the

IEEE 802.16 standards are sufficient for fixed WiMAX access service. Another
challenge for the Fixed WiMAX access air interface is the need to set up high
performance radio links capable of data rates comparable to wired broadband service,
using equipment that can be self installed indoors by users, as is the case for Digital
Subscriber Line (DSL) and cable modems. IEEE 802.16 standards provide advanced
physical (PHY) layer techniques to achieve link margins capable of supporting high
throughput in NLOS environments.
Figure 5: Basic Fixed WiMAX station
 Mobile WiMAX. 802.16a extension, refined in January 2003, uses a lower
frequency of 2 to 11 GHz, enabling NLOS connections. The latest 802.16e task group is
capitalizing on the new capabilities this provides by working on developing a

specification to enable Mobile WiMAX clients. These clients will be able to hand off
between WiMAX base stations, enabling users to roam between service areas.
1.9 WiMAX Architecture
WiMAX is based on IEEE standard for high layer protocol such as TCP/IP, Voice over Internet
Protocol (VoIP), and SIP etc. WiMAX network is offering air link interoperability. The
Architecture of WiMAX is based on all IP platforms. The packet technology of WiMAX needs
no legacy circuit telephony. Therefore it reduces the overall cost during life cycle of WiMAX
deployment. The main guidelines of WiMAX Architecture are as under
 It support structure of packet switched. WiMAX technology including IEEE 802.16
standard and its modification, suitable for IETF and Ethernet.
 Offers flexibility to accommodate a wide range of deployment such as small to large
scale. WiMAX also support urban, rural radio propagation. The uses of mesh topologies
make it more reliable. It is the best coexistence of various models.
 Offers various services and applications such as multimedia, Voice, mandated dogmatic
services as emergency and lawful interception. Provides a variety of functions such as
ASP, mobile telephony, interface with multi internetworking, media gateway, delivery of
IP broadcasting such as MMS, SMS, WAP over IP.
 Supports roaming and Internet working. It support wireless network such as 3GPP and
3GPP2. It support wired network as ADSL.
 Supports global roaming, consistent use of AAA for billing purposes, digital certificate,
subscriber module, USIM, and RUIM.
 The range is fixed, portable, nomadic, simple mobility and fully mobility.
The WiMAX architecture consists of three logical entities: BS, ASN, and CSN.
All three correspond to a grouping for functional entities which may be single or distributed
physical device over several physical devices may be an implementation choice. The
manufacturer chooses any implementation according to its choice which is may be individual or
combine.

 Base station (BS): The responsibility of Base station (BS) is to provide that the air
interface to the MS. The other functionality of BS is micro mobility supervision
functions. The handoff prompting, supervision of radio resource, classification of traffic,
DHCP, keys, session and multicast group management.
 Access service network (ASN): The ASN (Access Service Network) used to
describe an expedient way to explain combination of functional entities and equivalent
significance flows connected with the access services. The ASN offers a logical boundary
for functional of nearby clients. The connectivity and aggregation services of WiMAX
are personified by dissimilar vendors. Planning of functional to logical entities
represented in NRM which may execute in unusual ways. The WiMAX forum allows
different type of vendors implementation that is interceptive and well-matched for a
broad variety of deployment necessities.
 Connectivity Service Network (CSN): CSN is a set of functions related to
network offering IP services for connectivity to WiMAX clients. A CSN may include
network fundamentals such as AAA, server, routers, and user database and gateway
devices that support validation for the devices, services and user. The Connectivity
Service Network also handled different type of task such as management of IP addresses,
support roaming between different NSPs, management of location, roaming, and mobility
between ASNs
The WiMAX architecture is offering a flexible arrangement of functional entities when
constructing the physical entities, Because AS may be molded into BTS, BSC, and an ASNGW,
which are equivalent to the GSM model of BSC, BTS and GPRS Support (SGSN).
1.10 How WiMAX Works?
WiMAX make possible the broadband access to conservative cable or DSL lines. The working
method of WiMAX is little different from Wifi network, because Wifi computer can be

connected via LAN card, router, or hotspot, while the connectivity of WiMAX network
constitutes of two parts in which one is WiMAX Tower or booster also known as WiMAX base
station and second is WiMAX receiver (WiMAX CPE) or Customer Premise Equipment.
Fig WiMAX network
The WiMAX network is just like a cell phone. When a user send data from a subscriber device to
a base station then that base station broadcast the wireless signal into channel which is called
uplink and base station transmit the same or another user is called downlink. The base station of
WiMAX has higher broadcasting power, antennas and enhanced additional algorithms.
WiMAX technology providers build a network with the help of towers that enable
communication access over many kilometres. The broadband service of WiMAX technology is
available in coverage areas. The coverage areas of WiMAX technology separated in series of
over lied areas called channel.

When a user sends data from one location to another the wireless connection is transferred from
one cell to another cell. When signal transmit from user to WiMAX base station or base to user
(WiMAX receiver) the wireless channel faces many attenuation such as fraction, reflection,
refraction, wall obstruction etc. These all attenuation may cause of distorted, and split toward
multi path. The target of WiMAX receiver is to rebuild the transmitted data perfectly to make
possible reliable data transmission.
The orthogonal frequency division multiplexed access (OFDMA) in WiMAX technology, is a
great technique used to professionally take advantage from the frequency bands. The
transmission frequencies of WiMAX technology from 2.3MHz to 3.5 GHz make it low price
wireless network. Each spectral profile of WiMAX technology may need different hardware
infrastructure. Each spectrum contains its bandwidth profile which resolved channel bandwidth.
The bandwidth signal is separately in OFDMA (Orthogonal Frequency Division Multiplexed
Access) which is used to carry data called sub carrier.
Transmitted data divided into numerous data stream where every one is owed to another sub
carrier and then transmitted at the same broadcast interval. At the downlink path the base station
broadcast the data for different user professionally over uninterrupted sub-carriers.
The independency of data is a great feature of OFDMA (Orthogonal Frequency Division
Multiplexed Access) that prohibit interfering and be multiplexed. It also makes possible power
prioritization for various sub carriers according to the link quality. The sub carrier having good
quality carry more data since the bandwidth is narrow.
WiMAX is providing quality of service (WiMAX QoS) which enables high quality of data like
Voice over Internet Protocol (VoIP) or TV broadcasts. The data communication protocol from
base station is alternative of quality of service (WiMAX QoS) application and offering video
streaming. These types of data translated into parameters or sub carriers per user.
All type of technique is carrying out together to speed up coverage, bandwidth, efficiency and
number of users. The base station of WiMAX has ability to cover up 30 miles. WiMAX
technology supports various protocols such as VLAN, ATM, IPv4 Ethernet, etc.

1.11Comparison between Wi-Fi and WiMAX
WiMAX which is based on the IEEE802.16 standard provides a wireless broadband technology.
Both the technical execution and the business case show the differences between WiMAX and
traditional Wi-Fi technology. As they are both wireless technology, most of the people consider
WiMAX as the robust of Wi-Fi. The simple comparison shows the advantage of WiMAX in
larger network coverage area and the faster transfer speed than Wi-Fi. Because of the
technologies reason and standardization issues, WiMAX does not present its better performance
in market position. Besides the newborn technology and the standardization issues, the relatively
high price also decreases the speed of WiMAX to occupy the market.
The physical layer for WiMAX at the start stage is IEEE 802.16 which limits the physical layer
will be operated in 10 GHz to 66 GHz. During go through the standard of IEEE 802.16a and
IEEE 802.16e, WiMAX obtain benefits from the network coverage, self installation, power
consumption, frequency reuse and bandwidth efficiency. The standard IEEE802.16d is used on
WMAN fixed and IEEE 802.16e is used on WMAN Portable. The throughput for Fixed WiMAX
is up to 75 Mbps with the 20MHz bandwidth while the portable WiMAX is up to 30Mbps with
10MHz bandwidth. Also, the network coverage of fixed WiMAX and the portable WiMAX is 4-
6 miles and 1-3 miles respectively.

Figure 2: Wi-Fi and WiMAX network convergence
The conflict between WiMAX and Wi-Fi is the resistance for WiMAX to develop. In order to
extend the reach of WiMAX technology, redundant efforts have been done to cooperate with the
traditional Wi-Fi. This is the only way to satisfy both the Wi-Fi supporters and those who focus
on the higher speed and larger range. While the Wi-Fi is playing a smaller role in the wireless
industry, the opportunity for wireless technologies to grow up and offer the high speed appears.
The following graph gives a outline of how Wi-Fi and WiMAX is integrated to work together to
approach a better performance in either distance or transfer speed. It is no doubt that there are
several ways to deliver the broadband service. The next figure shows the various technologies.
WiMAX Wireless Local Loop based on IP and OFDM, including wireless voice over IP Satellite
Smart antenna The next figure shows the example of wireless communication start from Wi-Fi
users physically located inside traditional Wi-Fi network area to request a broadband service
provided by WiMAX network.
1.12 Advantages of WiMAX Technology:
 Coverage: The single station of WiMAX can operate and provide coverage for
hundred of users at a time and manage sending and receiving of data at very high speed
with full of network security.
 High Speed: The High speed of connectivity over long distance and high speed voice
makes it more demanded in hardly populated areas plus compacted areas.
 Multi-functionality: WiMAX perform a variety of task at a time such as offering
high speed internet, providing telephone service, transformation of data, video streaming,
voice application etc. WiMAX is a great invention for new era because WiMAX has
enough potential for developing and opportunity to offer various types of services for
new generation. Now you can connect Internet anywhere and browse any site and make
possible online conference with mobile Internet, multimedia application never let you
bored, IPTV stay you up to date etc.

 Stay in touch with End user: WiMAX network always keep stay in touch with
your friends and all others using same WiMAX network because it provide absolute
communication service to the end users to make possible rich communications.
 Infrastructure: WiMAX infrastructure is very easy and flexible therefore it provides
maximum reliability of network and consent to actual access to end users.
 Cheap Network: WiMAX is a well-known wireless network now days because it
provides a low cost network substitute to Internet services offered via ADSL, modem or
local area network.
 Rich Features: WiMAX is offering rich features which make it useful. WiMAX
offers separate voice and data channel for fun, the semantic connection make your
network more secure then before, fast connectively, license spectrum, liberty of
movement etc.
 WiMAX vs Wifi: The WiMAX network providing much higher speed and very long
range as compared to Wife Technology.
 Smart antenna and Mesh Topology: The use of smart antenna in WiMAX
network offering high quality widest array which enable you to make possible
communication on long route without any encryption. It offers 2.3, 2.7, 3.3 and 3.8 GHz
frequency ranges. The use of Mesh topology in WiMAX network for the expansion is an
extensive spectrum of antennas for commercial as well as for residential users.
 Ultra wide Band: The unique and excellent infrastructure of WiMAX is offering
Ultra-Wideband. Its exclusive design is providing range from 2 to 10 GHz and
outstanding time response.
 Homeland Security: Security options of WiMAX Technology also offer very high
security because of encryption system used by WiMAX. Now you can exchange your
data on whole network without any fear of losing data.
1.13 WiMAX Limitations:
 Low bit rate over Long distance: WiMAX technology offering long distance data
range which is 70 kilometres and high bit rate of 70Mbit/s but both features doesn‟t

work together when we will increase distance range the bit rate will decreased and if we
want to increase bit rate then we should reduce the distance range.
 Speed of connectivity: The WiMAX other drawback is that any user closer to the
tower can get high speed up to 30Mbit/s but if a user exist at the cell edge from the tower
can obtain only 14Mbit/s speed.
 Sharing of bandwidth: In all wireless technology the bandwidth is shared between
users in a specified radio sector. Therefore functionality could go down if more than one
user exists in a single sector. Mostly user have a range of 2- to 8 or 12 Mbit/s services so
for better result additional radio cards added to the base station to boost the capability as
necessary.
 WiMAX vs Wi-Fi: Any one can build up a Wi-Fi network but to set up a WiMAX
network is really expensive so it is very hard for everyone that they pay large mount for
the setup and frequency license of WiMAX in a region.
.
1.14 Reasons for VoIP Deployment:
There are two major reasons to use VoIP: lower cost than traditional landline telephone and
diverse value-added services. Zeadally et al., [14] introduce how these factors influencing VoIP
adoption. Each of these will be described in this section
Cost Saving: This can be achieved by reusing the devices and wiring for the existing data
network as most of the organizations already have their own networks. However, the most
attractive reason to adopt VoIP maybe is dramatically reduced phone call cost. Soft phones such
as Skype [5] enable PC-to-PC users can bypass traditional long-distance toll calls charge as voice
traffic over the Internet, they only need to pay flat monthly Internet-access fee. Soft phones also
allow a PC as a VoIP phone to call a mobile phone or a home line phone at a lower rate.
Advanced Multimedia Applications. Cost effective is only one of the good reasons to
use VoIP. VoIP also enables multimedia and multi-service applications that increase productivity
and create a more flexible work environment, e.g. real time voice-enabled conferencing systems

that may include white boarding, file transferring, etc. which combine both voice and data
features.
1.15 Challenges of VoIP
Though VoIP is becoming more and more popular, there are still some challenging problems
with VoIP:
Bandwidth: Network availability is an important concern in network. A network can be
broken down into many nodes, links, and generate a large amount of traffic flows, therefore, the
availability of each node and link where we only concentrate the bandwidth of the VOIP system.
An in a data network, bandwidth congestion can cause QoS problems, when network congestion
occurs, packets need be queued which cause latency and jitter. Thus, bandwidth must be properly
reserved and allocated to ensure VOIP quality. Because data and voice share the same network
bandwidth in a VOIP system, the necessary bandwidth reservation and allocation become more
difficult. In a LAN environment, switches usually running at 100 Mbps (or 1000 Mbps),
upgrading routers and switches can be the effective ways to address the bandwidth bottlenecks
within the LAN.
Power Failure and Backup Systems: Traditional telephones operate on 48 volts and
supplied by the telephone line itself without external power supply. Thus, traditional telephones
can still continue to work even when a power failure occurs. However, backup power systems
required with VOIP so that they can continue to operate during a power failure. An organization
usually has a uninterruptible power system (UPS) for its network to overcome power failure,
desktop computers and other network devices may need much of the power to continue their
functions during power outages, a backup power assessment is needed to ensure that sufficient
backup power is available for the VOIP system. This may increase the costs of backup power
systems; costs may include electrical power charge to maintain UPS battery, maintenance costs,
UPS battery etc.
Security: As VoIP becomes more and more popular, the security issues relate to VoIP network
systems are also increasingly arising [37]. W. Chou [16] analysis the different aspects of VoIP

security and gives some suggested strategies to these issues. In reference [17], the authors also
outline the challenges of securing VoIP, and provide guidelines for adopting VoIP technology.
Soft phone: Soft phones are installed on computers thus should not be used where security is a
concern. In today’s world, worms, viruses, Trojan houses, spy wares and etc are everywhere on
the internet and very difficult to defend. A computer could be attacked even if a user does not
open the email attachment, or a user does nothing but only visit a compromised web site. Thus
use of soft phones could bring high risks for vulnerabilities.
Emergency calls: Each traditional telephone connection is tied to a physical location, thus
emergency service providers can easily track caller’s location to the emergency dispatch office.
But unlike traditional telephone lines, VoIP technology allows a particular number could be from
anywhere; this made emergency services more complicated, because emergency call centers
cannot know caller’s location or may not possible to dispatch emergency services to that
location. Although the VoIP providers provide some solutions for emergency calls, there is still
lack of industry standards in a VOIP environment.
Physical security: Physical security for VoIP networks is also an important issue. An attacker
could do traffic analysis once physically access to VoIP servers and gateways, for example,
determine which parties are communicating. Therefore, physical security policies and controls
are needed to restrict access to VOIP network components. Otherwise, risks such as insertion of
sniffer software by attackers could cause data and all voice communications being intercepted.
Wireless Security: Wireless nodes integrated in VoIP network is getting more and more
common and popular [36]. Wired Equivalent Privacy (WEP) security algorithm for 802.11
wireless networks is very weak because WEP can be cracked with publicly available software.
Due to the weakness of the WEP, more recent Wi-Fi Protected Access (WPA and WPA 2) which
administered by the Wi-Fi Alliance provides significant security improvements, the WPA
protocol can be integrated with wireless technology in VoIP.

1.16 VoIP QoS:
QoS is a very important aspect for IP-based multimedia services. Many IP services without QoS
guarantees from network providers are also very successful because transport quality is sufficient
to meet customer demands. However, QoS for these services cannot be guaranteed when services
grow and customer demands increase. For instance, IP-based voice and video services within
organizations usually do not have explicit QoS support because usually the LANs provide
enough bandwidth for real-time voice and video services. However, it is very hard to assure QoS
for real time multimedia services across worldwide networks. There are many factors affect
voice quality, which includes the choice of codec, delay, packet loss and jitter.
 Delay: High QoS should be assured by control delay so that one-way communication
delay should be less than 150 ms. (ITU states that one-way, end-to-end telephony
applications should have less than150 ms delay in echo-free environments to ensure user
satisfaction [31]). Delay mainly comes from three components [13]: (1) delay caused by
voice codec algorithms (2) delay caused by queuing algorithms of communications
equipment (3) variable delay caused by various factors (i.e. network conditions, VoIP
equipments, weathers etc). It is very important to minimise the voice traffic delay. Thus,
a codec algorithm and queuing algorithm needs to be carefully considered. Although
traditionally think the end-to-end delay of 150 ms was considered as acceptable for most
applications. However, in reference [35], the authors state that a delay of up to 200ms is
considered as acceptable. Moreover, a one way end-to-end delay between 150ms to
400ms is considered as acceptable for planning purposes. In this study, 200ms will be
considered as the maximum acceptable one way end-to-end delay, high end-to-end delay
can cause bad voice quality perceived by the end user.
 Jitter: Delay variation also called Jitter. Jitter is the difference value between the delays
of two queuing packets. Root causes of jitter including network conditions and packet
loss; it is very difficult to deliver voice traffic at a constant rate. In order to minimize
jitter a jitter buffer (also known as play out buffers) is needed. A jitter buffer is used to
trade off delay and the probability of packet interruption play out. Jitter value is

considered acceptable between 0ms and 50 ms and above this is considered as
unacceptable [11].
 Packet loss: Packet loss is also an important factor VoIP QoS. Packet loss occurs when
more transmitted packets on the network then causes dropped packets. VoIP packets are
very time sensitive. Therefore, packet loss can significantly affect VoIP quality. For
instance, a dropped conversation, delay between communicating clients, or noise on a
VoIP call. Acceptable packet loss rate is 1 % and it will be considered as unacceptable if
above this ratio [26]. However, an early study shows that the tolerable packet loss rates
are within 1-3% and the voice quality becomes intolerable when voice packet loss rate is
more than 3%.
Therefore, all these factors need to be properly controlled by QoS mechanisms. When these
factors are properly controlled, VoIP voice quality can be even better through lower speed
connections. In the meantime, data applications in the network can be also prioritized and
assured with limited and shared network resources. The quality VoIP is the key factor of VoIP
service to achieve success.

CHAPTER 2
LITERATURE REVIEW
2. Literature Review:
There have been an increase in global demand for wireless data services as well as real-time
applications like VoIP, audio and video streaming (Sengupta, Chatterjee & Ganguly, 2008); this
increasing demands is as a result of rapid growth which has been massively witnessed in several
wireless technologies recently. Countless researches are on-going in areas of wireless
technologies deployment (especially WiMAX) using Voice over IP based network system, all in
a bid to come up with a communication system that will be able to provide optimal wireless
services so as to meet the increasing user demands. As self-reliant units, holons have a degree of
independence and handle circumstances and problems on their particular levels of existence
without reaching higher level holons for assistance. The self-reliant characteristic ensures that
holons are stable, able to survive disturbances.

In their respective researches in (Salah, 2009) and (Yanfeng & Aiqun, 2006), the authors argue
that it is necessary that the capabilities of a network to support VoIP applications be measured
prior to its deployment with such network. According to them, the network’s readiness to support
deployment with VoIP system could be investigated by using network modeling and simulation
approaches, measuring for voice packet end-to-end delay, voice packet delay variation,
throughput and voice jitter after injecting real time (VoIP) traffic into the network. The author’s
argument if adhered to, will help in solving a great deal of problem as it will save both time and
resources instead of just deploying real-time applications such as VoIP with just any wireless
access technology without prior investigation of whether such network has any real-time
application support capabilities or not. With reference to Halepovic, Ghaderi & Williamson,
(2009), VoIP system has become increasingly popular more than ever even as WiMAX
Networks are been deployed in several countries across the globe. Hence, many researchers in
recent years as well as currently have focused extensively on different features of VoIP services
over WiMAX networks, all focused on investigating and identifying network add-on
performance criteria that will enhance the quality of service delivery of VoIP system over
WiMAX networks.
In (Flizikowski, Majewski, & Przybyszewski, 2010), the authors have investigated to a
remarkable extent the audio, data and video support features in WiMAX Networks. Their
research was focused on examining the QoS deployment over WiMAX Networks and
comparison of the performance achieved using WiMAX service classes like Unsolicited Grant
Service (UGS) and Extended real time Polling Service (ertPS). The studies carried out by these
authors have confirmed that WiMAX Networks supports real-time application more compared to
other wireless access technologies like WLAN and 3G.
A traffic-aware scheduling algorithm for the deployment of VoIP applications over WiMAX
Networks have been proposed in (Ansari & Haghani, 2008), the authors critically examined the
performance of the proposed method in comparison with various notable conventional methods.
They further explained how the efficiency of VoIP over WiMAX networks performance can be
improved upon by the application of their proposed scheduling methods. But their proposed

algorithm was not investigated under known performance metrics to ascertain and establish its
robustness in QoS supports.
The authors in (Shrivastava & Vannithamby, 2009) maintain that though WiMAX Networks are
efficient in supporting data traffic, the capacity of VoIP when deployed over IEEE 802.16e
WiMAX system is not impressive as a result of overwhelming MAP overhead always generated
by dynamic scheduling of VoIP traffic. In their work, they adopted persistent scheduling as a
mechanism in IEEE 802.16e WiMAX system in order to minimise MAP overhead occurrence.
The only deficiency inherent in their proposed persistent/group scheduling mechanism is that it
creates sort of “resource hole” in the frame at the data allocation region which leads to inefficient
resource allocation. Majority of the VoIP QoS investigations have been conducted on Ethernet
LAN, Wireless LAN as opposed to WiMAX access networks. In most of the occasions where it
has actually been done with the deployment of WiMAX networks, the researchers have failed to
look at some notable complex codec algorithms/schemes with reduced value of voice frame size
per packet that can be applied on voice/video calls/conference to enhance the quality of VoIP
performance when deployed over WiMAX Networks. These complex codec schemes provide
tremendous compression efficiency that saves network bandwidth essentially in wireless
technologies like WiMAX networks. Moreover, majority of the researchers that have done some
related work on investigation of VoIP system performance when deployed over WiMAX
networks used other network simulators such as NS-2, E-modeling, NetSim, etc for designing,
modeling and simulation of the network which most computer networking students and
professional industrial practitioners are not quite very much conversant with. This research will
use OPNET Network Simulator which is rather universal among IT students and communication
professional to look at the performance criteria as well as some codec algorithms approved and
specified by ITU-T for voice and video conferencing using VoIP over IEEE 802.16e standard
WiMAX networks in a campus network.
2.1VoIP Protocols:
2.1.1 H.323:

There are two standard protocols used in VoIP network: Session Initiation Protocol (SIP) and
H.323, (Skype [1] and some others use proprietary signalling and messaging protocols). H.323
[6] is ITU (International Telecommunication Union) standard based on Real-time Protocol (RP)
and Real-Time Control Protocol (RTCP); H.323 is a set of protocols for sending voice, video and
data over IP network to provide real-time multimedia communications. H.323 is reliable and
easy to maintain technology and also is the recommendation standard by ITU for multimedia
communications over LANs [8], [9]. Figure 1.2 shows the H.323 architecture. There are four
basic entities in a default H.323 network [9], [10]: terminal, gateways (GW), gatekeepers (GK)
and multipoint control units (MCU): H.323 terminal also called H.323 client is the end-user
device. It could be IP telephone or a multimedia PC with another H.323 client. That provides
real-time two-way media communication. A Gateway (GW) is an optional component that
provides inter-network translation between terminals. A Gatekeeper (GK) is an optional
component provides address translations and access control services. A Multipoint Control Unit
(MCU) functions as a bridge or switch that enables three or more terminals and gateways in a
multipoint conference.

Figure1.2: H.323 Architecture
2.1.2 SIP
H.323 has some limitations such as lack of flexibility, thus another protocol SIP is getting
popular in VoIP [41]. SIP (stands for Session Initiation Protocol) was developed by the Internet
Engineering Task Force (IETF) and published as RFC 3261 [12]. SIP is a signalling control
protocol which is similar to http, it’s designed to initial and terminate VoIP sessions with one or
more participants [11]. It is less weight and more flexible than H.323 that also can be used for
multimedia sessions such as audio, video and data. Figure 1.3 shows the architecture of SIP
protocol.
SIP has two components: User Agents and SIP servers. User agents are peers in a SIP. User
agents could be either an agent client or an agent server. A user agent client initiates by sending a
SIP request. A user agent server can accept, terminate or redirect the request as responses to this
SIP request. There are three types of SIP servers include SIP proxy servers, SIP registrar servers,
and SIP redirect servers. A SIP server functions as a server that handles these requests, e.g.
requests transferring, security, authentication, and call routing.

Figure 1.3: SIP Architecture
SIP is not only popular in VoIP applications but also widely used in applications include instant
messaging and some other commercial applications, e.g. Microsoft MSN Messenger, Apple
iChat.
2.2 VoIP Compression Algorithms:
Codecs generally provide a compression capability to save network bandwidth. Currently, there
are many different audio codecs available for voice applications. The simplest and most widely
used Codecs are G.711, G.723 and G.729 [7]. The simplest encoder scheme is G.711 (64 kb/s).
G.711 is the sample based which uses Pulse Code Modulation (PCM). The acceptable packet
loss factor of G.711 is up to 0.928%. G.723 and G.729 are frame based encoder scheme with
higher compression and smaller data rates (8 kb/s for G.729, 5.3 and 6.4 kb/s for G.723.1).

2.2.1 VoIP Compression Schemes
The quality of voice and video services in VoIP system deployed over WiMAX network are
affected by many factors ranging from hardware, software, bandwidth, broadband connection
and even the technology we use (Muntean, Otesteanu, & Muntean, 2010). These factors are all
under our control as we can change, replace or even improve on them; but the VoIP system voice
quality over wireless technologies such as Wi-Fi, WLAN, 3G and WiMAX are not under the
users’ control. Hence, this research intends to use network simulation methodology to identify
the best data compression algorithms and other relevant network performance add-on parameters
that will improve the QoS of VoIP services deployed over WiMAX network. VoIP data
compression according to Cignoni et al (2008) is a process through which voice data are
rendered less unnecessarily large using compression software (also known as codecs) for easy
transmission over IP-based networks as well as to enhance the voice quality upon reception.
These codecs encode and transform the voice analog signals into digital data that are further
compressed into much lighter packets which are therefore transferred over the Intranet or
Internet. Decompression (or decoding) process is used at the destination point to decompress the
packets and realise the original analog voice signals which the user (receiver) can hear. Given
that analog voice and video signals cannot be transmitted over IP-based networks, they are
encoded before transmission using codec schemes.
2.3 VoIP over WiMAX:
Voice over Internet Protocol (VoIP) provides an alternative to the telephone service offered by
the traditional Public Switched Telephone Network (PSTN) by using an IP network to carry
digitized voice. Packet switched air interfaces that support flat IP architectures have now made it
possible to run VoIP applications over wireless technology. Compression/Decompression
(CODEC) techniques for VoIP transform audio signals into digital bit streams. While preserving
voice quality, speech samples are further compressed to produce bit streams of 8–12 kbps that
are carried over the IP network. The compressed speech sample is then transmitted using the
Real-time Transport Protocol (RTP) over the User Datagram Protocol (UDP) over the Internet
Protocol (IP). VoIP over wireless networks is affected by the choice of CODEC and packet loss,

delay and jitter. Fluctuating channel conditions typically cause packet loss and increased latency.
In order to keep mouth-to-ear round trip latencies to reasonable levels of 250–300 ms, the delay
budget for transmission over the air interface is 50–80 ms. The CODEC, jitter buffer and
backbone account for the remaining delay. Channel aware scheduling with Quality of Service
(QoS) differentiation, Hybrid Automatic Repeat Request (HARQ) and dynamic link adaptation
are used to keep delays within acceptable limits. Jitter buffers are used to compensate for delay
jitter experienced by packets due to network congestion, timing drift or route changes.
2.4 Features to Support VoIP over WiMAX:
WiMAX provides a number of features to support VoIP. Prioritization of delay-sensitive VoIP
traffic is achieved through the classification of flows into scheduling classes. Voice activity
detection and Extended Real-Time Polling Service (ertPS) conserve air link resources during
periods of silence. HARQ and channel aware scheduling are used reduce transmission latency
over the air link. Protocol header compression is supported to transport the speech sample
efficiently.
1. Silence Suppression using ertPS: Mobile WiMAX supports QoS requirements for
a wide range of data services and applications by mapping those requirements to
unidirectional service flows that are carried over Uplink (UL) or Downlink (DL)
connections. Table below describes the five QoS classes, Unsolicited Grant Service
(UGS), Real-Time Polling Service (rtPS), ertPS, Non-Real-Time Polling Service (nrtPS)
and Best Effort (BE) service, used to provide service differentiation by the Medium
Access Control (MAC) scheduler. In the absence of silence suppression, service
requirements for VoIP flows are ideally served by the UGS, which is designed to support
flows that generate fixed size data packets on a periodic basis. The fixed grant size and
period are negotiated during the initialization process of the voice session.
Service flows such as VoIP with silence suppression generate larger data packets when a
voice flow is active, and smaller packets during periods of silence. rtPS is designed to
support real-time service flows that generate variable size data packets on a periodic
basis. rtPS requires more request overhead than UGS, but supports variable grant sizes. In
conventional rtPS, a bandwidth request header is sent in a unicast request opportunity to

allow the Subscriber Station (SS) to specify the size of the desired grant. The desired
grant is then allocated in the next UL sub frame.
Although the polling mechanism of rtPS facilitates variable sized grants, using rtPS to switch
between VoIP packet sizes when the SS switches between the talk and silent states introduces
access delay. rtPS also results in MAC overhead during a talk spurt since the size of the VoIP
packet is too large to be accommodated in the polling opportunity, which only accommodates a
bandwidth request header. The delay between the bandwidth request and subsequent bandwidth
allocation with rtPS could violate the stringent delay constraints of a VoIP flow. rtPS also incurs
a significant overhead from frequent unicast polling that is unnecessary during a talk spurt.
The ertPS scheduling algorithm improves upon the rtPS scheduling algorithm by dynamically
decreasing the size of the allocation using a grant management sub header or increasing the size
of the allocation using a bandwidth request header. The size of the required resource is signalled
by the Mobile Station by changing the Most Significant Bit (MSB) in the transmitted data.
2. HARQ: In addition to link adaptation through channel quality feedback and adaptive
modulation and coding, HARQ is enabled in 802.16e using the „stop and wait‟ protocol,
to provide a fast response to packet errors at the Physical (PHY) layer. Chase combining
HARQ is implemented to improve the reliability of a retransmission when a Packet Data
Unit (PDU) error is detected. A dedicated Acknowledgment (ACK) channel is also
provided in the uplink for HARQ ACK/Negative Acknowledgment (NACK) signalling .
UL ACK/NACK messages are piggybacked on DL data. A multi-channel HARQ
operation with a small number of channels is enabled to improve the efficiency of error
recovery with HARQ. Mobile WiMAX also provides signalling to allow asynchronous
HARQ operation for robust link adaptation in mobile environments. The one-way delay
budget for VoIP on the DL or the UL is limited between 50 and 80 ms. This includes
queuing and retransmission delay. Enabling HARQ retransmissions for error recovery
significantly improves the ability of the system to meet the stringent delay budget
requirements and outage criteria for VoIP. Hybrid automatic repeat request (Hybrid ARQ
or HARQ) is a combination of forward error-correcting coding and error detection using

the ARQ error-control method. In standard ARQ, redundant bits are added to data to be
transmitted using an error-detecting code such as cyclic redundancy check (CRC). In
Hybrid ARQ, forward error correction (FEC) bits are added to the existing Error Detection
(ED) bits to correct a subset of all errors while relying on ARQ to detect uncorrectable
errors. As a result Hybrid ARQ performs better than ordinary ARQ in poor signal
conditions, but in its simplest form this comes at the expense of significantly lower
throughput in good signal conditions. There is typically a signal quality cross-over point
below which simple Hybrid ARQ is better, and above which basic ARQ is better.
3. Simple Hybrid ARQ: The simplest version of HARQ, Type I HARQ, adds both ED
and FEC information to each message prior to transmission. When the coded data block is
received, the receiver first decodes the error-correction code. If the channel quality is good
enough, all transmission errors should be correctable, and the receiver can obtain the
correct data block. If the channel quality is bad, and not all transmission errors can be
corrected, the receiver will detect this situation using the error-detection code, then the
received coded data block is rejected and a retransmission is requested by the receiver,
similar to ARQ. In a more sophisticated form, Type II HARQ, the message originator
alternates between message bits along with error detecting parity bits and only FEC parity
bits. When the first transmission is received error free, the FEC parity bits are never sent.
Also, two consecutive transmissions can be combined for error correction if neither is
error free. To understand the difference between Type I and Type II Hybrid ARQ,
consider the size of ED and FEC added information: error detection typically only adds a
couple bytes to a message, which is only an incremental increase in length. FEC, on the
other hand, can often double or triple the message length with error correction parities. In
terms of throughput, standard ARQ typically expends a few percent of channel capacity
for reliable protection against error, while FEC ordinarily expends half or more of all
channel capacity for channel improvement. In standard ARQ a transmission must be
received error free on any given transmission for the error detection to pass. In Type II
Hybrid ARQ, the first transmission contains only data and error detection (no different
than standard ARQ). If received error free, it's done. If data is received in error, the second

transmission will contain FEC parities and error detection and error correction can be
attempted by combining the information received from both transmissions. Only Type I
Hybrid ARQ suffers capacity loss in strong signal conditions. Type II Hybrid ARQ does
not, because FEC bits are only transmitted on subsequent retransmissions as needed. In
strong signal conditions, Type II Hybrid ARQ performs with as good capacity as standard
ARQ. In poor signal conditions, Type II Hybrid ARQ performs with as good sensitivity as
standard FEC.
4. Hybrid ARQ with Soft Combining: In practice, incorrectly received coded data
blocks are often stored at the receiver rather than discarded, and when the retransmitted
block is received, the two blocks are combined. This is called Hybrid ARQ with soft
combining. While it is possible that two given transmissions cannot be independently
decoded without error, it may happen that the combination of the previously erroneously
received transmissions gives us enough information to correctly decode. There are two
main soft combining methods in HARQ: Chase combining: every retransmission contains
the same information (data and parity bits). The receiver uses maximum-ratio combining
to combine the received bits with the same bits from previous transmissions. Because all
transmissions are identical, Chase combining can be seen as additional repetition coding.
One could think of every retransmission as adding extra energy to the received
transmission through an increased Eb/N0.
Incremental redundancy: every retransmission contains different information than the
previous one. Multiple sets of coded bits are generated, each representing the same set of
information bits. The retransmission typically uses a different set of coded bits than the
previous transmission, with different redundancy versions generated by puncturing the
decoder output. Thus, at every retransmission the receiver gains extra knowledge. HARQ
can be used in stop-and-wait mode or in selective repeat mode. Stop-and-wait is simpler,
but waiting for the receiver's acknowledgment reduces efficiency. Thus multiple stop-and-
wait HARQ processes are often done in parallel in practice: when one HARQ process is
waiting for an acknowledgment, another process can use the channel to send some more
data.

5. Channel Aware Scheduling: Unidirectional connections are established between
the BS and the MS to control transmission ordering and scheduling on the mobile
WiMAX air interface. Each connection is identified by a unique Connection Identification
(CID) number. Every MS, when joining a network, sets up a basic connection, a primary
management connection and a secondary management connection. Once all of the
management connections are established, transport connections are set up. Traffic
allocations on the DL and the UL are connection based, and a particular MS may be
associated with more than one connection. In every sector, the Base Station (BS)
dynamically schedules resources in every Orthogonal Frequency Division Multiple Access
(OFDMA) frame on the UL and the DL in response to traffic dynamics and time-varying
channel conditions. Link adaptation is enabled through channel quality feedback, adaptive
modulation and coding and HARQ. Resource allocation on the DL and UL in every
OFDMA frame is communicated in Mobile Application Part (MAP) messages at the
beginning of each frame. The DL-MAP is a MAC layer message, which is used to allocate
radio resources to Mobile Stations (MS) for DL traffic. Similarly, the UL-MAP is a MAC
layer message used to allocate radio resources to the MS for UL traffic. The BS uses
information elements within the DL-MAP and UL-MAP to signal traffic allocations to the
MS. The BS scheduler also supports resource allocation in multiple sub-channelization
schemes to balance delay and throughput requirements with instantaneous channel
conditions.
2.5 VoIP Traffic Characteristics
There are several characteristics of VoIP traffic that make VoIP packet scheduling challenging:
 VoIP packets are small in size.
 Number of VoIP users supported in a given frequency band is large compared with the
number of high data rate users that can be supported.
 The packet inter-arrival time is roughly constant.
 Speech includes periods of silence for roughly half the time and activity during the rest of
the time.

The fact that the VoIP packet size is small makes the ratio of the resources needed for
transmitting control information to schedule VoIP to the resources needed for actual VoIP traffic
transmission much higher than that observed in data-only systems. Moreover, the high number of
VoIP users supported in a given frequency band also adds to the total overhead required to
transmit the control information related to the VoIP resource allocation.
In supporting high data rate applications, the focus is on optimizing the throughput, but in
supporting VoIP the focus shifts towards delay sensitivity and minimizing the control overhead
associated with the VoIP resource allocation.
2.6 Dynamic Resource Allocation for VoIP:
To support VoIP in an OFDMA system, VoIP packets need to be scheduled on the DL and the
UL within a fixed delay bound every time a packet arrives at the BS and at the MS, respectively.
The OFDMA resources in frequency and time as well as transmit power and transmission mode
need to be specified in each allocation. Furthermore, the MS identification and HARQ
transmission related information also need to be specified.
All this information is sent using a robust Modulation and Coding Scheme (MCS), thereby
consuming additional resources. In WiMAX, control information associated with resource
allocation is signalled through MAP elements. Compressed MAPs can be used with sub MAPs to
reduce MAP overhead. The compressed MAP header is coded with the most robust MCS and
sub Maps can be coded with higher order MCSs. Although compressed MAPs and sub MAPs
conserve resources compared to conventional MAPs, MAP overhead associated with the larger
number of allocations for VoIP can be considerably high. Dynamic scheduling for every VoIP
packet incurs a significant amount of MAP overhead. The motivation for persistent scheduling
comes from the fact that the VoIP traffic is periodic and generates constant size packets. As the
name suggests, persistent scheduling conserves resources by persistently allocating resources
that are required periodically. We discuss two different ways of persistently allocating the
resources, namely individual persistent scheduling and group scheduling.
 Individual Persistent Scheduling: The basic idea behind individual persistent
scheduling is that a user is assigned a set of resources for a period of time and the
necessary information for the packet transmission are sent only once at the beginning of

the assignment. For the rest of the period of allocation, the MS is assumed to know all of
the information for data reception on the DL and data transmission on the UL. Note that
the allocation period can be infinite. In other words, persistent scheduling is in effect until
updated.
Figure compares the operation of dynamic and persistent scheduling operation. In the
case of dynamic scheduling, a MAP element is required to specify resource allocation
information every time a VoIP packet is scheduled. On the other hand, in the case of
persistent scheduling, resource allocation information is sent once in a persistent MAP
element and not repeated in the subsequent frames. The additional resource that becomes
available due to MAP overhead reduction can be used to increase VoIP capacity.
Resource allocation/deallocation for talk spurts/silence periods
As discussed earlier, VoIP users switch between talk spurts and silence. On the average, users in
a typical VoIP call will be in either mode for duration of the order of a second.
Every time a user goes into a talk spurt, resources need to be allocated with all of the information
necessary to identify the allocation. The resource is allocated periodically with persistent
scheduling as long as the user is in the active state. Similarly, every time the user goes into the
silence mode, resources need to be deallocated. Since the frequency of allocation and
deallocation of resource for conversational voice (50% voice activity factor) is typically once
every 250 WiMAX frames (1.25 s), the overhead associated with a persistently scheduled
allocation is small compared with the overhead in dynamic scheduling.
 Link Adaptation/MCS Changes: In a mobile environment, the channel
conditions are time varying. In order to be spectrally efficient, the MCS used for data

transmission and reception needs to be adapted according to channel variations.
Adjustment in MCS requires changes in the amount of allocated resources. As a result,
every time the MCS needs to be adapted, the BS needs to deallocate or allocate a
persistently scheduled resource. Depending on the frequency at which the MCS changes,
signalling the changing persistent allocation could result in considerable overhead.
Consequently, for fast link adaptation, individual persistent scheduling is not
recommended, since the overhead involved in adapting to the channel variations will
defeat the purpose of persistent scheduling.
 HARQ Retransmission: Depending on the operating point chosen by the vendor or
the network operator for initial transmissions, HARQ retransmission rates are typically in
the range of 10–30%. Allocation of resources for HARQ retransmissions is an important
consideration. Although persistent scheduling for HARQ retransmissions may be
possible, dynamic allocation of resources is recommended.
 Resource Holes: One issue with individual persistent scheduling is the associated
resource packing inefficiency in the data portion of the frame. As different users
transition between active spurts and periods of silence, or MCS changes occur due to link
adaptation, resources need to be deallocated and allocated dynamically in addition to
scheduling persistent allocations. Every time a resource is deallocated it may or may not
be possible to find a user with the same resource request. If there is no match, holes can
be created in the data region. Figure shows an example of the creation of resource
allocation holes. Hole creation can also defeat the purpose of using persistent scheduling
for efficient resource allocation.

 Implicit and Explicit Allocation: Deallocation and allocation does not always
need to be explicit. It is possible to implicitly allocate/deallocate resources to reduce the
MAP overhead, for example user N1 is allocated at location L1 with MCS1. When the
user is deallocated and another user N2 with the same MCS need to be allocated, the BS
can simply allocate user N2 at location L1. User N1 interprets this allocation to user N2
as a deallocation of resources, thereby eliminating the need for explicit signalling of the
deallocation for user N1. The broadcast nature of the 802.16e MAP offers this advantage
and provides a mechanism to further reduce the overhead.
 Resource Shifting/Repacking: A mechanism to remove the holes that have been
created using individual persistent scheduling is resource shifting or repacking. Basically,
the BS broadcasts the size and location of the holes/empty spaces. Using this information,
the remaining user allocations can be shifted upwards to account for the holes.
Broadcasting information related to the empty space results in some overhead. When the
benefit of making the resources available may be higher than the cost of broadcast
overhead, it is desirable to perform the shifting operation. It is also possible to tie the
shifting operation with each of the deallocation operations, that is, every time there is a
deallocation and there is no allocation in its place, all allocations following the
deallocated space are automatically shifted up. Figure shows an example of a resource
shifting mechanism to pack allocations more efficiently and make more resources
available.

 Reliable MAP Reception: Since persistent scheduling allocates resources not only
for the current frame, but also for future frames, the impact of losing MAP information
associated with persistent scheduling is much higher than it is with dynamic scheduling.
Hence, it is important to make the MAP transmission reliable. A MAP ACK channel can
be used to ensure that the MS has received the persistent allocation. This ACK channel is
very similar to the HARQ ACK channel. One issue with this approach is that the ACK
channels increase overhead in the uplink. In order to reduce the ACK channel overhead
in the uplink, shared NACK channels can be used to acknowledge subsequent MAP
reception, that is, a MAP ACK channel is used for the initial persistent scheduling
assignment, and a shared MAP NACK channel is used for the subsequent assignments. In
the case of the shared MAP NACK channel, multiple users use the same NACK channel
resource. If the BS receives a MAP NACK signal from one or more users, the incorrect
reception of the MAP is detected. The BS can either retransmit the MAP information to
all users sharing the NACK channel or intelligently retransmit the MAP information to
only those users who signaled the loss of the MAP information. For the latter case, the
BS can monitor the HARQ ACK, etc. for the MSs sharing the MAP NACK channel.
Figure shows an example of the operation of the error recovery procedure.

 Group Scheduling: In order to overcome the inefficiency of Persistent Scheduling,
we propose Group Scheduling of VoIP connections, in which MSs are clustered into
multiple groups based on a given criteria. An MS has some persistence within the group’s
resources. The location of group’s resources is persistent within the frame as long as

there are no changes to any of the allocations in the frame. If there are any changes in
allocation, the group’s resources as a whole can be moved within the frame to fill
resource holes. Since the individual MS‟s location is fixed with the group’s starting
location, each individual user need not be signaled this change in allocation. A bitmap
provides flexibility in scheduling within the group itself by indicating which users of the
group are scheduled in a frame. This allows the users to efficiently pack their resources
and prevents resource wastage if some users don’t have packets to send in that frame.
CHAPTER 3
PROJECT ESSENTIALS
3 Network Designs and Implementation:
3.1 Network Design:
OPNET modeller 16.1 is the software tool used for designing the networks and implementing
them. This tool is very user friendly and easily understandable for everyone. Numerous
advantages of this tool make this more popular when compared to the other network simulation
tool like NS1, NS2, OMNET, etc. Structure of OPNET simulation tool is divided into three main
classifications or domains: network domain, node domain and process domain. Basic computing
languages like C and C++ are used for building the components of OPNET, so these codes can
be changed according to our requirements and we can build our own components based on the
requirements. These network simulation tools are very useful for the researchers and network
developers for developing the new network and also to test the network that are going to be built
in the real world. These procedures save the money for the organization.
Network administrators can find the problems in the network and they can be rectified before
going into the real time. Existing networks can also be analyzed in regular time periods and they

can be updated in efficient manners with the use of network simulators. In the present thesis, two
scenarios are built for testing the performance of the encoding techniques. Scenario 1 is the small
scale network, which can be treated as one country. This network resembles the network of the
organization or a company which has few branches in same country. Scenario 2 is the large scale
network, which can be treated as worldwide network. This network resembles the network of the
organization or a company which has few branches in different countries around the world.
During constructing the networks, different scenarios are designed by using the facilities of the
OPNET. From the designed network efficient networks are presented in this documentation.
Following sections gives the detailed explanation of both scenarios, there configurations and the
implementation process. Results collected from these scenarios are analyzed in the next section.
3.2 Requirements and Components
General requirements for this implementation are OPNET 16.0 modeller and a PC. OPNET 16.0
modeler can’t be installed in our personal computer, so the lab resources of University of
Bedfordshire are used. Few books are referred before starting the implementation process, which
can be helpful for me to get better implementation steps. This OPNET is the included syllabus
for wireless technologies of my masters’ course, so the practical sections helped me mostly in
getting the basics about this simulation tool. All the components required for the
implementations and design processes are available in the object palette of OPNET modeler.
There is vast availability of objects in object palette, which resembles the real time components
and performs the same functionality as the real time components.
General components required here are application definition node, profile definition node, IP
QoS node, subnets, end users (WLAN workstations), connecting links, routers, and servers. For
any network application definition, profile definition and IP QoS nodes are the configurations
nodes. These nodes are enough to be configured for providing the additional features to the
network. These nodes contain the info like applications provide to the network, different profiled
applications for the specific users and maintain the quality of the network. Applications and
profile definition nodes are needed to be configured according to our requirements, whereas the

IP QoS node is self configured. All the components are available in the object palette, it is
enough to drag and drop the required components into the work space. After the required objects
are dragged into the workspace we need to configure them for making the network to function.
Configuration of components must be done carefully and correctly, or else the functionality of
the network changes.
3.3 Chosen Methodology:
3.3.1 Methodology:
OPNET is the simulation tool used for designing the network and deploying VOIP technology.
Theoretical research is done on H. 323, SIP and security solutions. Market survey is done on the
awareness and requirements of the users from the VOIP service providers.
3.3.2 Scenarios:
Small and large scale network scenarios are designed for implementation. Traffic in the networks
is varied by using the various combinations of links.
Scenario 1:
Scenario 1 is the small scale network, which can be treated as one country. This network
resembles the network of the organization or a company which has few branches in same
country. For the organizations like University of Bedfordshire or any another organization which
has few branches in same country can use this network for implementing the VOIP technology
into their existing network or they can also build a new network by following the configuration
and implementation steps presented in this document. Along with the VOIP application, for
making the network little complicated, few other applications are provided for the users. File
transfers and Video conversations are other additional application along with voice over IP
application. As the main aim of this presentation is to test the performance of the encoding
techniques, so from the available encoding techniques of OPNET G.711, G.723 and G.729 are
selected for implementation, as these are the mostly used encoding techniques in the real time
applications.

Scenario 2:
Scenario 2 is the large scale network, which can be treated as worldwide network. This network
resembles the network of the organization or a company which has worldwide branches. For the
organizations like Universities or any another organization which has few branches throughout
the world can use this network for implementing the VOIP technology into their existing
network or they can also build a new network by following the configuration and implementation
steps presented in this document. Along with the VOIP application, for making the network little
complicated, few other applications are provided for the users. File transfers database storage
and http browsing are other additional application along with voice over IP application. As the
main aim of this presentation is to test the performance of the encoding techniques, so from the
available encoding techniques of OPNET G.711, G.723 and G.729 are selected for
implementation, as these are the mostly used encoding techniques in the real time applications.
3.3.3 Simulation parameters:
Network area, packet size, node type, DES statistics, encoding techniques, network topology and
time of simulation

REFERENCES:
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going," JISC Technology and Standards Watch, 2006.
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and A. A. Richard III, PSTN Architecture for Video-on-Demand Services, 1993.
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Internet Protocol (VoIP), and voice over wi-fi," in Global Telecommunications Conference,
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[6] V. Rakocevic, R. Stewart and R. Flynn, "Voice over Internet Protocol (VoIP) Network
Dimensioning using Delay and Loss Bounds for Voice and Data Applications," Techincal
Report, 2008.
[7]S. Sengupta, M. Chatterjee and S. Ganguly, "Improving quality of Voice over Internet
Protocol (VoIP) streams over WiMax," Computers, IEEE Transactions on, vol. 57, pp. 145-156,
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[8]E. Halepovic, M. Ghaderi and C. Williamson, "Multimedia application performance on a
WiMAX network," in Proc. Of Annual Multimedia Computing and Networking Symposium,
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[9] I. Adhicandra, "Measuring data and Voice over Internet Protocol (VoIP) traffic in wimax
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[11] E. Haghani and N. Ansari, "Voice over Internet Protocol (VoIP) traffic scheduling in
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2006.
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[14] S. Jadhav, H. Zhang and Z. Huang, "Performance evaluation of quality of Voice over
Internet Protocol (VoIP) in WiMAX and UMTS," in Parallel and Distributed Computing,

Applications and Technologies (PDCAT), 2011 12th International Conference on, 2011, pp. 375-
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