How to use IPTV

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A Detail Performance Evaluation of Internet
Protocol Television (IPTV) Triple Play Services
over IP Based Networks
A Synopsis Submitted in the Partial Fulfilment of
The Award of the Degree of
MASTER OF TECHNOLOGY
IN
COMPUTER SCIENCE AND ENGINEERING
Under Guidance of: Submitted By:
Name of Internal Guide Name of Students
(Designation) Roll No

Abstract
In telecommunications, Triple Play service is a marketing term for provisioning of two
bandwidth-intensive services, high-speed Internet access and television, and a less bandwidth-
demanding (but more latency-sensitive) telephone service, over a single broadband connection.
In this thesis, the effect of mobility of mobile WiMAX subscribers on video on demand (VOD)
over WiMAX is analyzed by considering the scalable video coding (SVC) codes for video
streaming. This experiment has been carried out using OPNET modeller 14.5. To compare the
performance of Internet Protocol television (IPTV) over WiMAX, the packet delay variation,
packet end to end delay, delay and load matrices are used. The result shows that after certain
speed, the load increases, the delay again decreases and there is no change in the packet delay
variation and packet end to end delay.
Key words: WiMAX, OPNET, scalable video coding (SVC), wireless networks, IEEE 802.16,
internet protocol television (IPTV).

1. Introduction:
Worldwide Interoperability for Microwave Access (WiMAX) technology is the only wireless
system capable of offering high QoS at high data rates for IP networks. It supports data rates of
70Mbps over ranges of 50km with mobility support at vehicular speeds [1]. An attractive growth
rates in WiMAX subscriber base and equipment revenues in market studies last years, 133
million subscribers will be supported at the end of 2012 [2]. There is an increasing trend to
deploy WiMAX technology for offering different application, such as voice, data, video, and
multimedia services. Each of these applications has different QoS requirements.
Figure 1: IPTV application over WiMAX
Internet Protocol Television (IPTV) has become popular as it promises to deliver the content to
users whenever they want. IPTV is a set of multimedia services that are distributed throughout an
IP network, where end of user receives video streams through a set-top-box (STB) connected to a
broadband connection. IPTV is often combined with the services of VoD. VoD services contents
are not live but pre-encoded contents available at any time from servers. These services must
possess an adequate level of quality of service, security, interactivity, and reliability. From the

perspective of the provider, IPTV includes the video acquisition, video processed and video
secure distribution on the IP network infrastructure [3], [4]. WiMAX technology is one of the
access technologies that enables transmission of IPTV services. Transmitting IPTV over
WiMAX aims to make IPTV services available to users anywhere, anytime and on any device.
The QoS for delivering IPTV services depends especially on network performance and
bandwidth [5]. The generic network topology of the IPTV application over WiMAX is shown in
Figure 1
1.1 Problem Statement
In this thesis, the effect of mobility of mobile WiMAX subscribers on video on demand (VOD)
over WiMAX is analyzed by considering the scalable video coding (SVC) codes for video
streaming. This experiment has been carried out using OPNET modeller 14.5. To compare the
performance of Internet Protocol television (IPTV) over WiMAX, the packet delay variation,
packet end to end delay, delay and load matrices are used. The result shows that after certain
speed, the load increases, the delay again decreases and there is no change in the packet delay
variation and packet end to end delay.
1.2 Introduction to IPTV
When IPTV stands for Internet Protocol television and the IPTV main role is to deliver audio ,
video and any other data called Triple Play services over an IP network. i.e IPTV includes
services such as voice over IP(VOIP),video on demand and web access services that commonly
known as Triple Play services. The IP network for IPTV might be public IPTV network. Such as
internet or private IP network, such as LAN based network. IPTV has a numerous no. of features
such as to way capability of IPTV system which allows service provider to deliver a whole ,
interactive television application.( e.g., Interactive games, high speed internet browsing, and high
definition television)

Figure 1: IPTV technology overview as part of the Triple Play solution
1.2.1 Functional Components of the IPTV Architecture:
Content Sources:- It receives video content from producers and other sources, encodes them and
stores content to an acquisition database.
Services Nodes:- It receives video stream in various formats and then reformats and encapsulates
them for transmission with that appropriate Quality of Services(QOS) indicates to the
(WAN)wide area network for delivery to the customers. Its also communicate to the Customer
Premises Equipment (CPE) for the service management. And it may be centralized in metro
area(for e.g., at the Central Offices).

Figure 2: IPTV System Architecture
Wide Area Distribution Networks:- it provides the distribution services ,quality of services, and
other capabilities, such as multicast and other timely distribution of IPTV data streams from the
Service Nodes to the Customer Premises.
Customer Premises Equipment (CPE):-This device located at the customer premise provides
that the Broadband Network Termination(B-NT) that the functionality at the minimum .And its
may be include some other integrated functions like that set-top box ,routing gateway and some
other home networking capabilities.
IPTV Client:- The IPTV Client is the functional unit, basically it performs the functional
processing ,which includes setting up the connection and Quality of Services, decoding the video
streams such as set-top box.
1.3 ARCHITECTURE OF IPTV SYSTEM
In this architecture is described in Figure. Some major components of IPTV given in below:
a) Acquisition servers:- They add DRM metadata and encode video.

b) IP routers:- They provide fast reroute in that case if routing are failures. Basically they
route IP packets.
c) Distribution Servers:- They are provide QOS control and caching.
d) Set-Top –Boxes(STB):-This device on the customer side in that interfaces with the user
terminal (e.g., PC, TV, laptop and others)with a Digital Subscriber Line(DSL).
e) VoD creators and servers:- They retain the library of encoded VoD content to provide
VoD services
Figure 2: Basic IPTV System
A. IP-multicast IPTV Architecture
IP-multicast is a method of sending IP packets to a group of interested receivers. As shown in
Figure , AT&T U- Verse adopts the IP multicast architecture. When a TV program is encoded at
the super hub office (SHO), and then delivered via multicast through video hub offices(VHO),
intermediate offices(IOs), central offices(Cos), Digital Subscriber line access
multiplexers(DSLAM), & residential gateway(RG), to the TV Set-Top-Boxes(STB).

Figure 3: IP-multicast IPTV Architecture
B. Peer-to-Peer IPTV Architecture:
In this Architecture for a Peer-to-Peer(P2P) IPTV distribution, there is a group and a source of
peers as a torrent. Each peer receives many packets from the source and from other peers as
shown in Figure.

. Figure 4: Peer-to-Peer IPTV Architecture
1.4 Types of Triple Play Services
The term "Triple Play service" covers a large collection of voice, video and data services,
including: Video telephony / IPTV, which is multicast video (T.V channel) / Video on Demand
(VoD), which is unicast video / Voice over IP (VoIP) / Gaming / Internet access (HTTP, FTP
traffic) / E-mail / etc..
In the following subtopics a description for each service of the Triple Play services is given.
However, our attentions focus on delivering TV service. As a result, research analysis is focused
on IPTV; believe that effectively transmitted video is the greatest challenge. Later, a brief review
of the two other services, i.e. VoIP and data are continued. Finally, introductory section to
Quadruple Play services is discussed.
1.4.1 IPTV
Delivering high-quality video content is one of the latest and most demanding challenges faced
by the IP. With the deployment of these new IPTV services, the existing network infrastructures
will be pushed to their limits. To accommodate the needs of IPTV services, networks must be
able to scale to millions of customers, maximize bandwidth resources, and provide QoS and
security on an end-to-end basis. For these and other reasons, network intelligence is critical when
deploying video over broadband [29].
IPTV services may be classified into three main groups [31]:
 Live television, with or without interactivity related to the current TV show.
 Time-shifted television: catch-up TV (replays a TV show that was broadcast hours or days
ago), start-over TV (replays the current TV show from its beginning).
 Video on Demand (VoD): browse a catalog of videos, not related to TV programming.
IPTV is distinguished from Internet television by its on-going standardization process (e.g.,
ETSI) and preferential deployment scenarios in subscriber-based telecommunications networks
with high-speed access channels into end-user premises via set-top boxes or other customer-
premises equipment [32]. However, IPTV is a sensitive service to the packet loss and delays for
unreliable streamed data. IPTV has strict minimum speed requirements in order to ensure
delivering the right number of frames per second for moving pictures. This means, limited

connection speed and bandwidth available for a large IPTV customer can reduce the quality of
delivered service. Their impact is briefly summarized below:
 The latency inherent by the use of satellite Internet is often held up as a reason why
satellites cannot be successfully used for IPTV. However, in practice latency is not an
important factor for IPTV. An IPTV service does not require a real-time transmission, as
is the case with the telephony or videoconferencing services [33].
 Bandwidth requirements, for high-speed data transfer the needed bandwidth for the
viewer is increased. For example, at least 2 Mb/s is needed for web-based applications on
computer. Additionally, 64kbps is required for using landline telephone. In a minimal
usage, 13 Mb/s is requires to process in a household with an IPTV Triple Play service.
 Privacy implications due to limitations in bandwidth, an IPTV channel is delivered to
user one at a time, as opposed to the traditional multiplexed delivery. Changing a channel
requires a request from the head-end server to provide a different broadcast stream; much
like VoD (for VoD stream is delivered using Unicast, while for normal TV signal
multicast is used). This could enable the service provider to accurately track each and
every program watched with its duration for each viewer. Broadcasters and advertisers
could then understand their audience better, subsequently programming with accurate
data and targeted advertising [33].
 Service bundling for residential users, IPTV is often provided in conjunction with video
on demand and may be bundled with Internet services such as Internet access and VoIP
telecommunications services. Commercial bundling of IPTV, VoIP and Internet access is
sometimes referred in marketing as Triple Play service. When these three services are
offered with a cellular service, the combined service may be referred to as Quadruple
play.
 Regulation historically, broadcast television has been regulated differently than
telecommunications. As IPTV allows TV and VoD to be transmitted over IP networks,
new regulatory issues arise [34]. Professor Eli M. Noam highlights in his report [35], "TV
or Not TV: Three Screens, One Regulation?" some of the key challenges with sector
specific regulation that is becoming obsolete due to convergence in this field.

Chapter 2
Background Information and Literature Review
2.1 Introduction
This chapter introduces the Triple Play system and also explains how its network architectures
work. It also demonstrates how the system can delivers its services over different broadband
access connection with different necessities. Triple Play system of advantages and disadvantages
of the access broadband and the skills that are required to deliver these services are also
discussed in this chapter.
2.2 Next Generation Triple Play Network
The full package of this service includes: line rental and telephony with a combination of
Internet access, IPTV, VoD, entertainment applications and, eventually, cellular phones. In other
words : Multiple services, multiple devices, but one network, with different vendor and one bill.
It is a consequence of the important changes the industry is undergoing, such as technological
innovations, social changes and new regulations. Moreover, In next generation Triple Play
networks are capable of connecting wired and wireless subscribers, and its providing flexible and
fast service, provisioning high bandwidth with high QoS, and reduced overall service cost.
• The backhaul or the central transport connecting other networks or the Internet.
Core networks that handle high volume aggregated transmissions between the network and the
backhaul.
• Distributed networks that extend the line of sight (LoS) coverage area of the core network.
• Local networks that interface directly with the end users.

Figure: 2.1 Next Generation Triple Play Network
2.3 Triple Play Services Delivery Architectures
Its faces some new network challenges. Triple Play services are just not like conventional
Internet services. Conventional Internet is "best-effort" service. In that users are not online in the
same time, but aggregating traffic from multiple users increases network efficiency. On the other
hand, Triple Play includes real-time services, like video and voice. These kinds of services have
strict end-to-end delay, jitter and bandwidth requirements. If a real-time service confronts packet
delay or packet loss the connection is immediately dropped. From the above discussion it is
obvious that network delivery architectures are required for Triple Play services. However, with
the video service which includes IPTV, VoD and HDTV, things are not so apparent. Users may
now need up to 20Mbps to satisfy their needs. As a result, the Triple Play is postulates a new
kind of delivery infrastructure architecture as can be shown in Figure 2.2 [13].

Figure 2.2: The building blocks of the Triple Play services delivery infrastructure
It will be useful to get an idea of how technology providers try to address the challenges of
multiservice. Triple Play is considered to become a "killer application" as soon as its
requirements are satisfied and market matures enough to provide it in a low cost. Increasing
revenues lead many big companies to enter to the fields of IPTV. There are different
architectural approaches which declare to be able to approach together all the types of services in
a flexible and scalable manner [22].
2.3.1 Centralized Edge Design
In this type of architecture, the L2 metro Ethernet aggregates the traffic from multiples access
points before the IP edge network, as shown on Figure 2.3. Some of the characteristics of this
architecture are given below:

1) All types of traffic are backhauled to the Broadband Network Gateways (BNGs) and then to a
single P-router or PoP location, which is connected to the ISP backbone.
2) Subscriber execution functionality, multicast replication and IP QoS policies are executed in
the BNG deeper in the network.
3) IP multicast traffic for broadcast video is transmitted from the edge router over L2 multicast
Virtual Local Area Networks (VLANs) to all customer premises.
Figure 2.3: Centralized single edge overlay architecture
2.3.2 Distributed IP Edge Design
A distributed IP edge approach is being considered by many SPs as an alternative architecture to
satisfy the bandwidth requirements for future applications. As shown I Figure 2.4, the edge
network is comprised by both L2 and L3 routers. Video and HSI are backhauled over VLANs to
the edge routers, where services and access to the IP network are controlled. The scalability is

increased, since the amount of state information in the BNG is decreased (less subscribers are
terminated per BNG) and IP QoS is enforced closer to the last mile.
Figure 2.4: IP distributed single edge overlay architecture [22]
2.4 Types of Triple Play Services
The term "Triple Play service" covers a large collection of video, voice and data services,
including: Video telephony / IPTV, which is multicast video (T.V channel) / Video on Demand
(VoD), which is unicast video / Voice over IP (VoIP) / Gaming / Internet access (HTTP, FTP
traffic) / E-mail / etc. As a result, research analysis is focused on IPTV; believe that effectively
transmitted video is the greatest challenge. Later, a brief review of the two other services, i.e.
VoIP and data are continued.
2.4.1 IPTV
Its delivering high-quality video content is one of the latest and most demanding challenges
faced by the IP. To accommodate the needs of IPTV services, networks must be able to scale to

millions of customers, maximize bandwidth resources, and provide QoS and security on an end-
to-end basis. For these and other reasons, network intelligence is critical when deploying video
over broadband [29].
IPTV services may be classified into three main groups [31]:
 Time-shifted television: catch-up TV (replays a TV show that was broadcast hours or
days ago), start-over TV (replays the current TV show from its beginning.
 Live television: with or without interactivity related to the current TV show.
 Video on Demand (VoD): browse a catalog of videos, not related to TV programming.
 IPTV is distinguished from Internet television by its on-going standardization process
(e.g., ETSI) .However, IPTV is a sensitive service to the packet loss and delays for
unreliable streamed data. IPTV has limited connection speed and bandwidth available
for a large IPTV customer can reduce the quality of delivered service. Its briefly
summarized below:
 The latency inherent by the use of satellite Internet is often held up as a reason why
satellites cannot be successfully used for IPTV. An IPTV service does not require a real-
time transmission, as is the case with the telephony or videoconferencing services [33].
 Bandwidth requirements, for high-speed data transfer the needed bandwidth for the
viewer is increased.
 Privacy implications due to limitations in bandwidth, an IPTV channel is delivered to
user one at a time, as opposed to the traditional multiplexed delivery. Changing a channel
requires a request from the head-end server to provide a different broadcast stream; But
that is enable the service provider to accurately track each and every program watched
with its duration for each viewer. Broadcasters and advertisers could then understand
their audience better, subsequently programming with accurate data and targeted
advertising [33].
 Service bundling for residential users, IPTV is often provided in conjunction with video
on demand and may be bundled with Internet services such as Internet access and VoIP

telecommunications services. Commercial bundling of IPTV, VoIP and Internet access is
sometimes referred in marketing as Triple Play service. When these three services are
offered with a cellular service, the combined service may be referred to as Quadruple
play.
2.4.2 Voice over Internet Protocol
VoIP is defined as in this technology that allows users to make telephone calls over an IP data
network (Internet or Intranet) instead of traditional PSTN. Therefore; VoIP provides a solution
that merges both data and voice, which gains benefits include; cost savings, high quality and
value added services. The simplest encoder scheme is G.711 (64 kb/s). The acceptable packet
loss factor of G.711 is up to 0.928% [37]. So, the encoder scheme G.711 (64 kb/s) is used for
VoIP service in the simulation project.
Moreover, it is important to state that Ethernet and IP layers are critical to the deployment and
troubleshooting of VoIP service. They have significant impact on the overall QoE parameters,
including dropped calls and call quality. Items that affect the overall transport quality of VoIP
service are; IP packet delay, loss, jitter, and out of sequence (OoS) packets. These items are
briefly summarized below [29]:
 Packet delays can have varying effects on voice quality, so it is important to measure the
delay at the time of service installation to provide a benchmark for verification against
potential problems.
 Packet loss can occur for a variety of reasons inside a network. Periodic losses in excess
of 5 to 10 percent of all transmitted voice packets will degrade voice quality significantly.
 Packet jitter will make speech choppy . For high-quality voice, the average inter-arrival
packet time at the receiver should be nearly equal to the inter-packet gaps at the
transmitter and the standard deviation should be low.
 It is also important to note that lost packets and QoS packets are always measured on the
local link and can help in the segmentation of the overall problem.
 However, in the enterprise market, IP is opening up major new VoIP-based opportunities.
Forrester Research [38] predicts that by 2015, 95 percent of enterprise voice calls will be
VoIP-based. However, the fastest growing VoIP market is "hosted IP voice" or "IP
Centrex" which is expanded from about US$60 million in 2004 to more than US$7.6

billion in 2010, representing a CAGR of 282 percent. By that time, VoIP technologies are
expected to be handling 45 percent of the total voice telephony market [38].
2.4.3 Data Service
Data service refers to activities familiar to all of us like web browsing, e-mail, file downloading
electronic purchases, electronic games and other applications using HSI. The Internet has a
major and a successful cultural phenomenon in the past few years. The drivers to its success is
the convenience and benefits of applications like e-mail and web browsing, combined with low
cost and wide reach. However, today Internet needs something else to remain successful and
maintain its promise as a everywhere and universal means of accessing any type of information.
It must be a profitable business or, at least, economically self-sustainable. In order to be able to
properly bill for such services, they must be reliable, and feature high and persistent qualities.
Furthermore, a Triple Play solution with a low technological complexity is needed to support
single network infrastructures, which include traditional Internet application (e.g., web, e-mail,
and file sharing), telephony, and video. Current solutions for integrating traffic of heterogeneous
nature on the same network infrastructure are leverage on over provisioning to ensure
satisfactory quality. This leads to poor utilization of network resources, thus conflicting with the
objective of minimizing costs [23]. This thesis presents an optimal solution for supporting Triple
Play through IP networks, and specifically the Internet, as it offers:
 Quality Of Services(QOS) guarantees (deterministic delay and jitter, no loss) for (UDP-
based) CBR and VBR streaming applications.
 Low complexity, hence high scalability of routers.
 The service received by elastic, e.g., TCP-based applications.
2.4.4 Quadruple Play
A Quadruple play service is the little brother of the Triple Play service. It combines the Triple
Services (HSI, voice and video) with wireless services. In this thesis, when wireless services are
mentioned, rather than just having a cellular-like service, it is the ability to have wireless access

to all the aforementioned services. WiMAX is a wireless technology that has the ability to
provide broadband connections over long distances. Moreover, advances in both CDMA and
GSM standards, and utilizing 3G, 4G or UMTS allow the service operators to enter into
Quadruple Play and gain competitive advantage against other providers [13].
2.5 The Requirements of the Investigated Triple Play Services
There are many different services that are currently available in home, and even more when
looking towards the near future. These services can be audio/video entertainment, games, news,
education, personal communications, home control, financial management, telemedicine, asset
management and teleporting. These numerous (end-user) services each have their specific
requirements [39]. In the following subtopics, the types of requirement for Triple Play services
towards broadband networks are discussed.
2.5.1 Bandwidth Requirements
In the Next generation services of Triple Play (data/voice/video) require far more bandwidth than
the existing technologies like dial-up, ISDN, cable, and DSL that could offer. The gulf between
the access and metro networks is widening due to bandwidth difference. This problem is termed
as "access bottleneck". However there are more challenges for traffic/speed related requirements
to deliver Triple Play services efficiently with cost effectively, these are explained below [40]:
• Data rate and traffic direction: Symmetric/Asymmetric
The most important bandwidth requirement of services is data rate, this may differ in the up- or
downstream direction from narrowband (<128 kbit/s) to mid-broadband (128kbit/s – 10Mbit/s)
and super broadband (10Mbit/s-1Gbit/s) or even ultra-broadband (>1 Gbit/s). If a service uses
more / less in either the up- or downstream direction this connection is called asymmetric (e.g.
watching a video which uses more downstream capacity), other services use approximately the
same speed in both up as downstream (e.g. placing a telephone call over IP or video-
conferencing).

• Traffic Duplication: Point-Point, Point-Multipoint, Multipoint-Multipoint
Those Services can correspond in sequence purely to one user (Point-Point) like on demand
video (specifically requested by one person)S, other services can correspond in sequence to
multiple users (Point-Multipoint) like broadcast television over IP, or gaming (Multipoint-
Multipoint). Traffic pattern: constant, variable or burst services sometimes require a CBR
connection (like a 128 kbit/s MP3 audio stream) or generate traffic varying between a certain
range of bit rate(like certain VBR streaming video content).
2.5.2 Network Quality of Service Requirements
QoS is a concept in which promises to balance the conflict between different services. To
achieve this goal, it is necessary to priorities individual service and to allow a grouping in the so-
called class of service. Now a days, there are several mechanisms available for QoS skills; the
most common mechanisms include [19]: IEEE 802.1p/Q standards, DiffServ, MPLS and RSVP.
These mechanisms implement the tagging (marking) of certain types of traffic (e.g. voice, video,
Internet-traffic) and use it to serve, for e.g. voice traffic with a higher priority than the video and
Internet traffic [40].
In relation to the Triple Play services, these parameters are: packet loss, network delay, jitters,
throughput, FTP downloads response time, HTTP page response time, and buffer overflow
percentage. These parameters are explained below.
i. End to End Delay:
In that time duration in seconds from the instant when a packet arrives at the source node until it
is received at the destination node. This metric can be calculated as follows [41]:
Where: Q: This is the number of network elements (like switches, routers and firewalls) among
sender and receiver.
dproc: This is the processing delay at a given network element.
dqueue: is the queuing delay at given network element.

dtrans: is the transmission time of a packet on a given link and (d prop ) is the propagation
delay across a given network link
ii. Jitter:
It is defined to be the variation of delay between consecutive packets. If two consecutive packets
leave the source node with time stamps t1 and t2 and arrive the destination at time t3 and t4 after
reassembly and play back, then the jitter is represented by equation (2.2) [42]:
Jitter = (t4 - t3) - (t2 - t1) (2.2)
Where :(t4 - t3) is the expected packet reception time and (t2 - t1) is the actual packet reception
time. Negative jitter means that the packets where received in different time range i.e (t4 - t3) <
(t2 - t1).
iii. Packet Loss Ratio:
PLR is the number of corrupted, dropped, or excessively delayed packets in relation to the total
number of packets expected at the client station. PLR can be calculated as follows [43]:
𝑷𝑳𝑹=𝒍𝒐𝒔𝒕_ 𝒑𝒂𝒄𝒌𝒆𝒕𝒔𝒍𝒐𝒔𝒕_ 𝒑𝒂𝒄𝒌𝒆𝒕𝒔+ 𝒓𝒆𝒄𝒆𝒊𝒗𝒆𝒅_𝒑𝒂𝒄𝒌𝒆𝒕𝒔 ……………………….(2.3)
Another variation of this metric is the media loss rate (MLR) which track packet loss over time:
iv. Throughput:
it is defined as the traffic load that the media stream will impress upon the network. It can be
measured in bytes/sec (Bps) or bits/sec (bps). For CBR content, the throughput is constant
and it can be calculated as [41]:
Consequently, VBR traffic loads are typically quoted in throughput ranges.
v. FTP downloads response time:

It is the time elapsed between sending an FTP request to the FTP server and receiving the
complete response packet. The download response time is an basically indicator of the user-
perceived latency of the downloaded FTP file. In this experiment all time specific
information regarding download has been collected.
vi. HTTP page response time:
HTTP page response is the time elapsed between sending the HTTP request to the HTTP server
and receiving the complete response of the entire web page with all contained embedded objects.
And the main page object and the embedded objects are downloaded using a single TCP
connection with HTTP 1.1. The HTTP paging response time is an indicator of the user-perceived
latency of the web page retrieval [47].
2.6 Broadband Infrastructures and Their Support for Triple Play
In this thesis the latest developments in the leading broadband access technologies are reviewed
and the ability of those technologies to meet the future requirements of the broadband consumer
is assessed. Figure 2.6 shows several types of broadband access technologies that are used to
support Triple Play services bandwidth [49]. In general broadband solutions can be classified
into two groups: fixed line technologies and wireless technologies. The fixed line solutions
communicate via a physical network that provides a direct "wired" connection between customer
and service supplier. (for e.g. POTS, where the customer is actually connected to the operator
by a pair of twisted copper cables. Wireless solutions use in radio or microwave frequencies to
provide a connection between the customer and the network operator‘s; mobile phone
connectivity is a prime example. These two types are explained below:-
2.6.1 Fixed Line Technologies
Fixed line broadband technologies rely on a direct substantial connection from services supplier
to subscriber‘s residence or business. In Cable modem systems use existing hybrid fiber-coax
cable TV networks. xDSL systems use the twisted copper pair traditionally used for voice
services by the POTS. Broadband power line

broadband technology uses the power lines feeding into the subscriber‘s home to carry
broadband signals. In general, all three aforementioned technologies strive to avoid any upgrades
to the existing network due to the inherent implications of capital expenditure. By contrast,
FTTH or FTTC networks require the installation of a new (fiber) link from the local exchange
(CO) directly to or closer to the subscriber. Consequently, although fiber is known to offer the
ultimate in broadband bandwidth capability, the installation costs of such networks have, up until
recently, been prohibitively high. The fixed line technologies evaluated of wire include: Hybrid
Fiber Coax (Cable TV & Cable Modems), xDSL, BPL and Fiber to the Home/Curb [48].
However, in this research, the term ‗wire‘ applies only few technologies which are designed for
access to Triple Play networks requirement, such as xDSL and FTTx that are described in the
following subtopics.
A. Digital Subscriber Line (xDSL)
DSL is a very high-speed connection that uses the same wires of a regular telephone line. DSL
allows making regular phone calls and surfing the Internet simultaneously. It is a point-to-point

medium that manages to compress more information through a standard copper wire. DSL is a
distance sensitive technology, user's distance from the closest CO should be less than 18000 ft,
see Figure 2.7. The most important advantage of Triple Play services through DSL, the
bandwidth of copper wires is capable of carrying more than the phone conversations. DSL
exploits this property. In particular, Asymmetric DSL (ADSL) divides up the available
frequencies in a line such that upstream and downstream data as well as voice can travel together
in the same copper wire. Delivering video using DSL over twisted pair is a booming technology
called IPTV. A device called STB is used by the subscriber to control and order video services,
like video on demand. Internet data can be delivered using ATM or DOCSIS. Voice can be
distributed in two ways; using the traditional POTS interface or using VoIP [13].
Figure 2.7: Network architectures for various forms of xDSL
B. Fiber to the x (FTTx)
Fiber to the x (FTTx) is a generic term for any broadband network architecture that use optical
fiber to replace all or part of the usual metal local loop used for the last mile telecommunications.
The generic term was initially a generalization for several configurations of fiber deployment

(FTTN, FTTC, FTTB, FTTH...), all starting by FTT but differentiated by the last letter, which is
substituted by x in the generalization [50].
However, fiber which virtually has unlimited bandwidth, it is more economical than copper in
terms of maintenance, upgrade, and manpower requirements. Optical networks have been seen
by many as a promising solution for the last mile problem. Figure 2.8 shows a fiber optical
network that consists of one optical line terminal (OLT) located at the provider central office
(head end) and multiple optical network units (ONUs or stations) at the customer side [51].
Figure 2.8: Triple Play over FTTx
2.6.2 Wireless Technologies
All communications based on a transmission media other than copper wire, optical fiber etc..
That definition includes communications based on satellite microwave links, terrestrial radio
links and free space optical communications [27]. The technologies evaluated of wireless
include: Microwave links, MMDS, LMDS, FSO, Wi-Fi, WiMAX, Satellite, and 3G. This kind
of wireless communication is currently dominated by some technologies defined by the IEEE.

The most important one is the IEEE 802.11 family (Wi-Fi) and the IEEE 802.16 family
(WiMAX), which are described in the following subtopics.
2.7 Wireless WAN-WiMAX
The WiMAX technology, based on IEEE 802.16–2004 standard [52], defines a fixed broadband
wireless metropolitan area network. Mobile WiMAX, based on IEEE 802.16e-2005 [53], adds
functions and features to the original standard to support mobility. The most current IEEE
802.16–2009 standard [54] is a revision of IEEE 802.16–2004. It also consolidates material from
IEEE 802.16e–2005 and other previous 802.16 standards. Mobile WiMAX has a target
transmission range of up to 31 miles and a target data rate exceeding 100Mbps [55]. Compared
to mobile WiMAX, 3G data services provide a relatively low bandwidth and high price while
WiFi suffers from limited transmission ranges and security issues.

Figure 2.8 shows the system model of a WiMAX deployment for Triple Play services over
all IP backbone network
WiMAX is part of the progression from voice-only wireless relations systems to ones that
provide additional services like web browsing, streaming media, instant messaging, gaming, and
other content. The 802.16 standard offers adjustable data rate to and from each user while
maintaining the required QoS. However, WiMAX is a serious competitor for delivery of Triple
Play services. With advanced antenna techniques, it offers data rates up to 70 Mbps and ranges
up to 50 km, ensures secure delivery of content, and supports mobile users at vehicular speeds of
up to approximately 100 km/hr. Figure 2.10 shows the system model of a WiMAX deployment
for Triple Play services over all IP backbone network.
2.8 Wireless LAN –WiFi (802.11g)
WiFi like a trademark of the WiFi alliance that becomes used in wireless technology today.
Technically, WiFi is referred as the 802.11 communications standard for WLAN. IEEE 802.11b,
IEEE 802.11a, IEEE 802.11g and IEEE 802.11n are the most common used protocol in today‘s
environment [57]. The IEEE 802.11g is used in the simulation project. The g extension of the
802.11 protocol has a peak data rate of 54 Mbps, on averages around 22 Mbps. The data rate is
adaptable as the standard is backward compatible with extension a and b. Despite being wireless,
WiFi is designed to have low mobility. Current mobile devices such as iPhone 3G supports Wifi
extensions of a/b/g [58]. Today, nearly pervasive WiFi delivers high-speed WLAN connectivity
to millions of offices, homes, and public locations, such as hotels, cafés, and airports.
Worldwide, more than 223 million homes have WiFi connections, and there are over 127 million
WiFi hotspots. The integration of WiFi into notebooks has accelerated the adoption of WiFi to
the point where it is nearly a default feature in notebooks. Over 97% of laptops are integrated
with WiFi, and the number of handhelds and Consumer Electronics (CE) devices provided with
WiFi capabilities are increased [59]. One of the advantages of WiFi is that it operates over
unlicensed bands of radio spectrum. This fact eases deployment of Wi-Fi networks. However,
interference from external devices is more likely to happen in unlicensed frequencies than in the
licensed spectrum. Specifically, some WiFi devices operate at the same frequencies as
microwave ovens, bluetooth devices and cordless phones [28].

2.9 Integration WiFi and WiMAX
WiFi and WiMAX are both IEEE wireless standards And that are designed for IP based
applications. WiFi is optimized for a very high speed WLAN though WiMAX is optimized for a
high speed. Currently few users have WiMAX enabled devices. They either need to buy a
compatible device or promote their current electronic device (i.e. Desktop or Laptop) to enjoy
WiMAX capability [59]. In the simulation of the current project of users could be purchase a
WiMAX-WiFi router and then send data to their computers via WiFi; Table 2.3 summarizes the
implementation and deployment of WiMAX and WiFi.

Figure 2.11: The area covered by each of the WiMAX and Wi-Fi
2.9.1 Multicast and Triple Play Network
Today, with the junction of packet-based and circuit-switched technologies, it is possible to
deliver video, voice and data using a single IP network. Many of the new multimedia
applications, like IPTV or VoIP, are multicast. However, multicast service model of the e-mail
application is not adequate for these applications. And it does not have the scalability and
simplicity needed to deliver real-time voice or video to hundreds or maybe thousands of
receivers, see Figure 2.12.

Figure 2.12: Service model for IP data applications:
(a) The WWW service model – unicast and client/server-based;
(b) The e-mail service model – multicast application; multicast delivery is solved at the
application layer [28]
To enable an IP network to support Triple Play, it is necessary to implement multicasting in the
network layer of the protocol stack. Multicast delivery must be supported by the intermediate
routers in addition to the terminal equipment, see Figure 2.13. The aim of these protocols is to
choose the correct outgoing interfaces for incoming multicast packets. Then a challenge is raised
on how to manage multicast groups. This requires implementing a mechanism to add or remove
recipients dynamically. This is done by using multicast addresses and the IGMP [28].

Figure 2.13: Service model for the IPTV application. Multicast delivery is solved at the IP layer
and not at the application layer of the protocol stack.
2.10 IPTV SET-TOP BOX (STB)
Set-Top box is an IPTV device that is used to receive and decode digital television broadcast and
to turn that signal into content which is then displayed on a TV screen or any some other display
devices.
A. Professional Set-Top box:
It‘s referred to as Integrated Receiver Decoders (IRD) is professional method of the
broadcasting audio and video contents to the display devices. They are technically superior. (For
e.g. Tornado M10 Set-Top box)

Figure 2.14: Professional Set-Top box
B. Hybrid IPTV Set-Top box
It is designed basically to deliver video contents. This allows viewers to view broadcast
television and internet video on their display along with IPTV services such as VOD. A new
consumer television experience can created internet browser.
Figure 2.15 : Hybrid IPTV Set-Top box
C. Wireless IPTV Set-Top box:
It is to deliver IPTV contents through a wireless network connection. Its connect to displays
using high definition multimedia interface (HDMI) and S-video interface that can take input
from the internet and deliver it to the display device. It can deliver high quality audio and video
those contents to the displays with high speed network connection.

Figure 2.16: Wireless IPTV Set-Top box
2.11 PROTOCOLS USED FOR IPTV
A. RTP: Real –time Transport Protocol provides many packet format for delivering video and
audio contents through IP network. RTP is commonly used in some streaming media services
like including telephony & video teleconference applications services. This is used for streaming
channels and it‘s controlled by the Real Streaming Time Protocol (RSTP).
B. RTSP: Its stand for Real Time Streaming Protocol or controlling network protocol It‘s used to
establish and control media sessions from the server to the user. The media servers issue pauses
and play commands to easily control that those transmissions of media files from the server. In
RTSP Streaming of data is one –directional. Data streams can be sent from the server to the user
RTSP is used to control all unicast and multicast streams.
C. PIM: Protocol-Independent Multicast (PIM) is a collection of multicast routing protocols that
can provide data distribution through IP network such as the internet, LAN (Local Area
Network), and WAN (Wide Area Network). PIM is independent and it uses routing information
provided by various routing protocols such as BGP (Border Gateway Protocol) In IPTV, PIM is
commonly used to route IPTV multicast streams between networks.
D. IGMP: IGMP stands for The Internet Group management protocol. Its used to manage
membership in IP multicast groups. This protocol is basically used in online streaming video and
gaming. IGMP is main part of the multicast specification over IP network. It is used to change
from one TV channel to another TV channel.

2.12 Simulation and Simulation Tools:
Simulation is three phase process which includes the designing of a model for theoretical or
actual system followed by the process of executing this model on a digital computer and finally
the analysis of the output from the execution. Simulation is learning by doing which means that
to understand/ learn about any system, first we have to design a model for it and execute it. To
understand a simulation model first we need to know about system and model. System is an
entity which exists and operates in time while model is the representation of that system at
particular point in time and space. This simplified representation of system used for it better
understating. In wireless sensor network there are many simulation tools are used for simulation
purpose describe as below:
2.12.1 NCTUns:
NCTUns (National Chiao Tung University Network Simulation) is a simulator that combines
both traffic and network simulator in to a single module that built using C++ programming
language and support high level of GUI support. It is a highly extensible and robust network
simulator in no need to be concerned about the code complexity.
Features:
 It can simulate many standards such as IEEE 802.11a, IEEE 802.11b, IEEE
802.11e,IEEE 802.16d, IEEE802.11g and IEEE 802.11.
 It supports large number of nodes.
 It includes directional, bidirectional and omni directional communation.

Figure 2.17: Graphical user interface of NCTUns.
Figure 2.18 NCTU ns simulator

2.12.2 NS-2(Network Simulator):
Network Simulator (Version 2), called as the NS-2, is simply an event driven, open source
portable simulation tool that used in studying the dynamic nature of communication networks.
Basic Architecture of NS-2: In the Figure2.19 represent the basic architecture of NS-2. It
provides ns executable command to its users to take input argument .Users is feeding the name
of a TCL simulation script as an input argument of NS-2 executable command ns.
Fig. 2.19 Architecture of NS
NS-2 consists of two key languages one is the C++ and second is the Object-oriented Tool
Command Language (OTCL). In NS-2 C++ defines the internal mechanism (backend) of the
simulation objects, and OTCL defines external simulation environment (i.e., a frontend)for
assembling and configuring the objects. After simulation, NS-2 gives simulation outputs either in
form of text-based or animation-based.
2.12.3 OPNET (Optimized network engineering tool):
OPNET is a commercial network simulator environment used for simulations of both wired and
wireless networks. It allows the user to design and study the network communication devices,
protocols and also simulate the performance of routing protocol. This simulator follows the
object oriented modelling approach. It supports many wireless technologies and standards such
as, IEEE 802.11, IEEE 802.15.1, IEEE 802.16, IEEE 802.20 and satellite networks.

OPNET Architecture: OPNET provides a comprehensive environment to model and do
performance evaluation of networks and distributed systems. The OPNET package includes
numbers of tools. Those tools fall into three categories corresponding to the three phases of
modelling and simulation projects: Specification, Simulation and Data Collection, and Analysis.
These phases should necessarily be in sequence and form a simulation cycle as in Figure 2.6.
OPNET uses the concept of modeling domains to represent its modeling environments, and
graphical editors for editing the Network, Node and Process models. Specifically, there are
several editors in OPNET: project editor, node editor, process editor, external system editor, link
model editor, packet format editor, Interface Control Information editor, and probability density
function editor.
Re-Specification
Data Collection
And
Simulation
Analysis
Initial Specification
Figure 2.20 : Simulation Cycle in OPNET
Network Domain is used to define the network topology of a communication network. The
communicating entities are called nodes. Network domain is created by using the Project editor
tool of the OPNET modeller.

Node Domain describes nodes‘ internal architecture in terms of functional elements in the node
and data flow between them.
Process defines the behaviour of processes, including protocols, algorithms and application,
specified using infinite state machines and an extended high-level language.
External System specifies the interfaces to the models provided by other simulators running
concurrently with an OPNET simulation (a co-simulation).
OPNET Modeler Wireless Support
The Wireless module in OPNET provides a flexible and scalable wireless network modeling
environment, including a broad range of powerful technologies. The Wireless module integrates
OPNET‘s full protocol stack modeling capability, including MAC, routing, higher layer
protocols, and applications, with the ability to model all aspects of wireless transmissions,
including:
- Radio Frequency propagation (path loss with terrain diffraction, fading, and atmospheric and
foliage attenuation)
- Interference
- Transmitter/receiver characteristics
- Node mobility, including handover
- Interconnection with wire-line transport networks
The wireless module has rich protocol model suites to optimize the R&D processes, and more
effectively design technologies such as MANET, 802.11, 3G/4G, Ultra Wide Band, 802.16,
Bluetooth, and Transformational Communications systems. Wireless network planners,
architects, and operations professionals can analyze end-to-end behaviour, tune network
performance, and evaluate growth scenarios for revenue-generating network services.
Why use OPNET
A good modeling tool should closely reflect the true behavior of a network or computer system.
It should support a wide range of network protocols and applications. It must be easy to use and

master, especially for beginners. On the other hand, a good modeling tool should provide
comprehensive technical support and maintenance assistance. In summary, we believe that a
good modeling tool should have the following properties:
Versatile: able to simulate various network protocols/applications under a wide range of
operating conditions [26].
Robust: provide users with powerful modeling, simulation and data analysis facilities.
User Friendly: easy to use and master [26].
Traceable: easy to identify modeling problems and simulation faults [26]. OPNET is hailed by
network professionals because it has all these properties. OPNET is a software package that has
been designed with an extensive set of features. It can be tailored to suit almost every need of
network protocol designers, network service providers, as well as network equipment
manufacturers. OPNET supports most network protocols in existence, both wire line and
wireless. It can be used to model and analyse a complex system by performing discrete event
simulations [26].
OPNET Capabilities
OPNET has a lot of capabilities. Some of these capabilities are the following:
Hierarchical Network Models: Manage complex network topologies with unlimited sub-
network nesting [27].
Object Oriented Modeling: Nodes and protocols are model as classes with inheritance and
specialization [27].
Clear and Simple Modeling Paradigm: Model the behavior of individual objects at the process
level and interconnect them to from devices at the ―Node Level‖ ; interconnect devices using
links to form networks at the ―Network Level‖ [27].
Finite State Machine Modeling: Finite state Machine modeling of protocols and other
processes. Simulate arbitrary behavior with C/C++ logic in FSM‘s states and transitions. You
control the level of detail [27].
Comprehensive Support for Protocol Programming: 400 library functions support and
simplifying writing protocol models [27].
Wireless, Point-to-Point and Multipoint Links: Link behavior is open and programmable [27].

Geographical and Dynamic Mobility Modelling: It is for mobile and satellite systems [27].
Total Openness: API‘s from program-driven construction or inspection of all models and result
files. Easily integrate existing code libraries into your simulations [27].
Integrated Analysis Tools: Display simulation results. Easily plot and analyze, time series,
histograms, probability functions, parametric curves, and confidence intervals. Export to
spreadsheets [27].
Animation: Animation of model behavior, either during or after simulation [27].
Integrated Debugger: Integrated debugger to quickly validate simulation behavior or track
down problems [27].
Import Data from Some Popular Tools: Such as HP Open View and Network Associated
Sniffer [27].
Comprehensive Library of Detailed Protocol Models: Including ATM, Frame Relay, TCP/IP,
RIP, OSPF, BGP4, IGRP, Ethernet, FDDI, Token Ring, and many more. Provided as FSM‘s
with source code [27]
Run Time Environment (Modeler XE): Deliver proprietary protocol and device models to end-
users, working and running simulations at the network level only [27].
Solaris, Windows NT, and HP-UX: Supported (Transparent cross platform usage)
Flexible Licenses: Floating license (concurrent use based), and loan able license
c) GloMoSim (Global Mobile Information System Simulator); GloMoSim (Global Mobile
Information System Simulator) is a scalable simulation environment especially designed of
MANET and its applications. It is open source, portable and includes a large set of routing
protocols and several physical layer implementations. It was retired in 2000 but it is still possible
to download for educational purposes only. On the other side, Scalable Network Technologies
introduced the commercial version of GloMoSim (Global Mobile Information System Simulator)
named as QualNet (Quality Networking) simulator. The main merits of QualNet simulator
(Quality Networking), is that it is open source portable, highly scalable and extremely powerful
simulator. One of the main merits of QualNet, is that it is run on both Windows and Unix/Linux
platforms.

d.) QualNet (Quality Networking): QualNet is a highly scalable, fastest simulator for large
heterogeneous network It supports the wired and wireless network protocol. QualNet execute
any type of scenario 5 to 10 times faster than other simulators. It is highly scalable and simulate
up to 50,000 mobile nodes. And this simulator is designed as a powerful Graphical User
Interface (GUI) for custom code development. one of the main advantage of QualNet is that it
supports Windows and Linux.
Figure 2.20 QualNet simulator
h) SWANS: SWANS (Scalable Wireless Ad hoc Network Simulator) was proposed to be a best
alternative to the NS-2 simulator for simulating the wireless and ad hoc networks. On the basis
of comparative study of simulators like SWANS, GloMoSim, and NS-2,it is found that SWANS
simulator is the most scalable and more memory efficient. SWANS takes Java file as a input. It

is a scalable wireless network simulator built top on the JIST platform and good capabilities like
NS-2 and GloMoSim.
1.4 Lecture Survey
P. Kampanakis et al., in 2006 [13] studied the basics of Triple Play and Quadruple Play
services, and a few common and popular applications for the Triple Play architecture. They also
included on their study the pricing model adapted to support Triple Play services followed by a
simulation study on OPNET 11.5 to analyze and understand QoS parameters, i.e. end-to-end
delay and jitter experienced by voice, video and data traffic as they traverse routers configured
with various scheduling policies.
C. A. Papagianni et al., [14] examined in particular the performance of common packet
scheduling techniques used in DiffServ Triple Play architecture. The performance of PQ, WFQ
and WFQ-LLQ schemes was assessed, in order to find the most appropriate solution for the
underlying network.
N. Zotos et al., in 2010 [15] presented an experimental network infrastructure providing E2E
QoS, using a combination of MPLS and DiffServ technologies in the core network and WiMAX
technology as the wireless access medium for high priority services (VoIP, High Quality Video
Streaming) transmission.
R. K. Kalle et al., in 2007 [16] discussed network system architecture in which the 802.16e
operating in PMP mode was used for last mile access and the authors investigated some of the
Triple Play applications which include e-Education and Infotainment. A simulation study of the
delivery of Triple Play service over 802.16e was reported and the performance evaluation for a
typical emerging market scenario indicated that 6-8 simultaneous video sessions can be
supported for over an 802.16e network operating in PMP mode of operation.
L. Shi et al., in 2008 [17] studied how to achieve Network Utility Maximization (NUM) in NGN
running Triple Play services. By investigating the characteristics of most of its traffic classes,
they explicitly presented their utilities as the function of allocated bandwidth. They also further
formulated the NUM objective as a nonlinear programming problem with both inequality and
equality constraints. Several useful results are presented on the new features of the NUM-based
scheduling.

K. Ozdemir et al., in 2009 [18] analyzed user capacity of mobile WiMAX systems for each of
these three services (Triple Play) and considered various link characteristics, interference
scenarios, modulations, QoS classes, and QoE requirements. He also analyzed the impact of
header compression and suppression techniques and their effect on capacity.
F. Wan et al., [19] considered for home networks a heterogeneous wired and wireless network
architecture to support Internet Protocol TV (IPTV), voice, and data, the so-called Triple Play
services. To satisfy the quality of service (QoS) requirements for different traffic classes, class-
based queuing (CBQ) is deployed at home gateways and routers. Simulation results over wired
and multi-hop wireless paths are given which validate the analysis. The results presented provide
important guidelines for the planning of future home networks for Triple Play services. They
also provide important insights into how to efficiently support heterogeneous traffic with
stringent QoS requirements over wireless and wired networks.
T. Uhl, in 2009 [20] focused on theoretical and practical demonstration of the current methods to
evaluate the QoS during Triple Play services transmission.
Also in 2010 Y. Zhang et al., [21] offered a good solution for Triple Play integrated automation
network. It integrates the surface network, underground network and wireless communication
network in the same unified platform, so as to get maximum information sharing function and
achieving stronger spot monitor, in addition to control the capabilities of the network. The
application on Xinglong Coal Mine turned out to be a big success of the Triple Play integrated
automation network in practice.
I. Papapanagiotou and M. Devetsikiotis, in 2010 [22] the choice of an appropriate architectural
approach and sizing model for the aggregation network is studied through cost optimization
models, which encompass aspects of non-stop delivery, service flexibility, policy management
and cost allocation for Triple Play services.
M. Baldi and P. d. Torino, in 2011 [23] presented a low complexity solution for proving Triple
Play services that enables integrating different applications, so that each received required
service is received, possibly in a guaranteed fashion, while ensuring high resource utilization
efficiency.

F. Khan et al.,[24] presented an enhancement to our proposed hybrid (WDM/TDM) architecture
to support Triple Play services and employs EPON within last mile to address the access
bottleneck issue. Also, he used digital transmission for data and voice services with analog video
broadcast service is realized using RF overlay model.
Y. Hao-wei et al., in 2011 [25] briefly introduced the position of Triple Play system in the whole
government emergency management system, and discussed the development aim of Triple Play
system, the flow of police alert receipt and dispatching, the construction content and the
functions to be achieved.
O. Schilke et al., in 2012 [26] focused on the bundle of broadband Triple Play. Technological
advances have given telecommunications service providers a way to offer a full array of Internet
services, which they often combine into one bundled package. Triple Play has emerged as a term
in the business press that describes such offerings; it denotes the bundling of three broadband
services: Internet access, telephone, and television.
A. E. Garcia et al., in 2012 [27] proposed a cost model which considers QoS parameters for
Triple Play services. This model is based on the ―Total Element based Long Run Incremental
Cost‖ (TELRIC), which is applied to the wholesale access and interconnection paradigm. Three
traffic engineering methods were considered and studied for network dimensioning. Hereby the
aim was to guarantee QoS of different services: complete traffic segregation under virtual
tunnels, complete traffic integration by over engineering, and partial traffic integration using a
priority queuing scheme. The proposed method enables the development of a specific cost
scheme based on a complete scenario taking into consideration different types of users. The
variety of used IP applications supposes direct implications over different levels of
interconnection, mainly at the low-level Metro access and the high-level edge node.
Recently, there have been some works based on performance studies of video streaming over
WiMAX networks. Many of research workers have explored WiMAX in the context of real-time
and stored video applications.
For example, Pandey et al. [5] Developed a model to dimension the network for IPTV service
providers that offer VoD services to their customers in heterogeneous environments. The
proposed modelling and simulation technique allows us to determine the optimum deployment

conditions for a given number of potential IPTV users while satisfying predefined QoE
measures. On other hand, Shehu et al. [10] Discussed issues regarding challenges for delivering
IPTV over WiMAX. These issues include the challenges of QoS requirements. Also, they
describe the transmission of IPTV services over WiMAX technology, and the impact of different
parameters in WiMAX network when deploying this service. An intelligent controller has been
designed based on fuzzy logic to analyze QoS requirements for delivering IPTV over WiMAX in
[11] is used to analyze three parameters: jitter, losses and delays that affect the QoS for
delivering IPTV services. The aim is to define a maximum value of link utilization among links
of the network.
Hrudey et al. [12] used OPNET Simulation to design, characterize, and compare the performance
of video streaming to WiMAX and ADSL. The simulation results indicate that ADSL exhibits
behaviour approached the ideal values for the performance metrics while WiMAX demonstrates
promising behaviour within the bounds of the defined metrics. The work in [13] is extending the
work in [12] to include generation and integration of a streaming audio component, also
enhances the protocol stack to include the real time protocol (RTP) layer. Network topology is
redesigned to incorporate WiMAX mobility. Also, include characterization of WiMAX media
access control (MAC) and physical (PHY) layer. Simulation scenarios are used to observe the
impact on the four performance metrics. Gill et al. [14] used OPNET Simulation to compare the
performance metrics between ADSL and WiMAX by varying the attributes of network objects
such as traffic load and by customizing the physical characteristics to vary BLER, packet loss,
delay, jitter, and throughput. Simulation results demonstrate considerable packet loss. ADSL
exhibits considerably better performance than the WiMAX client stations.
Hamodi et al. [8] Used OPNET Simulation to design, characterize, and deployment the
performance of video streaming over WiMAX under a different video codec (SVC, and AVC).
The simulation results indicate that SVC video codec is an appropriate video codec for video
streaming over WiMAX. The work in [9] is extending the work in [38] to investigate the
performance of video streaming over WiMAX under two different terrain environments, namely
Free Space, Outdoor to Indoor and Pedestrian. The simulation results indicate that, free space
path loss model is a basic path loss model with all other parameters related to terrain and
building density set as constant.

However, many of recent works explore the performance studies of WiMAX under different
modulation and coding schemes. For example, Telagarapu et al. [15] analyzed the physical layer
of WiMAX with different modulation techniques like BPSK, QPSK, QAM and comparison of
QPSK modulation with and without Forward Error Correction methods. Islam et al. [4]
Evaluated WiMAX system under different combinations of digital modulation (BPSK, QPSK,
64-QAM and 16-QAM) and different communication channels AWGN and fading channels
(Rayleigh and Rician), and the WiMAX system incorporates Reed-Solomon (RS) encoder with
Convolution encoder with ½ and 2⁄3 rated codes in FEC channel coding.
Bhunia et al. [16] presented an in-depth performance evaluation of mobile WiMAX is carried out
using adaptive modulation and coding under the real-like simulation environment of OPNET.
They have evaluated the performance parameters of mobile WiMAX with respect to different
modulation and coding schemes. Their performance has been evaluated in terms of average
throughput, average data-dropped, the MOS value of voice application and the BW usage in
terms of UL data burst usage when deployed VoIP on WiMAX Networks. It has been observed
that using lower order modulation and coding schemes, the system provides better performance
in terms of throughput, data dropped and MOS at the cost of higher BW usage.

Chapter 3
PROJECT ESSENTIALS
3.1 Front End Tool Used (OPNET 14.5):
In our dissertation work we are using the Optimized Network Engineering Tool (OPNET v16.0)
software for simulating selected routing protocols. OPNET is a network simulator. It provides
multiple solutions for managing networks and applications e.g. network operation, planning,
research and development (R&D), network engineering and performance management.
OPNET 16.0 is designed for modelling communication devices, technologies, and protocols and
to simulate the performance of these technologies. It allows the user to design and study the
network communication devices, protocols, individual applications and also simulate the
performance of routing protocol. It supports many wireless technologies and standards such as,
IEEE 2002.11, IEEE 2002.15.1, IEEE 2002.16, IEEE 2002.20 and satellite networks. OPNET IT
Guru Academic Edition is available for free to the academic research and teaching community.

Fig. 3.1 OPNET Modeler 14.5 Opening Screen
It provides a virtual network environment that models the behaviour of an entire network
including its switches, routers, servers, protocols and individual application. The main merits of
OPNET are that it is much easier to use, very user friendly graphical user interface and provide
good quality of documentation.
OPNET Modeler constitutes a network simulation program based on C and C++, which offers a
convenient GUI in order to facilitate users to conduct network experiments. OPNET Modeler
includes model libraries that represent various network hardware devices from many vendors and
various communication protocols. Thus, the OPNET Modeler users are able to simulate large
network environments with network devices and routing protocols of will, without the need of
pursuing real equipment, saving this way cost. The specific program also gives the capability to
add or modify existing models, and bases its simulations on the Discrete Event Simulation
system which uses defined processes to model network events. Additionally, traffic patterns can
be simulated by the use of network layer traffic flows, by well-defined applications or by
transport layer application demands.

Fig: 3.2 Flowchart of OPNET
The sequence of the needed acts needed for a network simulation, includes the design and
configuration of the network topology, the selection of the desired measured metrics, the
simulation run and the analysis of the calculated statistics. Eventually, OPNET Modeler is
considered a reliable program when it comes to network evaluation, usually met on computer
networking publications and also used by industry. These advantages of the program led the
author to select it as the tool to facilitate the intended experiments.
3.2 Back End Tool Used (Visual Studio 2010):
Visual Studio is a complete set of development tools for building ASP.NET Web applications,
XML Web Services, desktop applications, and mobile applications. Visual Basic, Visual C#, and
Visual C++ all use the same integrated development environment (IDE), which enables tool
sharing and eases the creation of mixed-language solutions. In addition, these languages use the
functionality of the .NET Framework, which provides access to key technologies that simplify
the development of ASP Web applications and XML Web Services. Microsoft Visual Studio is
an integrated development environment (IDE) from Microsoft. It is used to develop computer
programs for Microsoft Windows, as well as web sites, web applications and web services.

Visual Studio uses Microsoft software development platforms such as Windows API, Windows
Forms, Windows Presentation Foundation, Windows Store and Microsoft Silver light. It can
produce both native code and managed code.
Visual Studio includes a code editor supporting IntelliSense as well as code refactoring. The
integrated debugger works both as a source-level debugger and a machine-level debugger. Other
built-in tools include a forms designer for building GUI applications, web
designer, class designer, and database schema designer. It accepts plug-ins that enhance the
functionality at almost every level—including adding support for source-control systems
(like Subversion) and adding new toolsets like editors and visual designers for domain-specific
languages or toolsets for other aspects of the software development lifecycle(like the Team
Foundation Server client: Team Explorer).
3.3 Modeling Methodology of OPNET
The environment of OPNET is shown in the following Figure 3.3.
Figure.3.3: OPNET Environment.
This section of the project contains the analysis of the buttons, which are located in the
environment of OPNET. In addition describes the basic modeling categories of OPNET,
which are the following:
 Network Editor

 Node Editor and
 Process Editor.
The toolbar, which is located on the top of the above figure 3.3, can be analyzed as follows.
Figure.3.4: The Main Toolbar of OPNET Environment.
3.3.1 OPNET Editors
The OPNET environment incorporates tools for all phases of a simulation study, including model
design, simulation, data collection and data analysis. Several OPNET editors represent these
phases. The very basic OPNET editors are the following:
 Network Editor
 Node Editor and
 Process Editor
A. Network Editor
The Network Editor graphically represents the topology of a communication network. Networks
consist of node and link objects, configurable via dial boxes. Drag and drop nodes and links from
the editor‘s object palettes to build the network, or use import and rapid object deployment
features. Use objects from OPNET‘s extensive Model Library, or customize palettes to contain
your own node and link models. The Network Editor provides geographical context, with

physical characteristics, reflected appropriately in simulation of both wire line and
mobile/wireless networks. Use the protocol menu to quickly configure protocols and activate
protocol specific views [27].
Figure. 3.5: Example of the Network Editor.
B. Node Editor
The Node Editor captures the architecture of a network device or system by depicting the flow of
data between functional elements, called ―modules‖. Each module can generate, send, and
receive packets from other modules to perform its function within a node. Modules typically
represent applications, protocol layers, algorithms and physical resources such as buffers, ports,
and buses. Modules are assigned process models (developed in the Process Editor) to achieve
any required behavior [27].
C. Node Editor Environment
The environment of a Node Editor is shown in the following Figure 3.6.

Figure. 3.6 : The Node Environment.

Figure. 3.7: The Node Editor Toolbar
Process Editor
The Process Editor is used to define the behavior for the programmable modules. In this way, it
is possible to control the underlying functionality of the node models created in the node editor.
These models are used to simulate software subsystems, such as a communication protocol, and
also to model hardware subsystems, such as the CPU of a MT. A process is an instance of a
process model and operates within on module. Initially, a process model contains only one
process, this is referred to as ―the root process‖. However, a process can create additional ―child
processes‖ dynamically. These can in turn create additional processes themselves. This is well
suited to model certain protocols. Processes respond to interrupts. These interrupts indicate that
events of interest have occurred like the arrival of a message or the expiration of a timer. An
interrupted process takes actions in response to interrupts and then blocks, waiting for a new
interrupt. It may also invoke another process and its execution is suspended until the invoked
process blocks [28]. Finite state machines, named State Transition Diagrams (STDs) in OPNET,
represent the process models. An example of a STD is shown in Figure 3.8. These STDs consist
of icons representing states and lines that represent the transition between the states. The

operations performed in each state or for a transition are expressed in Proto-C (embedded C/C++
code blocks and library of Kernel Procedures providing commonly needed functionality for
modeling communications and information processing) [28].
The main features of a STD are:
Initial State: is the first state the process model enters upon invocation. This state is easily
identified by a large arrow on its left-hand side (i.e. the INIT state in Figure.22). It usually
performs functions such as the initialization of variables [28].
The Transition Arc: describes the possible movement of a process from one state to another
and the conditions under which such a change in state may take place. A transition with no
attached condition is depicted with a directed solid line, while one with an attached condition is
depicted using a directed dashed line [28].
The Transition Conditions: Transition conditions are specified as Booleans. If no possible
transition or more than one possible transition exists then the simulation halts. A‖ default
transition‖ ensures that a situation where a simulation halts due to the fact that no transition
evaluated to TRUE never occurs [28].
The Transition Executive: The transition executive is carried out when a transition is taken. As
a transition is made from one state to another, actions can be executed when leaving the first
state (exit executives) and upon entering the next state (enter executives) [28].
Unforced States: Unforced States represent true states of the system. A process blocks after the
enter executives of an unforced state have been executed. The exit executives are executed when
a new interrupt causes the process to be reinvaded. The unforced states represent the possible
stable states of process. These states have a red colour in the process editor [28].
Forced States: Forced States do not allow a process to wait or block. When a transition is
followed that leads to a forced state, the enter executives are executed and another transition is
followed. This chain continues until finally an unforced is entered. Forced states are useful when
attempting to simplify a complex task by sub-dividing the task into multiple forced states. The
forced states are easily discriminated from the unforced states by its green colour [28].
Variables: OPNET processes not only include the facility to define variables for use during
process invocations, ―temporary variables‖, but also maintain a set of ―state variables‖. While the
values of the temporary variables are lost between process invocations, the values of state

variables are maintained. State variables are typically used to model counters, statistical
information and retransmission timer values while temporary variables are simply used to
complete tasks such as packet handling [28].
State Attributes: State attributes define a set of parameters, which can be used to tailor process
instance beha viour. This allows generic specification of a process, which can be used in many
different scenarios [28].
Process Editor Environment
The environment of the Process Editor is shown in the following figure 3.9
Figure 3.9: Process Editor Environment.

Figure. 3.10: The Toolbar of Process Editor Environment.
Finally, the sequence of the three basic editors of OPNET can be represented by the following
figure 3.11

3.3 Calculated Performance Metrics
Using OPNET Modeler the performance of OSPF and IS-IS on a dual-stack network will be
evaluated by calculating and comparing network performance metrics. Performance metrics are
values that can be measured and give clues about the speed, the scalability, the adaptability to
changes and the overall capacity and capability of the network. OPNET Modeler allows the
measurement of several metrics and the production of statistics that can lead to conclusions
regarding the performance of the network when using one or the other routing protocol. OPNET
Modeler itself divides the available metrics to Global, Node and Link metrics, where the first
concern the overall network function and the other two are measured on specific network nodes
or links. For the needs of the experiments, the overall performance metrics that were measured
were divided in two additional sub-categories, the pure network metrics and the end-to-end
Quality of Service metrics. Both categories and the specific metrics that they include are deeper
explained below.

Pure Network Metrics: As pure network metrics are described in this thesis the network
performance metrics that are measured on the initial network topology when no traffic is
implemented or when the traversing traffic does not have an effect on them. These metrics‘
values remain the same regardless of the application and traffic rates that are running on the
network. Metrics of this type can give an overall view of the performance of the configured
routing protocols configured on the network, and depend on the specifically selected topology.
During the experiment conduction of this project, the following pure network metrics were
calculated:
Convergence Duration: In routing protocol terms, convergence is the state that routers
configured with a dynamic routing protocol reach, where they all have the same topological
knowledge of the network or AS that they run on. Different routing protocols follow different
procedures until they converge. In any case, the convergence state represents the phase where the
appropriate information have been exchanged between participating routers, routing tables have
been built, and thus, all routers are in stable state and can begin routing. The time needed for a
network configured with a routing protocol in order to reach convergence is called convergence
duration. Convergence duration can lead to conclusions about the protocol‘s speed, because the
sooner convergence is achieved, the sooner data can be forwarded. Furthermore, convergence
duration can indicate how fast a network can return to a functioning stable state after a link or
router failure has occurred.
Convergence Activity: Based on the above mentioned information, convergence activity can also
be measured in order to show the timestamps during the experiment, where convergence
procedure routing information is being exchanged between the network‘s nodes. The importance
of this metric is similar to this of the convergence duration metric.
Routing Table Size: Another metric that can be calculated on the initial pure network topology,
is the routers‘ routing table size. This metrics literally shows the number of route entries that a
routing table holds. This specific value can also give a hint about the routing protocols‘ speed.
As routers search their routing tables sequentially when seeking an available route to forward a

packet, a routing protocol that introduces less entries in the routing table, will self-evidently
achieve a higher routing speed. Again, the routing table size is depending on the size of the
topology and not on the type or amount of sent traffic.
End-to-End QoS Metrics: Apart from the pure network metrics, network performance evaluation
demands the measurement of metrics that have to do with the actual performance of the network
when transferring data. In order to achieve a realistic network simulation, traffic flows have to be
emulated. The end-to-end QoS metrics concern the reaction, speed and efficiency of the routing
protocol configured network, when specific type of traffic is forwarded from a source to a
destination. During the experiment conduction of this project, the following end-to-end QoS
metrics were calculated:
Throughput: Throughput is one of the more important and common network performance
metrics. Measured in bits/sec or in packets/sec, it represents the amount of bits or packets that are
successfully transferred over a link. High throughput values indicate efficient network function
as packets sent reach their destination without being dropped and retransmitted for various
reasons. Low throughput on the other hand shows lower speed and more utilization of the
network capacity.
End-to-End Delay: Another metric exactly corresponding to network speed, is end-to-end delay.
This metric is calculated for every client-server pair where a traffic flow is running between
them. End-to-end delay is measured in seconds and represents the actual time passed from the
creation of the packet until its receipt at the destination. Obviously, lower end-to-end delay
values indicate a better network performance. The lower the end-to-end delay, the faster the
receipt of a packet.
End-to-End Delay Variation: This metric, also known as ―jitter‖, refers to the dispersion of
delay between different IP packets of a traffic flow. An average value for this metric can be
calculated for every client-server peer group. Obviously, high jitter is not desirable and it has a
negative effect especially on UDP traffic and real-time applications.

CPU Utilization: CPU utilization actually represents CPU overhead. A routing protocol can be
evaluated based on this metric, according to the burden that it applies to the participating routers.
The CPU utilization metric in OPNET Modeler shows the percentage of the CPU part that deals
with IP packet forwarding. High CPU utilization values can lead to latencies as well as in
dangers such as a router overload and failure.
Chapter 4
Simulation Results and Analysis
4.1 EXPERIMENTAL SETUP
In this experiment the effect of mobility of mobile WiMAX subscriber on VOD over WiMAX is
analyzed by using OPNET simulator. OPNET simulator 14.5 (OPNET official website,
http://www.opnet.com) was used to analyze the performance of WiMAX. We used OPNET
modeller, as OPNET modeler provides a comprehensive development environment supporting

the modeling of communication network and distributed systems. OPNET modeler provides
better environment for simulation, data collection and data analysis (OPNET official website,
http://www.opnet.com). The basic model of this experiment is shown in Figure 1. In this
experiment 4 scenarios with name WiMAX6, WiMAX7, WiMAX8 and WiMAX9 are taken. In
these scenarios subscriber is moving with different mobility. In WiMAX 6 subscriber have 60
km/h mobility, in WiMAX 7 has 70 km/h, in WiMAX 8 is 80 km/h and in WiMAX 9 nodes have
90 km/h mobility. In each scenario two hexagonal cells are taken. Each cell has a radius of 2
Km, in each cell there is one base station and 10 mobile nodes. These nodes are circularly
placed. The BS connected to the IP backbone via a DS3 WAN link. The node 0 is connected to
backbone through ppp_sonet_oct1 link. The node 2 is also connected to video server through 100
base T Ethernet link.
Table 4.1: Simulation Parameters
No. of Wimax Station WiMAX6, WiMAX7, WiMAX8 and WiMAX9
Cell Radius 2 km
No. of Subscriber Stations per BS 10
No. of Mobile nodes in the network 10
Speed of the mobile nodes 60, 70, 80, 90 km/hr
Simulation time 600 sec
Base Station Model wimax_bs_ethernet4_slip4_router
Subscriber Station Model wimax_ss_wkstn
ASN Gateway Model ethernet4_slip8_gtwy
IP Backbone Model ip32_cloud
Voice Server Model ppp_server
Link Model (BS-Backbone) PPP_DS3
Link Model (ASN - Backbone) PPP_SONET_OC12
Physical Layer Model OFDMA 20Mhz

MAC Protocol IEEE 802.16e
Multipath Channel Model ITU Vehicular A
Traffic Type of Service
Scheduling Type ertPS, nrtPS
Application FTP
Voice Codec G 711
FTP Load High
4.1.1 Video application configuration
The frame inter-arrival time and the frame size parameters of the video conferencing application
are shown in Figures 2 and 3. In this experiment the frame interval time was 30fps is taken for
incoming and none for outgoing. In this experiment the SVC video codes are used which are
shown in Table 1. Figure 4.1 shows the profile configuration of VOD over WiMAX. In this
experiment operation mode was simultaneous and start time was 1 s. The subscribers and video
server is configured with this profile.
Figure 4.1 Profile configuration
4.1.2 WiMAX configuration

In WiMAX model RTPS scheduling class was created for the downlink and uplink to support the
real time video streaming. The scheduling was configured with 5 Mbps, maximum sustainable
traffic rate, and 1 Mbps minimum sustainable traffic rate as shown in Figure 4.2. The mobile
WiMAX subscribers and base station is configured with match property of type of service
(TOS). In each WiMAX subscribers the 64 QAM modulation and coding scheme is used for
downlink and 16 QAM is used for uplink as shown in Figures 4.3 and 4.4. In this research, the
matrix we measured is packet delay variation, packet end to end delay, delay and load.
Figure 4.2 Classes configuration.

Figure 4.3 Base station parameters

Figure 4.4 WiMAX subscriber station parameters
4.2 RESULTS
Here the result of VOD over WiMAX is calculated by changing mobility of nodes. Figures 4.5 to
represent the result of packet delay variation, packet end to end delay, delay and load.
4.2.1 Packet delay variations

It is the variance among end to end delays for video packets. End to end delay for a video packet
is measured from the time it is created to the time it is received. Figure 4.5 shows the packet
delay variation at different mobility.
Figure 4.5 Packet delay variation.
Figure 4.5 shows that as the speed of the subscriber is increasing packet delay variation is
decreasing. For 60 km/h the packet delay variation is 0.024 for 70 and 80 km/h it is 0.023 and for
90 km/h it is 0.017.
4.2.2 Packet end to end delay
It is the time taken to send a video application packet to a destination node application layer.
This statistic records data from all the nodes in the network. Figure 4.6 shows that as speed is

increasing, the packet end to end delay is decreasing. From Figure 4.7, for 60 km/h, the highest
value of packet end to end delay is 0.33 and for 70 and 80 km/h it is 0.30 and for 90 km/h it is
0.26.
Figure 4.6 Packet end to end delay.
4.2.3 Delay:
It represents the end-to-end delay of all the packets received by the WiMAX MAC‘s of all
WiMAX nodes in the network and forwarded to the higher layer. Figure 4.7 shows the result of
Delay. Fig 4.8 shows that with increase in speed, the Delay decreases. Fig shows that at the end

of simulation 60 km/hr have high delay which is 0.11, the delay for 70 km/hr and 80 km/hr is
0.09 and 90 km/hr has delay of 0.08.
Figure 4.7 Delay
4.2.4 Load
This represents the total load submitted to WiMAX layers by all higher layers in all WiMAX
nodes of the network. Figure 4.8 represents the result of Load. This figure shows that with
increase in speed, the load also increases. From Figure 4.9, it has been observed that, 90 km/h
having the highest load which is 6534877, 80 and 70 km/h having load of 6500000 and 60 km/h
having load of 63916336.

Figure 4.8 Load.

CONCLUSION AND FUTURE SCOPE
In this research, analysis of the performance of IPTV (VOD) over WiMAX by varying mobility
of mobile WiMAX Subscriber in terms of packet delay variation, packet end to end delay, delay
and load is carried out. In this experiment the placement of nodes are circular within hexagonal
cell of radius 2 km. Here the speed of each node is varying from 60 to 90 km/h. For video
streaming SVC codes are used. Simulation is carried out for three minutes. The results show that
with increase in the speed, packet delay variation, packet end to end delay and delay are
decreasing but the load is increasing, no doubt this increment of load is very little. The result also
shows that for 70 and 80 km/h, the delay variation, packet end to end delay, delay and the load
has equal values. In future, one can analyze the IPTV (VOD) over integrated WiMAX and
MANET by varying different parameters like network area, number of mobile WiMAX
subscribers and power

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