Fred Biko Otieno Mboya

Internet Protocol Television (IPTV) Services

Helsinki Metropolia University of Applied Sciences

Bachelor of Engineering
Information Technology

Thesis
20 April 2016
 Abstract

Author(s) Fred Biko Otieno Mboya

Title Internet Protocol Television (IPTV) Services
Number of Pages 45 pages
Date 20 April 2016

Degree Bachelor of Engineering

Degree Programme Information Technology

Specialisation option Telecommunication and Networking

Instructor
Dr. Tero Nurminen, Principal Lecturer

This thesis mainly deals with IPTV (Internet Protocol TV) technology and how it changes
the business of television; its development and architectural design; its applications and
progress into the future. The goal of the project is to enhance professional networking on
both live TV and radio platform, know how the IPTV functions and how it differs from inter-
net TV, how it is formatted, transported and delivered to the end users. Equally important,
how providers charge for it and make a living.

The study itself was carried out by retrieving information from different sources such as the
library, the Internet, through self-observation, and discussions with the chief supervisor
and instructor. Different aspects of IPTV are discussed in different phases of the thesis.
First, the study introduces IPTV technology, its background and means of transmission.
Then, the study entails the architectural design of IPTV, multimedia methods and applica-
tions, compression techniques and finally its purpose and role to the growing technology
services.

The purpose of the project was to gain adequate practical experience, skills, techniques,
and theory by applying previous classroom knowledge to actual principal-like situations in
a strategic, organized and supervised environment. The end result came from the fact that
in the near future it is likely that IPTV can replace traditional TV technology since it deliv-
ers a good supplement business model for service providers, offers better quality of ser-
vice to consumers, and play a significant role on the fast growing and evolving interactive
TV applications such as VOD.

Keywords IPTV, VOD, DSL, QoS, IP, RTP, STB

List of Abbreviations and Acronyms
HDTV High Definition Television
IPTV Internet Protocol Television
VOD Video-On-Demand
TV Television
QoS Quality of Service
IP Internet Protocol
EPG Electronic Program Guide
RF Radio Frequency
DSL Digital Subscriber Line
ADSL Asymmetric digital subscriber line
HDSL High data rate Digital Subscriber Line
SDSL Symmetric Digital Subscriber Line
VDSL Very high bit rate Digital Subscriber Line
IETF Internet Engineering Task Force
IntServ Integrated Services
DiffServ Differentiated Services
SLS Service Level Specification
PHB Per-Hop Behavior
STB Set-Top Box
NVOD Near Video-On-Demand
QVOD Quasi Video-On-Demand
TVOD True Video-On-Demand
USB Universal Serial Bus
RIP The Routing Information Protocol
RTCP Real-time Transport Control Protocol
RTSP The Real Time Streaming Protocol
PIM Protocol Independent Multicast
SM Sparse mode
DR Designated Router
RP Rendezvous Point
DM Dense mode
RPF Reverse Path Forwarding
ICMP Internet Control Message Protocol
IGMP The Internet Group Management Protocol
DSAD Digitally Sampled Analog Data
AAC Advanced Audio Coding
Table of Contents
1 Introduction 1

2 Theoretical Background 2

2.1 IPTV Vs Internet TV 2

2.2 Mobile IPTV 3
2.2.1 Cellular Network Approach 4
2.2.2 Dedicated Broadcast Network 4
2.2.3 Mobile IPTV Benefits 5
2.3 Signal Transmission 5
2.3.1 Analog Signalling 6
2.3.2 Digital Signalling 8

3 Architecture of IPTV Systems 9

3.1 Digital Subscriber Line (DSL) 9

3.1.1 ADSL (Asymmetric Digital Subscriber Line) 11
3.1.2 HDSL (High Data Rate Digital Subscriber Line) 12
3.1.3 SDSL (Symmetric Digital Subscriber Line) 12
3.1.4 VDSL (Very High Bit Rate Digital Subscriber Line) 13
3.2 Quality of Services 13
3.2.1 Quality of Service Conceptual Model 13
3.2.2 Best Effort 14
3.2.3 Integrated Services 14
3.2.4 Differentiated Services (DiffServ) Model 15
3.3 Video on Demand 16
3.4 Triple Play Services 17
3.5 IPTV Set-top Box 18

4 Multimedia over IP 19

4.1 Video Conferencing 19

4.1.1 Purpose and Benefits of Video Conferencing 19
4.1.2 Components of Video Conferencing System 20
4.1.3 Types of Video conferencing 21
4.2 IPTV Networks 22
4.2.1 Unicast 22
4.2.2 Multicast 24
4.2.3 Broadcast Transmission 26
4.3 IPTV Protocols Network 28
 4.3.1 Real-Time Protocol (RTP) 28
4.3.2 Real-Time Transport Control Protocol (RTCP) 30
4.3.3 Real Time Streaming Protocol (RTSP) 32
4.3.4 Protocol Independent Multicast (PIM) 34
4.3.5 Internet Group Management Protocol (IGMP) 38
4.3.6 Internet Control Message Protocol (ICMP) 40

5 Compression in IPTV System 41

5.1 Compression Algorithm 41

5.1.1 Lossless Compression 41
5.1.2 Lossy Compression 42
5.2 Compression Techniques 42
5.2.1 Spatial Compression (Intra-frame) 42
5.2.2 Temporal Compression (Inter-frame) 43
5.3 Audio Compression and Video Compression 44

6 Results and Conclusions 45

 1

1 Introduction

Over the past decade, the only way to watch television was through over-the-air broad-
cast and cable signals. The emerging of satellite, digital cable, and High Definition Tel-
evision (HDTV) services have made it possible for telecommunication providers to dis-
cover a new technology in the television broadcast system. The innovation of digitiza-
tion of television technology around the globe has facilitated access to multiple ser-
vices, with better quality of service on all devices at all point of time.

Internet Protocol Television (IPTV) has provided the means to securely deliver high
quality triple play services to the end usersover a private or managed network. IPTV
functions just like a standard pay TV (Television) service and one of its key benefits is
to offer IP (Internet Protocol)based services in one integrated package, for example
receiving and displaying live or pre-recorded audio and video, as well as covering live
TV or Video on Demand (VOD).

IPTV systems are a significant aspect in the telecommunication field of technology as

they enhance professional networking on both live TV and radio platform. The author's
interest to have a better understanding of how to build a video on demand, know how
the television signals are transmitted, how they are formatted, transported and deliv-
ered to customers was a key drive to choosing this topic. Equally the importance to
know how providers charge for it and make a living.

This thesis is based on a project the author worked on during an internship period at
Streamafrik. The study itself was carried out successfully by retrieving information from
different sources such as the library, the Internet, through self-observation, and discus-
sions with the chief supervisor.
 2

2 Theoretical Background

The aim of the project was to explore the development of internet protocol television
and its different phases, as well as the transmission distribution mechanism that allows
for immediate interactivity and multimedia experience.

2.1 IPTV Vs Internet TV

IPTV is often mistaken for internet TV since they both use IP technology for video de-
livery. This section discusses the key differences between this two IP technology ser-
vices as shown in Table 1. IPTV services are delivered via private and managed net-
work using the internet protocol suite whilst Internet TV services are distributed over
open, public or global internet [1, 21-25]. This enables IPTV delivery to allow for higher
quality of delivery with secure delivery of content to the end users. Internet TV video
delivery, by contrast, can be subjected to longer waiting times due to lower bandwidth,
high traffic or poor connection quality.

Table 1 illustrates some of the key difference between IPTV and Internet TV. Modified
from [1, 26].
 3

In Table 1 above it is clearly arguable to note that both IPTV and internet TV play a
significant role in delivering video across a network platform. They both rely on IP tech-
nology for delivery, their approaches in delivery differs in the way in which the signal
travels and how the content is delivered over the internet. Internet TV model is open to
any rights holder and anyone can create an endpoint and publish on a global basis
offering a direct communication between the provider and the consumers.

2.2 Mobile IPTV

Mobile IPTV is a wireless mobile transmission platform that enables users to receive
multimedia content such as audio, graphics, video, and text over a wireless IP network
to a mobile medium with support for mobility, security, quality of service (QoS), quality
of experience, and reliability functions [2].

Mobile IPTV supports the following capabilities and features:

 It supports browsing of IPTV content information by using the Electronic Pro-
gram Guide (EPG).
 It has mobility capabilities to end users.
 Mobile IPTV allows for streaming of high image quality TV on mobile device
platform.
 It supports multiple languages and enhances faster interactivity and optimiza-
tion.
 End user device of mobile IPTV services can be a Smartphone or a tablet using
iOS (Originally iPhone OS) or Android.

Figure 1 shows a representation of mobile IPTV Architecture [33, 24]

 4

The Next Generation Networks (NGN) enable unrestricted access for users to networks
with a wide range of services offered by different service providers as shown in Figure
1. In this scenario both the sender and receiver are assumed to using a mobile device
[33]. This mobility capability enables communication between the sender (service pro-
vider) and the receiver (at the mobile terminal) over a wireless interface.

There are two main approaches used to deliver mobile TV, that is, across a cellular
network and across a dedicated broadcast network.

2.2.1 Cellular Network Approach

In cellular network approach, any broadcasting distribution network can be adopted as

it may incur the loss of quality of the mobile operator’s services since it competes with
data and voice services for bandwidth. Cellular network use a secure infrastructure that
would essentially reduce stationing to users of mobile TV services.

2.2.2 Dedicated Broadcast Network

Dedicated broadcast network on the contrary is a traditional digital broadcast network

that combines broadcasting service and internet to deliver IP based broadband, data
services, and optimize the provision of mobile TV via IP networking. The system deliv-
ery mechanism can be terrestrially based, satellite based, or a combination of both[3].
 5

2.2.3 Mobile IPTV Benefits

There are many advantages that mobile IPTV offers, such as:
 Mobile IPTV provides variety of real-time streaming data such as VOD and vid-
eo services much better on mobile phones.
 It allows for mobility of services based on Multimedia Subsystems (IMSP) and
wireless characteristics to IPTV.
 It provides digital television experience via interactivity.
 Mobile IPTV allows channel switching and casting.
 It provides Information access and entertainment for the users.
 It allows for content synchronization and offers an opportunity for watching TV
everywhere and anytime.

It is easy to see that mobile IPTV provides variety of new interaction level between In-
ternet, voice and video. Its wireless capability help to speed deployments and reduce
costs in the best possible way to reach out to users. The users only need a normal col-
ored screen phone with fair display resolution and mobile data connectivity to use this
technology service.

2.3 Signal Transmission

A signal is an electric or electromagnetic pattern of one or two independent variable

and representation of data to convey information. Signaling is a mechanism in which
data is transmitted across different medium ways such as twisted-pair, coaxial cable,
fiber-optic cable, radio frequency (RF) waves, satellite, and cellular telephony [4].

Signaling is carried out with the help of signals that indicate to the connected end de-
vice what data is requested across a media platforms. It plays a significant role in re-
ceiving information such as video, audio or encoded data.

Types of signal transmissions

There are two types of signal transmission used to transmit information in form of audio
or video usually through electric signals, that is, analog and digital signals. A well-
known example of analog vs. digital is that of clocks as illustrated below (Figure 2).
 6

Figure 2 Analog vs. Digital. Modified from [5]

Dubbing analog signals can often be skewed by just a few frames or by service sec-
onds. Every dub might be different due to generational loss. For example, in Figure 2,
digital clock specifically indicates time as 7.00, but analog signal time appears to be
either 7.00 or closer to 7.01 [5].

2.3.1 Analog Signalling

An analog signal is a continuous signal that contains time varying amplitude, voltage,
current, and frequency. Information is converted into an analog signal of varying ampli-
tude of high and low physical property (such as voltage or current) over time. For ex-
amples sound, voice, temperature vary continuously in frequency and amplitude.

In Figure 3, the sine wave's amplitude value can be seen to be either positive or nega-
tive between higher and lower points of the wave respectively, while the frequency
(time) value is amplified in the sine wave's physical length from left to right. Each time
the signal is amplified, the noise is also amplified.
 7

Varying Amplitude
0 frequency(time)

Figure 3 Sine wave of analog signal

Signals can be periodic or non-periodic analyzed in the frequency (time). Sine wave
and square wave are the common representations of analog signals. Square wave is
distinctive from digital signal by a negative minimum value.

Advantages of analog signaling

 Analog signaling suffers less attenuation than digital signal over long distances
[6].
 Analog signaling defines infinite amount of signal resolution. Analog devices are
equipped to handle the infinite values between 1 and 0 [6].
 It is a simpler implementation for easy processing and reproducibility.
 Analog signaling has a much higher density which can be multiplexed to in-
crease bandwidth at the same time make good use of the bandwidth [34, 8].
 Analog is better for higher frequency applications, where low cost and computa-
tion portable are required in real time.

Most sounds such as music and speech are analog signals. The main advantage of
analog signal is the potential for an infinite amount of signal resolution. Compared to
digital signals, analog signals are of higher density.
 8

Disadvantages of analog signaling

 Analog systems are less immune to noise, that is random unwanted variation,
over long distances. The noise becomes dominant creating disturbance and
distortion [34, 8].
 Analog systems are more likely to get affected by generation loss.

Even though analog signals is used in many systems today, its uses are declining with
the introduction of the more reliable digital signal.

2.3.2 Digital Signalling

A digital signal can be defined as a series of pulses consisting of discrete (discontinu-

ous) values of binary format(zero's and one's), where each bit is representative of two
distinct amplitudes. Digital signals can be uniform, square, or discrete in shape. Figure
4 shows an example of digital signal format.

Figure 4 an example of digital signal format

Each pulse represents a signal element and monitored periodically by network. Binary
data are transmitted by the presence or absence of signal elements. '1' represents the
presence of transition and '0'represents the absence of transition [6].
 9

Advantages of digital signal processing

 In digital signaling, the quality of the signal is maintained due its higher interfer-
ence immunity to external background noise. Noise does not accumulate on a
digital signal as it does on an analog signal during transmission.
 Digital signaling is compatible with integrated digital data and telephone signal-
ing which can be implemented with a relatively low equipment cost.
 Digital signaling offers various transmission options over long distances due to
its linear and nonlinear capabilities.

Disadvantages of digital signaling

 Digital signals like radios can be costly and not easily portable.
 Digital signals are intolerant to RF noises, where sampling can be complex re-
sulting in signal error [34, 11].

It is often recommended to convert analog signals to digital signals for more effective
signal processing. Video and audio transmissions are often transferred or recorded
using analog signals.

3 Architecture of IPTV Systems

3.1 Digital Subscriber Line (DSL)

Digital Subscriber Line is a broadband connection technology that enables telecommu-

nication network providers to offer high speed internet connection over existing 2-wire
or 4-wire standard copper telephone lines, thus eliminating the need for costly infra-
structure upgrades. DSL modem's ability to transmit data simultaneously in uplink and
downlink communication allows telecommunication providers to use their existing net-
works to provide high speed connectivity for on-demand streaming audio and video
entertainment to their subscriber bases. DSL connections are point-to-point dedicated
circuits that are cost effective, fast and reliable.
 10

In DSL the computer connects to the phone line which then connects to the DSL mo-
dem that has filters in place for different frequencies of voice and data. In other words,
the data connection is on the same line as the phone while travelling at different fre-
quencies. The data and the voice goes back and forth through the internet at the same
time. This way, DSL makes it is possible for the user to experience high speed internet
connection even when talking on the phone.

DSL modems establish a connection from one end of a copper wire by utilizing more of
the bandwidth on the analog line, thus allowing for greater bandwidth to the other end
of that copper wire and connecting digitally on both the uplink and downlink connection.
DSL modems can enable downlink connection speeds greater than 6Mbps and uplink
speeds up to 1MHz of bandwidth in both directions, thereby preventing the signals from
interfering with each other [7,2]. Figure 5 illustrates a connection setup of the DSL net-
work.

Figure 5 connection setup of the DSL network. Copied from [16, 5]

DSL modem's digital signal is not limited to 4 kHz of voice frequencies, making it much
faster than 56K analog modems in bandwidth capacity. The bandwidth rate available
are more consistent to the end user that are within 18000 feet [7,2]. Longer distances
must operate at lower bit rates to allow more subscribers to be served from a single
central office at a lower price.
 11

Variants/subtype of DSL
There are many different DSL service types’ options for broadband and IPTV, such as
ADSL, HDSL, SDSL, and VDSL referred collectively as XDSL. These variants of DSL
technology provide different data communication capabilities to different users. The
following section briefly states how each variant functions.

3.1.1 ADSL (Asymmetric Digital Subscriber Line)

ADSL is a subtype of DSL technology that uses a single pair and transmits higher data
rate downstream than upstream, implying that the download speed is greater than the
upload speed [17, 273].Upstream in this scenario refers to the transmission of data
from the subscriber back to the network or central office, and downstream refers to the
transmission of data in the direction towards the subscriber. An example of this type of
application is VOD. Figure 6 shows a connection setup of the ADSL network.

Figure 6 connection setup of the ADSL network. Copied from [16, 11]

Installation maintenance is a major problem for ADSL network operators. In order to

make installation easy, a version of ADSL known as ADSL Lite was developed by the
 12

International Telecommunication Union (ITU). ADSL Lite does not require a filter in the
customer premises, it can reach speeds of up to 1.5Mbps downstream and an up-
stream rate of 640Kbps [17, 275], enough to provide internet surfing access, remote
LAN access, multimedia access, software downloads, video-on-demand and home
shopping. ADSL is ideally suited to home and small office users who are downloading
more rather than uploading contents.

ADSL is the commonest form of DSL and by farther the most stable and affordable way
to access broadband internet. It uses existing telephone lines but splits it into two
channels, one for voice and one for data. This way the user can use the phone while
accessing the web at the same time.

3.1.2 HDSL (High Data Rate Digital Subscriber Line)

HDSL is a particular type of SDSL, symmetrically delivering 1.544 Mb/s in both down-
stream and upstream directions over two sets of copper twisted pair lines of up to
12000 feet, which is the same rate as T1 digital line type connection. It is possible to
extend the distance by using repeaters along the line to the customer.

HDSL is a better way of provisioning and transmitting T1/E1 over copper wires, using
less bandwidth and requires no repeaters up to the standard range [9]. It is heavily
used in cellular telephone build outs.

3.1.3 SDSL (Symmetric Digital Subscriber Line)

SDSL is similar to HDSL since it transmits same data rates (1.544 Mb/s) for the up-
stream and downstream channels simultaneously in both directions across a single
telephone line. SDSL connections typically allow transmission of up to 6 Mbps in both
directions, but usually require a 4-wire connection. This limits SDSL's reach of approx-
imately 3km. SDSL service is more expensive than ADSL, therefore it is ideally suited
to individual subscriber premises for connecting LAN's over short distances and video
conferencing.
 13

3.1.4 VDSL (Very High Bit Rate Digital Subscriber Line)

VDSL provides the highest data rates of the DSL technologies, being able to deliver
data at a transmission speed of up to 52 Mbps across a single copper cable but only
over short distances. VDSL is limited to distances of up to 2 km [17, 276]. This type of
DSL technology is particularly useful for supplying high data rate services for hotels,
university campuses and business parks that are closer to the telephone company’s
central office.

VDSL can also be used to connect premises distribution network to the optical network
unit to handle a whole range of high bandwidth applications, such as multichannel of
high definition TV broadcasting, VPNs, file downloading or uploading, video on demand
and surveillance systems.

3.2 Quality of Services

This section covers the requirements of QoS, conceptual model, implementations and
management of various QoS mechanisms to enable network administrators and archi-
tectures to deliver good quality traffic, full duplex communication, low levels of delay,
and allocate bandwidth in a way that improves applications performances across a
network in both directions.

3.2.1 Quality of Service Conceptual Model

QoS is a crucial element of any administrative policy since it measures the ability of
network to deliver data (end-to-end) with predicted results and computing systems to
provide different levels of services to networked applications such as, video conferenc-
ing, internet telephony, or voice over IP applications and associated network flows.
This services include error rates, network traffic loads, up-time, latency, and band-
width[12, 89].These applications require explicit quality of service guarantees in terms
of good quality traffic, full duplex communication, and low levels of delay.

The Internet Engineering Task Force (IETF) started working on a new group to develop
a framework for defining the services and service model and at the same time, an ar-
chitecture for an internet which can give quality of service guarantee [12, 89]. There are
 14

three different types of service models for providing QoS on a network namely Best-
effort, IntServ, and DiffServ.

3.2.2 Best Effort

The internet routing architecture is based on a best effort approach in which network
delivery of IP packets are treated in the same way to provide a scalable and reliable
network foundation. In best effort model, QoS is not applied to packets, the packets
arrives anytime in any order and no preferential treatment is guaranteed. For example,
critical data is treated the same as email. Best effort is the best suitable model for non-
real time applications such as telnet, file transfer or web browsing email in which QoS
is not necessarily needed.

3.2.3 Integrated Services

The Integrated Services (IntServ) model was designed to supplement the best-effort
delivery by reserving bandwidth, buffer and central processing unit time for applications
that require bandwidth and guaranteed packet delivery to end-to-end QoS over the
network [12, 89]. IntServ expects applications to signal their requirements to the net-
work to provide very high QoS to IP packets.

The signaling protocol which is used to set up the resource reservation in the IntServ
model is known as Resource Reservation Protocol (RSVP). RSVP is a general signal-
ing protocol where the receiver can specify its traffic characteristics and reserve net-
work resources through a network for an IntServ service. This quality of service attrib-
utes can be either at the individual application flow level or at aggregate level.

One drawback of this type of service model is that the individual applications and flows
must be maintained in the intermediate nodes and routers, making it impossible to
maintain large number of states. A new architectural framework known as differentiated
services model was introduced to solve this problem.
 15

3.2.4 Differentiated Services (DiffServ) Model

DiffServ framework was designed to overcome the limitations of both the best-effort
and IntServ models. It provides an almost guaranteed implementation QoS to a variety
of end-to-end services across the network IP packets while still being flexible, cost-
effective and highly scalable [31, 585]. The Differentiated Service model provides the
ability to assign different levels of services and QoS treatment to different network traf-
fic.

Essential features requirement for scalable QoS model

 Resources must be provisioned at aggregate class level but not at per-flow lev-
el.
 End to end quality of service should evolve gradually.
 DiffServ should not be independent on any dynamic signaling.

Requirement of differentiated services model/architecture

 It needs to accommodate a very wide variety of services and provisioning poli-
cies. Diffserv micro flows are subjected to policing and marketing according to
service level specification (SLS). SLSs specifies the service to be given to each
aggregate in the transmitter.
 It need a Per-Hop Behavior (PHB) which provides an appropriate way of traffic
conditioning and aggregate flow forwarding of traffic on a DiffServ interior node
to perform relatively coarse level of traffic classification within the network core
routers and switches in the path.
 No end to end services will be defined and no per flow or per customer or per
aggregate state will be maintained in the core router. Core routers will only look
at the index or label in the packet and router handles each packet differently.
 The resource will not be reserved by any hop by hop signaling.
 The admission control modules must ensure that new reservations do not ex-
ceed the aggregate traffic capacity to provide end to end services.
 It must be able to achieve scalability and gradual upgrade of the existing sys-
tem.
 16

With Differentiated Services, the scaling properties are achieved by marking each
packet's header with one of the standardized code point to deliver a particular kind of
service based on the QoS specified by each packet.

3.3 Video on Demand

Video on demand is an interactive TV technology that allows simultaneous real-time

access of audio and video materials to users and viewers over a cabled network. The
concept of VOD is based on the transmission of videos and audios in a coded and
compressed format and stored on hard disk or individual Set-Top Box (STB), and later
decoded and decompressed by set-top converters and then sent to the local server.

The main VOD system level consist of; a local database and server to store and pro-
vide access to programs, and a standard TV receiver along with a set-top box that al-
lows users to browse and play back a selected video as if they are watching from vide-
otape or a video player. Some of the main types of VOD systems are Quasi Video-On-
Demand (QVOD), True Video-On-Demand (TVOD), and Near Video-On-Demand
(NVOD). These systems and based on the amount of interactivity allowed[11, 4].

Quasi Video-on-Demand

QVOD is a service in which users are grouped based on their interest. Programming
will only be presented if a minimum number of subscribers sign up for it. Users can
choose between different programs by switching to a different group.

True Video on Demand

True video on demand is a service where the user receives an individual video stream
and has full control over requested playback media item. The user has full control of
continuous interactions such as start, stop, pause, forward, reverse at different speeds,
and full-function virtual video cassette recording capabilities [1, 36]. True video-on-
demand is achieved by paying a fee for each service request.
 17

Near video-on-demand

NVOD is a service in which a particular program is simulated to start in discrete time

intervals(of about 5 or 10minutes window) over a particular channel or multiple chan-
nels. NVOD are used for pay-per-view services, where a subscriber pays electronically
and selects time and day to start watching a particular program. Even though NVOD
only limits the subscriber to a 5-10 minutes waiting before the start of a program, it sig-
nificantly reduces the cost of services to subscribers [1, 36].

3.4 Triple Play Services

The concept of triple play service is considered as delivering telephony Voice over
IP(VOIP), IPTV, blended IP multimedia streaming services, and high speed internet
services over a single network using either fiber optic cable, copper cable, or satellite
transmitter.
Triple play services can be useful in multiple ways that include:
 Delivery of multiple services such as voice, video, and data over one single
network [12, 203-204].
 Flexibly to adapt to the next generation of multimedia-enabled networks and
scalable for future upgrade and maintenance.
 Triple play is cost effective enough to reduce operational and management
costs. This has made investors to maximize profit and increase return on in-
vestment.
 Triple play ensures a flawless user experience by offering mobility to enable
subscribers to do what they want anytime, anywhere.

To deploy a viable triple play service, the network must be more distributed to cost ef-
fectively deliver video and broadband; reliable, to allocate bandwidth to provide optimal
quality of experience for the subscriber; and flexible, to adapt to the next generation of
multimedia-enabled networks.
 18

3.5 IPTV Set-top Box

STB is a device that decrypt incoming signals into a synchronized format that directly
connects to an end-to-end IPTV services to enable subscribers to access a variety of
different types of digital entertainment content and video-on-demand programming con-
tent [12, 53-56]. A STB has a variety of TV interfaces at the back and front for connec-
tivity to a variety of different networking infrastructure [14].The set-top box back chan-
nel allow two-way communication to support interactive features like, adding premium
channels, option to play or stop live transmission, and the ability to record or save pro-
grams for future watchable purpose. Figure 7 shows a typical example of IPTV set-top
box.

Figure 7 typical example of IPTV set-top box. Modified from [15]

A typical digital set-top box has a physical height of 2.5 inches and a width of 18 inch-
es. As can be seen in Figure 7,the installation of STB is simple and involves plugging
one HDMI cable into the TV set and another into STB interface. Subscribers are pro-
vided with a handheld remote or wireless keyboards to choose what they want and
gain access to different channels and contents supported by the STB.

There are many different types of STBs based on different standards and geographical
locations. The most commonly known types are IP Set-Top Boxes(STBs), Hybrid IP
STBs, Hybrid IP Satellite STBs, Hybrid IP cable STBs, Multicast and Unicast IP STBs,
Digital STBs [14].
 19

4 Multimedia over IP

Multimedia is an important aspect in IPTV. It utilizes the combination of different media

types including text, audio, video, animation, and graphics interactivity content forms.
This section gives a clear picture of different multimedia transmission methods, appli-
cations and protocols used to transmit packets over a network.

4.1 Video Conferencing

IP Videoconferencing is a live communication technology that allows users to be seen,

be heard, share information, effectively interact and communicate face to face from
separate locations around the world without having to travel to a single location. It usu-
ally involves simultaneous transmission of audio, video and often text. The two most
popular videoconferencing software being used over the Internet are CUSeeMe and
Microsoft NetMeeting.

Videoconferencing is useful in general applications such as on demand meetings or

scheduled meetings, telemedicine, classroom practices and cross collaboration be-
tween partner institutions. It offers the ability to exchange ideas and knowledge, collab-
orate and communicate, share documents, share information, and even presentations
thus allowing decision making to be faster and easier. Videoconferencing purposes
have been found to be extensive and vary in nature.

4.1.1 Purpose and Benefits of Video Conferencing

The benefits and the most common purposes of videoconferencing are listed below.
 Video conferencing enhances face-to-face communication and collaboration in
real-time between two or more people regardless of location.
 It reduces cost of travel, maximizes productivity, improves work life and learning
experience, and accelerates decision making.
 Video conferencing provides live sharing of full-motion video images, text, and
high quality audio between two or more geographical locations providing an ex-
perience that is effective.
 20

 Videoconferencing is often used in educational institution such as colleges and

universities for live broadcasts of lectures and seminars to distance learning
students. It also promote cross cultural exchanges between students and tutors,
workforces and teams as well as stimulating collaborative learning.

 Keeps people in business organizations connected and unified and improves

communication between them. In this manner, employee training and group
work is facilitated.

It is easy to see that there is a growing need for videoconferencing both in business
and the educational fields.

4.1.2 Components of Video Conferencing System

In order to understand how video conferencing works it is important to recognize the

component parts of the system.

1. Camera

A webcam is used to record and send video signal that is required for adequate inter-
activity between other distant people during a live video conferencing session. These
can range from a simple desktop camera, small Universal Serial Bus (USB) camera to
other more high definition camera quality systems equipped with remote control pan,
autofocus, status indicator, automatic pan and zoom features.

2. Video Display /Monitors

The monitor displays the far end images to connected distant people received from the
videoconferencing codec. The monitor devices come in multiple options that is plasma
screens, liquid crystal display, projectors, and cathode ray tube.

3. Video Conferencing Codec Unit

The codec unit is used to digitize and compress video information into a digital signal
using encoding program to decompress the received transmission for play-
back[18].Common video codec's used in video conferencing applications are H.261,
H.263, H.264, MPEG2, and MPEG4.
 21

4. Microphone / Audio Sub-System

Many of stand-alone video conference systems automatically come with either a small
USB or analog microphones attached to a computer to enhance the audio capabilities
of the system and help with larger group interaction. Microphones can be of two types;
a unidirectional microphone picks up sound from one direction and an omnidirectional
microphone picks up sound from all directions [19, 68].

5. Computer with fast processor speed and large RAM

A computer with fast processor speed is needed to run the videoconferencing software.
It compresses and decompresses video streams and maintains the data link to the
network [18].

6. Speakers

A good set of speakers is essential to enable one to hear the audio from the far end of
the videoconference. Videoconferencing systems require a guaranteed symmetric
bandwidth for a point-to-point connection and multiple video performance (explained in
section 4.1.3).

4.1.3 Types of Video conferencing

There are two main types of videoconferencing systems; point to point videoconferenc-
ing and multipoint videoconferencing. This section briefly covers how these two types
of videoconferencing systems function.

1. Point-to-point video conferencing

Point to point videoconferencing is simplest scenario of videoconferencing system

where participant in different location communicate with each other using desktop vid-
eoconferencing software. Point to point video conferencing does not require a bridge to
function and each participant or group must use the same type of connection protocol.
Desktop systems are typically designed for a single user. They may be software client
based, web based, video-enabled IP telephone or dedicated standalone video appli-
ance [20].
 22

2. Multipoint video conferencing

Multipoint video conferencing are mostly commonly designed for more than two levels
of interactivity. It includes integrated audio, life-sized images, large flat panel display
devices and visual enhancements, which enhances the reality of the interaction. Mul-
tipoint conferences give the participants the feeling of being present in an actual meet-
ing, as well as the ability to see any content being shared during the meeting even
though the participants are geographically dispersed.

Multipoint conferences are created using a multipoint conference unit (MCU). The mul-
tipoint control unit either sends or receives calls from participants who dial the network
ID of the MCU to initiate a multipoint videoconference [20].

4.2 IPTV Networks

There are three methods used to transmit packets over a network: unicast, multicast,
and broadcast. They are introduced in detail below.

4.2.1 Unicast

Unicast by definition is one-to-one communication private session usually between a

server and a client. During a unicast session, each distinct stream is sent to only those
recipients that request the stream. VOD is an example of unicast application for IPTVs
in which the source servers establish an individual communication line to each request-
ing end user.

The end users during the VOD sessions will have the ability to pause, rewind and have
overall control to the video being steamed because of the established direct session
with the source server. A big concern for the video of demand services is higher band-
width demand requirements due to multiple individual direct connection between the
source server and the requesting end user. Figure 8 illustrates how data flows under
unicasting.
 23

Figure 8 shows how data flows under unicasting [23].

The information source sends a separate packet to a single host over the IP network as
shown in Figure 8. In this scenario, the server creates a separate transmission channel
for each host (E, D, and B) and the packets are received only if the destination address
matches one of its own IP addresses.

Unicast routing
Unicast routing is the process of forwarding unicasted traffic from a source to a unique
address on an internetwork [21]. Its goal is to determine a good path (sequence of
routers) through the network from source to destination. There are two main types of
unicast routing used. In distance vector routing, each node can only know the distance
between itself and shares its routing table with its immediate neighbors periodically.

In link state routing everyone gets a copy of a topology and computes their own routes,
including the type, cost (metric), and the condition of the links (up or down). All nodes
run the same algorithm (Dijkstra algorithm) concurrently to compute their forwarding
table in the same distributed setting as for distance vector.

The nodes know only who they are connected to, their neighbors and the cost to their
neighbors. They do not know the whole topology. The nodes can talk only to their
neighbors using messages to find what is going on for the network at large since they
 24

have no other ways to gain information about the network. The Open Shortest Path
First (OSPF) protocol is based on link state routing.

4.2.2 Multicast

Multicast is typically the opposite of unicast. With IP multicasting communication

mechanism, the host uses a specific IP address that send packets to a set of multicast
recipients to get traffic. The source server will broadcast the same session at the same
time to multiple users. A real time example of multicast session will be standard TV
programming whereby, the same prime time show will be playing at the same time on
all TVs that are tuned to the same channel.

IPTV provides the ability to set up multicast and stream videos via multicast as op-
posed to unicast. This capability maintains a session with same stream of data which
will also result to lower network congestion, lower bandwidth requirement and a reduc-
tion in the load of the sender and the overall load demands on the source server.

Multicast is a controlled and managed network thus making it much easier to predict
the bandwidth capability by having a certain backbone going through the network. Mul-
ticast sessions offers a couple of advantages over unicast sessions. Multicasting sup-
port is optional in IPv4, but mandatory in IPv6. It supports User Datagram Protocol
(UDP) only and its address can only be used as a destination address. Figure 9 illus-
trates how data flows under multicasting.
 25

Figure 9 shows how data flows under multicasting [23]

Suppose that Hosts E, D and B need receive information from the source server, they
need to join a receiver set or a multicast group as shown in Figure 9. The routers on
the network duplicate and forward the information based on the distribution of the re-
ceivers in this set [23].

The information source sends packets over the IP network to hosts (E, D, and B) that
joins the multicast group as shown in Figure 9. In this scenario, the server creates a
single transmission channel for all the hosts.

IP multicast addresses
IP addresses are defined in RFC 1112 and class D address are used as multicast in
the destination IP addresses field to identify a multicast packet. The Class D address
range is 224.0.0.0 to 239.255.255.255 and the numeric overall range of multicast ad-
dresses is 224.0.0.0 through 224.0.0.

The source IP address of a multicast packet is always a unicast address (Class A, B, or

C) [23]. Smaller address ranges are reserved for special use within the Class D ad-
dress range.
 26

Table 2 shows common multicast addresses reserved for various communications pro-
tocols. Modified from [23].

Reserved multicast addresses are used by network protocols on network routers for
different purposes as listed in Table 2. For example, an Open Shortest Path First
(OSPF) router must sends a "hello" packet to an assigned multicast address, which
is 224.0.0.5, and the other routers will respond. In case of multicast communication , by
pinging 224.0.0.1 address, all multicast capable hosts on the network must join that
group at start-up on all its multicast capable interfaces. A group of clients listening to
same multicast address is known as host group.

4.2.3 Broadcast Transmission

In broadcasting a single packet is delivered from one sender to all connected receivers
on the local network simultaneously. Each device that receives a broadcast packet
must process the packet in case there is a message for the device [24, 250].
Broadcast packets are normally restricted to the cable network and are undesirable for
streaming media, since even a small stream could flood every device on the local net-
work with packets that are not of interest to the device.
 27

Figure 10 shows how data flows under broadcasting [23].

In the diagram shown above, the source server sends a packet to all hosts (A, B, C, D,
and E) on the network segment. Suppose that hosts A,C and D do not need the infor-
mation and only B and E needs the information, the source will still broadcast the in-
formation to all the host, but only the one with that IP address will respond. To use a
broadcast transmission, map upper layer addresses to lower addresses, send a query
to request an address and then exchange routing information by routing protocols [23].

IP Broadcast Addresses
IP broadcast addresses can be used only as the destination IP address for single-
packet one-to-everyone delivery under the same LAN [24]. There are two different
types of IP broadcast addresses;

1. Limited Broadcast

The limited address is the broadcast limited to single LAN and represented by setting
all 32 bits of the IP address to the 255.255.255.255 [24]. It is used as the destination
address of an IP datagram during the automatic configuration process such as Boot
Protocol (BOOTP) or DHCP, and when the host does not know its subnet mask or net-
work ID. For example, with DHCP packets, the client must use the limited broadcast
address for all traffic sent until the DHCP server acknowledges the IP address lease
[25].This datagram is never forwarded by routers, it will only appear on the local net-
 28

work segment. The destination MAC address for such frames will be
FF:FF:FF:FF:FF:FF [25].

2. Directed Broadcast

Directed Broadcast address is the local subnet broadcast address sent to all hosts
FF:FF:FF:FF:FF:FF on an Ethernet interface [24]. The broadcast address uses the
highest address in the Ethernet interface range. For instance, if the subnet network ID
is 192.168.0.0, the directed broadcast address will be 192.168.255.255, which will be
heard by all in the same subnet hosts. NetBIOS Name Service (NBNS) uses directed
broadcast packets.

IP directed broadcast is used in implementing remote management or administration

task such as backups and Wake-on-LAN (WOL) on hosts in a subnet that is not directly
attached to the Internet. If no IP directed broadcast is configured, directed broadcasts
are dropped [26].

4.3 IPTV Protocols Network

There are many types of IPTV protocols used. The most commonly used are intro-
duced below.

4.3.1 Real-Time Protocol (RTP)

RTP usually runs over UDP and does not reserve bandwidth or guarantee QoS. RTP is
designed to support end-to-end delivery of real-time data such as voice and video from
the source to the receiver, and also supports a wide variety of media-on-demand appli-
cations such as internet telephony, IPTV services and online games.RTP randomly
picks even ports from UDP port or transport layer ports and encapsulate voice or video
data packets.
 29

RTP packet format

Table 3 illustrates the RTP packet format.

Bits 0 2 34 5 678 9 0 1 2 3 4 5 16 7 8 9 0 1 2 3 4 5 6 7 8 9 0 31
V P X CC M PT Sequence Number

Timestamp

Synchronization Source (SSRC)

Content Source (CSRC)

(0-15)

 V represents the version number 2.

 The padding bit (P) indicates if the padding bit is inserted at the end of the
packet containing padding octets, which are not part of the payload. Padding
may be required by some applications to fill up a block of certain size.
 The extension (X) bit indicates if the extension bit is set, there is an experi-
mental one extension header after the fixed header.
 The count field (CC) tells the count of contributing source identifiers that pre-
cedes the fixed header.
 The marker bit (M) is intended to indicate the frame boundaries of a speech
burst to be marked in the packet stream.
 The payload type (PT), 7 bits; this field identifies the payload format and deter-
mines its interpretation based on the information on the network feedback on
application quality. RTP defines a profile specification and one or more actual
payload formats, which may also describe some application specific extensions
or modifications to RTP. The profile provides a range of information that en-
sures a common understanding of the fields in the RTP header for a particular
application class [28]. These payload types include for example G.721, GSM
Full Rate, G.722 and G.728 speech codecs, Joint Photographic Experts Group
(JPEG) and H.261 video codecs.
 30

 The sequence number (16bits) is an incrementing counter which is started by a

source from a random number and used by the receiver to detect out-of-order
packet delivery.
 The timestamp (32 bits) corresponds to the generation instant of the exact time
at which first octet in the RTP data packet or payload are sampled. It provides a
mechanism that enable the source to playback samples at the appropriate time
intervals of the multimedia bit streams and to enable different media streams,
such as audio and video to be synchronized [12, 90].
 The SSRC 32 bits is a randomly generated value that uniquely identifies syn-
chronization sources within the same RTP session. It avoids two sources in the
same RTP session to have the same SSRC. Examples of synchronization
sources include the sender of a stream of packets derived from a signal source
such as a microphone or a camera, or a RTP mixer [27, 85].
 The Contributing Source (CSRC)is a list of the SSRC identifies of the sources
that contributes to the payload contained in the RTP header of a packet.

RTP is therefore responsibility for payload type identification, source identification, se-
quence numbering and time stamping. Translators and the mixers usually resides in
between senders and receivers to translate and forward RTP packets from one pay-
load to another. Mixer assigns itself as the sender of the packet, combines RTP
streams from different sources into a single stream and then forwards a new RTP
packet.

4.3.2 Real-Time Transport Control Protocol (RTCP)

RTCP is used during multicast audio or video transmission to receive streams of RTP
data packets from one or more sources and combines them into a single stream. RTCP
packets are distributed to all the participants using IP multicast. It is distinguished from
RTP through the use of distinct port numbers.

RTCP packets contain sender/receiver to exchange periodic reports such as;

 Number of packets send.
 Number of packets lost.
 Inter-arrival time jitters.
 31

It is up to the application to make use of RTCP packets. Different application may come
up with different algorithms and mechanisms to best use of these applications.

Receiver reception packets

Will contain information such as;
 SSRC of the RTP stream,
 fraction of packet loss,
 last sequence number received in the stream of RTP packets,
 Inter-arrival jitters.

Sender report packets

Will contain information such as;
 SSRC of the RTP stream,
 timestamp and wall clock time of the most recently generated RTP packet in the
stream,
 number of packet sent in the stream,
 Number of bytes sent in the stream.

Source description packets

Will contain information and about session participants;
 email address of the sender,
 senders name,
 SSRC of the associated stream.

RTCP is commonly used to create links on web sites that point to streaming media files
[1, 225].RTCP monitors periodic transmission statistics of control packets, reports qual-
ity of service (QoS) feedback, and helps to synchronize multiple streams. For example;
if RTCP packets are getting lost from the receiver, then it is obvious that the internet is
congested and it is recommended to appropriately adapt the sending rate or change
quantization levels.
 32

4.3.3 Real Time Streaming Protocol (RTSP)

RTSP establishes and controls the delivery of multimedia streams with real-time prop-
erties, such as audio and video across IP networks between client and server. RTSP
allows the client using a network remote control to communicate to the server infor-
mation to deliver channels such as UDP, multicast UDP and Transmission Control Pro-
tocol(TCP) [12, 92].

It is designed to work with lower-lever multimedia streaming protocols such as RTP. A

great protocol tool for VOD applications that have a unicast session between client and
the VOD servers. Some applications that use RTSP include YouTube, Spotify, Win-
dows Media Player, VLC player, and Skype.

Methods of control request

RTSP supports the following methods of control request made from the client to the
server.
1. Record

Initiates recording of a range of media being streams.

2. Option

An option request tells the client what request types the server accepts.

3. Describe

A describe request provides the client with a description of the media to start the ap-
propriate media applications. Figure 11 shows an example of a describe request.
 33

Figure 11 example of a describe request [29].

4. Play

Tells the server to begin sending the bit stream for playing the media file. Figure 12
illustrates an example of a play request

Figure 12 example of a play request [29].

5. Pause

Tells the server to pause media streams temporarily.

 Announce

Announce control request is used to send information to the client to register a new
entry or description of a media stream. From the client's side, it displays the description
of the media while from server's side, it updates the description in real time.
 34

6. Setup

Tells the server how to transport the media for an identified media stream and which
port to use.

7. Teardown

Terminates the media streaming session delivery and frees all associated network re-
sources associated with the session.

8. Get_parameter

Retrieves the value of a parameter a specified stream.

9. Set_parameter

Changes a parameter of a stream from a URL by enabling the client to issue request to
set the value of a parameter of a specified stream.

10. Redirect

Informs the client that it must connect to a different server and then moves it to that
server.

RTSP enhances interactions between the client and the server by using a network re-
mote control and adds a number of new requests to the existing HTTP requests as
shown in the list above. The client first requests the description of the media using the
DESCRIBE method, then requests that the session is SETUP and receives a session
identifier in return. The client requests that the media streams of the session are
PLAYED and at any point the client may PAUSE the media stream temporarily. When
the client has completed, the client issues a TEARDOWN request to terminate the me-
dia streaming session.

4.3.4 Protocol Independent Multicast (PIM)

The protocol independent multicast is a type of multicast routing protocol that does not
depend on any particular protocol for unicast traffic for its operation but it can leverage
 35

whichever unicast routing protocols used to populate the unicast table. All PIM routers
multicast group, or unicast to a specific destination [30].There are two main modes of
PIM that allow one-to-many and many-to-many transmission of information:

Sparse mode (SM)

PIM Sparse Mode (PIM-SM) is a multicast routing protocol designed on the principle
that recipients for any particular multicast stream will be sparsely distributed throughout
the internet with periodic multicast traffic [30]. PIM-SM by default uses shared trees
distribution, which are multicast distribution trees rooted in multiple Internet domains
designed to provide efficient communication, limit multicast traffic and used by all recip-
ients to join and leave multicast distribution trees. Figure 13 illustrates the PIM-SM
mode design with sparsely distributed streams being sent.

Figure 13 PIM-SM mode design with sparsely distributed streams being sent. Cop-
ied from [35].

One router is elected the 'querier' on each local/physical network, querier periodi-
cally sends membership query messages to 'all system group' (224.0.0.1) with TTL
= 1.

When active receivers actively request to join a specific multicast group, routers
along the path of these receivers register to join that group. Host sends a leave
group message to group address G if it was the most recent host to report mem-
 36

bership in that group. Router sends join messages to RP and sources register with
RP, intermediate routers update state and forward join messages. RP can send
stop messages to source if no receivers joined the group.

Characteristics of sparse mode (SM)

Characteristics of SM are listed below in details.

1. Designated Router

There must be one PIM Designated Router (DR) in each subnet in the network.
Any PIM-SM interfaces on the subnet elect the DR with the highest DR priority. If
there is more than one router with the same priority, or no priority, they choose the
interface with the highest IP address. If the current DR becomes unavailable, the
remaining switches elect a new DR for the subnet by DR priority or IP address [32].

The DR on the subnet containing a multicast source sends multicast packets to-
wards the Rendezvous Point (RP). DRs with group members connected RP sends
join messages towards the group’s RP.

2. Rendezvous Point

Each multicast group must have a RP. The RP for a group or range of groups is
found by an election process. The lowest preference value is elected from all the
RP group range of multicast addresses.

To create a routing tree for a group with rendezvous point as a root for the tree a
receiver send join messages towards the RP and the sender sends register to-
wards the RP.

3. Bootstrap Router

Each PIM-SM network must have at least one Bootstrap Router (BSR) candidate.
The Bootstrap Router for a network is chosen by election. The highest priority is
elected to be the BSR. The elected BSR listens to PIM Candidate RP bootstrap
messages to determine the RP for each multicast groups.
 37

Dense mode (DM)

PIM dense mode also known has push mode is assumed that all downstream sys-
tems wants to receive multicast feed or view the multicast feed.PIM dense mode
flooded across the network uses Reverse Path Forwarding (RPF) interface to receive
multicast traffic. It forwards the multicast traffic through every single segment
whereby some segments don't have group members interested in a multicast feed.
Packets arriving via the non-RPF interface are discarded. PIM-DM will prune off the
data packet destined for the group and forwarded by instantiating prune state.

PIM dense mode is recommended for small networks, to avoid more configurations
and easy management. Figure 14 shows the PIM-DM mode flooding example,
pruning unwanted traffic.

Figure 14 PIM-DM mode flooding example, pruning unwanted traffic. Copied from
[35].

Figure 14, the (S, G) state is created in every router in the network, multicast traffic
is flooded throughout the entire network. Figure 15, on the other hand, illustrates
the PIM-DM results after pruning.
 38

Figure 15 PIM-DM results after pruning. Copied from [35].

Other modes of PIM are source-specific multicast (SSM) and bidirectional, which
are not widely used throughout a multicast domain.

4.3.5 Internet Group Management Protocol (IGMP)

The Internet Group Management Protocol (IGMP) is used on secure stack to snoop the
multicast traffic only to those ports that need it. IGMP operates on a physical network
that is a single Ethernet segment. It is used by multicast router to manage membership
in IP multicast groups. It supports joining a multicast group, query membership and
sending membership reports.

Multicast router will send queries to the host from time to time when it joins a multicast
group. Report is sent only for the first process about multicast group membership relat-
ed to neighboring router interface. This means that the host has the right to respond or
not respond in receiving transmissions addressed to a specific multicast group. If there
is no response or response time expires, then it is treated as if the host left the group
and does not respond to the next query. Therefore, it will remove that host’s router in-
terface from the group.
 39

IGMP protocol is widely used in online streaming video and gaming. In IPTV it is used
to connect to a TV channel and to change from one TV channel to another.

IGMP host reports

Host sends a report when it joins a multicast group and no report is sent when a pro-
cess leaves a group.
For example:
IGMP report, TTL = 1
IGMP group addr = group address
dest IP addr = group address
src IP addr = IP address

IGMP router query

Router sends query at regular intervals to determine if known group members are still
active. Hosts belonging to a multicast group must reply to query if wishing to join or
stay in the group.
For example:
IGMP query, TTL = 1
IGMP group addr = 0.0.0.0
dest IP addr = 224.0.0.1
src IP addr = router IP address

IGMP message Types

Figure 16 shows an example of the components of IGMP messages.

 40

IGMP
Messages

Membership Leave
Query Report
Report

General Special

Figure 16 Components of IGMP messages.

The host may join a multicast group at their own will by sending a report message.
There is no restriction as the host can choose to leave a group at any time by sending
a leave report. Hosts can join as many group as they want to at a time. Membership
query are used to discover which hosts are members of a particular multicast group.

4.3.6 Internet Control Message Protocol (ICMP)

ICM Presides in the IP layer and it is part of the TCP protocol used for error handling in
the network layer. Since different types of errors can occur in the network, ICMP pro-
vide information messages concerning the routing of IP datagrams. It monitors, con-
trols network traffic and reports the errors after network debugging to diagnose prob-
lems within the network layer.

For example; if the router cannot send the packet to a particular destination, ICMP
communicate layer information between end hosts and routers sends an error mes-
sage indicating that it cannot deliver the message to the end destination. Error report-
ing messages are used when an end host or router wants to report an error using
 41

ICMP, it puts the information that it wants to send back to the source into an ICMP pay-
load and delivers it to IP to be sent as a datagram.

5 Compression in IPTV System

The goal of this section is to provide an understanding of how compression works and
types of compression techniques used. A good compression involves removing infor-
mation from a file without someone hardly telling the difference between the com-
pressed file and the original file. The compression techniques have made it possible to
transmit multimedia signals via the internet.

Compression is the discarding of information by using different algorithms from a video

or audio file in order to reduce overall file size, save storage capacity, speed file trans-
fer, and lower cost for network bandwidth. The higher the quality of the source file, the
higher the potential quality of the compressed files derived from it. Compressed file can
be made look better than the source file by making the frame size smaller. The size
and format of the source file have to have almost no relevance, all compression soft-
ware automatically determines the specs of the source file and creates the transcoded
file based on compression techniques applied. There are two compression algorithm
techniques used.

5.1 Compression Algorithm

The compression algorithm consists of two different kinds of compression, lossless and
lossy compression. They are both detailed below.

5.1.1 Lossless Compression

In lossless compression, data is compressed and the algorithm does not lose any sin-
gle bit of data when the file is uncompressed. Algorithms stores and transmits data into
smaller encoded files to restore the original information when uncompressed. Lossless
data compression is ideal for situations where any loss of textual information cannot be
tolerated. Example of this type of compression are ZIP compression and LZW com-
pression. Formats such as GIF and PNG use lossless compression.
 42

5.1.2 Lossy Compression

Lossy compression is a technique where a file is compressed or encoded to make it

smaller by minimizing the bit rate of original data and irretrievably discarding away any
unimportant data or unnecessary data. The data discarded from the original can never
be restored. Therefore it is recommended to compress files as few times as possible to
preserve as much data as possible.

This compression technique is used Digitally Sampled Analog Data (DSAD), where a
loss of quality of data can be tolerated. DSAD consists of picture files, audio or video
and graphics. Lossy compression is delivered in the form of Advanced Audio Cod-
ing (AAC), MP3,MP2, and many more. Digital audio is most often served in formats
that use lossy compression to save bandwidth transmission cost and storage space.

5.2 Compression Techniques

The following compression types, spatial and temporal compression, are commonly
used to compress audio and video.

5.2.1 Spatial Compression (Intra-frame)

Spatial compression is applied only to individual frames and mainly used to compress
still images such as JPEG by removing spatial redundancy that exist in each individual
frame. When a JPEG is created, color information of the image is reduced in a process
called chroma sub-sampling and then the image is split in a session of 8 by 8 pixels
called macro-blocks. Discrete cosine, transform and quantization is done to further re-
duce the file size.

Freeze frame is the pausing of a moving video which results in the view resting on one
frame. The computer reviews video in slow motion and looks at each frame individually,
and goes through a process of compare and contrast. It then reviews and take notes of
the elements that are similar or not and on the new changes that are taking place.
 43

Each frame does not have to be repeated making it much easier to reduce the overall
file size for upload.

5.2.2 Temporal Compression (Inter-frame)

In temporal compression a series of frames or pixels are looked at and see what is
different through the next ten frames. This compression technique looks at the data of
the current frame and then goes to the succeeding frame. It does not keep track of eve-
ry single pixel if the succeeding frames are identical or correlated, but only keeps track
of the changes over time by reducing redundancy. Temporal compression is to motion
compress and mainly uses lossy compression techniques such as temporal prediction
to reduce the file size significantly without too much quality loss.

The key goal of compression is to get the highest possible image and video quality
from the smallest possible bit rate. Image and video quality is a balance of five factors:

 Codec: Different codecs require different levels of efficiency settings for signifi-
cantly similar quality.
 Larger frame sizes require faster bit rates.
 Faster frame rates require faster bit rates.
 More movement between frames in the master file from camera, transitions,
and effects require higher bit rate needed to properly compress the frames for
higher quality.
 Video compression yields the smallest file sizes when there is very little move-
ment between the frames.
 Higher bit rates yield higher quality, but much larger file sizes.

The majority of television broadcast consists of 30 frames per seconds. YouTube for
example broadcasts video at 25 frames per second. Discarding those 5 frames per
second leads to a reduction of the overall size of video file. Many of live video streams
such as Skype are 15 frames per second which is a reduction of half of the original
frames.
In general, Intra-frame will always be better than inter-frame as long as the bandwidth
is affordable. Inter-frame normally comes of importance in situations where there is not
adequate bandwidth.
 44

5.3 Audio Compression and Video Compression

Audio and video compression is a very important aspect of IPTV that indicates how
audio and video streaming aspects e.g. in YouTube works. Video compression works
by minimizing redundancy in the video data and offers a number of standards encod-
ings as shown in Table 4 below.

Table 4 Image and Video Compression Standards. Copied from [36, 33].

Audio compression is typically the compression of audio signals into formats such as
MP3 or AAC, which reduces the size of the audio signal or file. Audio compression is
used to remove useless audio data from its dynamic range that goes from absolute
high to absolute low.
 45

6 Results and Conclusions

While cable companies worked on moving their content on the web devices, telecom-
munication companies streamed television content over private network to a set-top
box that is connected to a TV. IPTV is an important aspect in Television today dramati-
cally changing the way people communicate and operate especially when its full poten-
tial is implemented and converged with mobile TV. It is a highway through which end
users will see whatever video, whenever they want and on what device they prefer, be
it on a television, a computer or mobile device.

With the emergence of internet, the TV experience has undergone a fundamental shift
enabling cable operators and phone services to offer triple play of cable TV and high
speed internet in digital voice services. With triple play services, consumers are now
able to watch high quality programming over the internet using free or low priced over
the top services from the like of Hullu, Netflix, YouTube and Apple iTunes.

To match up cable quality, IPTV providers rely on content delivery networks that store
content across a geographically distributed network server rather than just one loca-
tion. Distributing content close to the end customer, helps reduce bottleneck that result
in startup delays or streaming video at a lower quality resolution. For instance when a
request is made from Finland to view a video that originates from the United States, it
can take a long time to cover that distance as compared to Sweden. This contributes to
the importance of QoS to the end user satisfaction and experience.

To assure the highest quality viewing experience, IPTV providers must monitor their
content delivery networks to isolate and quickly fix problems that might lead to sub-
scriber defections. Phone companies such as Sonera can optimize streaming video
performance, reduce costs and display an encoded video stream of IP packets via a
STB. Some Telco network operators such as AT&T utilize DSL technologies to deliver
IPTV and broadband services to users over their access network.

IPTV systems have the advantage over traditional cable TV services in comparison in
that they are relatively independent on the operating point and offer quality of network
solutions. The compression techniques of digital television data allows for more storage
of content with less space. IPTV can be applied in different institutions and organiza-
tions such as education, finance, and medical fields to stream live operations with
 46

higher quality of content delivery, protection of content and control of video quality over
a private network. It frees up bandwidth for distributers allowing them to deliver greater
content to their customer.
 47

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 Appendix 2
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