Chapter 1

INTRODUCTION

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 Chapter 2

THEORETICAL BACKGROUND

This chapter discusses the theoretical basis of a Public Address System, an

overview of the operation of the components used in each type of PA system,
relevant wireless technology to be used for real-time transmission, and a summary
of related works and literature.

2.1 Theories

2.1.1 Public Address System

A public address system (PA system) is an electronic sound amplification and

distribution system with a microphone, amplifier and loudspeakers, used to allow
a person to address a large public. This may be used for systems which may
additionally have a mixing console, and amplifiers and loudspeakers suitable for
music as well as speech, used to reinforce a sound source, such as recorded
music or a person giving live announcement or distributing the sound throughout a
building or venue. Public address systems consist of input sources, amplifiers, and
loudspeakers. The primary input sources are microphones for live announcements
and a source of recorded sound [PA].

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 Figure 2.1: Public Address System Basic Components [PA].

The Fig. 2.1 shows the basic components of a PA System and how they are
interconnected with each other.

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 Figure 2.2: Public Address System via LAN [PA].

Fig. 2.2 is the system allowing operators, or automated equipment, to select

from several standard prerecorded messages. These input sources are fed into
preamplifiers and signal routers that determine the zones to which the audio signal
is fed. The preamplifier signals are then passed into the amplifiers. These am-
plifiers will usually amplify the audio signals to 50 V, 70 V or 100 V speaker line
level [PA].
Function of Different Blocks of PA System:

• Microphone The microphone is an input source manually set up by the Multi-

media Solutions and Documentation Office (MSDO) during announcements.
The wireless microphone system currently used in CIT-U is the Sennheiser
EW100 G2. It is an Ultra High Frequency (UHF) wireless microphone sys-

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 tem featuring 1440 tunable frequencies, and four frequency presets. This
system includes the EM100G2 D True Diversity Receiver, SK100G2 body-
pack transmitter, a handheld wireless microphone, and ME2 omnidirectional
lavalier microphone.

• Amplifier
Two amplifiers are being used in CIT-U and are located in the Safety Security
Office (SSO). Fig. 2.3a is the A-2060 Mixer Power Amplifier (H version) and
Fig. 2.3b is the Yamaha XH200 2-Channel Power Amplifiers.
The A-2060 Mixer Power Amplifier (H version) is a high cost-performance
mixer power amplifier.
The Yamaha XH200 features a 2-channel high-impedance power amplifier
(200 W per 24 ohms at 70 V, 200 W per 48 ohms at 100V).

Table 2.1: Specifications for A-20160 [amp1] and Yamaha XH200 [amp2].

A-2060 H Yamaha XH200

Power Source 220 – 240 V AC, or

24 – 30 V DC

S \N Ratio Over 60 dB 20 Hz–20 kHz

Power Consumption 72 W (EN60065), 40 W (idle)

150 W (AC opera- 250 W (1/8, 4 Ohms)
tion at rated output),
4 A (DC operation at
rated output)

Frequency Response 50 – 20,000 Hz (3 -0.5 dB, 0 dB, 80 Hz-

dB) 20 kHz

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 (a) PA Amplifier.

(b) Power Amplifier.

Figure 2.3: Amplifiers used in CIT-U.

• Hornspeaker These horn speakers in Fig. 2.4 are currently installed in some
buildings in CIT-U. The model of the said horn speaker used is SC-630TU, its
main function is to produce sound created after an announcement is created.

Table 2.2: Specifications for SC-630TU [hspeaker].

Rated Input 30 W

Sound Pressure Level 113 dB (1 W, 1 m at 500 Hz to 2.5 kHz peak level)

Frequency Response 250 Hz – 10 kHz

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 Figure 2.4: SC-630TU[hspeaker].

Categories of PA System:
PA system is required in addressing large crowds of people to communicate
effectively. The whole concept of PA system revolves around making a sound
louder over a considerable distance. The following are the various PA systems
existed:

• Indoor PA Systems - This type of PA system is usually used in a small event

such as an art exhibition. Here, the volume occupied by the space is the key
to determining the power needed when setting up PA systems. This will call
for a less powerful system than that of a building with an extensive ceiling.
In some cases, sound can be absorbed instead of being reflected by things
such as thick carpets, and this calls for Pa systems as well. The same case
applies when it comes to huge crowds as the sound is absorbed [CatPA].
Indoor PA systems still consist of input sources such as microphone with
786 MHz to 822 MHz frequency range, 80 Hz to 20 kHz frequency response
for amplifiers, and 200 Hz to 10 kHz frequency response for loudspeakers.
These components are shown in Figure 2.1.

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 Figure 2.5: Audio Performer Pack [compoPA].

• Fixed PA Systems - Some locations may prefer having fixed PA systems as

they are used now and then, for example, a movie theater. Here, adhering
the layout of the area is needed. This will help in determining whether more
hardware is necessary to cover the sound needs. Fixed PA systems also
offer the chance to alter the design of a specific location to look professional
and allows for extras such as lighting. Designing is also vital in ensuring the
place is tidied up, such as hiding exposed wires [CatPA].

• Portable PA Systems - PA system that can be moved from one place to an-
other easily, example of which is a school where the same PA system is used
for various functions. These are mostly systems that have simple and light
equipment that you can transport using your own vehicle. The advantage of
small portable systems is that they can be easily stored as they do not oc-
cupy a lot of space. They also do not take up a lot of time to set up. However,
you may need extra help moving some PA systems that are heavy and have
a lot of equipment and this may lead to extra costs that you had not budgeted
for such as hiring a scissor lift to help move the equipment [CatPA]. Fig. 2.3a
and Fig. 2.3b shows the conventional setup for PA system. These amplifiers
can be moved easily from one place to another. Portable PA systems also
consist of input sources such as microphone with 786 MHz to 822 MHz fre-

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 quency range, 80 Hz to 20 kHz frequency response for amplifiers, and 200
Hz to 10 kHz frequency response for loudspeakers.

2.1.2 TCP/IP

The Transmission Control Protocol (TCP) and the Internet Protocol (IP) are the
foundational protocols in the Internet protocol suite.

Figure 2.6: Layers of TCP/IP model [IPlayers].

Fig. 2.6 shows the different layers of TCP/IP model.

If we are to follow the path of a message being sent from the transmitter to the
receiver, it would happen like the diagram shown in Fig. 2.7.

Figure 2.7: VoIP Architecture [IPlayers].

The message starts at the top of the protocol stack on the transmitter and work
its way downward. If the message to be sent is long, each stack layer that the

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message passes through may break the message up into smaller chunks of data
called packets. The reason for this is that data sent over the Internet are sent in
manageable chunks.
Next, each packet assigned with a port number goes through the Application
Layer and continue to the TCP layer. Knowing which program on the receiver
needs to receive the message because it will be listening on a specific port.
After going through the TCP layer, the packets proceed to the IP layer. This is
where each packet receives its destination address. Once the message packets
have a port number and an IP address, they are ready to be sent over the Internet.
The hardware layer is now then responsible of turning the packets containing the
alphabetic text of the message into electronic signals and transmitting them over
the device.
The Internet Service Providers (ISPs) router examines the destination address
in each packet and determines where to send it. Often, the packet’s next stop is
another router.
The packets now reach the receiver side. Here, the packets start at the bottom
of the destination computer’s TCP/IP stack and work upwards. As the packets
go upwards through the stack, all routing data that the transmitter stack added is
stripped from the packets. The packets now reassemble into their original form
when the data reaches the top of the stack.

IP Address

An Internet Protocol (IP) address is a unique number assigned to every device

on a network. Network devices use IP addresses to communicate with each other.
When a device is assigned a Static IP address, the address does not change. On
the other hand, most devices use Dynamic IP addresses, which are assigned by
the network when they connect and change over time [IPadd].

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Two versions of Internet Protocol:

• Internet Protocol Version 4 (IPv4): It is the most widely deployed IP used

to connect devices to the Internet. It uses a 32-bit address scheme allowing
for a total of 232 addresses (just over four billion addresses).

• Internet Protocol Version 6 (IPv6): In IPv6, the address size was increased
from 32 bits in IPv4 to 128 bits, thus providing up to 2128 (approximately 3.403
× 1038 ) addresses.

2.2 VoIP over WLAN

VoIP contain three components [VoIP]:

• CODEC
The process to sample analogical waves into digital information is made by
an encoder-decoder (CODEC). The voice is compressed and encoded into a
predetermined format using voice codec when the sender send voice signal.

• Packetizer
In packetization process in which fragment encoded voice into equal size of
packet, each packet contain some protocol header from different layers.

• Playout Buffer
Suitable layout buffer algorithm is developed that minimizes audible artifacts.

2.2.1 RTP and RTCP

The Real-Time Transport Protocol (RTP) is a protocol to carry data that has the
real-time properties. It is the main transport protocol used for IP telephony me-
dia streams and it defines a standardized packet format for delivering media over
the internet. RTP provided end-to-end network transport functions of applications

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transmitting real-time data, such as interactive audio and video over multicast or
unicast network services [VOIP1].
The Real-Time Transport Control Protocol (RTCP) provides the control services
for data stream that uses RTP. The RTCP provides feedback on the quality of
the transmission link. The other function of RTCP includes carrying a persistent
transport-level identifier for an RTP source and this identifier is used be receivers
to synchronize audio and video and convey minimal session control information
such as participant identification to be displayed in the user interface [VOIP1].

Figure 2.8: VoIP over WLAN Protocol Architecture [VoIP].

Fig. 2.8 shows that RTP and RTCP found on the Application Layer are designed
to support real-time application, although TCP transport layer is used in TCP pro-
tocol suitable for less delay sensitive data packet. This scheme introduced delay
as receiver has to notify the sender for each received packet by sending an ac-
knowledgement. On the other side, User Datagram Protocol (UDP) does not apply
this scheme, so it is more suitable for VoIP applications [VoIP].

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 Figure 2.9: VoIP End to End Communication [VoIP].

The signaling protocols of VoIP shown in Fig. ?? are Session Initiation Proto-
col (SIP) and H.323. SIP establish end to end close media streams between the
clients. While H.323 is standardized by ITU-T specially for smoothly working to-
gether with SIP and PSTN while SIP was introduced by Internet Engineering Task
force (IETF) to support application layer such as telephony. IP addresses can be
changed from one session to another, especially in dial-up-case, therefore there
is need for a common meeting point shared among user to discover each other
at the establishment stage of communication. The meeting point is called as Call
Server [VoIP].

2.2.2 Media Transport using TCP

Once the session layer is established using signalling protocol, the media (au-
dio, video or other media streams) is ready to be exchanged between parties. In
order to transfer real-time application data over internet, it is not suitable to use
a common stream-oriented protocol such as TCP. TCP is optimized for a reliable
transfer of bulk data and cannot handle real-time traffic. In real- time scenario us-
ing TCP, if there is a packet loss or reordering in the network, the delivery to the

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recipient will be delayed until the gap is filled. Because of this, User Datagram
Protocol (UDP) is widely used as transport layer protocol in VoIP [QoS].
UDP datagrams are further encapsulated using Real Time Transport Protocol
(RTP) in order to provide reliability for real-time traffic. RTP header contains a
sequence number and time stamp to correctly reassemble the media stream at the
receiving end. RTP is an unreliable and provides no mechanism for retransmission;
as a result the received media stream contains gaps or glitches. RTP maintains a
hitter buffer for incoming media at the receiving end to avoid the glitches imposed
by reordering. But this buffer also imposes latency, so its size should be chosen
carefully helping to provide better Quality of Experience (QoE) [QoS].

2.2.3 Quality of Service

QoS may be defined in different ways in telecommunication. From the Inter-

net service provider perspective, it can be traffic engineering to load balance the
internet traffic. The VoIP user evaluates the voice or video quality using quality
of experience (QoE). Better QoE is experienced in VoIP communication depend-
ing on the QoS provided by both control and data planes. It is mentioned above
how RTP provides better QoE for the voice/video data, but QoE does not entirely
depend on jitter or packet loss in media. SIP signaling or call set-up mechanism
has been emerged as Quality of Signaling (QoSg) focusing on QoS of different
parameters during call set-up process of SIP Protocol, and a separate IETF draft
has been evaluated only for SIP performance metrics [QoSanal].

2.2.4 Wireless Fidelity (Wi-Fi)

Wi-Fi is a technology that uses radio waves to provide network connectivity.

It is commonly used as wireless local area network (WLAN) allowing local area
network to operate without cable and wiring [WiFi].

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 The main characteristics of WLAN is simplicity, mobility, scalability, edibility
and cost effectiveness. It is same as LAN except that the transmission happens
via radio frequency (RF) of infra-red (IR) and not through physical wires and ca-
bles [VoIP].
Two ways of connecting and configuring WLAN:

• Infrastructure less mode: In Infrastructure less mode which is also called

Ad-hoc or P2P(peer-to-peer) network, there is no fixed point and each node
can directly communicate with all other nodes [VoIP].

• Infrastructure mode: In Infrastructure mode, there is the transmission be-

tween two or more nodes goes through a third node called Access Point (AP)
or Wireless Router [VoIP].

The terminal communicate with each other through AP forming a one-hop net-
work. When any terminal wants to send packet to other terminal, packet would be
sent to the AP first which will forward them to their destination [VoIP]. Lastly, IEEE
802.11 WLAN standards protocols deployed [VoIP].
Different operating frequencies of IEEE 802.11 Protocols:

• 802.11a support 5 GHz and 84 Mbps data rate.

• 802.11b support 2.4 GHz and 11 Mbps data rate.

• 802.11g support 2.4 GHz and 54 Mbps data rate.

• 802.11n support 2.4 GHz to 5 GHz and 300 Mbps (and can reach up to 450
Mbps when using three antennas) data rate.

• 802.11ac support 5 GHz and 433 Mbps up to several gigabits per second
data rate.

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 These standard is added by IEEE alphabetic character in each standard. Re-
cently, the integration of VoIP and IEEE 802.11 WLAN technologies. WLAN cannot
provide good service quality for almost real-time traffic. Therefore, deploying VoIP
over WLAN poses a challenge in term of performance is expected to be a good as
Public Switched Network (PSTN) performance or even better.

2.2.5 Raspberry Pi

Raspberry Pi is a credit-card sized computer capable of doing everything a

desktop computer would do. It is rampantly used in making different useful projects
and is also used in low-cost maintenance business ideas.
Types of Raspberry Pi:

• Raspberry Pi Zero

Fig. 2.10 is the Raspberry Pi Zero 512 MB RAM model with mini HDMI and
micro USB socket and BCM2835 1 GHz ARM11 CPU.

Figure 2.10: Raspberry Pi Zero [RPi0].

• Raspberry Pi Zero W

The Raspberry Pi Zero Wireless (W) is a 512 MB RAM model comes with
most of the same specifications as the standard Raspberry Pi Zero, but adds
the 802.11n Wireless LAN and Bluetooth 4.0.

• Raspberry Pi 1 Model A+

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 The Raspberry Pi 1 Model A+ is a 256 MB RAM model with HDMI, no Ether-
net and one USB 2.0 port and BCM2835 700 MHz ARMv6k CPU.

Fig. 2.11 is the Raspberry Pi1 512 MB RAM Model B+ comes with most of
the same specs as the standard Raspberry Pi1 Model A+, but has four USB
ports and has Ethernet.

Figure 2.11: Raspberry Pi 1 Model B+ [RPi1].

• Raspberry Pi 2 Model B

Fig. 2.12 is the Raspberry Pi2 1 GB RAM model B with HDMI, has Ethernet
and four USB 2.0 ports and BCM2836 Quadcore 900 MHz ARMv7 CPU.

Figure 2.12: Raspberry Pi 2 Model B [RPi2].

• Raspberry Pi 3 Model B

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 Fig. 2.13 is the Raspberry Pi3 1 GB RAM model B comes with most of the
same specs as the standard Raspberry Pi 2 Model B, but adds the 802.11n
Wireless LAN and Bluetooth 4.0 and has a Quadcore 64-bit 1.2 GHz ARM
Cortex A53 CPU.

Figure 2.13: Raspberry Pi 3 Model B [RPi3].

• Raspberry Pi 3 Model B+

Fig. 2.14 is the Raspberry Pi3 1 GB RAM model B+ with HDMI, four USB
2.0 ports, has Gigabit Ethernet (via USB channel), Camera Serial Interface
(CSI), Display Serial Interface (DSI), then adds the 2.4 GHz and 5 GHz
802.11b/g/n/ac Wi-Fi and Bluetooth 4.2 and has a Quadcore 64-bit 1.4 GHz
ARMv8 Cortex A53 CPU.

Figure 2.14: Raspberry Pi 3 Model B+ [RPi3plus].

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2.3 Literature Review

2.3.1 Wireless PA System used in Schools

The study [IPBasedPA] proposes an IP-based PA system which will be im-

plemented in schools. The administration office has a PC unit for pre-recorded
announcements and an external wireless microphone with input buttons for the
live announcement, which all together serves as the center console or server. The
center console transmits the data through a router going to a ready-made IP box.
This IP box serves as the medium of communication between the microphone unit
and the receiving speaker through the router. The IP box may be attached with
an amplifier for the speaker or be connected directly to the speaker. The system
includes a receiving class D speaker situated in each building. Operation of the
system starts when the user inputs the assigned numbers to the microphone in
the center control system which corresponds to a specific IP address of a certain
building. Once the user inputs these specific numbers, communication is triggered.
In another study [CN206948593U], the digital wireless transmitter single chip
input terminal is connected to a microphone, and an output terminal connected to
a transmission antenna. The said receiving means comprises a second micropro-
cessor, said second microprocessor connected to a digital single-chip radio and
infrared receiver Rx, the single digital wireless Rx input receiving antenna chip and
an output terminal connected to a subsequent stage of the audio processing circuit
is also connected. This covers the operating frequency of UHF band, and work is
performed between the single-chip digital wireless Tx and Rx single chip digital
wireless Rx through digital modulation and digital modulation-modulation identifi-
cation code comprising of operating frequencies. When the operating frequency of
the digital modulation identification code correspondence simultaneously, transmit-
ting device and the receiving device are successfully paired. This uses the power

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supply of polymer lithium cell, standard USB interface charges, and lower battery
use cost.

2.3.2 Internet Facilitated PA System

The study [CA2559267C] shows how the general paging system works over
the Internet. In one aspect, the Internet and VoIP processing transfers real-time
audio, voice, and instructions from a common control terminal or paging interface
to a plurality of zones in specified remote facilities. Next, the VoIP processing can
be combines with a data encryption protocol for real-time voice transmission and
messages be secured otherwise, decryption may also take place. Common input
source such as microphone can be used to forward real-time messages to remote
locations and selected zones. Through a GUI, available facilities and zones can
be identified for selection. Once selection is done, the audio message from the
operator can be transmitted real-time at the various remote facilities by activation
and deactivation of appropriate speaker zones specified by the operator. IP ad-
dress can be assigned in each destination for purposes of carrying out the Internet
communication process.
It is also stated in this study that using TCP/IP or UDP protocols, messages
can be transferred. Compression techniques and methods before transmission
are used to process the voice. Full duplex communication can be implemented by
providing a substantially identical paging interface at the destination for purposes
of transmitting messages back to the source interface.

2.3.3 Wireless PA System device

In the study [US20030053434A1], a system and method is designed by which

a wireless LAN may provide such emulation of a Push-to-Talk(PTT) device. IPPTT
is the preferred embodiment functioning over WLAN similar to that specified by

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IEEE 802.11, enhanced with a Quality of Service (QoS) protocol. An advantage of
WLAN with QoS is the ability to support phased migration to an all-LAN architecture
by eliminating the need to rapidly convert all conventional phones to VoIP.
A wireless device called Pi-Phone is invented in [PiPhone] where Raspberry
Pi 3 Model B+ , Resistive 2.8” PiTFT display, and an antenna are used. This
device functions like a cellular phone. The antenna being used in this study has a
microphone input and a speaker output, therefore an electric mic and a mini metal
speaker may be connected so that a handset is not a need any more.

2.3.4 Wireless Real Time Communication

The study [Tommi2014] specifies the most notable factors of a wireless lo-
cal area network impact on real-time, IP based data transmission. By studying
the specifications of a good quality voice call the main network afflicted causes for
user-end quality degradation were narrowed down to latency, jitter and packet loss.
The main tool for decreasing the effect of these three was found to be the prior-
itization of data. Deploying a Quality of Service (QoS) framework, the latencies
caused by link and queue delay can be significantly reduced.
In the study [Kotz2002], Mobile Voice over IP (MVOIP), an application-level
protocol is used to support terminal mobility in real-time applications such as voice
over IP, on a wireless local area network. Voice over Internet Protocol (VoIP) appli-
cations transmit real-time data. MVOIP provides a mechanism to maintain a VOIP
call even as the underlying network addresses of the hosts engaged in the call
need to change.

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