VOIP

1.1

Introduction of VoIP

The digitation of voice are being offered by local and long distance telephone operators, cable
television companies, Internet Service Providers, independent operators and mobile operators.
Voice are conversed and compressed into data packets that are transmitted over an IP network, to
be reassambled and converted back into voice communication at the receivers end. Digitation of
voice is known as VOIP (Voice over Internet Protocol). VOIP broadly includes : (1) Voice over
broadband (VoB); (2) Voice over digital subscriber line (DSL); (3) Voice over Internet; (4) Voice over
wireless local area network, and (5) Internet telephony. What distinguishes PSTN to VoIP? The
answers are the service offerings and capabilities. Some of the key differences between PSTN and
VoIP are summarized as follow :

Factor

PSTN

VoIP

Network/service relation

different networks providing multiple services

unified intergrated network providing multiple services

Mobility of services

limited mobility of end - user services

Broad - based generalized mobility

Interconnection

Circuit - switched interconnection (transit/termination)

Packet based, based on IP.

Structure

Vertically integrated

Services

Voice with some data in low bandwith offerings

Horizontally integrated
All existing services (including voice and data) with the
possibility to offer bundled packages in triple play in higher
bandwith offerings.

Scalability

Upgrades require purchasing more hardware and

dedicated lines, which can be very complex and costly

Upgrades usually require more bandwith and simple software

updates.

Table 1.1. The Differences between PSTN with VoIP

Sources : Various sources
1.1.1

How is VoIP work ?

VoIP works through the VoIP network. VoIP network consists of VoIP Server, switch, router, VoIP
Trunk, VoIP Client, VoIP Gateways, Protocol and Codec. VoIP server is the main server of VoIP
network that routes all the call in and out from and to VoIP clients. VoIP client is all the devices that
used by the end user to receive and send the call. VoIP System can connect to the VoIP client
through the switch. But to connect to the ethernet, VoIP server will connected to the switch and the
router. To make a call in and out, VoIP Server must be connected to the VoIP trunk which is
provided by VoIP Provider. In the VoIP trunk records all the IP address of the incoming and outgoing
call. VoIP server no longer has to be in the building. It can be hosted in the outside of the building

because of the ethernet network. The enterprise can rent the VoIP server from the VoIP Provider. So
the enterprise only needs switch and router and VoIP devices to use the VoIP call. This is the reason
why VoIP is cheaper than traditional telephony system. Customer doesnt need to build and
maintain their own PBX which costs lot of money.
The VoIP Clients which are consist of the all devices (smartphone, IP Phone, PC) to get connected to
VoIP server must create the account user name and password. The configuration of the account user
will be configure in the VoIP Server. Each user has it own extension. So, the end user must login with
the right username and password to make a call thorugh the VoIP server.
Another component of VoIP system is gateways. Gateways will connected the VoIP server to the
traditional phone line (PSTN). So, once the user make an outgoing call to the PSTN, the VoIP Client
will connect to VoIP server and then the VoIP server will connect to a gateway and then connect to
the PSTN.
All the VoIP communication uses Protocol to talk each other. The most popular protocol for VoIP is
SIP (Session Initiation Protocol). It works by sampling the sound by a computer at very high rates (at
least 8,000 times per second or more) and recording or stroing these samples. Then the computer
compresses the sound, so they require less space using a compressor / de-compressor algorithm
(CODEC). Once recorded and compressed, the sounds are collected into larger data packets and
sent over the IP network. Generally, an IP Packet will contain 20-30 more milliseconds of audio.
Codec is how the voice traffic incoded and incapsulated. The quality of the voice communication is
determined by the codec and the bandwith. To get the high definition of the sound, we can use the
the latest technology of the codec.
Packet loss during transmission is compensated for by filling in the gaps, a process called Packet
Loss Concealment (PLC). Other methods to ensure the completeness of packet include sending
duplicate packets multiple times (redundancy) and including some overlap of data within packets to
allow reconstruction of lost packets (Forward-Error Correction. Packets can also be delayed thorugh
jitter (which is a particular problem for voice. Jitter can result in choppy voice or temporary glitches,
so VoIP devices must implement jitter buffer algorithms to compensate for jitter. PLC algorithms can
also smooth the audio and compensate for late arrival of packets (which are then discarded). Video
works in a very similar way video information is broken into small pieces, compressed using a

CODEC, placed into small packets and transmitted over the IP network. This means that other media
including video can be added onto VoIP in a relatively simple manner.
In order for computers to communicate in this way, there must be agreed protocols for how
computers find each other and how information is exchanged in order to allow the flow of packets
between the communicating devices. Some of the main protocols for the different functions are :
a.

Call Signaling Protocols

These protocols find the remote device and negotiate the means by which media
can flow between the two devices to establish multi media communication. The two
most popular protocols are :
H.323 Protocol
The H.323 protocol was one of the first VoIP protocols that found widespread
implementation for long distance traffic, as well as local area network services.
This earlier protocol is recommended for the transmission of voice and video
over PSTN, ISDN, and ATM. It borrows heavily from legacy communication
systems and is a binary protocol, which means that it has better interoperability
with PSTN and offers better support for video and reliable out of band transport
of DTMF.

However, since the development of newer protocols, H.3232

deployments are increasingly limited to carrying existing long haul network

traffic.
Session Initiation Protocol (SIP)
SIP is the other main end-to-end

signalling protocol. It facilitates

communications between 2+ SIP supported devices, but further protocols may

be needed to make VoIP calls, including SIP extensions to carry necessary
information and provide the necessary functionality. SIP is the most popular for
instant messaging systems because it is easier to implement and troubleshoot.
b.

Device Control Protocols

These protocols governing the interface between the Media Gateway Controller
(MGC) containing the call control logic and Media Gateway (MG) interfacing with the
PSTN in the split gateway. These protocols include H.248 and Media Gateway
Controller Protocol (MGCP).

Figure 1.1 The Open System Interconnection (OSI)Reference Model of VoIP

Source : ITU Document

1.2

VoIP Industry

VoIP is an emerging trend in business communications, gaining ground because of its affordability.
Business VoIP platforms usually include a hosted or manager PBX phone system plus another
popular VoIP features While with traditional phone companies these features available only as
expensive add-ons.
1.2.1

Trend of VoIP Market and International Voice

Based on the research by market intelligence (http://point-topic.com/free-analysis/global-voipsubscriber-numbers-q2-2013/) that VoIP subscriber has reached 158.7 millions at the end of June
2013. The VoIP subscriber market is dominated by six leading countries, United States, Japan,
France, South Korea, Germany, and China. These six countries have nearly 80% of the reported
global VoIP market share.

Figure 1.2 Global VoIP subscriber numbers and net growth

Source : Point Topic
The growth in VoIP subscriptions continues to decline in 2013, but the numbers of subscriber still
grow. These number subscribers spread all over the world with the proportion as follows :

Figure 1.3 Regional VoIP Market Shares in Q2 2013

Source : Point Topic
From the figure above, we know that Asia-East Regional is the biggest market for VoIP. Singapore as
the part of South East Asia Regional have a growing market for VoIP Business. A Growing broadband
market and the declining of fixed lined market are the indicators that VoIP can be a promising
business of telecom services.