IMPLEMENTATION AND DEPLOYMENT OF IP PBX

(3CX)

BS Computer Science
Session 2016-2020 (spring)
Submitted To
Minhaj University Lahore
Department of Computer Science
Submitted By
Asma Batool
2016s-mubscs-03
Supervised By
Mr. Muhammad Sheraz Tariq
(Lecturer CS)
Department of Computer Science
 Contents
Chapter 1 9
Introduction 10
The Wireless LAN &VOIP 11
Why A Wireless LAN..........................................................................................................16
Problems with Wireless VoIP..............................................................................................16
Chapter 2 17
Signaling:.............................................................................................................................18
VOIP Options:......................................................................................................................18
SIP and H.323......................................................................................................................18
Database services:................................................................................................................19
Bearer control:......................................................................................................................19
Codecs:.................................................................................................................................19
VoIP Signaling Protocols 20
H.323....................................................................................................................................20
MGCP..................................................................................................................................20
SIP........................................................................................................................................20
SCCP....................................................................................................................................21
Components of a VoIP Network 21
Types of VOIP 23
Hosted PBX..........................................................................................................................23
Cloud PBX...........................................................................................................................24
On-Premise PBX..................................................................................................................24
Chapter 3 26
Overview 27
QoS 28
VOIP QoS Requirment 29
Hosted VOIP network..........................................................................................................30
Chapter 4 32
3cx 33
Why 3CX 33
Cost saving...........................................................................................................................33
Open plateform.....................................................................................................................33
Easy to Deploy.....................................................................................................................34
Easy to Manage....................................................................................................................34

2
 Mobility................................................................................................................................35
Advanced Customer Service functions................................................................................35
Tried and Tested...................................................................................................................35
Open-standards, software communications.........................................................................36
Save with affordable & transparent pricing.........................................................................36
Easy to get started, easy to maintain....................................................................................36
Increase efficiency with unified communications & collaboration.....................................37
Chapter 5 38
Step by Step Installation
Major configuation
Chapter 6 49
Conclusion 50
Overall analysis overview....................................................................................................50

3
 ACKNOWLEDGEMENT

Most importantly, acclaims and gratitude to the ALLAH the Almighty, for His showers of
endowments all through my exploration work to finish the examination effectively.

I might want to communicate my profound and earnest appreciation to my Project

Supervisor, Sir. Muhammad Sheraz Tariq, (Lecturer at Department of Computer Sciences),
at Minhaj University Lahore for allowing me the chance to do investigate and giving
significant direction all through this venture, notwithstanding of their bustling timetables. He
gave me various thoughts in making this task remarkable.

His dynamism, vision, and inspiration have profoundly motivated me. He has shown me the
philosophy to do the innovative work and to introduce the fills in as unmistakably as could be
expected under the circumstances. It was an incredible benefit and honor to work and study
under his direction. I am incredibly thankful for what he has offered me. I might likewise
want to express gratitude toward him for his kinship, compassion, and extraordinary comical
inclination.

I am incredibly appreciative to my folks for their affection, petitions, thinking about

instructing and setting me up for my future.

I thank the administration of Minhaj University for their help to accomplish this work. I
express gratitude toward Dr. Bilal Shoaib (HOD of Computer Sciences) at Minhaj
University Lahore for their certified help to complete this project effectively.

Student Name:

Asma Batool
Roll No: 03

4
 DEDICATED TO

My parents

I dedicated my project to my parents the strong souls who taught me to trust Allah and
believe in hard work. With whom motivation and guidance, I am able to perform this project
efforts and dedication.

My teachers

I also dedicated my project to my teachers the technical persons who taught e, the modern
knowledge of science and technology. The skills which may have given me will be very
beneficial for me in my professional career.

My fellows and juniors

It would be un justification not mentioned my fellows and my juniors of my department those

who helped me throughout my career of BSCS and my stay in university.

5
 Abstract

VoIP (Voice over Internet Protocol) telephone frameworks have been picking up prominence
in organizations as substitutions of existing PBX (Private Branch Exchanges). These IP put
together PBX frameworks typically depend with respect to the Ethernet structure promptly
accessible in a business office. The development of remote LAN (neighborhood) is likewise
increasing huge fame because of the extra portability given to the clients.

In this task, a PBX-style VoIP is executed and mimicked to certain focuses on account of
budgetary constraints, various items for the most part 3CX is utilized to reenact the earth in
any little, medium and large associations just as call focuses and execution factors were
investigated.

They are start to finish delay, defer jitter, and bundle misfortune. The variable boundaries of
the framework incorporated the speed of the remote organization, the kind of voice encoding
(G.711, G.723, G.729, and so forth.), and the quantity of stations in the organization.

6
 DECLARATION
Its stated that student of BS computer science session (S2016-2020) at Minhaj University
Lahore hereby declare that the matter printed in this documentation titles “Implementation
and Deployment of IP PBX” is my own work in the fulfill Meant of Bachelor Program
Under Minhaj University Lahore.
The supervision is provided by Head of Department Dr.Bilal Shoaib (Head of Department
CS) and guidance is provided by my project supervisor Mr. Muhammad Sheraz Tariq
under there very good supervision and command I am able to turn out this work.
The information and Data in this document is authentic and legitimated to the best of My
knowledge. I have performed this project with my own effort. I have mention the resources in
the reference list from where I have taken help and guidance regarding my project.

Name of student:
Asma Batool
Signature of candidate: —————————
Registration No: 2016s-mubscs-03

7
 CERTIFICATE

This is to certify that the research work contained in this titled “Implementation and
Deployment of IP PBX” has been carried out and completed by “Asma Batool” under
supervision of Muhammad Sheraz Tariq.
It is now my judgment that this project and this documentation is of sufficient standard to
warrant its acceptance by Minhaj University Lahore for BS degree in the subject of Computer
Science

Date: ——————————— Supervisor: ——————————————

Submitted Through: Lecturer (Computer Science)

Dr. Bilal Shoaib: ————————————

(Head, Faculty of CS)

8
 Chapter 1
Introduction To VOIP

9
1. Introduction
Voice over Internet Protocol (VoIP), is a technology that allows you to make voice calls
using a broadband Internet connection instead of a regular (or analog) phone line. Some VoIP
services may only allow you to call other people using the same service, but others may allow
you to call anyone who has a telephone number - including local, long distance, mobile, and
international numbers. Also, while some VoIP services only work over your computer or a
special VoIP phone, other services allow you to use a traditional phone connected to a VoIP
adapter.

VoIP services convert your voice into a digital signal that travels over the Internet. If you are
calling a regular phone number, the signal is converted to a regular telephone signal before it
reaches the destination. VoIP can allow you to make a call directly from a computer, a special
VoIP phone, or a traditional phone connected to a special adapter. In addition, wireless "hot
spots" in locations such as airports, parks, and cafes allow you to connect to the Internet and
may enable you to use VoIP service wirelessly.

A broadband (high speed Internet) connection is required. This can be through a cable
modem, or high speed services such as DSL or a local area network. A computer, adaptor, or
specialized phone is required. Some VoIP services only work over your computer or a special

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VoIP phone, while other services allow you to use a traditional phone connected to a VoIP
adapter. If you use your computer, you will need some software and an inexpensive
microphone. Special VoIP phones plug directly into your broadband connection and operate
largely like a traditional telephone. If you use a telephone with a VoIP adapter, you'll be able
to dial just as you always have, and the service provider may also provide a dial tone.

Some VoIP services offer features and services that are not available with a traditional phone,
or are available but only for an additional fee. You may also be able to avoid paying for both
a broadband connection and a traditional telephone line.

If you're considering replacing your traditional telephone service with VoIP, there are some
possible differences:

1.1. The Wireless LAN &VOIP

We can deploy VoIP on a wireless local area network (LAN) if you have one or if you plan to
set one up for communication, just like you can on a wired LAN. Wireless connections are
increasingly popular and have replaced many wired networks, particularly in residences. As a
result of this trend, more wireless networks are being used for VoIP communication.

Until relatively recently, LANs have been wired with RJ-45 jacks on an Ethernet network,
but with the advent of Wi-Fi and the steadily increasing speed it offers, network
administrators are leaning more toward wireless connection for their internal LANs. In most
cases, instead of a hub with wires that run to the different machines in a wired network, you
have a wireless router that connects to the machine's wireless adapter.

The caller, who may be using an IP phone or any other communicating device, such as a
PDA or smartphone, can make calls through the wireless LAN whenever the device is within
range of the wireless network. This is particularly handy for smartphone users who cannot
receive a cellular signal.

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Lower Costs
Cost savings is one of the benefits of VoIP that virtually any business can appreciate. You
can only install so many phone lines and costs quickly add up, especially if your business
regularly makes long-distance calls.

With communication data being modified into data packets and sent over the IP network, the
issue of a single phone line being able to be utilized by only two callers is eliminated. The IP
network could be a direct IP connection to your phone service provider or simply your
existing internet connect (or a combination of both).

Traditional phone lines typically charge for each minute of call time, where with VoIP your
only costs are your monthly charges from your ISP. In fact, many providers offer inexpensive
or even free calling to.

Cost-Effective Hardware and Software

One of the additional cost benefits of VoIP is the limited costs associated with hardware and
software required to operate the system. Quality providers ensure their clients always have
the most up-to-date software and provide current hardware. This eliminates the need for
businesses to purchase their own phones and infrastructure, which can result in additional
cost savings.

Simplified Conferencing
Without the need for dedicated phone lines, conferencing is simplified considerably.
Traditional phone systems allow for conferencing, but you’ll end up paying for an additional
service and hosting multiple callers each time you need to conference.

With a converged data network, these features are typically native and the cost is built into
the already lower price of the VoIP service that you’re already paying for.

An additional benefit of VoIP is that it makes video conferencing far simpler as well. In fact,
you can transfer various media formats (images, video, text) during your phone or video calls
to dramatically improve your ability to conduct presentations or solve issues on the fly.

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Worldwide Access
More employers are discovering the benefits of having their staff work from home in
exchange for smaller office spaces, decreased utilities costs, etc.

What they’re also discovering are the benefits of VoIP that allow their employees to
telecommute so effectively. VoIP allows employees to remotely utilize the voice, fax, and
data services of your office via your intranet.

VoIP technology has become extremely portable, allowing users to connect from home
offices and abroad. What’s more is that your employee’s number follows them to their new
home office when they make the change.

Mobility of Your Service

While telecommuting is one thing, one of the lesser mentioned benefits of VoIP is that the
entire service is highly mobile. Whereas traditional phone systems require a unique number
to be assigned to each line and transferring those numbers can be complicated, VoIP is
different. In the event that you outgrow your current office or need to change locations for
any reason, your VoIP system can be easily transferred.

Better Use of Bandwidth

One of the little known benefits of VoIP is that it makes for a more efficient use of your
existing bandwidth. As roughly half of voice conversations are made up of silence, VoIP
continues to fill those information gaps with other data from other bandwidth consumers to
make better use of your resources. What's more is that VoIP allows for compression and
elimination of speech redundancies to further improve efficiencies.

Extensive Additional Features

Many businesses don’t fully understand all of the benefits and additional features that are
included in a VoIP service. VoIP systems allow you to connect a wide variety of devices to
keep your business’ productivity high.

VoIP services typically include features like caller ID, virtual numbers, contact lists,
voicemail etc., but these features can all be used in more sophisticated ways to boost
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operational efficiency. For example, voicemails and messages can be forwarded to multiple
colleagues with a single click, and voicemail-to-text transcriptions can be sent directly to
your inbox so they can be reviewed while on the go.

Many features are included in various provider packages and, due to the flexible nature of the
service, custom VoIP services can be designed based on the unique needs of your business.

Network Flexibility
One of the benefits of VoIP that your IT team will enjoy is that its underlying network need
not be a part of a specific technology layout. That means your existing Ethernet, SONET,
ATM, or even your Wi-Fi can be used as the foundation for your network.

The complexity of PSTN (traditional) phone networks is virtually eliminated. This allows for
a more standardized system to be implemented that supports a variety of communication
types while being more tolerant of faults and requiring less management of equipment.

Fax over IP
One of the additional benefits of VoIP is that most providers include Fax over IP as a part of
their service. Fax over IP all but eliminates the high costs of long-distance facsimile, as well
as improves compatibility between machines and reliability of service.

Once again, fax information is transmitted via data packets that dramatically improve
efficiency. In fact, VoIP doesn’t even require a fax machine to send or receive a fax.

More Effective Communication

With your personnel working from various points within the office, their home, or around the
world, keeping them within reach is critical. One of the more interesting benefits of VoIP is
that you can have a single call ring to your desk phone for the first few rings, then to your
mobile phone, tablet, or laptop if the call goes unanswered. This way urgent calls are
answered more often and less time is spent checking voicemail or corresponding over other
platforms.

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Highly Reliable
One of the most common (and inaccurate) objections to VoIP is that if a business finds
themselves without internet for whatever reason, they’d be without phone as well. One of the
benefits of VoIP flexibility is that in the event of an office phone going down due to lack of
network, calls can always be forwarded to mobile phones and other devices. That also means
weather issues and power outages no longer present the risk they once did.

Ease of Installation, Configuration, Maintenance

One of our favorite benefits of VoIP is that IP phones are incredibly easy to install—even for
those who are less technically savvy. There is no need to have expert technicians running
phone wiring throughout your office. Instead, IP phones are virtually plug-and-play.

Hosted VoIP software also makes it incredibly simple to add new users, and a web portal
makes moving, adding, or changing your systems configuration much easier. All of this
simplicity means maintenance is straightforward and rarely requires professional support.

Scalability
Highly efficient business systems scale with the needs of the business, but traditional phones
systems are far more difficult to scale. Scalability is one of the benefits of VoIP that supports
your efficiency and productivity while remaining highly cost effective at the same time.

VoIP systems allow you to add a line as you hire a new employee and eliminate lines in the
case of downsizing. You’re only ever paying for what you need.

Easy Integration with Other Business Systems

Your business likely utilizes various other systems and technologies to enhance your
operational efficiencies. VoIP easily integrates with a wide variety of existing business
systems.

That means you’ll realize all of the benefits of VoIP without requiring modification of your
existing applications or IT infrastructure. For example, outbound calls can be placed via

15
Outlook or other email systems and customer records can even be viewed during the inbound
call with said customer.

1.1.1. Why A Wireless LAN

The main idea behind going wireless is mobility. The ability to make a call from anywhere
you can access a wireless network connection is convenient and productive. Consider a few
scenarios:

● A medical team in a clinic needs to be able to communicate internally and externally

while attending to emergencies, which implies being on the move. VoIP on a wireless
LAN makes this possible for each responder who has a smartphone.
● A factory floor team finds it difficult to either remain glued to a fixed phone set or
going to and from one for communication. Here again, VoIP service deployed on a
wireless LAN within the company premises saves time, energy and nerves, while
boosting productivity.
● VoIP at Wi-Fi hotspots is incredibly convenient. Just like you take your laptop
computer along with you for a business lunch or a study group with classmates, you
can take an IP phone or your smartphone along.

1.1.2. Problems with Wireless VoIP

There are reasons wireless VoIP is not readily accepted everywhere:

● VoIP on LANs is deployed mostly in corporate environments—in companies rather

than houses. Wireless VoIP poses problems of scalability for enterprises.
● As is the case with nearly all wireless networks, the quality of service is not as good
as with wired networks, although that is rapidly improving.
● The cost of setting up and maintaining a wireless network, in terms of money, time,
and skills, is higher than to set up and maintain a wired network.
● The security threats posed by the use of VoIP are even more inherent over a wireless
network because access points are more numerous within the perimeter of the
network.

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17
In the traditional PSTN telephony network,Chapter 2 required to complete a call are
all the elements
VOIP
transparent to an end user. Migration Fundamentals
to VoIP requires an awareness of these required
elements and a thorough understanding of the protocols and components that provide the
same functionality in an IP network.

Required VoIP functionality includes these functions

Signaling

Signaling is the capability to generate and exchange control information that will be used to
establish, monitor, and release connections between two end-points. Voice signaling requires
the capability to provide supervisory, address, and alerting functionality between nodes. The
PSTN network uses Signaling System 7 (SS7) to transport control messages. SS7 uses out-of-
band signaling, which, in this case, is the exchange of call control information in a separate
dedicated channel.

VOIP Options

VoIP presents several options for signaling, including H.323, Session Initiation Protocol
(SIP), H.248, Media Gateway Control Protocol (MGCP), and Skinny Client Control Protocol
(SCCP). Some VoIP gateways are also capable of initiating SS7 signaling directly to the
PSTN network Signaling protocols are classified as either peer-to-peer or client/server
protocols.

SIP and H.323

SIP and H.323 are examples of peer-to-peer signaling protocols where the end devices or
gateways contain the intelligence to initiate and terminate calls and interpret call control
messages. H.248, SCCP, and MGCP are examples of client/server protocols where the
endpoints or gateways do not contain call control intelligence but send or receive event
notifications to a server commonly referred to as a call agent. For example, when an MGCP
gateway detects a telephone that has gone off hook, it does not know to automatically provide
a dial tone. The gateway sends an event notification to the call agent, telling the agent that an
off-hook condition has been detected. The call agent notifies the gateway to provide a dial
tone

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Access to services, such as toll-free numbers or caller ID, requires the capability to query a
database to determine whether the call can be placed or information can be made available.

Database services

Database services include access to billing information, caller name delivery (CNAM), toll-
free database services, and calling-card services. VoIP service providers can differentiate
their services by providing access to many unique database services. For example, to simplify
fax access to mobile users, a provider can build a service that converts fax to e-mail. Another
example is providing a call notification service that places outbound calls with prerecorded
messages at specific times to notify users of such events as school closures, wake-up calls, or
appointments.

Bearer control

Bearer channels are the channels that carry voice calls. Proper supervision of these channels
requires that appropriate call connect and call disconnect signaling be passed between end
devices. Correct signaling ensures that the channel is allocated to the current voice call and
that a channel is properly deallocated when either side terminates the call. Connect and
disconnect messages are carried by SS7 in the PSTN network. Connect and disconnect
message are carried by SIP, H.323, H.248, or MGCP within the IP network.

Codecs

Codecs provide the coding and decoding translation between analog and digital facilities.
Each codec type defines the method of voice coding and the compression mechanism that is
used to convert the voice stream. The PSTN uses TDM to carry each voice call. Each voice
channel reserves 64 kbps of bandwidth and uses the G.711 codec to convert an analog voice
wave to a 64-kbps digitized voice stream. In VoIP design, codecs might compress voice
beyond the 64-kbps voice stream to allow more efficient use of network resources. The most
widely used codec in the WAN environment is G.729, which compresses the voice stream to
8 kbps.

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2. VoIP Signaling Protocols
VoIP uses several control and call-signaling protocols. Among these are:

H.323

H.323 is a standard that specifies the components, protocols, and procedures that provide
multimedia communication services, real-time audio, video, and data communications over
packet networks, including IP networks. H.323 is part of a family of International
Telecommunication Union Telecommunication Standardization sector (ITU-T)
recommendations called H.32x that provides multimedia communication services over a
variety of networks. H.32x is an umbrella of standards that define all aspects of synchronized
voice, video, and data transmission. It also defines end-to-end call signaling.

MGCP

MGCP is a method for PSTN gateway control or thin device control. Specified in RFC 2705,
MGCP defines a protocol that controls VoIP gateways that are connected to external call
control devices, referred to as call agents. MGCP provides the signaling capability for less-
expensive edge devices, such as gateways, that might not have implemented a full voice-
signaling protocol such as H.323. For example, anytime an event, such as off-hook, occurs on
a voice port of a gateway, the voice port reports that event to the call agent. The call agent
then signals the voice port to provide a service, such as dial-tone signaling.

SIP

SIP is a detailed protocol that specifies the commands and responses to set up and tear down
calls. SIP also details features such as security, proxy, and transport control protocol (TCP) or
User Datagram Protocol (UDP) services. SIP and its partner protocols, Session
Announcement Protocol (SAP) and Session Description Protocol (SDP), provide
announcements and information about multicast sessions to users on a network. SIP defines
end-to-end call signaling between devices. SIP is a text-based protocol that borrows many
elements of HTTP, using the same transaction request and response model and similar header

20
and response codes. It also adopts a modified form of the URL addressing scheme used
within e-mail that is based on Simple Mail Transfer Protocol (SMTP).

SCCP

SCCP is a Cisco proprietary protocol used between Cisco Communications Manager and
Cisco IP Phones. The end stations (telephones) that use SCCP are called Skinny clients,
which consume less processing overhead. The client communicates with the Cisco Unified
Communications Manager (often referred to as Call Manager, abbreviated UCM) using
connection-oriented (TCP-based) communication to establish a call with another H.323-
compliant end station.

2.1 Components of VOIP Network

The following is a description of these basic components:

2.1.1. IP Phones:

Cisco IP Phones provide IP endpoints for voice communication.

2.1.2. Gatekeeper:

A gatekeeper provides Call Admission Control (CAC), bandwidth control and management,
and address translation.

2.1.3. Gateway:

The gateway provides translation between VoIP and non-VoIP networks, such as the PSTN.
Gateways also provide physical access for local analog and digital voice devices, such as
telephones, fax machines, key sets, and private branch exchanges (PBX).

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 2.1.4. Multipoint Control Unit (MCU):

An MCU provides real-time connectivity for participants in multiple locations to attend the
same videoconference or meeting.

2.1.5. Call agent:

A call agent provides call control for IP phones, CAC, bandwidth control and management,
and address translation. Unlike a gatekeeper, which in a Cisco environment typically runs on
a router, a call agent typically runs on a server platform. Cisco Unified Communications
Manager is an example of a call agent.

2.1.6. Application servers:

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Application servers provide services such as voice mail, unified messaging, and Cisco
Communications Manager Attendant Console.

2.1.7. Videoconference station:

A videoconference station provides access for end-user participation in videoconferencing.

The videoconference station contains a video capture device for video input and a
microphone for audio input. A user can view video streams and hear audio that originates at a
remote user station.

2.2 Types of VOIP

There are two type of VOIP systems both have their own edges and drawbacks and their
detailed discussion and functions are mentioned below

There are many points to consider when making a business communications upgrade and it’s
probably a good idea to start with whether to go hosted or on-premise. This depends on a few
factors; the size of the company, existing infrastructure, available budget, management
resources, and what they wish to gain from their PBX.

To make a decision, it’s important to understand the differences between the two and the
benefits that each option could provide.

2.2.1. Hosted PBX

A hosted PBX allows you to retain control of your phone system whilst remotely hosting the
software from either the vendor or a third-party hosting provider. This is a great option for
smaller companies who may not have the infrastructure available but still want to manage
their communications.

● Host your PBX in your cloud account with the likes of Google, Amazon and
Microsoft Azure or have it hosted by 3CX
● Maintenance, operation and installation costs are reduced and possibly even avoided.
● You get to choose your hosting provider and SIP trunk provider for a tailored solution
that fits your needs and budget.

23
Maintenance and upgrades all taken care of. This is the preferred deployment for SMBs that
have limited IT resources.

● A third party provider handles the PBX and all responsibility of running and
upgrading the Hosted PBX is shifted onto them.

2.2.2. Cloud PBX

Cloud and hosted are often used interchangeably but when talking about specific
deployment options, Cloud refers to a managed service. With a Software-as-a-Service
(SaaS) subscription, companies can go hosted with their communications system and
have the management,
● There’s no need to worry about any network issues, like bandwidth, with a Cloud
PBX; especially important for SMB’s that don’t have the bandwidth to accommodate
Unified Communications and VoIP.
● Dedicated personnel to manage the phone system are not required; less costs.
● No training required on how to run the phone system, how to add extensions and so
on. The PBX provider does all of that for the end user.
● All upgrades are included in the maintenance costs and are automatically done by the
PBX provider.

2.2.3. On-Premise PBX

An on-premise PBX is deployed on servers belonging to the business, and thus is managed by
them entirely if they so wish. For this type of installation, the company must have in place the
appropriate infrastructure, including servers, network, devices and so on, or they must factor
this into their budget. On-premise PBXs are more suited to larger enterprises that have the
infrastructure and resources to run and manage the phone system, and are possibly in sectors
that require strict security and confidentiality.

CLOUD BASED

ON PREMISES
24
25
 Chapter 3
VOIP Quality of service

26
Overview

The quality of a VoIP call is heavily dependent on the network environment. Factors include
the device the client is running on, the network characteristics and firewall/router
configuration and more. A VoIP deployment requires careful consideration of the end to end
experience. This document is intended to share the best practices in configuring and selecting
the best environment for VoIP calling.

Local network conditions have the biggest impact on voice quality. Packet loss, most
frequently jitter-induced packet loss can cause the biggest impact. WiFi can be particularly
bad for creating jitter.

Callers start to notice the effect of latency around 250ms, above ~600ms the experience is
unusable. There will always be some latency, the objective is to minimize it and keep total
trip time well below 250ms.

Ideally latency should be below 100ms because while it's noticable at 250ms, other services
and issues beyond your control might add delay causing the cumulative total to be over
250ms.

27
3. QoS
QoS (Quality of Service) is a significant issue in VOIP executions. The issue is the way to
ensure that packet traffic for a voice or other media association won't be deferred or dropped
because of obstruction from other lower need traffic.

we are currently working in Timeliness category and its parameters are mentioned below

3.1. Latency

Latency is the time it takes the RTP (media) packets to traverse the network. Too much
latency causes callers to speak over the top of each other.

3.2. Packet loss

Packet loss is very common in IP networks, but certain networks such as WiFi can be
particularly prone to high levels of packet loss. This causes sections of media to be missing,
and can cause the ‘robot’ distortion effect of media.

3.3. Jitter

Jitter is when packets don’t arrive in the same order they were sent. For small amounts of
jitter, this can be resolved in the jitter buffer – a queue of media packets waiting to be played
which can be shuffled into the correct order while they wait in the queue. The length of the
jitter buffer introduced must be traded off against the impact of increased latency. Too much
jitter cannot be resolved by a reasonable length jitter buffer without introducing too much
delay, so instead results in jitter induced packet loss causing choppy audio.

Latency and Delay are similar terms that refer to the amount of time it takes a bit to be
transmitted from source to destination. Jitter is Delay that varies over time or when packets
don’t arrive in the same order they were sent.

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3.2 VOIP QoS Requirment

Callers usually notice roundtrip voice delays of 250ms or more. ITU-T G.114 recommends a maximum of
a 150 ms one-way latency. Since this includes the entire voice path, part of which may be on the public
Internet, your own network should have transit latencies of considerably less than 150ms.

Jitter can be measured in several ways. There are jitter measurement calculations defined in:

But, equipment and network vendors often don’t detail exactly how they are calculating the
values they report for measured jitter. Most VOIP endpoint devices (e.g. VOIP Phones and
ATAs) have jitter buffers to compensate for network jitter.:

Jitter buffers (used to compensate for varying delay) further add to the end-to-end delay, and
are usually only effective on delay variations less than 100ms. Jitter must, therefore, be
minimized.

VOIP is not tolerant of packet loss. Even 1% packet loss can “significantly degrade” a VOIP
call using a G.711 codec and other more compressing codecs can tolerate even less packet
loss.

The default G.729 codec requires packet loss far less than 1 percent to avoid audible errors.
Ideally, there should be no packet loss for VoIP

Parameters Average Quality Ideal Quality

End-to-End Delay <150ms <50ms

Delay-Jitter <60ms <20ms

Packet Loss Rate <5% <1%

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Hosted VOIP network

This approach is known as a cloud phone system or a “cloud PBX.” Administrators can
manage the permissions and features for each employee along with more sophisticated VoIP
features with an online interface.

When an employee calls a customer, they pick up the handset and dial them just as they
normally would. The IP phone (or app) travels through your Local Area Network (LAN)
switch and business router before reaching the VoIP service provider. From there, the VoIP
provider establishes the call.

If the network path to the called party supports a digital voice signal, then the call quality is
upgraded to high definition. Otherwise, a VoIP provider connects the call over the Public
Switched Telephone Network (PSTN).

Using a hosted VoIP system in your business is that simple. However, for established
businesses with a more sophisticated phone system, there are different needs.

01I0f0 the office uses a PBX, you probably also pay for trunked telephone lines. These
trunked lines handle voice calls from the PBX to the phone company—and they’re not cheap

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SIP Trunking for PBX Phone Systems

In some cases, companies are locked into their hybrid phone system due to the costs to
replace hardwired phone lines in every office. Since a VoIP uses the Session Initiation
Protocol, it can be used to establish multiple lines of calls through the internet.

You’ll see in the diagram below SIP trunks accept calls from the VoIP provider. Inbound and
outbound calls are funneled in much the same way until it reaches the business location and is
hardwired into a PBX. Like old school analog circuit-switches, the “trunk” acts as a switch to
control and funnel data.

The most significant benefit for large and small businesses alike is if they have a hardwired
telephone system on site. SIP Trunking allows you to mix analog phone systems and new
VoIP solutions to eliminate redundancy.

SIP Trunking provides multiple channels of voice service on-demand for any IP-based PBX.
A SIP Trunk can either be metered or unmetered for $15–$25 each month. They aren’t too
hard to set up, either. Simply provide the SIP username and password into your PBX.

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4. 3cx

3CX is a global leader in business VoIP and Unified Communications (UC) technology. It
offers customers a simple, flexible and affordable solution that dramatically cuts telephony
costs and management headaches. With 3CX you are guaranteed to increase productivity,
reduce business travel and telco costs, streamline operations and improve customer service.

4.1. Why 3CX

Cost saving
Highlight the cost savings with 3CX! They will be remarkable. Switching from a traditional
PBX to an IP PBX will already deliver significant call costs savings. But this will be
highlighted by all contenders for the system. Stress the low cost of 3CX – compared to other
traditional PBX vendors and hosted systems in the market today. You will be up against other
hosted systems – you might as well preempt and show how 3CX is much more cost effective.

● Unlimited Users
● Choose your own SIP Trunk
● FREE for unlimited extensions
● Low annual license price – starts at $2 per user per month!
● Slash your phone bill: Employees on the road/at home call free.
● Eliminate Interoffice Calls. Leverage low cost SIP Trunk pricing

Open platform
3CX does not tie your customer into particular SIP Trunks, phones or hosters. With 3CX, the
customer stays in control of their system and data.

It’s all about freedom of choice!

● Choose & Control your own phone numbers.

● Choose your own SIP Trunk and save.
● Choose your own cloud provider.
● Choose your own hardware.

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Easy to Deploy
Easy to Deploy Run 3CX On-premise or in the Cloud – whatever makes most sense for the
customer. Importantly, your customers can choose a Hosted by 3CX solution and benefit
from 1-year free hosting; or have it deployed in your cloud account on Google, Azure or
Amazon. Larger companies might want to manage it themselves in which case it’s a selling
point that

you can offer.

● Self-Host or save the hassle with Hosted by 3CX.

● Install On-Premise on Mini PC or Virtualize on an existing server.
● Linux or Windows.
● Auto configuration of phones & SIP Trunks.

Easy to Manage
3CX’s easy and quick deployment and simple day to day management is a huge selling
feature. Not only does 3CX make running a PBX easy in terms of upgrades, security and
scalability, but its management interface makes these tasks and the addition of new
extensions a walk in the park. 3CX’s zero admin will be a time, and therefore a cost, saver for
you and your customer.

Easily Add & Manage Extensions.

● Phone configuration and firmware are tested by 3CX.

● 3CX can manage bundled OS as well.
● 3CX Updates and Upgrades can be deployed automatically.
● Choose Hosted By 3CX to further reduce admin and maintenance.

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Mobility
3CX has the full range of features that modern businesses require today. Integrated video
conferencing at no charge, Mobility Apps for iOS, Android, Windows Client and modern
web client will deliver tangible productivity gains for your customers.

● Modern Android and iOS apps

● Web and Windows Communications Client
● Corporate Chat
● Browser based video conferencing for all users
● All at no extra charge

Advanced Customer Service functions

In today’s competitive market it’s important to never miss that call. 3CX includes advanced
call center features previously only available to enterprise customers. Calls can be linked to
customer records before the agent takes the call. 3CX integrates with the largest number of
CRM and Helpdesk packages.

● Call Queues, IVR

● Advanced Call Reporting
● Integration with leading CRM apps
● Contact Centre (including website customer service chat)

Tried and Tested

3CX is a tried and tested solution. With thousands of businesses using 3CX globally the
numbers talk for themselves.

● 250,000 installations
● 25,000 partners
● Leading customers including Wilson Sporting Goods, Mitsubishi, American Express,
MIT and Subaru of America.
● Supported globally by 3CX 24/5 and from 12 offices worldwide

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Open-standards, software communications

3CX is open-standards, which makes it much more flexible in terms of available options
when building a solution that suits your business. Choose your IP phones, SIP trunks and
hosting provider, and integrate with a wide range of applications such as CRM and
accounting software. With 3CX you can tailor a solution that suits your needs and budget.

● Deploy on-premise on Windows or Linux

o On standard server hardware or miniPC appliance
o Virtualize with VMware, Hyper V or KVM
● Run in the cloud in your own cloud account
o Google, Amazon, Azure, OVH and more
● Hosted by 3CX – Let us take care of hosting, maintenance and updates
● No need for expensive and difficult to maintain proprietary hardware

Save with affordable & transparent pricing

3CX offers three license options, three sets of features and zero add-ons. Simply select the
edition based on the features you need, and the license size based on the number of
simultaneous calls NOT per extension. There are no add-on modules and no need to purchase
separate licenses for each user, making it easy and cost-effective to scale.

● Choose from Standard, Pro & Enterprise

● No extra licensing for apps, web conferencing, contact center or live chat
● Unlimited extensions, just choose your call capacity
● Easily upgrade if you outgrow your license

Easy to get started, easy to maintain

Setup takes minutes whether on-premise with the download wizard or in the cloud with our
PBX Express tool. Once your installation is ready, use the web-based management console to
easily configure the settings of your PBX. Create extensions, set up your SIP trunk provider
from one of the ready-to-go templates and more in just a few clicks. What’s more, the

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intuitive dashboard displays the status of critical services while automated management tasks
are carried out to lighten the load.

● Pre-configured templates for supported IP phones, SIP trunks and gateways

● Perform network-wide firmware updates in seconds
● Easily and remotely reprovision or reboot IP phones
● Simple backup and restore for your peace of mind
● Inbuilt failover in Enterprise License

Increase efficiency with unified communications & collaboration

Improve business performance and enable employees to work more productively with user-
friendly communication and collaboration features. With integrated web conferencing, apps
for iOS, Android, Windows and the web, live customer chat, instant messaging, switchboard
and more, your employees have everything they need to effectively communicate externally
and internally within your organization

● See the status of colleagues

● Quick and easy audio and video conferencing
● Instant messaging and file sharing
● Use your extension anywhere with apps and web client

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5. Step by Step installation
Installation and Configuration steps are mentioned below briefly

Step 1

Sign up to get your installation file by verifying your email

Step 2

Choose your company name that will be mentioned in your external link

Step 3

Choose your platform to get your required setup file from customer portal

Step 4

Run downloaded file by u=administrator rights

Step 5

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Step 6

Step 7

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Step 8

Step 9

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Major Configurations

Step 10

Step 11

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Step 12

Step 13

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Step 14

Step 15

44
Step 16

Step 17

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Step 18

Step 19

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Step 20

Step 21

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Step 22

Result:

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49
6. Conclusion

to conclude performance of our PBX system I've sampled different calls and analyzed their
transmission via wire shark so that I can examine how they travel and their effect as well as
parameters fluctuations.

There is a whole analysis and an individual graph that clearly shows jitter and difference in
arrival of a voice packet.

6.1Overall analysis overview

Forward jitter
analysis

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Reverse jitter analysis

Forward difference in packet arrival

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Reverse difference in packet
arrival

However,

there is another analysis phase where we are using wire as a medium a CAT6 cable to
connect two PC's to communicate via standard modem and will deconstruct every aspect
independently whether our PBX performance is up to par or not.

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Call no 1
sample:

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call no 2
Sample

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Call no 3
Sample

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Call no 4
Sample

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Call no 5
Sample

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Conclusion

As per standards the jitter should be between 50 to 250ms and it lies between them actually
very low which is a good indication to a certain levels there is no such traffic on the network
when analyzing this procedure documented above.

Traffic will affect the overall performance.

Data loss is as low as 0 percent which is great.

Voice quality is greater than expected as my bandwidth is 16mb/s on a shared line.

Voice transmission is smooth and less echo included in it.

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References

https://www.voipbusiness.com/blog-post/what-is-a-hosted-pbx/
https://www.voiplid.com/what-is-sip-trunk-and-pbx-sip-trunking/
https://www.shutterstock.com/license

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