Implementation and Deployment of Ip PBX (3CX) : BS Computer Science Session 2016-2020 (Spring) Submitted To [PDF]

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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

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Introduction

10

The Wireless LAN &VOIP

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Why A Wireless LAN..........................................................................................................16 Problems with Wireless VoIP..............................................................................................16 Chapter 2

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Signaling:.............................................................................................................................18 VOIP Options:......................................................................................................................18 SIP and H.323......................................................................................................................18 Database services:................................................................................................................19 Bearer control:......................................................................................................................19 Codecs:.................................................................................................................................19 VoIP Signaling Protocols

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H.323....................................................................................................................................20 MGCP..................................................................................................................................20 SIP........................................................................................................................................20 SCCP....................................................................................................................................21 Components of a VoIP Network

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Types of VOIP

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Hosted PBX..........................................................................................................................23 Cloud PBX...........................................................................................................................24 On-Premise PBX..................................................................................................................24 Chapter 3

26

Overview

27

QoS

28

VOIP QoS Requirment

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Hosted VOIP network..........................................................................................................30 Chapter 4

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3cx

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Why 3CX

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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

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Step by Step Installation Major configuation Chapter 6

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Conclusion

50

Overall analysis overview....................................................................................................50

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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

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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.

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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.

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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

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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)

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Chapter 1 Introduction To VOIP

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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 10

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 13

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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2 required to complete a call are In the traditional PSTN telephony network,Chapter all the elements VOIP Fundamentals transparent to an end user. Migration 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-ofband 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 18

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), tollfree 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 lessexpensive edge devices, such as gateways, that might not have implemented a full voicesignaling 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

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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.323compliant 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.

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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

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Chapter 3 VOIP Quality of service

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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.

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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