call centre architecture

Call Centre

Architecture

Summer Project in Samsung Contact Centre

9/9/2014Created By APOORVA TYAGI

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AcknowledgementI am thankful to Mr. Sandeep Kaul, for providing necessary facility to carry out my

training successfully.

I like to take this opportunity to show my gratitude towards him who helped me in bringing the project to its present form. He has been a motivator & source of

inspiration for me to carry out the necessary proceeding for the project to be

completed successfully.

I am highly obliged to my family for constant help and encouragement. They

helped me a lot during training period and for successfully completing my report.

Finally I would like to take this opportunity to thank the organization Samsung who

helped me get proper knowledge and success in my training.

Mr. Sandeep Kaul9/9/2014Created By APOORVA TYAGI

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OverviewCall Centre applications are an integral part of almost any business – from

sophisticated call centres down to the smallest of operations.

A call centre is a complex integration of computers, human

operators(agents), telephone and packet network and their equipment.

The fully integrated call centre application enables the rapid delivery of

full-featured hosted call centres, meeting the needs of the most

sophisticated enterprise customers and providing new revenue

generating services.

Call Centre solutions allow service providers to not only differentiate

themselves in a crowded market, but benefit from recurring high margin

services.

Call Centres can be classified into three types, depending of whether they

handle inbound traffic, outbound traffic, or both inbound and outbound

traffic, and their scale can vary from a few thousands of agents.


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Types of Call Centre

Software

• Automatic Call Distributor (ACD)

• Computer Telephony Integration (CTI)

• Interactive Voice Response (IVR)

• Predictive Dialling

• Call Centre Monitoring

• Call Accounting Software

• Call Analytics


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ACD helps productivity by assigning inbound agents to incoming call.

The automatic call distributor uses a set of instructions to determine who

gets the call in the system. The algorithm can route calls based on agent

skill or whoever has an idle phone. ACD can use caller ID or automatic

number identification, but usually interactive voice response is enough to

help the system determine the reason for the call. An automatic call

distributor can also take advantage of computer telephony integration. Agents can receive relevant data on their computers along with the

incoming call.

CTI is a broad category of software that connects telephone and

computer systems. CTI software can have both desktop and server

functions. Various applications make up a system that can control phones, display call information, and route an report calls.


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G650 Gateways

Enterprise LAN

Control Network

PSTN

PRI’s

NICE CLS

Logger

Main AES

S 8800 Media servers CM 6

Main CMS

HA AES

PN #3 (8 PRI)

Duplicate power supply, MedPro and IPSI

Duplicate AIC

Telephony server

HA CMS

PN #1

8 PRI)

PN #2

(8 PRI)Port Net Work Setup

Call Centre Setup9/9/2014Created By APOORVA TYAGI

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Avaya

Avaya is a global provider of business collaboration and

communications solutions, providing unified communications, contact

centres, data solutions and related services to companies of all sizes

around the world. For more information please visit www.avaya.com.

All Avaya products are easy to integrate

Avaya CMS Supervisor supports your existing TCP/IP Ethernet LAN

connections for access to the CMS server Avaya CMS Supervisor can be

loaded onto your server and downloaded across the LAN to your

networked PCs. Instantly access your critical data anytime… anywhere.


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http://www.avaya.com/


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Avaya Call Management

SystemAvaya Call Management System (CMS) is an integrated analysis and

reporting solution to help you keep in touch with virtually everything that’s

going on in your contact centre — whether you want to evaluate the

performance of one agent, a group of agents, a single contact centre, or

multiple locations around the world. CMS provides robust real- timemonitoring and historical reporting, including custom reporting, task

scheduling, exception notification, threshold warning, administration and

configuration, and long term ACD data storage, working with one or more

of your Avaya Media Servers and Gateways.


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Call Flow In Avaya

Avaya IR system-to-agent transfers are accomplished by using the A_Tranexternal function within a voice application that is servicing a caller. The

use of A_Tran invokes ASAI Third Party Call Control operations to transfer a

call away from the telephony channel to which the caller is connected.

The caller is transferred to the destination that is identified in

the Destination Number field of the A_Tran external function.

The transferred call can be monitored by a monitoring application so that

data screen delivery applications can be supported for Avaya IR system-

to-agent transfers. The transferred call can be monitored in two different

ways:


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The call can be transferred to a VDN or ACD split domain that is

monitored by the Avaya IR system with a monitoring application. Call

events for the transferred call are passed to the application that is

monitoring the domain to which the call is transferred.

The call can be monitored using a CTL type monitoring application. In this

case, the call can be transferred to no monitored domains and individual

stations. Here, only call events for calls that are transferred from theAvaya IR system to agents are passed to monitoring applications. Other

direct calls to an ACD split, for example, are not monitored. Therefore, no

call events for the direct calls are passed to monitoring applications.


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Avaya Voice Solutions

OverviewAvaya’s open standards-based platform accommodates customers with

multi-vendor environments seeking to use their existing investments,

supplement their existing solutions with specific collaboration products

they need, and rapidly create and deploy applications. Our solutions

allow organizations to develop short- and long-term IT strategies and

deploy them at their own pace. Regardless of whether your business

consists of 10 employees or 100,000, we can help you to take all your forms

of communication and get them working together, to dramatically

improve collaboration and accelerate growth. Your company can be

more spontaneous and intuitive in communications, more customer-friendly, and ultimately more profitable.


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Avaya focuses on the following businesses :

• Unified Communications

• Contact Centres

• Small and Medium Enterprises

• Networking

• Avaya Client Services


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Voice Solution Offerings

• Traditional TDM-based PBXs with IP capability. These are known as the

Definity® range of products.

• IP-based solutions. These solutions migrate the call signaling and other

signaling information into IP networks. They are called Enterprise Class IP

Solutions (ECLIPS). They are built on server-gateway architecture, and

include the S8700, S8500 and S8300 Media Servers, as well as the SCC1,

MCC1, G350, G600, G650, G700 Media Gateways.


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Traditional Offerings

Definity G3r with MCC1

Gateways

Definity CSI (Prologue) with CMC1

gateways


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Newest Additions to the

Family of Solutions

The G350 Media Gateway with/without a S8300

Media Server

The S8500 Simplex Media Server

The G650 rack-mountable Media Gateway 9/9/2014Created By APOORVA TYAGI

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Avaya Definity Solutions

OverviewThe Avaya Definity telephone system is the flagship of the Avaya telephone system range. Applications such as CTI, Voice over IP & Call Centre solutions are commonplace. The system uses the Avaya 6400 series system phones.

We also supply Avaya Definity headset solutions.

Listed below is the are the Avaya Definity Telephones & Avaya Definity Cards & Accessories.

We have not listed the Avaya Definity telephone system themselves as they can be configured in many different ways. It is simpler if you email or call us with your system requirements. We can then come back to you with appropriate suggestions.


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Definity OverviewBasic System Components

• Switch Processing Element (SPE) or Processor Card

• Architecture

Cabinets and Carriers

• Compact Modular Cabinets (CMC1)

• Single Carrier Cabinets (SCC1)

• Multi Carrier Cabinets (MCC1) – 1 to 5 carriers

System Configurations

• Direct Connect Systems.

• Systems Connected using Center Stage Switch (CSS)9/9/2014Created By APOORVA TYAGI

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Concept of Port NetworksIn Definity, a Port Network is nothing but a stack of cabinets withextension and/or trunk termination port circuit packs. The name PortNetwork comes

from the fact that these port circuit packs provide a bank of physicalcircuits

onto which extensions and/or trunks can be terminated. There are 2 types

of Port Networks (PNs)

• Processor Port Networks :- These are the PNs that contain theprocessor card (SPE) and other control cards in addition to port cards.The SPE is a computer that handles call control and controls other PNs.

• Expansion Port Networks :- These carry additional port circuitpacks to increase the capacity of the PNs.


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The above PNs can be interconnected via fiber as• Direct connect :-Direct fiber connection between PNs. Applicable

when number of PNs is less than or equal to 3.• CSS connect :- Connection using Center Stage Switch. Necessary

when number of PNs exceeds 3.

Additionally, ATM connectivity (ATM-PNC) option is also there.


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Basic System Components• Processor Port Networks (PPN)

• Expansion Port Networks (EPNs)

• Center Stage Switch (CSS)

• Switch Processing Element (SPE) :- It contains several components

connected by a processor bus. They include a RISC processor (TN2404

for G3si, TN2402 for G3csi and UN331C for G3r), Memory (32 MB Flash

ROM and 32 MB DRAM for si/csi, and 4 TN1650B Memory circuit packs

to provide 128 MB memory in G3r), Storage (PCMCIA Flash in si/csi,

while r has separate storage drive with an optional optical backup

drive), I/O circuits acting as interfaces to the TDM and Packet buses,

and a Maintenance Interface (connects system to an administrativeterminal, monitors power failure, clock signals and temperature

sensors)


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Single Carrier Cabinets

(SCC)• This cabinet can be used both for PPN and EPN

• High and Critical Reliability is possible with this cabinet.

• This cabinet is used in G3si and G3r systems.

• Is a floor-mountable cabinet with universal slots

• SCCs can be stacked at a location. Up to 4 SCCs can be stacked

together at one location.

• In a particular stack, cabinet positions are labeled “A” through “D”.

• Lowest cabinet of the stack is called “A”. Subsequent cabinets above

it are called “B”, “C” and “D”.


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Port Network Connectivity

(PNC) Options


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– Provides an alternative to CSS. Supported only from R8

systems onwards. This PNC integrates delivery of voice, video and data via ATM over a converged large-bandwidth network. Uses standards

based open interfaces.


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Types Of Circuit PackFour types of circuit packs are installed in carriers:

1. Port circuit packs

•Provide links between analog and digital lines, trunks, networks

external communications equipment, and the TDM bus and packet

bus.

•These circuit packs install in any purple port slot.

•Form analog/digital interfaces between the PN and external trunks

and devices providing links between these devices and the TDM bus and packet bus.


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•Incoming analog signals are converted to pulse-code modulated

(PCM) digital signals and placed on the TDM bus by port circuits. Port

circuits convert outgoing signals from PCM to analog for external

analog devices. All port circuits connect to the TDM bus. Only specific

ports connect to the packet bus.

2. Control circuit packsInclude processor, memory, network control, disk control, tape control,

protocol interfaces, duplication, and maintenance.

• These circuit packs install in dedicated white slots in the control

carrier and do not operate in any other slots.


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3. Service circuit packs • Produce and detect tones, synthesize speech, classify calls, record

announcements, and allow system access for administration and

troubleshooting.

• Connect to an external terminal to monitor, maintain, and troubleshoot

the system.

• Also provide tone production and detection as well as call classification, modem pooling, recorded announcements, and speech synthesis.


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Port Circuit Packs provide physical termination circuits for termination of

trunks or extensions, adjunct links, LAN interfacing. Following are some of

the representative circuit packs used in Definity systems

•Analog Line Card :- Used to terminate 24 analog phones.

•Digital Line Card :- Used for 2-wire and 4-wire DCP digital phones. Each

type of telephone needs a different digital card. The 2-wire card supports

24 phones, while the 4-wire card supports 8 phones.

•CO Trunk Card :- Accepts analog CO trunks.

•T1/E1 Card :- Terminates a E1/T1 or PRI link. Can act as either T1 or E1

card via a dip-switch on the card. Called a DS1 card. Supports both A-

Law and mu-Law commanding.

•Expansion Interface (EI) Card :- Used for connecting expansion cabinets

or carriers. Connects a fiber transceiver at the back.


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Communication Manager:

Avaya Communication manager is application running on a variety of

Avaya Media servers and Definite servers and providing control to

Avaya Media Gateways and Avaya communication devices.

The software provides user and system management functionality,

intelligent call routing and application integration.

Nice Servers:

Nice call recording system is used for recording calls.

-It comprises of Nice Logger, Nice Application Server, Storage server

and AES.

Function of Servers


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AES (Avaya Enablement Server):

AES server is used to get CTI event for call information tagging. AES

provides adjunct control of telephone calls through its call control to

complete adjunct routing of calls, report various events to an

adjunct.

AIC (Avaya Interaction Centre):

AIC is used to for integration of soft phone and G-CIC with EPABX.

-It's main function is to provide all soft phone functionality viz. soft

phone login, call transfer, hold etc.


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CMS (Call Management System):

CMS is a database, administration and reporting application. CMS is

used for real time and historical reports.

It collects the call-traffic data, formats management reports and

provides an administrative interface to the ACD feature on the

Communication Manager system.


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Components of Media

Gateways IP Server Interface (IPSI):

• Provides communication interface between Avaya Communication

Manager Server and Media Gateways (Port Networks).

Control / Customer LAN (CLAN):

• It is the card where all IP endpoints, gateways and adjuncts have to

register.


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• It has 31 usable playback ports.

Val Announcement Card :

• This is integrated Announcement Card that play the pre Recorded

announcement, When the call is in Queue or the call is put on Hold to

encourage the caller Patience.

Media Processor (MEDPRO):

• Contains all the Digital Signal Processors (DSPs) for converting /

compressing digital / analog / IP Voice.

• IP phones get dial tone from MedPro card and one MedPro card

can handle 64 simultaneous voice calls.


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Call Classifier Card:

• Circuit Pack used to increase the Signal Amplification. 4 DS1 cards

require 1 Call Classifier.

PRI Card:

• DS1:- DS1 stand for Digital Signaling Level 1 standard term describing

1.544 Mbps digital signal carried on T1 facility or 2.048 Mbps on E1

facility.


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Logical Flow of Registration

and Signaling:

RAS h.225 Registration

H.323:

H.225/Q.931/H.245/FasStart

Call SignalingH.248 Control

RTP (Voice Stream)

Translation Transfer to LSP via

sync Embedded CCMS messaging

(within IP)

PIN Transmission Encrypted = 56-bit Diffie-Helman in 2.0,

1024-bit Diffie-Helman in 3.0

NOT ENCRYPTED in 2.0. In 3.0, 128-bit AES

Encrypted - 128-bit AES (as of 1.3.1)

Encrypted ** - 104-bit AEA in 1.3. In 2.0, also 128-bit AES. In

3.1, SRTP 128-bit AES

NOT ENCRYPTED in 2.0. In 3.0, TLS-tunneled (AES 120-bit

within TLS)Encrypted 128-bit AES for S8700 IP Connect Only

ENCRYPTIONPROTOCOLKEY

** Requires 1.8 IP Phone Firmware


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Example Flow of CM

Registration and

Signalling:


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IPS

I

CLA

N

Pro

wle

r

UDP 1719 – H.225 RAS

RegistrationTCP 2945 or 1039 – H.248

Control TCP 1720 – H.225/Q.931 Call

Control Listen

~500 TCP Control

Sockets

Private Control

LAN

Customer LAN/WAN

S870

0S870

0

Remote G350 Gateway

Remote G700 Gateway stack

w/LSPSCC1/MCC1/G650

Gateway

Avaya IP

Phone

TC

P 2

1873

Random

TC

P

Random

TC

P

Random

UD

P

Port

Contro

l Port

TCP

21873Contr.

Port


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Primary Rate Interface

(PRI)

The Primary Rate Interface (PRI) is a standardized

telecommunications service level within the Integrated Services Digital

Network (ISDN) specification for carrying multiple DS0 voice and data

transmissions between a network and a user.

PRI is the standard for providing telecommunication services to offices. It

is based on the T-carrier (T1) line in the US and Canada, and the E-

carrier (E1) line in Europe. The T1 line consists of 24 channels, while an E1

has 32.


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http://en.wikipedia.org/wiki/Telecommunication

http://en.wikipedia.org/wiki/Integrated_Services_Digital_Network

http://en.wikipedia.org/wiki/DS0

http://en.wikipedia.org/wiki/T-carrier

http://en.wikipedia.org/wiki/E-carrier

Each B-channel carries data, voice, and other services. The D-channel

carries control and signalling information. Larger connections are

possible using PRI pairing. A dual T1-PRI could have 24 + 23 = 47 B-channels and 1 D-channel (often called "47B + D"), but more

commonly has 46 B-channels and 2 D-channels thus providing a

backup signalling channel. The concept applies to E1s as well and

both can include more than 2 PRIs. Normally, no more than 2 D-

channels are provisioned as additional PRIs are added to the group.

•PRI, the Primary Rate Interface for large organisations, with one 64-

kbit/s D channel and 23 (1536 Mbit/s T1, a.k.a. "23B + D") or 30 64-kbit/s

B channels (2048 Mbit/s E1, a.k.a. "30B + D").


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http://en.wikipedia.org/wiki/T-carrier

http://en.wikipedia.org/wiki/B_channel

http://en.wikipedia.org/wiki/E-carrier


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VoIP (Voice Over Internet

Protocol) Voice-over-Internet protocol (VoIP) is a methodology and group of

technologies for the delivery of voice

communications and multimedia sessions over Internet Protocol (IP)

networks, such as the Internet. Other terms commonly associated with

VoIP are IP telephony, Internet telephony, voice over

broadband (VoBB), broadband telephony, IP communications, and broadband phone service.

The term Internet telephony specifically refers to the provisioning of

communications services (voice, fax, SMS, voice-messaging) over the public Internet, rather than via the public switched telephone

network (PSTN).


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http://en.wikipedia.org/wiki/Voice_communication

http://en.wikipedia.org/wiki/Multimedia

http://en.wikipedia.org/wiki/Internet_Protocol

http://en.wikipedia.org/wiki/Fax

http://en.wikipedia.org/wiki/Internet

http://en.wikipedia.org/wiki/Public_switched_telephone_network

VoIP systems employ session control and signalling protocols to control the

signalling, set-up, and tear-down of calls. They transport audio streams

over IP networks using special media delivery protocols that encode voice,

audio, video with audio codecs, and video codecs as Digital

audio by streaming media.

The steps and principles involved in originating VoIP telephone calls are

similar to traditional digital telephony and involve signalling, channel setup, digitization of the analog voice signals, and encoding. Instead of being

transmitted over a circuit-switched network, however, the digital

information is packetized, and transmission occurs as Internet Protocol (IP)

packets over a packet-switched network. Such transmission entails careful

considerations about resource management different from time-division

multiplexing (TDM) networks.


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http://en.wikipedia.org/wiki/Audio_codec

http://en.wikipedia.org/wiki/Digital_audio

http://en.wikipedia.org/wiki/Streaming_media

http://en.wikipedia.org/wiki/Telephony

http://en.wikipedia.org/wiki/Internet_Protocol

http://en.wikipedia.org/wiki/Packet-switched_network

http://en.wikipedia.org/wiki/Time-division_multiplexing

Various codecs exist that optimize the media stream based on

application requirements and network bandwidth; some

implementations rely on narrowband and compressed speech, while

others support high fidelity stereo codecs. Some popular codecs

include μ-law and a-law versions of G.711, G.722, which is a high-

fidelity codec marketed as HD Voice by Polycom, a popular open

source voice codec known as iLBC, a codec that only uses 8 Kbit/s

each way called G.729, and many others.


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http://en.wikipedia.org/wiki/Narrowband

http://en.wikipedia.org/wiki/Speech_coding

http://en.wikipedia.org/wiki/High_fidelity

http://en.wikipedia.org/wiki/%CE%9C-law

http://en.wikipedia.org/wiki/A-law

http://en.wikipedia.org/wiki/G.711


http://en.wikipedia.org/wiki/Polycom

http://en.wikipedia.org/wiki/ILBC



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Why VOIP ??Technological advances and consumer trends are interesting, but they

don’t answer the question: why you should use VoIP for your business

communications.

1. Same Quality as Always, More Benefits than Ever VoIP services can

be deployed to retain the same, if not better, quality and reliability that

you expect from traditional phone lines. Earlier, phone lines were

usually more reliable than Internet connections, but this is no longer the case.

With VoIP, calls to destinations around the globe can be made with no

difference in quality from traditional phone lines. When a professional

phone company implements VoIP service, dropped calls, crackling,

echoes or other problems are concerns of the past. Voice clarity using VoIP is excellent.


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2. Monthly Savings Why pay more for a service that offers less? When you're running your own

business or focused on the financial needs of your company, the bottom

line is a priority equal to securing reliable service from a trusted provider.

Since VoIP services and devices now utilize widely deployed IP and

broadband technologies, they are available to business customers at a price that’s several times cheaper than services relying on more costly

network infrastructures.

3. Reduce Capital Expense and Total Cost of Ownership for Your Office

Phone System VoIP clearly helps businesses realize savings on their monthly

voice services, and it also can help reduce capital expenditures. Depending on a company’s growth patterns and communications needs,

the life cycle of an office key telephone system or Private Branch

Exchange ranges anywhere from 5 to 10+ years.


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For businesses in the market for a new office phone system, VoIP service

strongly supports consideration of IP PBX or Hosted IP PBX — two options that can significantly reduce capital investment:

• Businesses purchasing IP PBXs can save on the cost of not only the

central processing unit but also the IP handsets, since both typically

cost less than their traditional PBX counterparts.

• Business customers can also use VoIP to completely eliminate the

major capital expense of purchasing a PBX, by choosing Hosted IP PBX,

a solution that has in some cases produced a 60% decrease in

operations and administration expenses over a five-year period.5 Hosted IP PBX vendors typically provide tools that empower

administrators and end users to manage their own line and desk

phone/softphone feature changes, via easy-to-use websites.


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How VoIP works for business is simple: By adding voice to a data network,

you’ll reduce costs, improve productivity, and enhance collaboration.

You'll save money by having one network to manage instead of two. You can easily add, move, or change phone extensions and locations, which

saves money and gives you more flexibility.

Your workforce can use your communications system from home or on the

road. Also, wireless IP phones connect users to your communications system and data resources, such as customer information, while they're in

the warehouse, on the sales floor, or anywhere they can access your data

network wirelessly.9/9/2014Created By APOORVA TYAGI

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http://www.cisco.com/cisco/web/solutions/small_business/products/voice_conferencing/index.html-tab-Phones

ATA (Analog Telephone Adapter)

An ATA is a simple device which lets you connect any standard telephone or fax machine so it can use VoIP through your internet

connection. The ATA converts the analog signal from your telephone into

digital data that can be transmitted over the internet. Providers usually

bundle this device with their service so that you can start making calls

right away.

Unified communications solutions for small businesses go beyond basic

VoIP capabilities in enhancing collaboration. With a unified communications solution, workers can easily collaborate through voice,

video chat, Web conference, and instant messaging. Employees can

collaborate using each technology individually or all of them

simultaneously, and from a single, easy-to-use interface.


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http://www.cisco.com/cisco/web/solutions/small_business/products/voice_conferencing/index.html-tab-VoiceSystems

IP Phones are special telephones which look and work like normal phones but connect directly to your internet connection without the use of ATA device (to convert analog signals to digital signals). An IP Phone

plugs directly into your internet router and comes in both wireless and

corded models. Business VoIP users generally opt for IP Phones because

they have special buttons which allow calls to be transferred put on hold

and have multiple lines.

Computer-to-Computer

Using software installed on your computer and a headset you can make

and receive VoIP telephone calls right on your desktop or laptop. You

can even place callers on hold, transfer them to another extension, or

answer multiple telephone lines. Some software also allows you to host conference calls.


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1. Total controlOn your computer you can see a complete list of staff members who use

VoIP. Call any of them with a single click, check missed calls and numbers

dialled or call lengths. Some systems will even let you listen in to calls. This

means that if you use VoIP for your contact centre environment, you can maintain complete control.

2. Speed dialSet up unlimited speed dial numbers, so you can quickly make calls

without searching for and dialling long numbers. One click puts you

straight through.


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4. Priority callsYou may not want to be disturbed, but if you are awaiting a call from a

colleague or client you can set priority alerts, so the phone rings only if

it's the person you're waiting for. This helps you maintain productivity

levels without missing important calls.

5. DiscussionIf you need to have a three-way conversation, you can have a three-

way VoIP conference call. Since the VoIP system uses the internet, you

shouldn't have to pay extra for this function.

3. Do not disturbOne of the most frustrating things in any office is never getting anything

done because the phone is always ringing. You don’t want to miss calls,

but you can’t have your productivity frequently interrupted. VoIP allows

calls to be sent straight to your voicemail without the phone ringing.


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7. Out of office messagesYou're working onsite with a client and not due back in the office until

next week. But a new sales lead has just left a message on your office phone. If you don't call back fast you might miss making the sale. What

do you do? With VoIP it’s simple. Just wait for the system to send a

message to your mobile letting you know you’ve received a voicemail.

Then log in on your laptop and check your messages.

6. MessagingVoicemail can be awkward. If you receive lots of calls, important

messages can get buried beneath more recent ones. And some systems

delete messages after a week or a month. With VoIP, you can receive voicemail messages as audio files in your email. Then you can listen to

them, store or share them with others.


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10. Selective forwardingYou can set rules so only important calls are forwarded to your alternate number. This means you can avoid unnecessary calls, yet still connect with

the important ones.

9. Remote officeMany VoIP systems have a remote office function, allowing you to route all

your calls to an alternate phone. So if you want to work from home this week, you won’t miss out. Plus, when you make a call, your work number

will appear in the recipient’s caller ID.


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9/9/2014 Created By APOORVA TYAGI 68

Manual Operator

Exchanges

Mechanical /Stronger

Exchanges

Electronic SPC

Exchanges

Avaya Definity® Communication Servers.

Supported traditional analog and digital telephones.

With evolution of IP technology, supported IP end-

devices as well. However, ultimately, all control

signals and voice was carried by a TDM and a

Packet bus.

IP PBXs

With robust IP infrastructure, corporates wanted telephony solutions to

totally come over the LAN, and a converged voice & data network to

be built. Avaya Media Servers and Gateways follow a server-gateway

architecture and brings voice, PBX control signals and data networking

on one network, thus building the ECLIPS range of products. Existing

analog or digital end-devices are supported using TDM buses inside the

Media Gateways.

Road to IP Telephony


IP Phone Registration

over WANNormally there are two ways to register IP Phones into a remote ACD over

WAN.

1. Register IP Phones over WAN directly to remote port network and ACD.

2. Register IP Phones to a Port Network locally and Port Network further register to remote ACD

Registering IP phones into media server over customer LAN/ WAN requires some key devices as follows:

a. IP Phone

b. DHCP Server

c. Switch/ Router

c. TFTP Server

d. CLAN (Gate Keeper)

e. IPSI 9/9/2014Created By APOORVA TYAGI

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IP Phone: This is an IP end point.

DHCP Server: We can configure IP phone by using static IP address or acquiring IP address from DHCP. But in most of the cases we use DHCP option for better administration.

Switch/ Router: Switch or router actually provide IP connectivity to an IP end point however DHCP provide an IP address from an IP address pool.

TFTP Server: IP phone require three codes normally for booting and protocol analyze.

CLAN: IP phone register to the Control LAN (gate keeper) in port network.

IPSI: In every port network there is a IPSI (IP server interface) who is a communication interface between media server and media gateway. This is required to provide control network signaling over a customer’s LAN/ WAN. IPSI is responsible for gateway control and tunneling call control message back to the server. Also this is the interface between server and IP endpoints. IPSI sent keep alive message to the media server in time interval set by Avaya communication manager.


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IPS

I

CL

AN

Med

Pro

Customer LAN

Customer WAN

S8700

S8700

G650 Gateway

Avaya IP Phone

Con

trol P

ort

IPS

I

CL

AN

Med

Pro

Customer WAN

Customer LAN

S8700

S8700

G650 Gateway

Con

trol P

ort

Register IP Phones over WAN Register Port Network over WAN

Avaya IP Phone Avaya IP Phone


IP Phone Boot up

Sequence


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6.Registration: The telephone registers with a media controller (gate

keeper) after all the required codes are successfully loaded. The following

packets are interchange between gate keeper and phones.

a. GRQ: RAS – gatekeeper request

b. GCF: RAS – gatekeeper confirm

c. RRQ: RAS – registration request

d. RCF: RAS – registration confirmRAS is registration admission status

4.Request 46XX Code File: Normally IP phone require three code files from TFTP server for boot up and protocol analyze. These are

46XXUPGRADE.SCR, boot code and application code. IP phone

download these files from TFTP server and upgrade itself.

5.Ext. and Password Prompts: The telephone prompts for the extension and the password.


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7.Telephone is operational: Administered display shows up on the

telephone.

8. Keep alive Message: The keep alive message sent by each phone to the

media controller at time interval set by Avaya communication manager.

The keep alive message are RRQ and RCF in time interval.


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A huge shift in customer service is underway, brought about by

changing demographics, new technologies and new forms of

communication and interaction-not the least of which are the still

nascent media networks.

While technologies and specific networks may wax and wane in

importance, there is no disputing in overall trend. To meet customer

expectations and mitigate customer satisfaction, companies need to

focus on how to evolve their customer service infrastructure and

manage customer satisfaction expectations, offering interactions across

various channels while delivering a consistent experience to manage

customer satisfaction expectation.Ultimately, customer experience management crosses the boundary

from contact centre to enterprise wide customer care and provides a

new way for companies and organisation to differentiate and grow.


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