Call Centre Architecture

Call Centre
Architecture
Summer Project in Samsung Contact Centre
Created By APOORVA TYAGI 9/9/2014
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Acknowledgement
I 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 Kaul
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Overview
Call 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
PN #2
(8 PRI)
Control Network
Enterprise LAN
PSTN
PRI’s
NICE CLS
Logger
Main AES
S 8800 Media servers CM 6
HA AES
Main CMS
PN #3 (8 PRI)
Duplicate power supply, MedPro and IPSI
Duplicate AIC
Telephony server
HA CMS
PN #1
8 PRI)
Port Net Work
Setup
Call Centre Setup
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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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Avaya Call Management
System
Avaya 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- time
monitoring 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_Tran
external 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 the
Avaya 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
Overview
Avaya’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
Created By APOORVA TYAGI The G650 rack-mountable Media Gateway 9/9/2014
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Avaya Definity Solutions
Overview
The 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 Overview
Basic 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)
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Concept of Port Networks
In Definity, a Port Network is nothing but a stack of cabinets with
extension and/or trunk termination port circuit packs. The name Port
Network comes
from the fact that these port circuit packs provide a bank of physical
circuits
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 the
processor 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 circuit
packs 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 administrative
terminal, 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 Pack
Four 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 packs
Include 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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Function of Servers
 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.
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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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 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.
 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.
• It has 31 usable playback ports.
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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:
KEY PROTOCOL ENCRYPTION
RAS h.225 Registration
H.323:
H.225/Q.931/H.245/FasStart
Call Signaling
H.248 Control
RTP (Voice Stream)
Translation Transfer to LSP via
sEymnbc edded 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
Ewnithcriny pTteLdS )128-bit AES for S8700 IP Connect Only
** Requires 1.8 IP Phone Firmware
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Example Flow of CM
Registration and
Signalling:
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IPSI
CLAN
Prowler
UDP 1719 – H.225 RAS
Registration
TCP 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
0
S870
0
Remote G700 Gateway stack
w/LSP
Remote G350 Gateway
SCC1/MCC1/G650
Gateway
Avaya IP
Phone
TCP 21873
Random TCP
Random TCP
Random UDP
Port
Control Port
TCP
21873
Contr.
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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•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").
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.
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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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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.
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.
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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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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.
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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.
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.
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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 control
On 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 dial
Set 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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3. Do not disturb
One 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.
4. Priority calls
You 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. Discussion
If 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.
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6. Messaging
Voicemail 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.
7. Out of office messages
You'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.
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9. Remote office
Many 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.
10. Selective forwarding
You 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.
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9/9/2014 Created By APOORVA TYAGI 68

Road to IP Telephony
Mechanical /Stronger
Manual Operator
Exchanges
Exchanges
Electronic SPC
Exchanges
IP PBXs
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.
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.

IP Phone Registration
over WAN
Normally 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. IPSICreated By APOORVA TYAGI 9/9/2014
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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.
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.
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Register IP Phones over WAN Register Port Network over WAN
IPSI
CLAN
MedPro
Customer LAN
Customer WAN
S8700
S8700
G650 Gateway
Control Port
Avaya IP Phone
IPSI
CLAN
MedPro
Customer WAN
Customer LAN
S8700
S8700
G650 Gateway
Control Port
Avaya IP Phone Avaya IP Phone

IP Phone Boot up
Sequence
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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.
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 confirm
RAS is registration admission status
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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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Call Centre Architecture

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