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8/11/2019 A CRM Model Based on Voice Over IP
http://slidepdf.com/reader/full/a-crm-model-based-on-voice-over-ip 1/5
A RMmodel based on Voice over IP
Y.S. Moon+,C.C. Leung+,K N Yuen’, H.C. Ho+,
X
Yu*
+Departmentof Computer Science and Engineering
*Dept. of System Engin. and Engin. Management
The Chinese University of Hong Kong
Shatin, N.T., Hong Kong.
Email: ysmoon, ccleung, knyuen, [email protected], [email protected]
Abstract
Based mainly on the VoIP techniques, a Customer
Relationship Management (CRM) application model is
proposed for online help desk services. In contrast to the
traditional call center, it does not use an analog Private
Branch Exchange (PBX) but utilizes both PSTN and the
Intemet
so
that people who need help desk services while
surfing a web page, can request a customer service orally
by a traditional phone or VoIP software. At the same
time, the customer service operator can also grasp the
web-surfing status specific to the customer through the
Intemet link. An analysis of the components of the system
is
presented in this paper.
1 Introduction
Seamless integration of computer and communication
to replace PSTN is the hope of voice over
IP
in this
millennium. Nevertheless, such a hope is still not quite
available
yet.
We
will
review the existing difficulties
and
study the technical advancements that are possible to
achieve under the present constraints.
There are several developing standards that aid the
transmission of voice over IP. For example, H.323
[l]
which is targeted to the transmission of multimedia over
packet-based networks; SIP [2] which aims at the creation
of sessions between different parties; RTP [3] which
transmits real-time packets over network: RSVP [4]
which tackles the network bandwidth reservation
problem. We will examine these standards and how they
shape the development of VoIP.
Based mainly on the VoIP techniques, a Customer
Relationship Management (CRM) application model,
Intelligent Call Center, is proposed for online help desk
services. Consider the following scenario. When a
customer visits a web site on the Internet that contains a
form to fill in, helshe may not know how to fill in a
particular part of the form. Although helshe may send an
email to ask about the form, the feedback may take time.
Moreover, it is often difficult to give the answer in written
form. In the worst case, the two parties communicating
with each other may not be focusing on the exact problem
to be solved, due to wordings and interpretations. In view
of such difficulty, we propose a VoIP based model to
tackle this problem. Instead of sending an email to ask
about the form, the customer will contact the call center
using phone or VoIP software and ask questions. During
the conversation, the customer service operator can
retrieve the web-surfing status of the customer through
the Intemet link, if necessary. In this way, the customer
can be promptly and correctly served by personalized
service.
In this architecture, data packets from the Intemet
and analogue voice signal from PSTN can enter the call
center through a digital voice gateway to replace a
traditional analogue Private Branch Exchange (PBX). In
such a way, we have a voice and data unified environment
151
The above is only one of the obvious advantages.
Features, like Predictive Dialling and Automatic Call
Distribution [ 6 ] are favoured based on the new version of
the Computer Telephony System.
2
Underlying Network Infrastructure
Our proposed model consists of a PSTNM.323
gateway, an Intranet and a cluster of customer service
operators’ PCs. This model is based on the ITU H.323 as
well as a set of other Internet Engineering Task Force
(IETF) standards serving as the underlying network
infrastructure. We will divide this underlying
Infrastructure into three parts: Codec, Transmission of
Voice Packet and Call Control.
2 1 H 323
ITU H.323 is a standard for multimedia
communications over local area network (LANs). H.323
supports both point-to-point and also broadcasting
0 7803 5957 7l 00l~10.00
Q 2000 IEEE
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communications. It addresses call control, multimedia
management, bandwidth management, and interfaces
between LANs and other networks. Although H.323 does
not provide Quality of Service (QoS), there are quality of
service protocols such as Resource Reservation Protocol
(RVSP) that may complement it.
There are four elements in H.323. These are User
Terminal, Gateway, Multipoint Control Unit (MCU) and
Gatekeeper (GK). User terminals are front-end
communication devices that enable users to communicate.
A gateway lies between H.323 and other network (such as
PSTN) to translate data and signal between networks.
MCUs are responsible for multiparty conferencing. GKs
are responsible for call authorization, address resolution,
and bandwidth management. They intercept call signaling
between endpoints and provide “signaling-based”
advanced services.
Under H.323, there are many protocols and standards
that address different issues related to VoIP. For instance,
H.225 [7] (another name for this protocol is Registration,
Admission, Status protocol) allows user terminals to
communicate with a GK. 4.931
[8]
is derived from ISDN
end-to-end call setup signaling and provides the logical
connection between the two endpoints he calling party
and the called party. H.245 [9] is used to exchange
capabilities between the caller and the callee. They have
to agree on the nature of the information they will
exchange through the media channels (audio, video and
data) and their formats. H.450 [lo] defines signaling
between endpoints for supplementary services that are
added on top of the H.323 protocol to provide additional
functionality.
2.2 Codec
The first step in transmitting voice over IP networks
is to digitize it
from
analog form using codecs. Pulse
Code Modulation (PCM) codec and Adaptive Differential
Pulse Code Modulation (ADPCM) codec are two mostly
commonly used technologies.
One of the codec in ITU, the ITU recommendation
G.711 [ l l ] , uses PCM to encode voice data. G.711
is
an
international standard for encoding telephone audio on a
6 kbps channel. It is a pulse code modulation (PCM)
scheme operating at a 8 kHzsample rate, with 8 bits per
sample. According to the Nyquist theorem, which states
that a signal must be sampled at twice its highest
frequency component, G.711 can encode frequencies
between
0
and
4 kHz
Several ITU codecs use ADPCM methods. For
example, G.721 (CCITT 32 kbps ADPCM codec) which
codes each difference value using 4 bits at
8 kHz
G.723.1
[12] (CCITT 5.3 kbps and 6.3 kbps ADPCM codec)
which provides low bit rate speech coding; G.727 [13]
(CCITT 40, 32, 24 and
16
kbps/s embedded ADPCM
codec).
Besides PCM and ADPCM, there are also other
waveform codecs such as ITU G.729 and G.729A [14]
which use CS ACELP (Conjugate-Structure Algebraic
Code Excited Linear Prediction) to support speech at
8
kbps.
As mentioned before, different codecs have different
data rates. Low bit-rate codec such as G.723.1 is suitable
for modems that support only low bandwidths. For such
low bit-rate codec, users have to tolerate lower quality of
voice. Medium bit-rate codec such as G.721 and G.727
will be suitable for higher bandwidth users who connect
to the Intemet through broadband network or corporate
Intranet. The quality of voice will be comparable to PSTN
telephone call. At last, G.711 is suitable for conventional
phone users as G.711 is currently used in telephone line
for transmission of voice. Since G.711 consumes high
bandwidth, the whole telephone line bandwidth is
dedicated to its consumption. The gateway will convert
voice signals in G.711 to G.721 or (3.723.1 so that these
voice signals will not consume large portions of the call
center’s Intranet bandwidth.
2.3 Transmission
of
Voice Packet
Given a data stream of digitized voice, the network
must
try
its best to transmit it in a lossless and timely
manner. In reality, packets will be delayed or lost in
transmission, these problems need to be addressed.
’
In our proposed system, we will use the Realtime
Transport Protocol (RTP) to transmit voice packets. We
use’ RTP instead of the User Datagram Protocol
(UDP)
[15] or Transmission Control Protocol (TCP) [16]
because RTP is particularly designed for transmission of
real time packets such as the voice packets in our system.
2.3.1 Realtime Transport Protocol RTP)
After codec is employed to digitize analog voice
signal into binary form, we need to &vide the resultant
data into segments and place these segments into RTP
packets to be transmitted over the network. At the
receiver end, segments will be extracted from the RTP
packets and decode into voice signal to be played. RTP
usually
runs
on top of UDP. As RTP
runs
on top
of
UDP,
it is possible that some RTP packets may be lost during
their joumey to destination. If a RTP packet is lost during
transmission, no attempt will be made to send that RTP
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packet again. It is because voice packets are generated in
real-time.
2.4 Call Control
This
part is targeted to solve two important issues in
VoIP application: Call Setup and Quality of Service.
2.4.1 Call Setup
There are two approaches in call setup. One approach
is to use H.323 standards to do the call setup. The other
approach is to use Session Initiation Protocol (SIP)
developed by Internet Engineering Task Force (IETF).
However, we prefer the SIP approach because the H.323
call setup sequences are quite complicated. It takes long
time during which several trips of messages (4.931,
H.245) before the call setup procedure is completed. As a
result, H.323 has another method called Fast Connect
which shortens the time to complete call setup.
Switched Telephone Network (PS ) or'VoIP software
to communicate with a customer service operator in the
call center. At the time, the customer also accesses the
web page. The PSTNlH.323 gateway connects two
different networks, the switched telephone network and
the Internet to the Intranet of the call center. The
PSTNlH.323 gateway digitizes, compresses, puts the
voice of the call into data packets and forwards the
packets to the operator. When a new telephone call comes
into the gateway, the gateway routes the call to the
Automatic Call Distributor (ACD). The ACD finds the
most appropriate operator to serve the customer. The
database server stores the information on the customer's
history with the company. The directory server maps the
operator's name to the IP address of the workstation the
OPERATOR is currently working. The web server is
integrated with the call center in the system. Also, the
web-surfing status of the customer can be transferred to
the operator when necessary. In this way, the customer
can receive a personalized service.
2.4.2 Quality
of
Service
It is necessary to ensure that the quality of voice
transmission will not be affected by varying network
traffic conditions. This is especially important as Internet
traffic rate changes continuously. Packet-loss and packet-
jitters are common phenomenon on today's Internet
traffic
so
that methods must be deployed to ensure the
quality of voice during conversation. In our system, we
will use Resource Reservation Protocol (RSVP) as our
Quality of Service (QoS) protocol to ensure the quality of
voice conversation between customers and customer
service operatorsis
good
The reservation mechanism is described as follows.
When both parties' (customer and customer service
operator) terminals have finished the call setup procedure,
both of them will send a path message containing the
flowspec to each other. Flowspec is a flow specification
that describes both the characteristics of the traffic stream
sent by the source and the service requirements of the
applications. Then, when each side receives such packets,
it will request resource reservation by sending a
reservation message using flowspec.
If
all the routers
between them accept such reservation, both sides can start
to send their packets. As RSVP has the soft-state
property, both sides need to send refresh messages
periodically to keep the reservation.
3 A
Proposed
Architecture
proposed architecture is depicted in Figure 1. A
customer who requests help desk services uses Public
1. The customer phones to the call center and the phone call
enters
the
gateway. r the customer uses
VoIP
software in hidher PC to talk to
the call centerand he request messageis sent to the gateway.
2 The gatewayforwards he phone call to the ACD.
3. The ACD retrieves the caller
ID of
the call and send it to the database
4
The database server checks whether there exists a customer
associated with
this
caller ID.
If
it exists, the database server
sends
these information back to the ACD.
5
According to the reply from the database server, the ACD may know
the name of operators who often served this customer
in
the past. The
ACD queries the directory server about the availability of these
operators and their IF addresses.
server.
6.
The directory server replies the query back to the ACD.
7 The ACD chooses an operator to answer the call. The choice and the
8 A QoS guaranteed connection is attempted to establish between the
IP address
of
the operator's workstation is sent to the gateway.
customer and the operator along the Intranet.
Figure 1 The proposed architecture
3.1 Customer
There are three ways for the customer to request the
service from the call center. While surfing a web site, a
customer can find the telephone number from the web
page. The customer can immediately make a phone call to
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the call center if hetshe has a spare telephone line.
Another approach is that the customer use VoIP software
in hisher PC to talk to the call center if hetshe does not
have a spare telephone line. The third approach is that the
customer clicks the “call me back” button in the web
page, the browser shows a form requesting the customer’s
name and telephone number, and asking when the
customer is available to be called. After the customer fills
in and submits the form, the call center will call back to
the customer according to the scheduled time.
During conversation, the customer can deliver hisher
web-surfing status to the operator at hisher
own
will.
This enables the customer and the operator to view the
same web page, even permitting the operator to change
the web page if the caller is willing to.
3.2 Customer Service Operator OPERATOR)
The workstation with which the operator works is
connected to an Intranet server that, in turn, connects to
the PSTN and the Internet through the PSTNfH.323
gateway. All sampling, compression and packetization of
the operator’s voice occur in the codec hardware and
software on the operator’s workstation and the server.
When the operator is assigned to answer a phone call,
the profile of the callerlcustomer, if available, will appear
on the operator’s screen
so
that helshe can well prepare
himselfherself to address the needs
of
the customer.
3.3 Automatic Call Distributor ACD)
When a customer dials the call center, the gateway
forwards this phone call to the Automatic Call Distributor
(ACD) which arranges an operator to handle this phone
call and informs the gateway of the IP address of the
operator’s workstation.
When the ACD receives a new phone call, it retrieves
the caller ID
of
the phone call. The ACD sends the caller
ID to the database server to ask whether there exists a
customer record associated with it. If
so
the database
server returns the customer profile information to the
ACD. The information may include the name, the address
and the credit card information of the customer, the name
of operators who often serves this customer in the past,
etc.. Then the ACD queries the directory server about
the availability of the operators and their IP addresses.
Based on their availability and experience as well as the
customer’s profile, an operators is chosen to handle the
call.
If the customer uses the “call me back” approach, he
can click the “call me back” button in the web page while
surfing. In response, the browser will send a Hypertext
Transfer Protocol (HTTP) message to the Web server.
The server reacts by directing the browser to point to an
URL containing a form requesting the customer’s name
and telephone number, and asking when the customer
should be called. When the web server receives the
completed form, it will inform the ACD to schedule the
“call back”. At the scheduled time, the ACD uses the
same procedures mentioned previously to choose an
operator to serve the customer. However, in this case, the
information about the customer is sent to the operator
before helshe makes a phone call to the customer.
3.4 Directory Server
The directory server records the mapping of operators
to the IP address of their assigned working workstations.
When the ACD sends an operator’s name to the directory
server, the server looks up the table to check whether the
operator is available now. If yes, the server sends the IP
address of the operator’s workstation to the ACD.
Otherwise, the server informs the ACD that the operator
is not available. If an operator leaves the working
workstation or works in a new workstation, the operator
or hisher supervisor will request the directory server to
update the corresponding record.
4
Advanced Features
Upon the proposed model, more application entities
can be built to enhance the features provided in the call
center system. Two advanced features, predictive dialling
and skill-based routing, are discussed below.
4.1 Predictive Dialing
Predictive Dialing is gaining widespread interest in
the call center industry. To implement this feature,
two
additional entities, a List Management System (LSM) and
a Telemation Manger (TM), are needed. The LSM is an
external process that determines which calls are to be
made by an operator. The TM provides management with
the tools necessary to optimize the performance of the
operators and manage the call lists in the predictive
dialling. The TM also accepts phone numbers from the
LMS in bulk and initiates calls without connecting the
calls to the operators. When an answer is detected, a
particular operator is connected and informed of the start
of the conversation. Simultaneously, the TM displays the
called party’s information on that operator’s display. The
TM uses its own predictive process to determine the
number of calls to place based upon the number of
operators and other statistics such as the number of
connections per dialed numbers, average phone call
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