a crm model based on voice over ip

8/11/2019 A CRM Model Based on Voice Over IP

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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

6



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

465



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

466



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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a crm model based on voice over ip

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