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The
Technology
Guide
Series
Next-Gen VoIP
Services and
Applications Using
SIP and Java
This Guide has been sponsored by
This Guide has been sponsored by
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For Service Providers,
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itÕs a
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2
TECHNOLOGY GUIDE
Visit our Web site
Table of Contents
to read, download,
Abstract
4
and print all the
Introduction
4
Architecture Models
6
Technology Guides
Technology Enablers for Next Generation
Voice Services and Applications
16
in this series.
Next Generation IP Voice Services
and Applications
29
Summary
33
Glossary
34
Appendix A: Session Initiation Protocol
(SIP) Concepts and Operation
38
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4
TECHNOLOGY GUIDE
Next-Gen VoIP Services and Applications Using SIP and Java
5
Abstract
caller ID, etc.), cannot provide the types of features
that are needed by a contemporary business in the
age of e-commerce. The traditional business
telephony solutions are complicated, for both the
service administrators and the users. Because of the
daunting complexity of PBX and CLASS/Centrex
user-interfaces, users typically know and use only a
fraction of the total feature set.
Now imagine telephony services in the context
of the current business need. The users would still
like to use a phone for making and receiving calls
and playing voice-mail messages. However, they
would also like to have the phone appliance
integrated with a browser-based PC for managing
phone books and seamlessly interfacing with other
applications, such as customer relationship
management (CRM), sales force automation (SFA),
supply chain management (SCM), time accounting,
etc. In other words, perform tasks most suitable
for the PC on the PC and those most suitable for
the telephone using a phone appliance and have
the two devices seamlessly integrated.
Today’s telephone just cannot deal with this new
business imperative.
In contrast, the Internet and Web-based
communications have revolutionized the business
environment and user personal life-styles by their
inexpensive, standards-based innovations. We already
have data, multimedia, video, and music applications
on the Internet. The Internet is already serving as the
underpinning of critical business and IT solutions.
Just in the last few years alone the Internet and the
Web have generated more innovations than
traditional telephony has produced in its entire
history. The next frontier for the Web is to apply the
same degree of innovation to telephony.
Most market surveys have verified that IP
telephony is already supplementing traditional
telephony and it is expected that the IP telephony
architecture will ultimately replace the traditional
telephony model.
This Technology Guide explains the unique
benefits of using the Web architectural model with
SIP and Java as the enabling technologies for next
generation IP voice services and applications.
Using the Web as a reference model for rapid
innovation, the Guide contrasts the limitations of
circuit-switched telephony and first generation
VoIP architectures with the Web model. It
summarizes limitations of centralized-processing
models such as traditional telephony, MGCP, and
Megaco as compared to peer-to-peer models such as
SIP and H.323.
This Technology Guide explains in more detail
the unique benefits of using SIP for call control
and Java for making phones intelligent. SIP is
compared with H.323 in terms of innovation,
scalability, simplicity, ease of deployment, and
standardization. The guide also includes an
explanation of SIP concepts and operation. A
description of Java features supporting new voice-
services and applications is also included.
The Guide concludes with examples of new
voice-services and applications made possible
exclusively by SIP and Java.
Introduction
Traditional telephony has hit a wall in terms of
innovation, ease of use, and cost reduction. The
core components of traditional telephony —
the terminal (telephone), PBX, the central office
switch, and the switching network — are struggling
and failing to keep up with the rate of innovations
on the Internet. The archaic telephony framework
with PBXs and Custom Local Area Signaling Services
(CLASS) switches providing Centrex and enhanced
residential services (call waiting, call forwarding,
6
TECHNOLOGY GUIDE
Next-Gen VoIP Services and Applications Using SIP and Java
7
This Technology Guide explains the architecture
of the new IP telephony model using Session
Initiation Protocol (SIP) and Java. The Guide also
demonstrates the power of SIP and Java in terms
of scalability, ease of use, and innovative services
and applications.
Figure 1B: First-generation
IP telephony architectures
LAN PBX
IP Centrex
Softswitch
"gatekeeper"
"call manager"
Architecture Models
Circuit-Switched and First-Generation IP
Telephony Architectures
The traditional telephony architecture is based
on a centralized processing model. First generation
IP telephony architecture uses a Media Gateway
Control Protocol (MGCP), Megaco, or vendor
proprietary protocols such as Cisco’s Skinny Client
Control Protocol (SCCP), which also are centralized
architectures similar to the traditional telephony.
Figure 1A: Traditional circuit-switched
telephony architectures
Both models have all of their intelligence in a
centralized switch or server, which performs all of
the telephony functions such as call setup, call
forwarding, conference calling, etc. All requests,
responses, and state changes must be processed
by the central switch/server with the end-station
being a dumb terminal.
The following are the salient characteristics of
the traditional telephony environment:
•
Archaic, Host-to-Dumb Terminal Architecture:
Voice service architecture has not changed for
generations. Today, PBX and Centrex services
are delivered using switches that contain all
application intelligence — just as mainframes
and minicomputers did for IBM 3270 or VT100
terminals in old computer systems.
•
Dumb Terminal — The Telephone:
Voice
service delivery assumes a dumb terminal in
telephony parlance — the telephone. The end-
PBX
Centrex
CLASS 5
switch
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