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Table of Contents
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Introduction 4
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Architecture Models 6
Technology Enablers for Next Generation
Technology Guides
Technology Guides
Voice Services and Applications 16
in this series. Next Generation IP Voice Services
in this series.
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
caller ID, etc.), cannot provide the types of features
Abstract
that are needed by a contemporary business in the
age of e-commerce. The traditional business
This Technology Guide explains the unique
telephony solutions are complicated, for both the
benefits of using the Web architectural model with
service administrators and the users. Because of the
SIP and Java as the enabling technologies for next
daunting complexity of PBX and CLASS/Centrex
generation IP voice services and applications.
user-interfaces, users typically know and use only a
Using the Web as a reference model for rapid
fraction of the total feature set.
innovation, the Guide contrasts the limitations of
Now imagine telephony services in the context
circuit-switched telephony and first generation
of the current business need. The users would still
VoIP architectures with the Web model. It
like to use a phone for making and receiving calls
summarizes limitations of centralized-processing
and playing voice-mail messages. However, they
models such as traditional telephony, MGCP, and
would also like to have the phone appliance
Megaco as compared to peer-to-peer models such as
integrated with a browser-based PC for managing
SIP and H.323.
phone books and seamlessly interfacing with other
This Technology Guide explains in more detail
applications, such as customer relationship
the unique benefits of using SIP for call control
management (CRM), sales force automation (SFA),
and Java for making phones intelligent. SIP is
supply chain management (SCM), time accounting,
compared with H.323 in terms of innovation,
etc. In other words, perform tasks most suitable
scalability, simplicity, ease of deployment, and
for the PC on the PC and those most suitable for
standardization. The guide also includes an
the telephone using a phone appliance and have
explanation of SIP concepts and operation. A
the two devices seamlessly integrated.
description of Java features supporting new voice-
Today s telephone just cannot deal with this new
services and applications is also included.
business imperative.
The Guide concludes with examples of new
In contrast, the Internet and Web-based
voice-services and applications made possible
communications have revolutionized the business
exclusively by SIP and Java.
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
Introduction
underpinning of critical business and IT solutions.
Just in the last few years alone the Internet and the
Traditional telephony has hit a wall in terms of
Web have generated more innovations than
innovation, ease of use, and cost reduction. The
traditional telephony has produced in its entire
core components of traditional telephony 
history. The next frontier for the Web is to apply the
the terminal (telephone), PBX, the central office
same degree of innovation to telephony.
switch, and the switching network  are struggling
Most market surveys have verified that IP
and failing to keep up with the rate of innovations
telephony is already supplementing traditional
on the Internet. The archaic telephony framework
telephony and it is expected that the IP telephony
with PBXs and Custom Local Area Signaling Services
architecture will ultimately replace the traditional
(CLASS) switches providing Centrex and enhanced
telephony model.
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
Figure 1B: First-generation
of the new IP telephony model using Session
IP telephony architectures
Initiation Protocol (SIP) and Java. The Guide also
IP Centrex
demonstrates the power of SIP and Java in terms LAN PBX
of scalability, ease of use, and innovative services
Softswitch
"gatekeeper"
and applications. "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.
Both models have all of their intelligence in a
Figure 1A: Traditional circuit-switched
centralized switch or server, which performs all of
telephony architectures
the telephony functions such as call setup, call
forwarding, conference calling, etc. All requests,
Centrex
PBX
responses, and state changes must be processed
CLASS 5
by the central switch/server with the end-station
switch
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-
8 TECHNOLOGY GUIDE Next-Gen VoIP Services and Applications Using SIP and Java 9
user interface for these services on the dumb
Web Architecture
telephone requires non-intuitive flash
The Web represents the most successful
sequences and star codes. No options exist for
application architecture in history. The Web
making telephony features easier to use and
features many intelligent servers located
increasing user productivity.
everywhere on the network and an intelligent,
" Hardware Specific Software: The voice features browser-based client device (a PC or a low cost
reside in software that is usually hardware- Internet appliance). It is the client device, not the
specific and/or proprietary. This environment server, that both initiates and controls all
requires highly-specialized software engineers communications with the server. When a user
that are expensive and hard to find. Even simply clicks on an icon to access an application,
simple software modifications require the the browser pulls content in the form of HTML
extensive regression testing of feature and applications (Java, Java script, Flash, Active X,
interaction. etc.) from the server and runs them on the PC.
There is a complete disaggregation of services in
" Limited Next-Generation Platforms: Next-
the Web model. Not only do the services come
generation voice service platforms still fall short
from different servers, they may be provided by
of business needs. Most first-generation IP
different and multiple service providers. Some of
telephony systems, for both service providers
the examples (shown in figure 2) include Yahoo
and enterprises, do exploit IP for transport and
for news; Amazon for shopping; MSN for instant
some feature a Java or XML software
messaging; ASP services (such as Corio) for
environment. However, this  open
customer relationship management (CRM), sales
environment is not easily made extensible by
force automation (SFA), enterprise resource
anyone other than the vendor or possibly a
planning (ERP); and MP3.com for music. An
service provider; certainly not the enterprise or
enterprise can outsource as few or as many
an independent software vender with a great
services as suits its business model.
idea. These systems, consequently, still
Key characteristics of the Web architecture
perpetuate the same 1960 s host-terminal
include:
architecture with a dumb telephone as the
endpoint: " Intelligent end devices (clients)
" The IP PBX is a host computer with all the " Distributed, intelligent servers (no central
smarts driving dumb IP phones. switch or server for services)
" VoIP gateways, softswitches, and their " An open architecture leading to innovation,
feature servers are merely physically rapid application development, and lower costs
distributed mainframes talking to dumb
terminals.
10 TECHNOLOGY GUIDE Next-Gen VoIP Services and Applications Using SIP and Java 11
voice-world are solely defined and developed by
Figure 2: Web application architecture
PBX and CLASS switch manufacturers, just as
mainframe applications were defined by the
CRM/SFA
Intelligent servers
vendors.
MSN Instant
MP3.com
Messenger
The PBX and CLASS switch vendors, their ideas,
amazon.com
their bureaucratic practices, and their business
doubleclick.com
motivations have held innovation in the voice-
yahoo.com
Virtualcart.com
world hostage. Voice features reside in software on
the switch that is hardware-specific and vendor-
specific. It is a proprietary environment that is not
openly extensible. Even modest new functions
require the onerous regression testing of feature
interaction.
Java Active X
The centralized, closed-software environment
MP3
offers no way for enterprises to add their own
HTML
Flash
Cookies innovations or enhancements to telephony
features, let alone individual users or software
Intelligent clients
developers with really good ideas. Some features
are impossible to implement because of the dumb-
telephone as the endpoint. Consequently,
innovation is and will remain dead, especially
when compared to the revolutions on the Web.
Comparing the Architectures
The Web has revolutionized the world of
Web
business. It has enabled a whole new business
Innovation on the Web occurs at the edges of
paradigm in the form of e-business, portals,
the network, where anyone  businesses and
e-tailers, and collaborative applications. The Web
individuals can create Web sites that are
has enabled businesses to reach business partners
immediately open for other users to interact with.
and customers worldwide with a click of the
On the Web, in contrast to traditional telephony, a
mouse. Telephony services must change
new page or  feature can be created in a few
dramatically to become a functional member of
minutes. More importantly, the Web page can be
this business revolution. However, given their
conceived, created, delivered and personalized by
limitations, it is virtually impossible for the current
anyone  yahoo, e-bay, GE, a company, an
telephony architectures to satisfy emerging
individual, their kids or their grandparents. Several
requirements.
million Web sites are in existence today, up from a
few thousand in 1993. These sites satisfy
Innovation
everyone s personal and business needs for news,
Traditional and First-Generation IP Telephony
buying, entertainment, chat, sports, sex, etc.
The telephone was invented more than 125
regardless of gender, race, religion, ethnic
years ago. Since then it has enabled people to talk
background, industry and occupation.
and do only a handful of other things, like use
Amazon.com would not have happened if the
voice mail. All of the features and services in the
world needed to rely on the data communications
12 TECHNOLOGY GUIDE Next-Gen VoIP Services and Applications Using SIP and Java 13
vendors such as Alcatel, Cisco, Lucent, or Nortel to browser s graphical user interface means that users
invent the  service and add the features to a do not have to memorize features as in the world
router or a switch. of telephony. The use of any Web site is an
intuitive discovery process, performed simply by
Ease of Use
pointing and clicking at images and words.
Traditional and First-Generation IP Telephony
Scalability and Capacity
For most telephone users, cryptic impossible-to-
remember flash sequences and * codes are the Traditional and First Generation IP Telephony
interface to thousands of PBX and CLASS features. In the telephony world, big centralized boxes
For the fortunate few with block character have all the smarts. Whenever the telephone, the
displays, even IBM 3270 and VT100 terminals  terminal in the parlance of telephone equipment
appear attractive. vendors, sends a flash sequence or * code, it s the
Users don t know what voice features exist and PBX or CLASS switch that figures out what it
if they do, they do not know how to use them. means. The PBX or the switch also must actively
While most voice service platforms such as PBX manage each and every call. Consequently, it just
and CLASS switches offer hundreds or thousands does not scale. Support for just one more user
of features (300-400 features in a typical PBX, may end-up requiring a hugely expensive
3000-4000 in a CLASS 5 switch), most users replacement or addition.
typically don t know any more than just a few 
Web
transfer, hold, last number redial. In research
A Web site, however, can support millions of
conducted by WorldCom, 9 out of 10 executives
users. Scalability is achieved not only through the
could not even transfer a call without resorting to
connection-less nature of IP and by adding more
the  help scream   Do I dial  flash first and
and bigger servers to the Web site. Scalability is
then the number, or the other way around? Trying
also achieved by exploiting an intelligent endpoint
to set-up just a 3-party conference call over a PBX
 the browser-based PC. In fact, it s the browser
is even a bigger nightmare. It s no wonder that the
software that interprets Web objects and puts a
assisted conference calling businesses of AT&T,
Web page together.
Sprint and WorldCom are so big and profitable.
For example, in accessing a typical e-commerce
For many, the most difficult part of changing jobs
site, it s the browser, not a server, that:
is learning a new phone system.  What do I dial
to get an outside line? Consequently, for the vast
" Retrieves and displays the source HTML page
majority, ignorance is bliss, yet very expensive in
and embedded product images individually
user productivity.
" Retrieves and runs a Java applet, Java script,
Web Flash, Active X or other application
On the Web, millions of sites with billions, components
perhaps trillions, of pages can be easily navigated
" Retrieves and displays a dynamic advertisement
by pointing and clicking at pictures or words
from DoubleClick.com
displayed on an intelligent, browser-based PC.
In contrast to telephony feature usage, anyone " Retrieves shopping cart services from a
from kids to their great grandparents can easily ShoppingCart.com
discover and use any site on the Web. The
14 TECHNOLOGY GUIDE Next-Gen VoIP Services and Applications Using SIP and Java 15
" Stores cookies to identify users and maintain An enterprise has the option of providing PBX
states services locally through a premises-based system
device or these could be outsourced to a network-
" Encrypts credit card numbers
based service. The outsourced service not only
eliminates capital costs but may actually provide
Manageability
richer services than those available from a PBX.
Traditional and First-Generation IP Telephony
The figure also shows some illustrative services
An expert  the equivalent of the proverbial
such as unified messaging, presence messaging,
rocket scientist  must perform all maintenance
instant messaging, and CRM integration, all of which
and management tasks for the PBX or the switch.
can be provided by separate service providers
Tools for managing moves/adds/changes tend to
offering best-of-breed solutions for an enterprise s or
be horrendous and, consequently, administrators
even an individual user s specific requirements.
learn only the basic coping skills. This makes it
extremely costly to administer the switch.
According to some estimates it can cost as much
Figure 3: Web architecture for next-generation
as $300-$500 per PBX move/add/change. For a
voice services and applications
Centrex line, it can take weeks for a change to be
Intelligent servers
implemented by the telephone company. Audio
Hosted
Auctions
CRM/SFA PBX
service
Web
Presence & IM
Self-service by users is the normal operative
IP PBX
model here  for registration, buying things,
Unified Messaging
PSTN
personalizing info, etc.
gateways
Every office device including printers, copiers,
and now intelligent IP phones have a built-in Web
server that enables remote configuration over the
net via browser interface.
Every office device and home appliance is
becoming more intelligent and capable of running
Phone-to-phone
Java
data & app exchange
PC app
HTML
automated diagnostics, reporting the findings, and integration
MP3
ordering replacements before service is disrupted.
Exploiting the Web Architecture for Next
Intelligent clients
Generation Voice Services and Applications
Figure 3 shows what telephony would look like
PCs and other phones are simply resources on
if migrated to a Web-like architecture. In this
the network that provide services to users. In this
model, services and applications are resources on
model, the PC may provide services for the phone
the network and are accessed and controlled by
such as integration with the desktop applications
the phone and not by a central-switch or a
or the phone may provide services for the PC such
gatekeeper. Nor does a central-switch or
as causing the phone to ring and automating
gatekeeper control what the phone can do.
conference calls in Microsoft Outlook.
16 TECHNOLOGY GUIDE Next-Gen VoIP Services and Applications Using SIP and Java 17
Phone Intelligence Technology
Technology Enablers for
An ability to support small footprint applications
is the key for incorporating intelligence in phones.
Next Generation Voice
A powerful yet easy to use programming language
Services and Applications
used widely for Web-enabling Internet appliances
is required. In addition to rich functionality for
Clearly, while the model in figure 3 is quite
traditional Web applications, features developed
pedestrian in the Web world, it is quite
specifically for telephony and security are
revolutionary in the context of traditional
mandatory. Lastly, the language must already be
telephony. The components needed to implement
used by hundreds of thousands of programmers
this model for telephony are as follows:
worldwide in order for innovation to happen
rapidly.
Intelligent Servers
These are distributed resources that interact with
Extensible, Scalable Call Control Protocol
intelligent clients (PCs and phones). In terms of
A call control protocol is used for call related
hardware and software, these servers are standard
functions such as setting up, monitoring, and
Unix, Linux, and Microsoft Windows platforms.
terminating calls. However, in the new IP
Compared to traditional PBXs, these servers offer
telephony model, the call control protocol must
choices of multiple vendors and competitive
differ from traditional telephony and the first
pricing with an open applications development
generation IP telephony protocols. For maximum
environment.
scalability, the new call control protocol must
support peer-to-peer communications whereby
Intelligent Phones
two or more phones can set up and communicate
These phones should provide much more than
directly without requiring anything more than
incoming call ringing. In order to maintain their
locations services from a call control server. In
independence from a central switch, they must
addition, the protocol must allow the peer-to-peer
also provide local capabilities such as call hold,
exchange of applications and data in addition to
transfer, forwarding, redial, caller ID, multi-party
voice communications.
conferencing, and many other traditional
The call control protocol must support a wide
telephony features.
range of environments  from home-office to the
The intelligent phones should be thin-client
largest enterprise and from the smallest to the
computing devices that can interoperate with PCs
largest services provider. Thus, the protocol must
and servers on the network. These devices must
be highly scalable as well as cost effective in a
support dynamic loading and management of
diverse range of configurations. Since it is not
applications such as Java applets. For ease of use,
possible to predict all future applications of IP
they should incorporate functions such as
telephony, the protocol must also be extensible in
graphical and audio helpers to ease the use of
order to accommodate unforeseen requirements.
traditional and next generation applications.
18 TECHNOLOGY GUIDE Next-Gen VoIP Services and Applications Using SIP and Java 19
H.323, the older of the protocols, was originally
SIP (Session Initiation Protocol)  The
designed for video conferencing over the LAN.
Call Control Protocol
Since then it has been morphed and used to
SIP introduces the benefits of the Web
support voice and video over then WAN as well.
architecture to IP telephony. It provides a
SIP, however, was designed from the beginning for
powerful, extensible, scalable, and easy-to-deploy
multimedia sessions and conferences over the
protocol for call control and media exchange.
WAN. Because of these differences in their design
Several standards are available for building IP
objectives, SIP offers numerous compelling
telephony solutions. These include the Session
advantages in the areas of extensibility, scalability,
Initiation Protocol (SIP) from the IETF; ITU-T
and ease of deployment over H.323.
H.323, an ITU-T umbrella standard; Media
Today there are more products available
Gateway Control Protocol (MGCP) from IETF;
supporting H.323 than SIP. However, since its
Media Gateway Control (Megaco), a joint protocol
introduction, SIP is rapidly becoming the preferred
by IETF and ITU-T; and proprietary protocols such
protocol. A January 2001 survey of Voice over IP
as Cisco s Skinny Client Control Protocol (SCCP).
vendors in Network World found that while 75%
A high-level comparison of these protocols is
of the vendors offered products based on one of
included in table 1.
the four H.323 versions, an approximately equal
number of them were already planning to offer
Table1: IP Telephony standards
SIP-based products by June 2001. However, the
SIP H.323 MGCP MEGACO PRO-
more telling statistic was that less than 25% of the
PRIETARY
vendors were planning to upgrade their products
Architectural Peer-to-peer Peer-to-peer Master/ Master/ Master/
from H.323 Version 2 to Version 3 and even fewer
Model slave slave slave
Media types Voice, video, Voice, video, Voice Voice, Voice
to Version 4, the latest version of H.323. According
data limited data video
to the same survey, most vendors expected H.323
Network Intra, Extra, Intra, Extra, Intranet Intranet Intranet
to become a legacy protocol. In contrast, the list
scope and Internet and Internet only only only
of vendors supporting or planning to support SIP
Extensibility High Low Medium Medium Low
Scalability High Medium Low Low Low is growing rapidly. Service providers embracing
Ease of High Low Medium Medium Medium
SIP include WorldCom, Level 3, Net2Phone, Telia,
deployment
Webley, Ibasis, LipStream, and TalkingNets as of
Standardization IETF ITU-T IETF IETF and None
March 2001 with many more anticipated.
ITU-T
The reasons for the rapid ascendancy of SIP
become obvious when we compare it with H.323
Why SIP
in the areas of innovation, scalability, ease of
Of the protocols listed in table 1, only SIP and
deployment, manageability, and the standardization
H.323 are peer-to-peer protocols. MGCP, Megaco
process. Appendix A provides additional details on
and Cisco s proprietary SCCP represent the old
SIP concepts, definitions, and operation.
centralized model and suffer from this model s
limitations discussed earlier. Thus, the real choice
for a protocol with Web-like benefits comes down
to one of the peer-to-peer protocols  H.323 or
SIP.
20 TECHNOLOGY GUIDE Next-Gen VoIP Services and Applications Using SIP and Java 21
Innovation protocols within H.323. These include Registration,
SIP enables new services and applications not Admission and Status (RAS), Q.931 for call control,
possible with H.323 (or other IP telephony and H.245 for transmission of non-telephony
protocols) and easily empowers service providers, signals on the line. As shown in the tables, SIP has
application developers, and enterprises to create a total of 5 methods (commands) and 8 responses
unique, differentiated services and applications. and H.323 has 21 commands/messages across the
For example, SIP uses a simple text-based three protocols. SIP can be implemented as a
encapsulation (based on the Internet standard stateless protocol and does not need to maintain
MIME) which enables it to transmit data and any call states, which further increases scalability
application programs with the voice call, making it of SIP. SIP also shows a substantially higher
easy to send business cards, photos, and/or MP3 efficiency than H.323 during call set-up by using
encoded information during a call. approximately 50% fewer messages. Figures 4 and
SIP also supports third-party call control through 5 show call set-up messages for H.323 and SIP,
simple applications to modify SIP messages and respectively. While H.323 requires a total 13
enable functions such as sending office calls to a message exchanges, SIP requires only 7
home phone after 5:00 PM or forwarding video exchanges.
calls to a PC. Lastly, SIP envisions the need to
SIP Methods and Response Codes
accommodate extensions  new protocol headers,
methods, bodies and parameters, to implement
Table 2: SIP methods
new and innovative applications. By design not all
products are required to support these extensions
SIP METHODS
(just the endpoints) servers or phones that want to
INVITE User or service is being invited to participate in a session.
use them.
ACK Client has received a final response to an INVITE request.
OPTIONS Server being queried about capabilities.
Scalability
BYE User agent client indicates to server to release the call.
Being peer-to-peer protocols, both SIP and
CANCEL Cancels a pending request.
H.323 eliminate the need for central servers to
REGISTER Client registers address with a SIP server.
control everything. Peer-to-peer protocols reduce
costs of network and server infrastructure
equipment necessary to support a user population
Table 3: SIP response codes
of a given size.
SIP RESPONSE CODES
Within peer-to-peer protocols, SIP is a much
1xx Informational: Request received, continuing to process request.
more efficient and less complex protocol,
2xx Success: Action successfully received, understood and accepted.
therefore, more scalable than H.323. H.323 is
3xx Redirection: Further action required to complete request.
actually an umbrella specification that includes
4xx Client Error: Request contains bad syntax or cannot be executed
several protocols from other ITU-T standards.
at server.
Tables 2  4 cover three categories of such 5xx Server Error: Server failed to execute an apparently valid request.
6xx Global Failure: Request cannot be executed at any server.
22 TECHNOLOGY GUIDE Next-Gen VoIP Services and Applications Using SIP and Java 23
H.323 Commands/Messages Table 6: H323/H.248 commands and responses
H.248
Table 4: H.323 RAS commands and responses
Command/Message Function
RAS
Master-Slave Determination Determines which terminal is the master and
which is the slave. Possible replies:
Command/Message Function
Acknowledge, Reject, Release (in case of a
RegistrationRequest (RRQ) Request from a terminal or gateway to register
time out).
with a gatekeeper. Gatekeeper either confirms
Terminal Capability Set Contains information about a terminal s
or rejects (RCF or RRJ)
capability to transmit and receive multimedia
AdmissionRequest (ARQ) Request for access to packet network from
streams. Possible replies: Acknowledge,
terminal to gatekeeper. Gatekeeper either
Reject, Release.
confirms or rejects (ACF or ARJ)
Open Logical Channel Opens a logical channel for transport of
BandwidthRequest (BRQ) Request for changed bandwidth allocation,
audiovisual and data information. Possible
from terminal to gatekeeper. Gatekeeper either
replies: Acknowledge, Reject, Confirm.
confirms or rejects (BCF or BRJ)
Close Logical Channel Closes a logical channel between two
DisengageRequest (DRQ) If sent from endpoint to gatekeeper, DRQ
endpoints. Possible replies: Acknowledge.
informs gatekeeper that endpoint is being
Request Mode Used by a receive terminal to request
dropped; if sent from gatekeeper to endpoint,
particular modes of transmission from a
DRQ call to be dropped. Gatekeeper either
transit terminal. General mode types include
confirms or rejects (DCF or DRJ). If DRQ sent
VideoMode, AudioMode, DataMode, and
by gatekeeper, endpoint must reply with DCF.
Encryption Mode. Possible replies:
InfoRequest(IRQ) Request for status information from
Acknowledge, Reject, Release.
gatekeeper to terminal.
Send Terminal Capability Set Commands the far-end terminal to indicate its
InfoRequestResponse (IRR) Response to IRQ. May be sent unsolicited by
transmit and receive capabilities by sending
terminal to gatekeeper at predetermined intervals.
one or more Terminal Capability Sets.
RAS Timers and Request Recommended default timeout values for
End Session Command Indicates the end of the H.245 session. After
in Progress (RIP) response to RAS messages and subsequent
transmission, the terminal will not send any
retry counts if response is not received.
more H.245 messages.
Table 5: H.323/Q.931 commands and responses
Ease of Deployment
Deploying and supporting SIP is similar to
Q.931
HTTP. It uses standard protocols and functions,
Command/Message Function
which already exist in the current IP networks and
Altering Called user has been alerted   phone is ringing .
Sent by called user.
are well understood by system administrators and
Call Proceeding Requested call establishment has been initiated and
technical support personnel. SIP has the following
no more call establishment information will be
HTTP characteristics:
accepted. Sent by called user.
Connect Acceptance of call by called entity. Sent from called
" Standard Internet addressing: SIP uses
entity to calling entity.
standard IP addressing format for both names
Setup Indicates a calling H.323 entity s desire to set up a
connection to the called entity.
and addresses, e.g., sip:[e-mail]
Release Complete Indicates release of call if H.225.0 (0.931) call
or sip:[e-mail]
signaling channel is open. Afterwards, call reference
value can be reused. Sent by a terminal
" Clear text protocol: SIP uses clear text for its
Status Responds to an unknown call signaling message or
protocol encapsulation unlike H.323, which
to a Status Inquiry message. Provides call state
information.
uses binary encoding, making SIP easier to
Status Inquiry Requests call status. Can be sent by endpoint or
diagnose and troubleshoot.
gatekeeper to another endpoint.
24 TECHNOLOGY GUIDE Next-Gen VoIP Services and Applications Using SIP and Java 25
" Simple error messages: SIP uses familiar error-
Figure 5: H.323 Call set-up sequence
messages with prefixes such as 10x, 20x, etc.
" Leverages other Internet protocols: SIP uses
Endpoint 1 Gatekeeper
Endpoint 2
other familiar Internet protocols such as MIME
Admission
1
Request
and Session Description Protocol (SDP), again
Admission
eliminating the need for new technical training
Confirm
2
or expertise.
3
Setup
Call Proceeding
4
Admission Request
5
Figure 4: SIP Operation in Proxy Mode
6
Admission Confirm
Altering Connecting
7
Site 1 Site 2
Terminal Capability Set
8
Master/Slave Determination
Endpoint Location Client 2
Proxy
1@Site 1 Server @Site 2
Terminal Capability Set + Ack
9
Master/Slave Determination + Ack
INVITE
Endpoint 2
@Site 2
1 10
2 Terminal Capability Set Ack
Master/Slave Determination Ack
Endpoint 2
11
Open Logical Channel
Client 2
3
@Site 2
Open Logical Channel + Ack
12
INVITE
4
Endpoint 2
@Site 2 13
Open Logical Channel Ack
100 Trying Media (RTP)
200 OK Close Logical Channel
End Session Command
100 Trying
5
200 OK
Close Logical Channel + Ack
6
End Session Command
Release Complete
7
Ack
Disengage Request Disengage Request
Ack
Disengage Confirm Disengage Confirm
Endpoint 1 Gatekeeper
Endpoint 2
RAS 0.931 H.245
Standardization
The ITU-T, organized under the auspices of the
United Nations, defines traditional telephony and
H.323 standards. It is a slow moving body with a
highly political process. Participation in ITU-T
activities is limited to paid members. Most of
26 TECHNOLOGY GUIDE Next-Gen VoIP Services and Applications Using SIP and Java 27
ITU-T documents are written using very dense can run on minimalist appliances. Simple Java
language, which make it virtually impossible for applets can be developed in anywhere from a few
the uninitiated to fathom their intent. Most ITU-T minutes to a few hours. Key features of Java
standards tend to be very complex. For example, include:
H.323 specification with its co-requisite protocols
Network Orientation
runs some 700 pages compared to about 150
Java applications, called applets, run on thin-
pages for SIP. The ITU-T specifications are not
clients. Java applets are network-aware and can
freely available and have to be purchased. As of
open and access objects across the Internet via
February 2001, you could not even buy the H.323
URLs. The Remote Method Invocation (RMI)
specifications from the ITU-T bookstore because
feature of Java allows the building of distributed
ITU-T still had not made them available for
applications. RMI-based applications can connect
purchase.
to other Java applications as well as legacy
In contrast, the Internet standardization process is
applications.
geared toward rapid innovation. It has an open and
Java Naming and Directory Interface (JNDI)
democratic process which draws architects from the
provides a unified interface to multiple
industry, academia, government, and individuals
heterogeneous naming and directory services
who are experts in specific technology areas. All
including LDAP directories. JNDI enables seamless
Internet specifications are available for free to
connectivity to these services. Developers can
anyone and can be simply downloaded from the
build powerful and portable directory-enabled Java
Internet. Lastly, the Internet standardization is
applications using this industry-standard interface.
rooted in the  proof-of-concept , i.e., there must
Java Database Connector (JDBC) is an application
exist a prototype implementation for a standard to
programming interface (API) that provides cross-
achieve approved status. The standard documents
DBMS connectivity to a wide range of SQL
often include model codes to document the
databases. Using JDBC, an application can establish
standard. Additionally, almost always, the actual
connectivity with nearly any enterprise or service
code to implement a prototype is available on the
provider database from a Java-enabled phone.
Internet for free download and use.
Java also features specifications and supports
products which can automate the process of
Java  the Applications Engine
distributing new versions of applications over the
A key element of the proposed architecture for
network. This includes Java Management
the next-generation IP voice services and
Extensions (JMX), the specification, and Java
applications is an intelligent phone. Java is the
Dynamic Management Kit (JDMK), Sun s product
ideal application engine technology for intelligent
which implements this specification.
phones. Java has already proven itself as one of
the most innovative technologies fueling the Powerful APIs for Telephony and Speech Applications
Internet innovations and Java applications that are Java has two APIs specially designed for
at the core of the contemporary Web-pages. telephony and speech applications:
Java applications do not reside permanently on
" Java Telephony API (JTAPI) defines interface to
thin-clients, thus, do not consume any resources
access the following functional areas: call
on the phone when not needed. They are typically
control, telephone physical device control,
designed with very small footprints so that they
28 TECHNOLOGY GUIDE Next-Gen VoIP Services and Applications Using SIP and Java 29
media services, and telephony administrative processor that is running Java runtime environment.
services. JTAPI functions can be used with both Consequently, a Java applet written for an IP phone
wired and wireless phones and its core appliance can run without modification on a PC-
functions can be extended to build applications based softphone supporting Java.
such as call logging and tracking, auto-dialing,
Ease of Development
screen-based telephone applications, call
Sun makes developing applications quick and
routing applications, automated attendants,
easy with great tools in their Java Development
interactive Voice Response (IVR) systems call
Kit. In addition, Java is supported by numerous
management center, voicemail, etc.
tools, components, and applications that are
" Java Sound API (JSAPI) allows developers to
available from many vendors. In fact, many are
incorporate speech technology into user
available for free on the Internet. These tools
interface for their Java applets and applications.
include application and user interface (UI)
This API specifies a cross-platform interface to
components, authoring and workflow tools, and
support command and control recognizers,
integrated development environments. A wide
dictation systems and speech synthesizers.
variety of Java training options ranging from
classrooms to web-based are also available. Lastly,
Security
due to Java s tremendous popularity, Java software
Java has a built-in security framework or
engineers are readily available on permanent or
 sandbox that can protect basic phone operation
contract basis to assist in development.
like making and receiving calls from rogue or
misbehaving applets. Java enables the construction
of virus-free, tamper-free appliances like phones. It
also incorporates authentication techniques based
Next Generation IP Voice
on public-key encryption. Java s security features
Services and Applications
also allow enterprises to control access to
resources via policy-based permissions.
SIP and Java also enable a whole new
generation of applications which are impossible
Support for a Wide Variety of Devices and User
with other telephony architectures. These
Interfaces
applications can generally be divided into three
Java applets can run on virtually any platform
categories:
due to their platform independence. A Java applet
can be written once and run on virtually any
" Personal productivity applications
operating system including cell phone OS, HP UX,
" Occupation specific and industry specific
IBM AIX, Palm OS, Sun Solaris, VxWorks, Microsoft
applications
Windows, and various other varieties of Unix and
Linux systems. To enable a Java application to
" Web-telephony integration (WTI) applications
execute anywhere on the network, the Java
Listed below are a few examples of each.
compiler generates an architecture-neutral object file
and the compiled code is executable on any
30 TECHNOLOGY GUIDE Next-Gen VoIP Services and Applications Using SIP and Java 31
Automated conference calling  create conference
Personal Productivity Applications
call appointments in Microsoft Outlook. The
Electronic business cards  send an enriched
application would automatically set-up the
electronic virtual business card (vCard) including
conference call at the specified time.
photo and audio file automatically with every call
as caller ID information (or selectively during the Distinctive rings  play unique rings from any
middle of call). This information can be added sound file based on caller ID or personal directory
into any personal contact database such as information. Separate rings could be set up for a
Microsoft Outlook, or a corporate CRM, or a boss, spouse, kids, or anyone else.
Supply Chain Management (SCM) database with
the push of a button.
Industry and Occupation-Specific
Presence and instant messaging  use an instant
Applications
messenger service to determine when
Telecommuters  get all office telephony
geographically distributed colleagues are available
functionality at home  extension dialing, call
for a quick conference call with a customer.
transfer, intranet intercom, call billing, etc.
Simply click or automatically  camp on your
 buddy list to create the conference call.
Consultants  start the  clock automatically for
time accounting or billing when picking up the
Call filters  have every call from that very
phone or dialing the number of a client using
important customer ring at every phone 
caller ID or contact database information.
business phone, cell phone, home phone, vacation
phone, etc. The call will get completed to the first
Sales reps  integrate voice and data information
device from where the user picks up the call.
collected during a call with sales force automation
applications such as ACT or Goldmine, or an ASP
Phone book  use multiple phone books 
like sales.com.
corporate, personal, Internet, etc., on the phone
and simply point to an entry to make the call. The
Public relations  click-to-dial personalized and
phone books can be synchronized with the data
up-to-date press, analyst and vendor contact lists,
on a PC or any server.
and track and report time on the phone by client
using a public relations ASP like mediamap.com.
Personalized music on-hold  play personalized
announcements or music from a favorite MP3
recording or Internet radio station while callers are
Web-Telephony Integration (WTI)
on hold.
Applications:
Voice tag elimination  deliver customized
Auction site for purchasing agents of electronic
messages to people trying to contact busy contacts
components  create a live audio auction for
and eliminate phone tag.
excess DRAM inventory and use the  heat of a
real-time event to pump-up prices and the
auctioneer s commission. Use Java applets on the
32 TECHNOLOGY GUIDE Next-Gen VoIP Services and Applications Using SIP and Java 33
phone to manage the bidding process and to track
Summary
who  raised a hand to bid first, etc.
Virtual call center ASP  support the integrated The Web has revolutionized the world of
voice and data requirements of call center agents business. Traditional telephony, however, cannot
working from their homes. fulfill the needs of the emergent e-business model.
The traditional telephony model is constrained by
Airlines reservations  use a Java applet to
an inflexible and inefficient architecture based on
visually display interactive voice response (IVR)
centralized processing and the dumb terminal.
options rather than forcing users to wait through
This environment inhibits innovation, is nearly
very long recorded instructions and go through
impossible to use, and simply perpetuates the old,
multi-level menus requiring the use of a telephone
cumbersome, and limited functionality services.
keypad.
IP telephony needs to embrace the Web
architectural model in order to achieve rapid and
cost effective innovation. Old definitions of
 enhanced services and features do not come
anywhere near even the simplest applications made
possible by technologies such as SIP and Java.
SIP, coupled with Java, can bring the same
revolutionary innovations and mindset to the
world of IP telephony that the Web has brought to
IT and the data world.
34 GLOSSARY Next-Gen VoIP Services and Applications Using SIP and Java 35
API: Application Programming Interface, a set of IVR: Interactive Voice Response, a system used for
programming functions and calls supported by a generating voice prompts and menus and for
language or a software product. APIs are used by accepting and processing user responses.
software developers to develop programs in a
specific language or to enhance or extend the JTAPI: Java Telephony API, an extension to Java that
capabilities of a product. provides telephony functions such as call control.
ASN.1: Abstract Syntax Notation 1, an object-oriented JSAPI: Java Speech API, an extension to Java that
language used by various architectures such as provides functions for controlling dictation
OSI, ITU-T, and SNMP to define objects including systems and speech synthesizers
data structures.
JNDI: Java Naming and Directory Interface, an
ASP: Application Services Provider, a service provider extension to Java that provides a unified
that provides applications over a network with a interface to multiple naming and directory
usage-based fee. services.
CLASS: Custom Local Area Signaling Services, services Megaco: Media Gateway Control, a VoIP protocol jointly
such as caller ID and ring back provided by a developed by ITU-T and IETF. It uses softswitches
telephone company. Devices in the telephone and gatekeepers for central control of calls and
central office that provide such services are conferences.
called CLASS switches.
MGCP: Media Gateway Control Protocol, a VoIP protocol
CPU: Central Processing Unit, the arithmetic and logic developed by and IETF. It uses softswitches and
unit in a computer. Examples include the Intel gatekeepers for central control of calls and
Pentium family, the AMD Atheon, and the IBM conferences.
RISC processors.
MIME: Multipurpose Internet Mail Extensions, an
CRM: Customer Relationship Management software, Internet standard used for encapsulating e-mail
used with application such as ACT or Goldmine to messages in clear text.
keep track of customer contacts and sales
information. PBX: Private Branch Exchange, a customer premise
based telephone switch for intra-campus and
H.323: An ITU-T specification for multimedia outside telephone calls.
conferences over IP for LAN attached stations. It
is a peer-to-peer protocol as opposed to MGCP PSTN: Public switched Telephone Network, a general
and Megaco which require central control reference to telephone networks using circuit
switching and time division multiplexing.
HTTP: Hyper Text Transfer Protocol, used for encoding
and transferring Web objects from Web servers Q.931: An ITU-T Call control protocol for ISDN, also used
to Web browsers. in H.323. It defines procedures for setting up and
clearing calls.
36 GLOSSARY Next-Gen VoIP Services and Applications Using SIP and Java 37
RAS: Registration, Admission, and Status, a component SIP: Session Initiation Protocol, IETF standard for
of H.323, defines procedures whereby users can peer-to-peer multimedia sessions and IP
register themselves with a gatekeeper as a telephony. An alternative to the ITU-T H.323
preliminary step to setting up a call. protocol.
RMI: Remote Method Invocation, a component part of VoIP: Voice over IP, a general reference to several
Java, allows building of distributed applications technologies and protocols that allow voice
that can connect to other Java applications as telephony implementation over IP networks.
well as legacy applications. Examples of components and technologies that
enable VoIP include codecs, IP PBXs,
RTCP: RTP Control Protocol, control protocol for RTP softswitches, gateways, H.323, SIP, MGCP, and
that allows multimedia session partners to Megaco.
monitor the quality of their sessions.
RTP: Real-time Transport Protocol, an IP standard for
encapsulating multimedia streams for
transmission over IP networks. It includes
information such as packet timestamps to help
implement quality of service for a session.
SCCP: Skinny Client Control Protocol, a Cisco proprietary
protocol for voice over IP that uses central
control with gatekeeper-like functions.
SCM: Supply Chain Management, used in reference to
application programs used for managing
purchases and suppliers.
SDP: Session Description Protocol, an IETF standard to
advertise multimedia conferences. SDP is
intended for describing multimedia sessions for
the purposes of session announcement, session
invitation, and other forms of multimedia session
initiation.
SFA: Sales Force Automation, used in references to
application programs used for managing sales
activities such as capturing customer contact
information, generating contracts, and generating
order forms.
38 APPENDIX A Next-Gen VoIP Services and Applications Using SIP and Java 39
cases of a multicast conference, a full-mesh
Session Initiation Protocol
conference and a two-party  phone call , as well
as combinations of these. Any number of calls can
(SIP) Concepts and Operation
be used to create a conference.
SIP is an Internet protocol defined under
Call
Request for Comment 2543 (RFC 2543). SIP is not
A call consists of all participants in a conference
just for voice communications  it supports data
invited by a common source. A SIP call is
and multimedia in its core specification.
identified by a globally unique call-ID.
Figure 6: SIP and other Internet Protocols
Figure 7: SIP clients and servers
Kerb
Gopher SMTP Telnet FTP SIP SNMP RPC
User Agent
User Agent
Client
TCP UDP Client
User Agent
User Agent
Server
IP Server
LAN or WAN Interface
SIP Servers:
Proxy
Redirect
Location
In TCP/IP terminology, as shown in figure 6, SIP
Registrar
is an application level protocol and runs over UDP
but may use TCP. SIP is based on existing and
well-understood Internet protocols and extends
SIP Components
them to support IP telephony.
User Agent Clients and Servers
SIP Concepts
A user agent is a program that runs on a SIP
Session
device (e.g., the phone). It contains a client
A SIP session is a multimedia session consisting
function and a server function.
of a set of multimedia senders and receivers and
The user agent client (UAC) is a program that
the data streams flowing from senders to receivers.
initiates SIP requests such as initiating a call. A
Session is the basic building block in SIP. All calls
UAC is also known as the calling user agent
and conferences are established by setting up
A user agent server (UAS) is a program that
sessions among users.
receives SIP requests such as an incoming call and
sends back responses to those requests. A UAS is
Conference
also known as the called user agent.
A conference is a multimedia session, identified
by a common session description. A conference
can have zero or more members and includes the
40 APPENDIX A Next-Gen VoIP Services and Applications Using SIP and Java 41
SIP Servers rwhois, LDAP, multicast-based protocols or
operating-system dependent mechanisms to
Location Server
actively determine the end system where a user
A location server is used to obtain information
might be reachable.
about a callee s possible location. A location is the
IP address of the domain where a user is located.
SIP Addressing
To locate a user, the name of the user is sent to
SIP uses traditional Internet names as addresses,
the location server and the location server returns
which consist of a user name and a domain name.
zero or multiple locations (IP addresses orf
This is an important issue because it means that
domains) where a callee may be found. If the
the existing Internet naming, addressing, and
caller already knows the IP address of the
routing services can process SIP addresses without
destination server, the caller can directly contact
modifications. Examples of SIP addresses include:
the callee s UAS.
SIP:[e-mail]
Proxy Servers
SIP:user@25.16.10.8
A proxy server is an intermediary program that
acts as both a server and a client for the purpose
SIP:[e-mail]
of making requests on behalf of other clients.
These addresses are similar to HTTP URL
Requests are serviced internally by a proxy server
addresses except that they start with SIP instead of
or forwarded, possibly after translation, to other
HTTP. The first example shows a user being
servers. A proxy interprets and, if necessary,
identified via a typical e-mail address. The second
rewrites a request message before forwarding it.
example shows an address where the IP address
Redirect Server
of the destination is known. The last example
A redirect server is a server that accepts a SIP
shows how we could use a phone number-like
request, maps the address into zero or more new
address under SIP.
addresses and returns these addresses to the client.
The major advantages of this addressing scheme
Unlike a proxy server, it does not initiate its own
are:
SIP requests. Unlike a user agent server, it does
" It invents no new directory structure and can
not accept calls.
be processed by existing IP servers
Registrar
" Users can use familiar e-mail or URL addresses
A Registrar is a server that accepts REGISTER
to make phone calls and have one less thing to
requests. A client uses the REGISTER request to let
remember, the phone number.
a proxy or redirect server know the location
where the client can be reached. It provides a
Domain Name Services (DNS)
means whereby users can register their locations
DNS is a standard Internet service to convert
with a SIP server dynamically. As users move to
user names, e.g., [e-mail] into IP
different locations, they can register their new
addresses, e.g., 172.30.10.20, that can be used for
locations with the local location server.
finding user locations and routing calls. Because
To supplement information obtained through
SIP uses standard IP naming and addressing, we
user registrations, a location server may also use
are able to use existing, standard DNS services for
one or more TCP/IP protocols, such as finger,
SIP without any modification.
42 APPENDIX A Next-Gen VoIP Services and Applications Using SIP and Java 43
SIP Messages When the callee sends a response to the INVITE
SIP messages include SIP methods and request agreeing to participate in the call, the
responses to the methods. These are listed in caller sends an ACK to confirm callee s response.
tables 5 and 6.
Call Setup Using A Proxy Server
SIP Message Encapsulation  MIME To initiate a SIP call, a caller first locates the
Multipurpose Internet Mail Extensions (MIME) is appropriate proxy server and then sends a SIP
the Internet standard for describing different types invitation request to the proxy server. The location
of content on the Internet, including video and of the proxy server is locally configured on the
image types. It is already used by HTTP for user station. The proxy server can also be
composing Web pages and by e-mail systems for discovered automatically by the caller using a
encoding e-mail messages. SIP uses this well- variety of mechanisms such as DHCP options, DNS
established standard for encoding information, SRV and others. Instead of directly sending the call
eliminating the need for inventing a new to the intended callee, the proxy server may
technique for encoding voice and multimedia over redirect the SIP request or trigger a chain of new
the Internet. SIP requests to other proxies or location servers.
Figure 5 shows detailed flows for SIP call setup
SIP Call Setup
using a proxy server and are describe below:
SIP is inherently capable of carrying voice,
video, and multimedia calls. In the examples
1. Endpoint1@Site1 sends an INVITE request for
below, the setup flows remain the same
Endpoint2@Site2 to the proxy server.
irrespective of the type of the call. In these
scenarios a call set up is illustrated where a caller
2. The proxy server contacts the location service
knows the name but not the IP address of a
for Endpoint2.
callee, necessitating the use of a SIP server. If the
3. The proxy server receives a more precise
caller knew the IP address of the callee, the caller
location for Endpoint2 as Client2@Site2 from
would not need services from the SIP servers.
the location server.
With a callee s destination IP address known, the
caller s user agent client only needs to select the
4. The proxy server issues an INVITE request to
protocol (UDP by default), port (5060 by default)
the address(es) returned by the location
and IP address of the SIP user agent server to
service. The INVITE request carries a Call-ID.
which the INVITE request should be sent.
(Upon receiving the INVITE request, the called
A successful SIP call setup consists of two
user-agent alerts the user by generating a
messages, an INVITE followed by an ACK. The
phone ring).
INVITE request asks the callee to join a particular
conference or establish a two-party conversation.
5. The called user agent returns a 100 Trying
It also includes information about the media types
response indicating that it is processing the
and formats that are allowed for the session. If the
INVITE request.
callee wishes to accept the call, it responds to the
6. The called user agent returns a 200 OK
invitation by returning a similar description listing
response to indicate successful processing of
the media and format it wishes to use.
the INVITE request.
44 APPENDIX A Next-Gen VoIP Services and Applications Using SIP and Java 45
7. The calling user agent sends an ACK to 3. The location server returns information that this
complete the handshake. The call is now in client can be found at Site3.
place.
4. The redirect server forwards precise location
information to the calling user agent using a
Call Setup Using Redirect Server
302 Moved Temporarily message: Contact
Again we assume that the IP address of the
Client2@Site3
caller is not known to the caller s agent, thereby,
necessitating services of the local SIP server, a
5. The calling user agent acknowledges the
redirect server in this case. The key difference
information with ACK
compared to the proxy server is that the redirect
6. The calling user agent sends an INVITE request
server cannot initiate an INVITE request.
directly to the called user agent.
7. The called user agent returns a 100 Trying
Figure 8: SIP Operation in Redirect Mode
response indicating that it is processing the
request.
Site 1 Site 2 Site 3
8. The called user agent returns a 200 OK
Endpoint 1 Redirect Location Client 2
response to indicate successful processing of
@Site 1 Server Server @Site 3
INVITE
Endpoint 2
the INVITE request.
@Site 2
Endpoint 2
302 9. The calling user agent sends an ACK to
Moved
Temporarily Site 3
complete the handshake. The call is in now
Contact:
Client 2
place.
@Site 3
Ack
INVITE
Client 2 @Site 3
100 Trying
200 OK
Ack
The flow of requests and responses for figure 8
is as follows:
1. Enduser1@Site1 sends an INVITE request to the
redirect server for Endpoint2@Site2.
2. The redirect server contacts the location server
for location information about Endpoint2.
46 NOTES Next-Gen VoIP Services and Applications Using SIP and Java 47
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