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Final Year Project Report i-talk

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 Final Year Project Report

i- talk
B.S. Computer Science Department

Project Advisor

Mr. Abdul Hameed

Prepared by
Abdul Rauf 2009-CS-049
M.Usman Shafi 2009-CS-044
S.M.Ghufran 2009-CS-177

i
 Table Of Contents

Chapter 1

1.1 Introduction Page 1

1.2Goals And Objective Page 2
1.2.1 Goal Page 2
1.2.2 Objectives Page 2

1.3 System Scope Page 2

1.4 Benefits of VoIP Page 3

1.5 Voip Applications Page 3

1.5.1 Skype Page 3

1.5.2 Gizmo Page 4

1.5.3 VoIP Stunt Page 4

1.5.4 Nimbuzz Page 4

1.5.5 Pakistan VoIP Call Center Page 4

1.6 History of VOIP Page 4

1.7 VOIP Market Overview Page 5

ii
Chapter 2

VOICE-OVER-INTERNET PROTOCOL (VOIP) Page 7

2.1 What is Voice Over Internet Protocol (VOIP)? Page 7

2.2 VoIP scenarios Page 8

2.2.1 Operational cost Page 8

2.2.2 Flexibility Page 8

2.3 Cost/Benefit Analysis Page 9

2.4 VoIP Protocols Page 11

2.4.1 Session Initiation Protocol (SIP) Page 11

2.4.3 H.323 Page 13
2.4.3 Media Gateway Control Protocol (MGCP) Page 15
2.4.4 Real-time Transport Protocol (RTP) Page 15
2.4.5 Session Description Protocol (SDP) Page 16
2.4.6 Inter-Asterisk eXchange (IAX) Page 16

Chapter 3

SESSION INITIAL PROTOCOL Page 17

3.1- Introduction Page 17

3.2 SIP Formats Page 18

3.3 Overview of SIP Functionality Page 18

3.4 SIP supports Page 18

iii
 3.5 Structure of the Protocol Page 19

3.6 Definitions Page 21

3.7 Ten Reasons to Use SIP-Enabled Solutions Page 25

Chapter 4
4.1 How SIP Transforms Communications Page 30

4.2 How SIP Transforms End-User Communications Page 30

4.2.1 Using presence to route communications Page 31

4.3 AORs — 0ne address to rule them all Page 32

4.4 How SIP Transforms Small Office Communications Page 34

4.4.1 Small offices Page 34

4.4.2 Capital costs Page 34

4.4.3 Speed Page 34

4.4.4 Business continuity Page 34

4.5 How SIP Transforms Enterprise Communications Page 35

Chapter 5

5.1 SIP Providers Page 36

5.1.1 Hosted SIP Page 36

5. 1.2 Get On Sip

5.1.2 IP Tel Page 38

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 5.1.3 Ekiga SIP Service Page 38

5.1.4 Sip2Sip Page 38

5.1.5 AntiSIP Page 38

5.1.6 VoIPUser Page 38

Chapter 6

SOCIAL NETWORKS Page 39

6.1 INTRODUCTION Page 39

6.1.1 Facebook Page 40

6.1.2 MySpace Page 41

6.1.3 Twitter Page 41

6.1.4 LinkedIn Page 41

6.1.5 Google+ Page 43

6.1.6 Hi5 Page 43

6.1.7 Orkut Page 43

6.1.8 Friendster Page 43

6.1.9 SkyRock Page 43

CHAPTER 7

HOW APPLICATION WORKS Page 44

7.1 INTERNET PHONE SETTING Page 44

7.2 USER CREDENTIALS Page 44

v
 7.2.1 DESCRIPTION OF STEPS Page 45

7.3 TECHNICAL WORK FLOW OF THE APPLICATION Page 46

7.4 Cost Estimation Page 48

7.5 SNAP SHOTS OF APPLICATION Page 49

Chapter 8
8.1 INTRODUCTION TO THE TOOL. Page 51
8.1 JAVA Page 51
8.1.1 32-bit Platforms Page 51
8.1.2 64-bit Platforms Page 53

8.2 Eclipse Page 53

8.2.1 RE/JDK Sources Page 54
8.2.2 Extending Eclipse Page 55
8.2.3 System requirements for Eclipse Page 56
8.3 Android Development Tool Page 56
8.3.1 ADT Plugin for Eclipse description Page 56

8.3.2 Requirements Page 57

8.3.3 DEPENDENCIES Page 57

Conclusion Page 61

vi
 Department of Computer Science

CERTIFICATE OF FYP COMPLETION

This is to certify that the following students

Abdul Rauf 2009-CS-049 ______________

M.Usman Shafi 2009-CS- 044 ______________
S.M.Ghufran 2009-CS-177 ______________

“ i-talk ”

___________
Advisor

vii
 Introduction To The Members

ABDUL RAUF
2009-CS-049

Immediate Contact

03463142523

Email:rauf019@yahoo.com

S.M.Ghufran Hussain
2009-CS-177

Immediate Contact

03433250774

Email:rauf019@yahoo.com

Mohammad Usman Shafi

2009-CS-044

Immediate Contact

03456264632

Email:mohammadusman_77@hotmail.com

viii
Preface

When our final year project was on our head we had so much to think
about and so much to deal with but don’t know from where we had this
idea of making this project. At the time of making this project we all
were amazed that how would we accomplished this project, but it’s only
done by the continuous support of our family and teachers who believe
that we can do it. Thanks to them and Almighty ALLAH. We fulfilled
our entire task which has been assigned to us.

We worked hard to achieve our goal in beginning it was a huge question

of how to do it and how to conquer this mountain but yes we did it.

The project i-Talk solves all the communication concerned it works by

converting analog voice signal into digitized data packets. The packets
are sent out across the internet the same way as any other IP packets,
using the internet’s TCP/IP protocol

ix
Acknowledgment

We praise to almighty Allah who showers the choicest of his blessing to

all those have taken part in the making of this project.
This project when assigned to us was a big challenge for us to
accomplish, because we haven’t done any project of this kind so far. By
the grace of Allah we were to accomplish this task quiet efficiently that
was a great experience for us and it will also help us in our future. We
would like to confiscate this opportunity to do our best to give credit
where it is used. First and foremost to our parents, whose prayers and
advices throughout our educational career has brought us to this in life
where we are completing our graduation.
Second thanks goes to our project advisor Mr. Abdul Hameed who
helped and guided us in this project and for his continuous support
regarding the various aspects of our project.

x
Abstract

In the Secure voice call, the human voice shall be digitized by the
Android APIs and the VOIP packets will travel over the SIP layer. The
digitization process also includes the encryption phase where in secure
call technique is used in order to generate unique keys every time a call
handshake is done. During the Secure Call key exchange, the caller party
sends a Secure Call hello packet. Once that packet is positively
acknowledged by the recipient party the handshake happens successfully
and the call packets get encrypted. Using secure call, digitized voice
data is transformed into cipher text form on third generation GSM data
or GPRS servers in android platform which results in a better encrypted
voice speed and clarity.

xi
xii
Chapter 1

1.1 Introduction

Voice over IP (Voice over Internet Protocol or "VoIP") technology converts voice calls from
analog to digital to be sent over digital data networks. Voice over IP (VoIP) defines a way to
carry voice calls over an IP network including the digitization and packetization of the voice
streams. IP Telephony utilizes the VoIP standards to create a telephony system where higher
level features such as advanced call routing, voice mail, contact centers, etc., can be utilized.

Voice over IP is revolutionizing the world of communications. It allows you to make and receive
phone calls over the Internet and IP networks for much cheaper than with the traditional landline
phone network. It also makes your communication experience much richer and nicer with a
series of enhanced features and extended possibilities. Here is what you need to know to get
started.

Phone calls from a regular home phone are made using the public switched telephone network
(the PSTN). When you pick up the receiver and hear a dial tone, you have access to a line on the
network. The line stays open between you and the person you are calling until the end of the call

VoIP calls don’t use the phone network. They route calls via the internet. To send voice across
the internet, the voice information is coded into a digital format and transmitted in packets of
information in the form of data. The data packets are then sent across the internet and
reassembled into sound at the other end for the receiver to hear.

1
1.2 Goals And Objective

1.2.1 Goal
 Lower Costs of Service – Since it is at the top of most lists, let’s start with saving
money! The charge structures for VoIP providers come in many shapes and sizes,
however it is not uncommon for a VoIP phone service to save a business 20 to 40
percent on their monthly phone bill. Many offer all distance calling programs, so if
you have a high amount of long distance calls, your savings will be higher.
 ALL AT ONE -- Since social networking has become very common now a days I-Talk
provide one platform for all social networking sites.

1.2.2 Objectives

1) Users will be able to communicate effectively, speedily and most importantly,

securely there b y enhancing the privacy and confidentiality o f mobile communication.

2) Implementing voice encryption on third generation GSM data or GPRS servers which
would in turn result in a better encrypted voice speed and clarity.

3) The configuration and usage of proxy server (SIP / Asterisk) thro ugh defining call
routing and handset registration mechanisms.

4) It will en able the users to communicate with each other in an encrypted fashion.

1.3 System Scope

This system works on the VoIP protocol. The technology establishes call and
sends/receives data over existing internet network using IP protocol.SIP is the application
layer protocol, a signaling protocol which is used for establishing multimedia session
over IP network in the internet telephony process.

2
1.4 Benefits of VoIP
The VoIP system offers UofL many benefits as a replacement for its current telephone
technology:

 VoIP uses a single communications network for both telephones and computers instead of
the separate phone and computer networks that are prevalent today

 VoIP uses programmable sets that provide new features, applications and capabilities such as
allowing the university to quickly relay alerts or messages to all locations.

 VoIP makes it easy to administer the system and individual features can be configured
through a simple web interface.

 The IP phones can access the university phone directory, allowing users to find the most up-
to-date telephone numbers right on their phones.

 Using VoIP positions the university for future technologies and future needs of students,
faculty, and staff.

1.5 Voip Applications

There are many areas of VOIP applications. A separate industry has taken birth within the few
years of its implementation.

Even in the sub-continent the light of VOIP has emerged. VOIP can become low-cost or toll-free
for people who are using only the internet to communicate. If both or all users are using the
internet to communicate, VOIP becomes a way of talking or conferencing without having to pay
any additional fee apart from the regular fee for internet.

Some examples of VOIP applications are

1.5.1 Skype

Skype is the most popular of all softphones, with hundreds of millions of subscribers worldwide.
The Skype community is well developed on the web. Skype is also a VoIP service provider. PC
to PC calls are free while PC to Phone calls (SkypeIn and SkypeOut) are paid. Skype's prices

3
have taken a slight rise compared to other services of its like. The application is downloadable
for free at www.skype.com

1.5.2 Gizmo

The Gizmo Project, launched by SIPPhone.com, is more or less like Skype, with both a
softphone and a service.

1.5.3 VoIP Stunt

VoIPStunt is similar to Skype and Gizmo, with the difference of offering free calls to
landline/mobile phones in a number of countries. Its rates are quite low and it is another
competitor for Skype.

1.5.4 Nimbuzz

Nimbuzz has interesting interfaces: for mobile phones (and any Java-enabled devices), for the
web and for installation on a PC (Windows only). Chat and SMS are free, and calls are cheap.
With Nimbuzz, you can call your friend on MSN or Yahoo with your mobile phone. The amount
of things that are free in Nimbuzz is interesting.

1.5.5 Pakistan VoIP Call Center

Inbound Outbound SIP Agents Customer Care ACD IVR CRM Recording.

1.6 History of VOIP

Since the telephone was invented in the late 1800s, telephone communication has not
changed substantially. Of course, new technologies like digital circuits, DTMF (or, "touch
tone"), and caller ID have improved on this invention, but the basic functionality is still the
same. Over the years, service provides made a number of changes "behind the scenes" to
improve on the kinds and types of services offered to subscribers, including toll-free
numbers, call-return, call forwarding, etc. By and large, users do not know how those
services work, but they did know two things: the same old telephone is used and the service
provider charges for each and every little incremental service addition introduced.

4
In the 1990s, a number of individuals in research environments, both in educational and
corporate institutions, took a serious interest in carrying voice and video over IP networks,
especially corporate intranets and the Internet. This technology is commonly referred to
today as VoIP and is, in simple terms, the process of breaking up audio or video into small
chunks, transmitting those chunks over an IP network, and reassembling those chunks at the
far end so that two people can communicate using audio and video.
Voice over IP (VoIP) defines a way to carry voice calls over an IP
network including the digitization and packetization of the voice streams. IP Telephony
utilizes the VoIP standards to create a telephony system where higher level features such as
advanced call routing, voice mail, contact centers, etc., can be utilized.

1.7 VOIP Market Overview

Aside from the need to develop a business case and distinguish cost-benefits by vendor, a
few key findings emerged from our VOIP research. First, deployment is impressive in terms
of the number of companies deployed; penetration within companies has plenty of room for
growth. Second, VOIP is commonly the first phase of a multi-application unified-
communications strategy. Finally, management and interoperability remain a concern which
helps explain why managed services continue to be on the rise.

5
 Figure 1:

VOIP deployment has stayed fairly constant in the past year. Only about 3% of the organizations
in Nemertes’ benchmark are doing nothing right now with VOIP. More than half (53.85%) are
either fully deployed (12.5%) or in the process of deploying the technology to the entire
organization (41.35%). Another 28.85% are engaged in a limited deployment, meaning they have
the technology rolled out in a tactical way—for specific job functions, applications, or types of
locations. The balance (14.42%) are evaluating the technology and vendors or conducting pilots.
(Please see Figure 1: VOIP State of Deployment)

6
Chapter 2

VOICE-OVER-INTERNET PROTOCOL (VOIP)

2.1 What is Voice Over Internet Protocol (VOIP)?

VOIP works by converting analog voice signal into digitized data packets. The packets are
sent out across the internet the same way as any other IP packets, using the internet’s TCP/IP
protocol. VOIP is often used abstractly to refer to the actual transmission of voice(rather than the
protocol implementing it).

Voice over IP (also called VoIP, IP Telephony, and Internet telephony) refers to technology that
enables routing of voice conversations over the Internet or a computer network. To place calls
via VOIP, a user will need a software based sip phone program OR a hardware based VOIP
phone. Phone calls can be made to anywhere / anyone: Both to VOIP numbers as well as people
with normal phone numbers.

For the past 50 years companies have been using conventional PBX systems which require
separate networks for voice and data communications. But with the new VOIP telephony
revolution, businesses are quickly migrating to VOIP PBX systems, which offer the huge
advantage of converging data and voice networks.

VOIP, which stands for Voice over Internet Protocol, is basically the transmission of voice
traffic over IP-based networks. Initially designed for data networking, the Internet Protocol (IP)
was adapted to voice networking following its successful positioning as the global standard for
data networking.

With VOIP phone systems users are not limited to making and receiving calls through the IP
network, traditional phone lines can be used to guarantee a higher call quality and availability.
With the use of a VOIP gateway incoming PSTN/telephone lines can be converted to VOIP/SIP.
This way the VOIP gateway allows the user to receive and place calls on the regular telephony
network.

7
VOIP PBX systems provide mobility to employees, flexibility when a business expands as they
are much easier to manage than the traditional PBX, and can also considerably reduce telephony
administration costs.

2.2 Voip scenarios

2.2.1 Operational cost

1.) Implementation Costs – This includes planning, installation, and troubleshooting the initial
implementation.

2.) Capital Costs – This includes the IP PBX and the handsets and/or softphones. Though we

gather data on gateways and other equipment, for consistency between vendors, the figures

forthcoming only include IP PBX and handset/softphone acquisition costs.

3.) Ongoing operational costs – This includes the staff resources (internal or external) to
operate the VOIP network. Additional operational costs, such as maintenance fees to vendors,
are not included in the forthcoming costs, though we do include those in our cost models.

8
2.2.2 Flexibility

VoIP itself was evolving significantly. During the past five to ten years, VoIP technology has
achieved the ability to offer:

• An entirely new level of clear, true voice quality

• Additional reliability, delivered by new networking technologies

• Easier setup and maintenance

• Security features for protection from fraud and other attack

• Excellent emergency response compliance with E911 (enhanced 911)*

• Better integration between mobile workers and the office.

2.3 Cost/Benefit Analysis

When conducting a VOIP analysis, there are several costs and benefits that apply to most
organizations and others that are industry- or company-specific. First, engage non-IT employees
to determine benefits the technology can bring. Interview business-unit leaders, managers, and
staff to determine the business problems that exist. Then, use IT expertise (or better yet, business
technology liaisons) to determine how VOIP can help resolve business problems,
ultimately leading to increased revenue, improved productivity, or decreased costs. Second,
determine the straight-forward IT costs, such as implementation, capital, and IT training against
which the cost benefits will be weighed. Determine the period of time for the cost analysis, as
well as the depreciation schedule and the Net Present Value percentage.
On the cost side, companies must consider:
 Implementation
Typically, companies spend about 20% more in the first two years of their VOIP
deployments on the actual implementation than they would have spent in TDM. After
they gain expertise, implementation costs are equivalent to TDM rollouts.

9
 Switches
This covers the cost of IP PBXs or the cost to IP-enable an existing PBX.

 Handsets/End-Unit Devices or Applications

This includes IP hardphones or softphones.

 Gateways
Often, companies require gateways for TDM-to-IP traffic, unless they’re using SIP
trunking throughout the organization (which is rare still).

10
 LAN upgrades
VOIP requires Power-Over-Ethernet switches, and most companies provide
Uninterrupted Power Supplies to provide for backup. When organizations upgrade
their LANs, the costs account for 32% to 47% of an overall VOIP project

 Management/monitoring tools
Many companies don’t budget for management and monitoring tools, which is a
mistake. Acquisition costs range from free (with open-source tools) to several million
dollars. On average, small and midsize companies spend about $20,000 for each third
party monitoring tool, and large companies spend about $200,000 per tool.

 Training
Many vendors are including training with the sale of equipment. But when they
don’t, companies spend between $1,000 and $5,000 per IT staff member for training,
and they find the most success by training their end users with internal IT staff.

 Equipment licensing & maintenance

Vendors are shifting more to a software model in which the initial acquisition cost is
lower, but maintenance and licensing is higher. Whereas vendors once charged about
10% to 14% for maintenance, those fees now are 16% to 22%.

 Ongoing WAN costs

This includes the cost of the converged WAN. Typically, this includes the circuit
costs for services such as MPLS,Ethernet, and/or SIP trunking.

 Ongoing operational costs

This includes the cost to manage and maintain the network from a staff perspective.
It includes the total compensation of internal staff members devoted to VOIP, plus the
cost of any third-party MSPs managing the VOIP system. Also included are
power and cooling costs.
11
2.4 VoIP Protocols

Voice over IP has been implemented in various ways using both proprietary and open protocols
and standards. Examples of the network protocols used to implement VoIP include:

 Session Initiation Protocol (SIP)

 H.323
 Media Gateway Control Protocol (MGCP)
 Real-time Transport Protocol (RTP)
 Session Description Protocol (SDP)
 Inter-Asterisk eXchange (IAX)
 Jingle XMPP VoIP extensions

2.4.1 Session Initiation Protocol (SIP)

The Session Initiation Protocol (SIP) is an IETF-defined signaling protocol widely used for
controlling communication sessions such as voice and video calls over Internet Protocol (IP).
The protocol can be used for creating, modifying and terminating two-party (unicast) or
multiparty (multicast) sessions. Sessions may consist of one or several media streams.
Other SIP applications include video conferencing, streaming multimedia
distribution, instant messaging, presence information, file transfer and online gamesThe SIP
protocol is an Application Layer protocol designed to be independent of the underlying
Transport Layer; it can run on Transmission Control Protocol (TCP), User Datagram Protocol
(UDP).

2.4.2 SIP for telephony and VoIP

SIP can be regarded as the enabler protocol for telephony and voice over IP (VoIP) services. The
following features of SIP play a major role in the enablement of IP telephony and VoIP:

12
  Name Translation and User Location:
Ensuring that the call reaches the called party wherever they are located.
Carrying out any mapping of descriptive information to location information. Ensuring
that details of the nature of the call (Session) are supported.

 Feature Negotiation:
This allows the group involved in a call (this may be a multi-party call) to
agree on the features supported recognizing that not all the parties can support the same
level of features. For example video may or may not be supported; as any form of MIME
type is supported by SIP, there is plenty of scope for negotiation.

 Call Participant Management:

During a call a participant can bring other users onto the call or cancel
connections to other users. In addition, users could be transferred or placed on hold.

 Call feature changes:

A user should be able to change the call characteristics during the course of
the call. For example, a call may have been set up as 'voice-only', but in the course of the
call, the users may need to enable a video function. A third party joining a call may
require different features to be enabled in order to participate in the call

 Media negotiation:
The inherent SIP mechanisms that enable negotiation of the media used in a
call, enable selection of the appropriate codec for establishing a call between the various
devices. This way, less advanced devices can participate in the call, provided the
appropriate codec is selected.

2.4.2 H.323

H.323 is a recommendation from the ITU Telecommunication Standardization Sector (ITU-T)

that defines the protocols to provide audio-visual communication sessions on any packet

13
network. The H.323 standard addresses call signaling and control, multimedia transport and
control, and bandwidth control for point-to-point and multi-point conferences.
It is widely implemented by voice and videoconferencing equipment
manufacturers, is used within various Internet real-time applications such as GnuGK and
NetMeeting and is widely deployed worldwide by service providers and enterprises for both
voice and video services over IP networks.

It is a part of the ITU-T H.32x series of protocols, which also address multimedia
communications over ISDN, the PSTN or SS7, and 3G mobile networks.
H.323 call signaling is based on the ITU-T Recommendation Q.931 protocol and
is suited for transmitting calls across networks using a mixture of IP, PSTN, ISDN, and QSIG
over ISDN. A call model, similar to the ISDN call model, eases the introduction of IP telephony
into existing networks of ISDN-based PBX systems, including transitions to IP-based PBXs.
Within the context of H.323, an IP-based PBX might be a gatekeeper or other
call control element which provides service to telephones or videophones. Such a device may
provide or facilitate both basic services and supplementary services, such as call transfer, park,
pick-up, and hold.

2.4.2.1 H.323 and Voice over IP services

Voice over Internet Protocol (VoIP) describes the transmission of voice using the Internet or
other packet switched networks. ITU-T Recommendation H.323 is one of the standards used in
VoIP. VoIP requires a connection to the Internet or another packet switched network, a
subscription to a VoIP service provider and a client (an analogue telephone adapter (ATA), VoIP
Phone or "soft phone"). The service provider offers the connection to other VoIP services or to
the PSTN. Most service providers charge a monthly fee, then additional costs when calls are
made.Using VoIP between two enterprise locations would not necessarily require a VoIP service
provider, for example. H.323 has been widely deployed by companies who wish to interconnect
remote locations over IP using a number of various wired and wireless technologies

14
2.4.3 Media Gateway Control Protocol (MGCP)

Media Gateway Control Protocol also known as MGCP is one of the implementation of the
Media Gateway Control Protocol Architecture[1] for controlling media gateways on Internet
Protocol (IP) networks and the public switched telephone network (PSTN).

MGCP is a signalling and call control protocol used within Voice over IP (VoIP) systems that
typically inter-operate with the public switched telephone network (PSTN). As such it
implements a PSTN-over-IP model with the power of the network residing in a call control
center (soft switch, similar to the central office of the PSTN) and the endpoints being "low-
intelligence" devices, mostly simply executing control commands. The protocol represents a
decomposition of other VoIP models, such as H.323, in which the media gateways (e.g., H.323's
gatekeeper) have higher levels of signalling intelligence.

2.4.4 Real-time Transport Protocol (RTP)

RTP is designed for end-to-end, real-time, transfer of stream data. The protocol provides facility
for jitter compensation and detection of out of sequence arrival in data, that are common during
transmissions on an IP network. RTP supports data transfer to multiple destinations through IP
multicast. RTP is regarded as the primary standard for audio/video transport in IP networks and
is used with an associated profile and payload format.

Real-time multimedia streaming applications require timely delivery of information and can
tolerate some packet loss to achieve this goal. For example, loss of a packet in audio application
may result in loss of a fraction of a second of audio data, which can be made unnoticeable with
suitable error concealment algorithms. The Transmission Control Protocol (TCP), although
standardized for RTP use,is not normally used in RTP application because TCP favors reliability
over timeliness. Instead the majority of the RTP implementations are built on the User Datagram
Protocol (UDP).Other transport protocols specifically designed for multimedia sessions are
SCTP and DCCP, although, as of 2010, they are not in widespread use.

RTP was developed by the Audio/Video Transport working group of the IETF standards
organization. RTP is used in conjunction with other protocols such as H.323 and RTSP. The RTP

15
standard defines a pair of protocols, RTP and RTCP. RTP is used for transfer of multimedia data,
and the RTCP is used to periodically send control information and QoS parameters.

2.4.5 Session Description Protocol (SDP)

SDP provides a standard representation for such information, irrespective of how that
information is transported. SDP is purely a format for session description -- it does not
incorporate a transport protocol, and it is intended to use different transport protocols as
appropriate, including the Session Announcement Protocol, Session Initiation Protocol, Real
Time Streaming Protocol, electronic mail using the MIME extensions, and the Hypertext
Transport Protocol.
SDP is intended to be general purpose so that it can be used in a wide range of network
environments not intended to support negotiation of session content or media encodings: this is
viewed as outside the scope of session description.

2.4.6 Inter-Asterisk eXchange (IAX)

IAX is the Inter-Asterisk eXchange protocol, which facilitates VoIP connections between
servers, and between servers and clients that also use the IAX protocol. IAX was created
through an open source methodology rather than through a traditional, standards-based
methodology. It is an open protocol originally used by Asterisk, a dual-licensed open source and
commercial PBX server from Digium. Independent IAX implementations may be open,
proprietary, or licensed in anyway the author seems fit without royalty to the protocol creators.

16
Chapter 3

SESSION INITIAL PROTOCOL

3.1- Introduction

There are many applications of the Internet that require the creation and management of a
session, where a session is considered an exchange of data between an association of
participants. The implementation of these applications is complicated by the practices of
participants: users may move between endpoints, they may be addressable by multiple names,
and they may communicate in several different media - sometimes simultaneously. Numerous
protocols have been authored that carry various forms of real-time multimedia session data such
as voice, video, or text messages. The Session

Initiation Protocol (SIP) works in concert with these protocols by enabling Internet endpoints
(called user agents) to discover one another and to agree on a characterization of a session they
would like to share. For locating prospective session participants, and for other functions, SIP
enables the creation of an infrastructure of network hosts (called proxy servers) to which user
agents can send registrations, invitations to sessions, and other requests. SIP is an agile, general-
purpose tool for creating, modifying, and terminating sessions that works independently of
underlying transport protocols and without dependency on the type of session that is being
established.

17
3.2 SIP Formats

SIP is typically offered in two formats, computer based and hardware based. Computer based
SIP is a system that allows you to make calls using your computer as the router and
communicating via a headset on your computer. The more practical and popular version,
however, actually provides you with new SIP enabled telephone handsets or converts your
existing phones to SIP. By eliminating any technical requirements, modern SIP providers have
made using the system as easy, or easier, than using a traditional phone.

3.3 Overview of SIP Functionality

SIP is an application-layer control protocol that can establish, modify, and terminate multimedia
sessions (conferences) such as Internet telephony calls. SIP can also invite participants to
already existing sessions, such as multicast conferences. Media can be added to (and removed
from) an existing session. SIP transparently supports name mapping and redirection services,
which supports personal mobility - users can maintain a single externally visible identifier
regardless of their network location.

3.4 SIP supports

SIP supports five facets of establishing and terminating multimedia communications:

User location: Determination of the end system to be used for communication.

User availability: Determination of the willingness of the called party to engage in

communications.

User capabilities: Determination of the media and media parameters to be used.

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 Session setup: "ringing", establishment of session parameters at both called and calling
party.

Session management: including transfer and termination of sessions, modifying session

parameters, and invoking services.

3.5 Structure of the Protocol

SIP is structured as a layered protocol, which means that its behavior is described in terms of a
set of fairly independent processing stages with only a loose coupling between each stage. The
protocol behavior is described as layers for the purpose of presentation, allowing the description
of functions common across elements in a single section. It does not dictate an implementation
in any way. When we say that an element "contains" a layer, we mean it is compliant to the set
of rules defined by that layer.

Not every element specified by the protocol contains every layer. Furthermore, the elements
specified by SIP are logical elements, not physical ones. A physical realization can choose to
act as different logical elements, perhaps even on a transaction-by-transaction basis. The lowest
layer of SIP is its syntax and encoding. Its encoding is specified using an augmented Backus-
Naur Form grammar (BNF).The second layer is the transport layer. It defines how a client sends
requests and receives responses and how a server receives requests and sends responses over the
network. All SIP elements contain a transport layer.

The third layer is the transaction layer. Transactions are a fundamental component of SIP. A
transaction is a request sent by a client transaction (using the transport layer) to a server
transaction, along with all responses to that request sent from the server transaction back to the
client. The transaction layer handles application-layer retransmissions, matching of responses to
requests, and application-layer timeouts. Any task that a user agent client (UAC) accomplishes
takes place using a series of transactions. User agents contain a transaction layer, as do stateful
proxies. Stateless proxies do not contain a transaction layer. The transaction layer has a client
component (referred to as a client transaction) and a server component (referred to as a server

19
transaction), each of which are represented by a finite state machine that is constructed to
process a particular request.

The layer above the transaction layer is called the transaction user (TU). Each of the SIP
entities, except the stateless proxy, is a transaction user. When a TU wishes to send a request, it
creates a client transaction instance and passes it the request along with the destination IP
address, port, and transport to which to send the request. A TU that creates a client transaction
can also cancel it. When a client cancels a transaction, it requests that the server stop
further processing, revert to the state that existed before the transaction was initiated, and
generate a specific error response to that transaction. This is done with a CANCEL request,
which constitutes its own transaction, but references the transaction to be cancelled.

The SIP elements, that is, user agent clients and servers, stateless and stateful proxies and
registrars, contain a core that distinguishes them from each other. Cores, except for the stateless
proxy, are transaction users. While the behavior of the UAC and UAS cores depends on the
method, there are some common rules for all methods (Section 8). For a UAC, these rules
govern the construction of a request; for a UAS, they govern the processing of a request and
generating a response.

Since registrations play an important role in SIP, a UAS that handles a REGISTER is given the
special name registrar. Certain other requests are sent within a dialog.A dialog is a peer-to-peer
SIP relationship between two user agents that persists for some time. The dialog facilitates
sequencing of messages and proper routing of requests between the user agents. The INVITE
method is the only way defined in this specification to establish a dialog. When a UAC sends a
request that is within the context of a dialog, it follows the common UAC rules but also the rules
for mid-dialog requests.
The most important method in SIP is the INVITE method, which is used to stablish a session
between participants. A session is a collection of participants, and streams of media between
them, for the purposes of communication. Section 13 discusses how sessions are
initiated, resulting in one or more SIP dialogs. Section 14 discusses how characteristics of that
session are modified through the use of an INVITE request within a dialog. Finally, section 15
discusses how a session is terminated.

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3.6 Definitions
The following terms have special significance for SIP.

Address-of-Record: An address-of-record (AOR) is a SIP or SIPS URI that points to a domain

with a location service that can map the URI to another URI where the user might be available.
Typically, the location service is populated through registrations. An AOR is frequently
thought of as the "public address" of the user.

Back-to-Back User Agent: A back-to-back user agent (B2BUA) is a logical entity that receives
a request and processes it as a user agent server (UAS). In order to determine how the request
should be answered, it acts as a user agent client (UAC) and generates requests. Unlike a proxy
server, it maintains dialog state and must participate in all requests sent on the dialogs it has
established. Since it is a concatenation of a UAC and UAS, no explicit definitions are needed for
its behavior.
Call: A call is an informal term that refers to some communication between peers, generally set
up for the purposes of a multimedia conversation.

Call Leg: Another name for a dialog; no longer used in this specification.

Call Stateful: A proxy is call stateful if it retains state for a dialog from the initiating INVITE to
the terminating BYE request. A call stateful proxy is always transaction stateful, but the
converse is not necessarily true.

Client: A client is any network element that sends SIP requests and receives SIP responses.
Clients may or may not interact directly with a human user. User agent clients and proxies are
clients.

Conference: A multimedia session (see below) that contains multiple participants.

Core: Core designates the functions specific to a particular type of SIP entity, i.e., specific to
either a stateful or stateless proxy, a user agent or registrar. All cores, except those for the
stateless proxy, are transaction users.
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Dialog: A dialog is a peer-to-peer SIP relationship between two UAs that persists for some
time. A dialog is established by SIP messages, such as a 2xx response to an INVITE request. A
dialog is identified by a call identifier, local tag, and a remote tag. A dialog was formerly known
as a call leg in RFC 2543.

Downstream: A direction of message forwarding within a transaction that refers to the

direction that requests flow from the user agent client to user agent server.

Final Response: A response that terminates a SIP transaction, as opposed to a provisional

response that does not. All 2xx, 3xx, 4xx, 5xx and 6xx responses are final.

Header: A header is a component of a SIP message that conveys information about the
message. It is structured as a sequence of header fields.

Header Field: A header field is a component of the SIP message header. A header field can
appear as one or more header field rows. Header field rows consist of a header field name and
zero or more header field values. Multiple header field values on a given header field row are
separated by commas. Some header fields can only have a single header field value, and as a
result, always appear as a single header field row.

Header Field Value: A header field value is a single value; a header field consists of zero or
more header field values.

Home Domain: The domain providing service to a SIP user. Typically, this is the domain
present in the URI in the address-of-record of a registration.

Informational Response: Same as a provisional response.

Initiator, Calling Party, Caller: The party initiating a session (and dialog) with an INVITE
request. A caller retains this role from the time it sends the initial INVITE that established
a dialog until the termination of that dialog.

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Invitation: An INVITE request.

Invitee, Invited User, Called Party, Callee: The party that receives an INVITE request for the
purpose of establishing a new session. A callee retains this role from the time it receives the
INVITE until the termination of the dialog established by that INVITE.

Location Service: A location service is used by a SIP redirect or proxy server to obtain
information about a callee's possible location(s). It contains a list of bindings of address-of-
record keys to zero or more contact addresses. The bindings can be created and removed in
many ways; this specification defines a REGISTER method that updates the bindings.

Loop: A request that arrives at a proxy, is forwarded, and later arrives back at the same proxy.
When it arrives the second time, its Request-URI is identical to the first time, and other
header fields that affect proxy operation are unchanged, so that the proxy would make the same
processing decision on the request it made the first time. Looped requests are errors, and the
procedures for detecting them and handling them are described by the protocol.

Loose Routing: A proxy is said to be loose routing if it follows the procedures defined in this
specification for processing of the Route header field. These procedures separate the
destination of the request (present in the Request-URI) from the set of proxies that need to be
visited along the way (present in the Route header field). A proxy compliant to these
mechanisms is also known as a loose router.

Outbound Proxy: A proxy that receives requests from a client, even though it may not be the
server resolved by the Request-URI. Typically, a UA is manually configured with an outbound
proxy, or can learn about one through auto-configuration protocols.

Parallel Search: In a parallel search, a proxy issues several requests to possible user locations
upon receiving an incoming request. Rather than issuing one request and then waiting for the
final response before issuing the next request as in a sequential search, a parallel search issues
requests without waiting for the result of previous requests. Provisional Response: A response

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used by the server to indicate progress, but that does not terminate a SIP transaction. 1xx
responses are provisional, other responses are considered final.

Proxy, Proxy Server: An intermediary entity that acts as both a server and a client for the
purpose of making requests on behalf of other clients. A proxy server primarily plays the
role of routing, which means its job is to ensure that a request is sent to another entity "closer" to
the targeted user. Proxies are also useful for enforcing policy (for example, making sure a user
is allowed to make a call). A proxy interprets, and, if necessary, rewrites specific parts of
a request message before forwarding it.

Recursion: A client recurses on a 3xx response when it generates a new request to one or more
of the URIs in the Contact header field in the response.

Redirect Server: A redirect server is a user agent server that generates 3xx responses to requests
it receives, directing the client to contact an alternate set of URIs. Session: From the SDP
specification: "A multimedia session is a set of multimedia senders and receivers and the data
streams flowing from senders to receivers. A multimedia conference is an example of a
multimedia session." (RFC 2327 [1]) (A session as defined for SDP can comprise one or more
RTP sessions.) As defined, a callee can be invited several times, by different calls, to the same
session. If SDP is used, a session is defined by the concatenation of the SDP user name, session
id, network type, address type, and address elements in the origin field.

SIP Transaction: A SIP transaction occurs between a client and a server and comprises all
messages from the first request sent from the client to the server up to a final (non-1xx) response
sent from the server to the client. If the request is INVITE and the final response is a non-2xx,
the transaction also includes an ACK to the response. The ACK for a 2xx response to
an INVITE request is a separate transaction.

UAS Core: The set of processing functions required at a UAS that resides above the transaction
and transport layers.

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User Agent (UA): A logical entity that can act as both a user agent client and user agent server.

The role of UAC and UAS, as well as proxy and redirect servers, are defined
on a transaction-by-transaction basis. For example, the user agent initiating a call acts as a UAC
when sending the initial INVITE request and as a UAS when receiving a BYE request from the
callee.
Similarly, the same software can act as a proxy server for one request and as a redirect
server for the next request.

3.7 Ten Reasons to Use SIP-Enabled Solutions

3.7.1 Easier Manageability

SIP networks are relatively easy to set up and administer, yet include many advanced features,
such as automated attendant, voice mail, and three-party conferencing. Because SIP is modeled
after HTTP as a text-based language, it is easy to learn, develop, troubleshoot, and support. From
analyzing network packets to developing custom applications, SIP’s structured language makes it
easier for IT systems engineers and developers to understand and interpret it. The ability to
rapidly deploy new technologies and applications will give your business a definite competitive
advantage. As the industry continues to evolve, so will interoperability and manageability across
third-party software, devices, and so on, and integrated management across the entire SIP
environment will become even easier.

3.7.2 User-Centric Communications

SIP works with the Extensible Messaging and Presence Protocol (XMPP), or presence , to
intelligently connect communicating parties based on their ability and willingness to participate
in a communications session, as well as their preference, based on parameters such as time of
day, day of week, desired mode, and type of caller. Unified addressing, through the use of a SIP
AOR ( address of record ), provides a single URI (Uniform Resource Identifier) that can be used
for routing all communications to a user.Simply put, an AOR allows for a single user identity to

25
be mapped across multiple devices so that people connect with people, without needing to know
which devices they have and are presently using. This eliminates the need for tracking users’
multiple phone numbers, e-mail addresses, and IM contact names. SIP also supports intelligent
forking — that is, the ability to route a communications session to the right person, using the
right medium (voice, video, text), on the right device (or application), and at the right time.

3.7.3 Native Mobility

SIP builds the foundation for native mobility in applications and devices for the (not-too-distant)
future. As more devices become SIP-capable, users will be able to pick up and go at will, but still
communicate as if they were in their office. Their presence and readiness to communicate will
still be visible to everyone For example, SIP’s awareness of a user’s communication capabilities
will aid international travelers who have to use different mobile phones and other messaging
devices and protocols in different countries. A caller trying to locate such a traveler need not
know the traveler’s availability or location: SIP will know how a person can be reached and will
facilitate the connection.

3.7.4 Unlimited Scalability

No matter the size of your business or the state of your cur-rent telecommunications
infrastructure, Avaya has a solution to fit your intelligent communications and your unified
communications needs and budget. SIP-enhanced communications solutions can easily scale
from a small branch office deployment to carrier-class enterprise networks spanning multiple
continents. Whether you’re looking to consolidate a mix of legacy TDM PBX
telecommunications equipment and evolve your IP communications network incrementally, or
“rip and replace” it all at once, SIP can easily scale to any deployment scenario and support the
future growth of your business. See Chapters 3 and 4 for more information about the flexibility
and scalability of SIP communications architectures.

3.7.5 Better Survivability

26
The ability to communicate is important to any business, but real-time communications are
critical to the survival of your business when disaster strikes. SIP-enhanced intelligent
communications architectures help ensure that disaster recovery and business continuity plans
actually work.

Through SIP, enterprises can control their redundant plat-forms and fail-over paths and, with the
advent of both User-to-User Information (UUI) and Network Call Redirect (NCR), SIP trunks
offer feature parity with transport mechanisms such as Integrated Services Digital Network
(ISDN).

3.7.6 Endpoint Flexibility

End users appreciate choice so they benefit from the fact that the SIP protocol works on a wide
variety of communications devices. Users preferring a desk phone, a unified communications
soft phone on a PC, or a favorite mobile phone can connect to a SIP-based communications
network seamlessly and easily. This means that the same communication functionality available
in the office extends to a user via any device. For example, your desk phone and mobile phone
can ring simultaneously. This enables you to remain in contact with customers and business
associates no matter where you are. SIP also gives you “mobility within the office,” allowing
you to sit down at any desk or in any office, log in, and automatically download your
communications profile to your current location making it appear as if you were sitting at your
own desk! Call logs, conference calling, call transfer and other features you use on your office
phone are extended to your other communication devices and your current location. With SIP,
you get endpoint flexibility without sacrificing endpoint functionality.

3.7.7 Unprecedented Interoperability

SIP is an open standard defined in RFC 3261 by the IETF, an international community of
network designers, operators, vendors, and researchers, all concerned with the evolution of the
Internet architecture and the development of standards to ensure the smooth operation of the
Internet.Several working groups, including SIP it, SIP connect, SIP Foundry, and the SIP Forum
the board of directors of which includes Avaya — arrange regular events where companies with
SIP hardware and software products can test interoperability with other SIP products. This

27
process helps to promote smoother integration of SIP products in enterprise networks. The SIP
connect Compliant designation helps customers identify solutions that provide interoperability
among multiple vendors.

3.7.8 Lower Total Cost of Ownership

SIP-enhanced intelligent communications architectures deliver lower total cost of ownership

(TCO) to businesses and enterprises through SIP trunks. SIP trunks are IP trunks from service
providers that use SIP for call control and routing, enabling enterprises to create a single, pure IP
connection to carrier clouds. Voice traverses the network just like other IP applications. SIP
trunks reduce operational costs by enabling the enterprise to eliminate hardware, software, and
recurring network charges associated with using traditional PSTN trunks for voice
communications. In fact, Avaya customers typically see a return on investment (ROI) of 6–12
months for their trunking solutions.

3.7.9 Enhanced Customer Service

More than ever, customer service is a competitive differentia-tor for any successful business.
Customers expect and deserve more than just a call center that routes them to the “next available
agent” or an impersonal interactive voice response (IVR) system that routes them in circles until
they get frustrated and take their business elsewhere. SIP has accelerated the migration from call
center to contact center, allowing businesses to truly leverage the power of real-time IP
communications, including voice, IM, chat, Web, and video, to ensure that customer service
agents and technical experts all have the features, functionality, and information necessary to
best serve their customers.

3.7.10 Increased Productivity

It’s easy to imagine how any of the features and advantages already described in this chapter
could increase individual productivity just by simplifying your communications world. But SIP
can actually do a lot more to improve productivity, increase efficiency, and save the whales!
Well, that may be a stretch, but SIP can help make your office “greener,” for example, by
facilitating virtual meetings using multimedia collaboration tools (voice, video, and data through
28
the Web) such as Avaya’s Meeting Exchange, Avaya Web Conference, and Avaya one-X portal
together, thereby reducing the need for costly travel. With rising fuel costs alone, many
companies can expect an ROI of six months or less using SIP’s conferencing/ video conferencing
and collaboration capabilities.
By deploying SIP-enabled intelligent custom applications, businesses can offer new services for
their customers and improve existing business processes. For example, we’ve already described
the advantages of SIP in the contact center in terms of customer service, but what about its
implications for employee productivity? With SIP-enabled applications, a customer service agent
can be more productive by improving his response and resolution times. Rather than simply
escalating calls to various technicians, sales reps, or supervisors, the agent can communicate with
all the necessary parties over a number of SIP-enabled applications and platforms while
simultaneously maintaining contact with the customer. This saves valuable time (and frustration)
for everyone involved since the customer doesn’t have to repeat the issue each time a call is
“handed off” to someone else.

29
Chapter 4

4.1 How SIP Transforms Communications

SIP can revolutionize real-time IP communications in any organization, large or small. This
chapter takes a look at how SIP-enabled features like presence and addresses of record can
enhance the productivity and quality of communications for end users. We also examine SIP for
small and medium businesses and discuss some of the unique challenges for small and
distributed business environments. Finally, we see how SIP scales to even the largest enterprises
and service provider networks, providing the framework for new and evolving architectures such
as IMS and SOA, and making SIP a truly universal solution for intelligent communications.

4.2 How SIP Transforms End-User Communications

Are you spending more time managing your communications devices and looking up phone
numbers or addresses than actually communicating with others? If so, SIP is about to simplify
your life and let you control your communications devices, rather than having them control you.
When a user activates a communications device (user agent, or UA), the device registers its
presence on the network, indicating its ability to communicate. The concept of presence is
somewhat analogous to the telephone network’s busy signal, signaling to a caller that you are
unable to talk right now because you’re already talking with someone else. But, SIP takes
presence a step further. Presence distributes the following information:
- User status (that is, online or offline)
- User availability (such as Available, Away, In a Meeting, On the Phone, and Busy)
- User’s desired contact method (such as instant messaging, desk phone, mobile phone,
pager, and so on)

30
SIP’s presence states also permit predictable rules-based routing decisions to be made.
These decisions are based on a user’s specific presence state, and on customizable
preferences that include any information the user wants to share.

Presence doesn’t just apply to people and need not only apply to a single entity;
presence can also be associated with a device or group. For example, a presence status
might capture:

- the status of a device (Phone Status = Off-Hook) or the

- status of a user (User Status = Online).

Presence for composite entities like contact center groups or shared documents
can be similarly represented.

4.2.1 Using presence to route communications

SIP can make call-routing decisions based on presence information by enabling users to inform
others of their status, availability, and how they can be contacted — before a communication
session even begins. A user can communicate status and availability to others through multiple
devices such as IP phones, mobile phones, softphones, instant messaging, pagers, video
conferencing, e-mail, wireless devices, and even TDM phones connected to an intelligent IP
PBX.

Presence can span a number of different communication channels and provide an aggregate view
of a user’s presence (that is, availability across all of an individual’s SIP-enabled devices).
Possibilities include:

- Setting the user’s status to Away when his phone and keyboard are inactive for some
time
- Making inferences about a user’s presence through mobile device location information
- Checking a user’s calendar to see whether he is in a meeting or on vacation
- Checking a user’s e-mail to see whether he is reading or sending e-mail, or whether he
has an Out of Office setting

31
SIP uses presence to make routing decisions for a variety of incoming communications
including:

- Routing incoming calls from a desk phone to a mobile phone if the user has indicated that
he is roaming and prefers calls routed as such.
- Classifying non-urgent incoming communications as polite calls that the user can choose
to answer, forward, or ignore
- Routing urgent incoming calls and e-mail to others if the user is on vacation or in an
extended meeting. When a SIP proxy (a server that processes and forwards SIP requests
between calling and called parties) receives an

INVITE (request to communicate), it uses the called party’s

presence to make a routing decision, sometimes called forking. The forking decision may be to a
specific party (an intelligent fork), or it may send several INVITEs to different addresses
(parallel forking).

Forking is an old UNIX term where a process “clones” itself into two or more new processes. In
the SIP context, forking refers to sending multiple simultaneous INVITEs to other parties to
initiate a communication session.

4.3 AORs — 0ne address to rule them all

Another key feature of SIP is its ability to use an end-user’s address of record (AOR) as a single
unifying public address for all communications. With SIP-enhanced communications, a user’s
AOR becomes her single address that links the user to all of the communication devices or
services that she uses. For example, Eileen Dover’s AOR might be
sip:eileendover@company.com. Using this AOR, you can reach Eileen on any of her multiple
communication devices (her UAs) without having to know each of her unique device addresses
or phone numbers.

Presence means “being there” for your customers. Every successful company strives to provide
superior customer service, with call centers or contact centers (you can read more about contact
centers in Chapter 5), which are often the first contact an upset customer may have with your

32
company when dealing with a particular issue. But how can your customer service agents get to
all the information they need to provide quick, accurate responses for your customers? A
credit card company that provides ongoing support for its customers through a contact center
offers one example of how SIP presence can help you deliver superior customer support. The
names, places, and events are fictitious, but the possibilities are real: A customer planning an
overseas trip calls with a question about monetary conversion rate policies. The customer service
agent checks her “finance expert” presence tab and sees that an internal resident expert is off the
phone and available for consultation. The agent clicks the IM tab and is automatically routed to
the available expert. The agent then gets, and quickly relays, the expert’s answer to the customer.
The customer then asks about a disputed transaction with a merchant. The agent brings up the
merchant information, which displays the presence and availability of the merchant’s customer
service agents for phone calls or IM, and contacts an available agent who can look up details of
the transaction and send it back via a Web-page push. Complete customer service with a smile
(or maybe just a smiley face icon)! 07_381144-ch03.qxp 9/26/08 8:52 PM Page 24To
complement AORs, SIP supports Uniform Resource Identifiers (URIs) that establish a common
addressing scheme for all of an individual’s user agents. A URI address follows the same basic
format as a Web or e-mail address: contact-address@ domain. Using this format, SIP can map
the unique addresses of a user’s multiple devices and services to a communication domain, and
then link all the user agents to a user’s single AOR for that domain. Some examples of how a
URI might be applied include:

A phone: sip:908-555-1212@company.com;

user= phone

A fax: sip: 908-555-1214@company.com;user=fax

An IM user: sip:eileendover@company.com

A user typically has just one SIP AOR, such as sip:

eileendover@domain. Each of the user’s devices then has

its own URI, such as sip:908-555-1214@company.com;

user=fax.

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Because a SIP URI supports both numeric (phone numbers) and alphanumeric (Internet-style
addresses) formatted contact addressing, the public switched telephone network (PSTN) and the
Internet can be seamlessly linked together. With SIP, users can potentially contact any user,
whether they are on the PSTN or the Internet. As with e-mail addresses, users probably won’t
memorize other users’ SIP AORs. Instead, they’ll use address books and buddy lists, just like
they do on their e-mail systems, mobile phones, and IM clients today. A SIP AOR will be just
another data field associated with each person or group. When used by a SIP device, the URI
will be retrieved and used to communicate with another party

4.4 How SIP Transforms Small Office Communications

4.4.1 Small offices — including small or mid-sized businesses and small branches of large
enterprises — are becoming mor dynamic in form and function and are becoming increasingly
distributed. These work environments must address challenges that although not unique, can be
nonetheless daunting for small offices, including:

4.4.2 Capital costs: As small offices seek to maintain a more dynamic form that focuses on
the localized needs of their markets, they often find themselves balancing the need for
adaptability with the upfront capital costs of communications solutions. Operating and
administrative costs: Communications solutions often require on-site technical installation and
maintenance services. Additional costs are incurred when local support is required to fix
problems, add capacity, or perform basic administrative tasks.

4.4.3 Speed: For many small offices, competitive advantage is all about speed — time to
deployment drives time to market. Developing new applications that integrate with complex
communications architectures through open but proprietary APIs (application programming
interfaces) can be a time-consuming event requiring detailed planning, staging, testing, and
debugging.

4.4.4 Business continuity: Small businesses are often more sensitive to disruptive events than
larger businesses. The survivability of a small business may be threatened by even a relatively
minor or short-term event lasting only a few days.

34
4.5 How SIP Transforms Enterprise Communications

SIP fundamentally improves the efficiency of communications between enterprises and their
partners, suppliers, and customers. The initial benefit of IP communications has been primarily
limited to intra-enterprise communications. Communications between enterprises, even those
that are VoIP-enabled, still largely require a circuit-switched handoff that impacts voice quality,
adds complexity, and introduces additional expense through intermediate carriers. SIP changes
all of that by interconnecting SIP communications architectures and the PSTN, and with SIP
trunking and federation services. Enterprises can benefit from the simplification of enterprise
networks through SIP standardization for both internal and external communications. As SIP
becomes ubiquitous in both service provider and enterprise networks, a single standard interface
for all connectivity is available for adding endpoints, deploying contact center adjunct services,
or even connecting trunk services for external communications. Proprietary signaling protocols,
including variants of voice-centric T1 and E1 standards, and hardware-intensive digital/analog
interfaces give way to a simple, logical SIP interface that connects application servers residing
on industry-standard platforms. With SIP as a unifying protocol, you can dramatically reduce the
need for dedicated hardware gateways and devices.

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Chapter 5

5.1 SIP Providers

The term “SIP Provider”, for example, means different things depending on who is using it. It
has its roots in “SIP Trunking” which is a service provided to businesses that allows them to
connect their in house IP PBX systems to the outside world - the public switched telephone
network or PSTN. Without it, devices within a PBX can only talk to other devices on the same
network or using the same system. The Internet Telephony Service Provider (ITSP) however,
gives such firms a pipe or a “trunk” which makes the connection to the PSTN, thereby
exponentially increasing the reach and usefulness of their PBXs. One example of a SIP trunking
service is our own, pstn.junctionnetworks.com.

This is the most common definition of an SIP provider as of now.

But recently lots of new entrants to the field are offering innovative services making use of the
SIP protocol as a backbone. These nimble ITSPs are able to build on the traditional SIP trunk
service and create entirely new business models. These new “SIP Providers” are changing the
face of the VoIP industry.

5.1.1 Hosted SIP

While a SIP trunk only caters to firms who already have their own IP PBX system built and
ready to hook up to the PSTN, the latest breed of SIP providers don’t require even that. These
ITSPs have their own PBXs hosted on their servers and allow businesses to make use of them in
order to receive VoIP services at extremely low costs. 'Hosted PBX' services come with built-in
connections to the PSTN network, eliminating the need to purchase it separately.

“Hosted PBX” providers are like a web hosting service. A business owns a website, but doesn’t
own the servers on which that site is stored. Neither does it have its own colocation facilities to

36
house those servers. Instead, they use the resources of a web host like “GoDaddy” or “Bluehost”.
This makes it easy for the smallest businesses or even a private blogger to maintain their own
website.

Of course, every SIP Provider offering hosted PBX services will also provide traditional trunking
services since it’s a subset of their existing operations.

There are also many SIP services which are targeted towards private individuals such as
sip2sip.info and getonsip.com. These offerings allow people to get their own SIP address for
free. Sip2sip.info offers a PSTN connection while getonsip is currently SIP only, meaning you
can only make calls to SIP addresses.

5.2 Additional Benefits for Hosted SIP customers

SIP hosting does the same thing with VoIP communications by putting it within the reach of
firms who have no SIP infrastructure of their own. These new SIP Providers are redefining what
it means to provide an SIP service. Along with the ability to make VoIP calls, businesses get SIP
accounts to distribute to their employees. This means that anyone using a SIP device can call
them for free as long as they know the SIP address associated with that account.

5.2.1 SIP Address

An SIP address looks just like an email ID. By default, the domain of the email - the part just
before the “.com” will belong to the ITSP providing the service. When we host your VoIP
communications, your SIP address will look something like
“employeename@yourcompany.onsip.com.” However, we also give you the ability to change
the SIP address domain to match the corporate email ID which your employees use. If you have
access to the DNS records (and you certainly should,) your SIP address will simply be your
email ID.

Having a free SIP account is a great way of making free calls on the Internet. You only
need to choose a SIP provider that gives you a SIP account for free. There are many of these.
Here is list:

5.2.1.1 Get On Sip

Get On Sip is a straight service offered by Junction Networks, the company behind OnSip.com,
offering free SIP accounts in a very simple and easy registration. The registration fields stand
right in your face as soon as you load the home page. You are not restricted on the amount of SIP

37
accounts, and you get free SIP voice and video calling service and XMPP messaging. Get On
SIP also offers paid business SIP accounts.

5.2.1.2 IP Tel

IP Tel provides IP telecommunications services and hosts several projects like the SIP Express
Router, SIP Express Media Server and the SIP Express Router Web. They also provide on their
site a wealth of information on Sip communication. The free SIP account they offer is of good
quality and is also straightforward to register.

5.2.1.3 Ekiga SIP Service

Ekiga is a VoIP service that offers a soft phone and free VoIP service, and it is known for its
Linux VoIP app. Ekiga also offers free SIP addresses that you can use with any SIP-supporting
soft phone app.

5.2.1.4 Sip2Sip

Sip2Sip is a rather straightforward SIP service, offer by ag-projects.com. It is a free SIP service
based on fair use policy and no support. Registration and account management are easy. AG
Projects has lots of products and offers this free SIP service for users to test the features present
in its products.

5.2.1.5 AntiSIP

AntiSIP offers a set of SIP-based services among which is the free SIP account, to which you can
even receive landline calls, for example through the free US numbers provided by IPKCALL.

5.2.1.6 VoIPUser

VoIP User SIP account is a community effort of VoIP enthusiasts. Do not expect much support
from them. The registration process, while being straightforward, is rather restrictive. But if you
believe in community efforts, check them out.

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Chapter 6

SOCIAL NETWORKS:

6.1 INTRODUCTION

Social Networking involves the use of the internet to connect users with their friends, family and
acquaintances. Online social networks facilitate connections between people based on shared
interests, values, membership in particular groups (i.e., friends, professional colleagues), etc.
They make it easier for people to find and communicate with individuals who are in their
networks using the Web as the interface.

When it comes to online social networking, websites are commonly used. These websites
are known as social sites. Social networking websites function like an online community of
internet users. Depending on the website in question, many of these online community members
share common interests in hobbies, religion, politics and alternative lifestyles. Once you are
granted access to a social networking website you can begin to socialize. This socialization may
include reading the profile pages of other members and possibly even contacting them.

The friends that you can make are just one of the many benefits to social networking online.

Another one of those benefits includes diversity because the internet gives individuals from all
around the world access to social networking sites. This means that although you are in the
United States, you could develop an online friendship with someone in Denmark or India. Not
only will you make new friends, but you just might learn a thing or two about new cultures or
new languages and learning is always a good thing.

As mentioned, social networking often involves grouping specific individuals or

organizations together. While there are a number of social networking websites that focus on
particular interests, there are others that do not. The websites without a main focus are often
referred to as "traditional" social networking websites and usually have open memberships. This
means that anyone can become a member, no matter what their hobbies, beliefs, or views are.

39
However, once you are inside this online community, you can begin to create your own network
of friends and eliminate members that do not share common interests or goals.

Fig. growth of social networking sites

There are several different online social networks, but for our purposes, Each of these sites have
their own unique style, functionality and patterns of usage. You will also find that different
people are active in these different sites.

6.1.1 Facebook:
To access Facebook.com, you must create an account on the site which is free. Facebook's terms
of use state that members must be at least 13 years old with valid email ID’s. After updating
you're details, your Facebook profile is generated. Using Facebook.com you can:
Browse and join networks, which are organized into four categories: regions, colleges,
workplaces and high schools.

40
Pull contacts from a Web-based e-mail account, into Facebook.com.

Find friends in several ways, including search engine to look for a specific person and lot more.
Facebook has recently crossed 500 million users and is the most popular Social Networking site
of the world.

6.1.2 MySpace:
On MySpace, your social network starts growing from the first day. When you join MySpace,
the first step is to create a profile. You then, invite friends to join there and search for your
friends on already profiled on MySpace these friends become your initial Friend Space. Once the
friendship is confirmed all the people in your friends' Friend Space become part of your network.
In that sense, everyone on MySpace is in your Extended Network. As part of terms of MySpace,
the user must be at least 14 years old to register.

6.1.3 Twitter:
Twitter is a very simple service that is rapidly becoming one of the most talked-about social
networking service providers. When you have a Twitter account, you can use the service to post
and receive messages to a network of contacts, as opposed to send bulk email messages. You can
build your network of contacts, and invite others to receive your Tweets, and can follow other
members' posts. Twitter makes it easy to opt into or out of networks. Additionally, you can
choose to stop following a specific person’s feed.

6.1.4 LinkedIn:
LinkedIn is an online social network for business professionals, which is designed specifically
for professional networking, to help them find a job, discover sales leads, connect with potential
business partners. Unlike most of the other social networks, LinkedIn does not focus on making
friends or sharing media like photos, videos and music. To start using LinkedIn you need to
register and create a profile page. To register to LinkedIn, you need to provide personal
information. You can update the profile with your education and job details and a summary.
Additionally, you can also give and receive recommendations from co-workers and bosses.
There are more than 75 million professionals registered on LinkedIn.
41
6.1.5 Google+ :

As it made its debut in the early summer of 2011, Google+ was the fastest growing social
network the web has ever seen. It seems as if Google has finally created something that people
are actually excited to use – unlike previously failed attempts at jumping on the social media
bandwagon with projects like Google Wave and Google Buzz.

6.1.6 Hi5:
Hi5 shares many similarities with many social network sites; however, it introduces some twists
that make it worthwhile for people who love trying out new and interesting online communities.
However, it is not one of the popular sites in the United States. This was a strategic move from
the founder, therefore, Hi5 claims around 60 million members from more than 200 countries
other than the US. One of the site's biggest transformations is the addition of many entertainment
options, including games.

6.1.7 Orkut:
Orkut is a free social networking website where you can create a profile, connect with friends,
maintain an online scrapbook and use site features and applications to share your interests and
meet others. The prerequisite for logging on to Orkut is that the user must be over 18 years old.
Currently, Orkut is the most popular in Brazil. The number of orkut users in India is almost
equivalent to those in its original home in the United States.

6.1.8 Friendster:
Friendster was one of the first Web sites to bring it into mass culture. It was designed as a place
to connect with friends, family, colleagues and new friends over the Internet. However, it went
beyond just a one-way communication. Using Friendster, you can connect with friends and
family, meet new people through the connections you already have.

6.1.9 SkyRock:
SkyRock.com is a social networking site that offers its members free web space where they can
create a blog, add a profile, and exchange messages with other registered members. The site also

42
offers a specific space for members who create blogs showcasing their original musical
compositions.

43
CHAPTER 7

HOW APPLICATION WORKS

7.1 INTERNET PHONE SETTING

Hit the settings/options button on your Android device. Select 'Settings' and then 'Call settings'.
Scroll down to the bottom of this menu and you will see 'Internet call settings'. Select 'Accounts'.
You will see the option to "add accounts" here.
Again you will need your SIP account credentials, which you can find in the email we sent you
when you signed up.

Register

Login

Instant Messaging

Call

USER

Social Update

7.2 USER CREDENTIALS

o Select 'Add account' and enter in your credentials as follows.

o Username > Username
o Password > SIP Password
o Server > Domain
o Set as primary account (used for outbound calls) >Your choice
o Open up 'Optional Settings'.

44
o Authentication username > Authentication Username
o Display name > Your choice
o Outbound proxy address > Outbound proxy: sip.onsip.com

Hit the 'back' button to return to you accounts menu. The status of your account will be published
here so you will know if you did anything wrong. You can activate the option "receive incoming
calls" here.

7.2.1 DESCRIPTION OF STEPS

Step1: Voice communication can occur in 2 ways namely:
(a)Wi-Fi (without the SIM card in Android handsets)

(b)GPRS (with SIM card in Android handsets)

In the proposed work GPRS mode will be used.

Step2:
There would be 1 SIP server (Asterisk) and minimum 2 Android handsets. The handsets need to
be registered at SIP server.

Step3:
The SIP server of callcentric.com is used. SIP client is configured on both the Android handsets.

Step4:

45
The Android application which will be developed needs to be installed on all the handsets which
are registered in the SIP server and wish to communicate with each other using encrypted IP
voice communication.

Step 5:
The dialer will launch the Android application on his/her handset and dial the receiver’s number.
The dialing interface will be developed for the Android application. Once the call is made, the
request will go the SIP server wherein the recipient’s number will be checked and the call will be
routed on its handset.
Step 6:

Once the recipient receives the call through the same Android application and the dialer starts the
conversation, all the digitized voice data will first be encrypted on the dialer’s handset by the
application and then sent to the SIP server for routing to the recipient’s handset. Logically, every
handset will be assigned a numeric no. in the SIP server during configuration through which it
will become identifiable.

Step7:
The digitized voice data will travel in an encrypted fashion through the GPRS medium.

7.3 TECHNICAL WORK FLOW OF THE APPLICATION

1) The caller handset uses Secure Call application to initiate the VOIP call to the recipient.

2) When the call is made, the caller handset also sends the logical number of both caller and
receiver
3) Via GPRS which is cross-verified at the call centric server for genuineness.

4) Once the genuineness of both the numbers is verified the analog voice signals are
digitized using the SIP stack APIs which is PJSIP used in application. SIP stack APIs are
a collection of packages and functions used for the transmission of voice over IP using
SIP protocol.

46
5) Once the call is established successfully, the caller sends “Hello Packet” in order to
exchange the secure key with the recipient. The caller sends the “Hello Packets” a couple
of times. When the recipient sends the acknowledgement for the hello packet, the secure
key exchange is accomplished successfully. When the secure key is exchanges,the call
packets become encrypted.

6) Before the call is connected, the sender and recipient parties have to register themselves
with the.
SIGNUP Call Dialog
Call Setup
Call accept,
Create Making
Account calls,Reject
Call
Calls SIP server
Connected

Call
Call Accepted Forwarded

Personal
Sign in Call recieved
information
Call accept,
Manage
Making
Personal Log in/Log out
calls,Reject
Information
calls

Social network
Database

Facebook
Twitter
Etc
Request

Update/
Instant Server
Msg,etc

”Data Flow Diagram”

7) Call centric server in order to attain a logical number which is also added in the SIP client
on Android in
8) Order to synchronize with the server so that the SIP account gets activated.

47
 9) Once the SIP account is activated the SIP call session is established using SIP stack
which is the collection of APIs and the encryption is performed using ZRTP. SIP stack is
written in C language.

10) The social network portion deals in a way when user write any post the application post
their message to the social networks wall.

There are two different projects in the application.One project consists of the full SIP Android
interface design and the SIP client development and the another deals with the social networking
portion.
7.4 Cost Estimation
Resources Cost

Documentation 5000

Web Hosting 2000

SIP * 10000

* For continue service user will have to pay for SIP.

48
7.5 SNAP SHOTS OF APPLICATION

49
50
51
Chapter 8

8.1 INTRODUCTION TO THE TOOL.

Eclipse is for android development. Installing Eclipse is relatively easy, but does involve a few
steps and software from at least two different sources.
8.1 JAVA
This information covers processor, disk space, and memory requirements for the following
platforms:

 32-bit Platforms
 64-bit Platforms

8.1.1 32-bit Platforms

The following topics are covered:

 Processor Requirements
 Disk Space Requirements
 Memory Requirements

Processor Requirements
Both the Java SE Development Kit (JDK) and Java SE Runtime Environment (JRE) require at
minimum a Pentium 2 266 MHz processor.

Disk Space Requirements

For the JDK, you are given the option of installing the following features:

 Development Tools
 Source Code
 Public Java Runtime Environment

52
8.1.2 64-bit Platforms
The following topics are covered:

 Processor Requirements
 Disk Space Requirements
 Memory Requirements

Processor Requirements
Both the JDK and JRE require at minimum a Pentium 2 266 MHz processor.

Disk Space Requirements

 The JDK features available for 64-bit platforms are the same as those for Windows 32-bit
operating systems.
 The disk requirement for development tools for 64-bit platforms is 181 MB. The disk space
requirements for source code and the public JRE are the same as those for Windows 32-bit
operating systems, except for the JavaFX SDK (68 MB) and the JavaFX runtime (32 MB).

8.2 Eclipse
Installing Eclipse is relatively easy, but does involve a few steps and software from at least two
different sources.Eclipse is a Java-based application and, as such, requires a Java runtime
environment (JRE) in order to run.

53
8.2.1 RE/JDK Sources

There are several sources for a JRE/JDK. Here are some of the more common/popular ones
(listed alphabetically):

 IBM JDK
 OpenJDK
 Oracle JDK

Eclipse 4.2 (Juno)

Eclipse 4.2 (Juno) was released in June 2012.

A Java 6 JRE/JDK is recommended for Eclipse 4.2. The download will be delivered as a
compressed (i.e. a ".zip", or ".tar.gz") file. Decompress this file into the directory of your choice
(e.g. "c:\Program Files\Eclipse" on Windows). You can optionally create a shortcut of the
executable file ("eclipse.exe" on Windows, or "eclipse" on Linux).

54
Note that there is a known problem with the built-in decompression utility on all current versions
of Windows. We recommend that you use a more robust decompression utility such as the open
source 7zip when decompressing an Eclipse download. Some people report success when
initially decompressing Eclipse into a root directory (e.g. c:\) and then moving it to a more
appropriate home (e.g. c:\Program Files\Eclipse)

8.2.2 Extending Eclipse

Use the Help > Install new software... menu option to add Juno features to your Eclipse
installation (you can, for example, use this option to add C/C++ development support).
Additionally, you can tap into a vast collection of extensions provided by the Eclipse community
and ecosystem via the Eclipse Marketplace Client (Help > Eclipse Marketplace)

55
8.2.3 System requirements for Eclipse

Table 1-1. System requirements for Eclipse

Requirement Minimum Recommended

Java version 1.4.0 5.0 or greater

Memory 512 MB 1 GB or more

Free disk space 300 MB 1 GB or more

Processor speed 800 Mhz 1.5 Ghz or faster

8.3 Android development tool

8.3.1 ADT Plugin for Eclipse description

Provides a powerful, integrated environment in which to build Android applications.

Android Development Tools (ADT) is an Eclipse IDE plugin, that provides developers with a
powerful, integrated environment designed to creating applications for Android.

ADT extends Eclipse's capabilities so you can quickly setup new Android projects, create an
application user interface, add components based on the Android API, debug applications using
the Android SDK tools, and even exported signed (or unsigned) .APK files, in order to distribute
your Android application.

Development of Eclipse with ADT is the fastest way to start and is highly recommended.

56
 The project configuration, which provides guidance and tool integration, custom XML editors,
and debugging output panel, ADT provides an incredible boost in developing Android
applications.

8.3.2 Requirements:

· Eclipse 3.6.2 or later

· Java 1.6 or later
· Android SDK 21.0.1

8.3.3 DEPENDENCIES:
· Java 1.6 or higher is required for ADT 21.0.1.
· Eclipse Helios (Version 3.6.2) or higher is required for ADT 21.0.1.
· ADT 21.0.1 is designed for use with SDK Tools r21.0.1. If you haven't already installed SDK
Tools r21.0.1 into your SDK, use the Android SDK Manager to do so.

Installing the Eclipse Plugin

Android offers a custom plugin for the Eclipse IDE, called Android Development Tools (ADT).
This plugin provides a powerful, integrated environment in which to develop Android apps. It
extends the capabilities of Eclipse to let you quickly set up new Android projects, build an app
UI, debug your app, and export signed (or unsigned) app packages (APKs) for distribution.

Download the ADT Plugin

1. Start Eclipse, then select Help > Install New Software.

2. Click Add, in the top-right corner.

3. In the Add Repository dialog that appears, enter "ADT Plugin" for the Name and the following
URL for the Location:

https://dl-ssl.google.com/android/eclipse/

57
4. Click OK.

If you have trouble acquiring the plugin, try using "http" in the Location URL, instead of
"https" (https is preferred for security reasons).

5. In the Available Software dialog, select the checkbox next to Developer Tools and click Next.

6. In the next window, you'll see a list of the tools to be downloaded. Click Next.

7. Read and accept the license agreements, then click Finish.

If you get a security warning saying that the authenticity or validity of the software can't be
established, click OK.

8. When the installation completes, restart Eclipse.

Configure the ADT Plugin

Once Eclipse restarts, you must specify the location of your Android SDK directory:

58
1. In the "Welcome to Android Development" window that appears, select Use existing SDKs.

2. Browse and select the location of the Android SDK directory you recently downloaded and
unpacked.

3. Click Next.
Your Eclipse IDE is now set up to develop Android apps, but you need to add the latest SDK
platform tools and an Android platform to your environment. To get these packages for your
SDK, continue to Adding Platforms and Packages.

Troubleshooting Installation

If you are having trouble downloading the ADT plugin after following the steps above, here are
some suggestions:

 If Eclipse can not find the remote update site containing the ADT plugin, try changing the
remote site URL to use http, rather than https. That is, set the Location for the remote site to:

http://dl-ssl.google.com/android/eclipse/

 If you are behind a firewall (such as a corporate firewall), make sure that you have properly
configured your proxy settings in Eclipse. In Eclipse, you can configure proxy information
from the main Eclipse menu in Window (on Mac OS X, Eclipse)
> Preferences > General > Network Connections.

If you are still unable to use Eclipse to download the ADT plugin as a remote update site, you
can download the ADT zip file to your local machine and manually install it:

1. Download the ADT Plugin zip file (do not unpack it):

Package Size MD5 Checksum

ADT-21.0.1.zip 13569302 bytes acfb01bf3fd1240f1fc21488c3dd16bf

2. Start Eclipse, then select Help > Install New Software.

59
3. Click Add, in the top-right corner.

4. In the Add Repository dialog, click Archive.

5. Select the downloaded ADT-21.0.1.zip file and click OK.

6. Enter "ADT Plugin" for the name and click OK.

7. In the Available Software dialog, select the checkbox next to Developer Tools and click Next.

8. In the next window, you'll see a list of the tools to be downloaded. Click Next.
9. Read and accept the license agreements, then click Finish.

If you get a security warning saying that the authenticity or validity of the software can't be
established, click OK.

10. When the installation completes, restart Eclipse.

To update your plugin once you've installed using the zip file, you will have to follow these steps
again instead of the default update instructions.

60
Conclusion
Though VOIP is widespread among most businesses, IT staffs must understand the costs and
benefits of a new deployment or the resurgence of a stalled deployment. In doing so, it’s
important to evaluate the factors common to most companies, as well as those specific to the
company or the industry in question. By evaluating the return-on-investment, net present value,
and total cost of ownership, it will be much easier to secure funding for the project—and to keep
it safe during economic downturns. Moving forward, examine vendors’ roadmaps to determine
how they align with your plans. For example, some questions to ask include:

 To what extent do single products provide multiple unified communications applications?

Do you need to add new servers, appliances, or equipment to acquire new capabilities, or
are they rolled into a single platform?
 Are they supporting virtualization? Can you use the applications in virtual servers in the
data center? What is the plan to extend communications and collaboration applications to
a virtual desktop? Do they have virtual machines running on their appliances, making it
easier for customers to add applications?
 What architectures do they support? Is there flexibility to use either a distributed or
centralized approach?
 How easy is it to increase the size of the deployment? Are there numerous products, or a
fairly smooth migration either within a product family or between a small number of
them?
 What kind of expertise does the vendor and its partners have in your particular industry,
company size, or solution requirement?

With any vendor selection, though, the key business issue is to know your numbers. Vendors and
their resellers often provide you with their best figures available, but operational costs are
extremely variable and are based on your internal expertise. What’s more, vendors may provide

61
 capital estimates for IP telephony equipment but ignore the fact that you need a LAN upgrade.
Make sure to include the LAN upgrade in your IP telephony project costs if it’s required.
Finally, make sure your team evaluates all potential savings with VOIP, including replacement
of hard phones with soft phones, network consolidation, and staff reductions, among other areas.

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