This document provides an overview of a project report on Voice over Internet Protocol (VoIP) submitted by two students, Amardeep Singh and Jaswinder Singh, at Chandigarh Engineering College in partial fulfillment of their B-Tech degree in Electronics and Communication Engineering. The report introduces VoIP technology, discusses software and hardware used in the project including Cisco routers and switches, and provides details on configuring an IP phone network with Cisco Call Manager Express including assigning IP addresses via DHCP and configuring phone directory numbers. Future enhancements discussed include integrating VoIP with wireless networks.

1. A
Project Report
Submitted in Partial fulfillment for the requirement for
award of B-Tech in
Electronics and Communication Engineering.
To
The Punjab Technical University,
Jalandhar.
Department of
ELECTRONICS & COMMUNICATION ENGINEERING
CHANDIGARH ENGGINEERING COLLEGE.
LANDRAN.
May-June 2014
Undertaken at
HCL CDC Mohali
Submitted By:
Amardeep Singh(1181771) Jaswinder Singh(1181786)

2. Project on VOIP
1.1 Introduction to VOIP
Voice over Internet Protocol (VoIP) is a methodology and group of technologies for the
delivery of voice communications and multimedia sessions over Internet Protocol (IP)
Networks, such as the Internet. Other terms commonly associated with VoIP are IP telephony,
Internet telephony, voice over broadband (Dobb), broadband telephony, IP communications,
and broadband phone service.
The term Internet telephony specifically refers to the provisioning of communications services
(voice, fax, SMS, voice-messaging) over the public Internet, rather than via the public switched
telephone network (PSTN). The steps and principles involved in originating VoIP telephone
calls are similar to traditional digital telephony and involve signaling, channel setup,
Digitization of the analog voice signals, and encoding. Instead of being transmitted over a
circuit-switched network, however, the digital information is packetized, and transmission
occurs as Internet Protocol(IP) packets over a packet- switched network. Such transmission
entails careful considerations about resource management different from time-division
multiplexing (TDM) networks.
Early providers of voice over IP services offered business models and technical solutions that
mirrored the architecture of the legacy telephone network. Second-generation providers,
such as Skype, have built closed networks for private user bases, offering the benefit of free
calls and convenience while potentially charging for access to other communication networks,
such as the PSTN. This has limited the freedom of users to mix-and-match third-party hardware
and software. Third-generation providers, such as Google Talk, have adopted the concept of
federated VoIP—which is a departure from the architecture of the legacy networks. These
solutions typically allow dynamic interconnection between users on any two domains on the
Internet when a user wishes to place a call.
Useful Terms
Understanding the terms is a first step toward learning the potential of this technology:
 VoIP refers to a way to carry phone calls over an IP data network, whether on the
Internet or your own internal network. A primary attraction of VoIP is its ability to help
reduce expenses because telephone calls travel over the data network rather than the
phone company's network.
 IP telephony encompasses the full suite of VoIP enabled services including the
interconnection of phones for communications; related services such as billing and
dialing plans; and basic features such as conferencing, transfer, forward, and hold.
These services might previously have been provided by a PBX.

3.  IP communications includes business applications that enhance communications to
enable features such as unified messaging, integrated contact centers, and rich-media
conferencing with voice, data, and video.
 Unified communication stakes IP communications a step further by using such
technologies as Session Initiation Protocol (SIP) and presence along with mobility
solutions to unify and simply all forms of communications, independent of location,
time, or device. (Learn more about unified communications.)
VoIP systems employ session control and signaling protocols to control the signaling, set-up,
and tear-down of calls. They transport audio streams over IP networks using special media
delivery protocols that encode voice, audio, video with audio codecs, and video codecs as
Digital audio by streaming media. Various codec exist that optimize the media stream based
on application requirements and network bandwidth; some implementations rely on
narrowband and compressed speech, while others support high fidelity stereo codecs. Some
popular codecs include μ-law and a-law versions of G.711, G.722, which is a high-fidelity
codec marketed as HD Voice by Polycot, a popular open source voice codec known as LBC, a
codec that only uses 8 Kbit/s each way called G.729, and many others.
VoIP is available on many smart phones, personal computers, and on Internet access devices.
Calls and SMS text messages may be sent over 3Gor Wi-Fi
1.2 Software & Hardware Requirements
To complete the work on VOIP NETWORK, I need a help from some software requirements.
Software requirements are as follow:
 Cisco Packet Tracer:
Used to do the project work easily & proper understanding.
 Windows 7:
It is an operating system. It is an interface unit between the user and hardware device.
 Microsoft Office:
It is used to save the work done on the project.
Hardware Used
 Routers : Cisco 2811 Series.
 Switches : Cisco 2960 Series.
 Devices : Computers, Servers, IP phones.
 Other Media : Console cables, Ethernet cables, Serial cable etc.

4. 1.3 Network Layout (Front end + Back end)

5. 1.4 IP Phones
A IP phone uses voice over IP (VoIP) technologies allowing telephone calls to be made over
an IP network such as the Internet instead of the ordinary PSTN system. Calls can traverse the
Internet, or a private IP network such as that of a company. The phones use control protocols
such as Session Initiation Protocol(SIP), Skinny Client Control Protocol(SCCP) or one of
various proprietary protocols such as that used bySkype. It is commonly refers to the
communication protocols, technologies and transmission techniques involved in the delivery
of voice communications and multimedia sessions over Internet Protocol (IP) networks, such
as the Internet.
Session Initiation Protocol (SIP) is a signaling protocol widely used [citation needed] 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 (Uni-cast) or
multiparty (Multi-cast) sessions. Sessions may consist of one or several media streams.

6. Skinny Client Control Protocol (SCCP) is a
proprietary network terminal controlprotocol. SCCP
is a lightweight protocol for session signaling with
Cisco CallManager.
Examples of SCCP clients include the Cisco 7900
series of IP phones, Cisco IP Communicator
softphone along with Cisco Unity voicemail server.
CallManager acts as a signaling proxy for call events
initiated over other common protocols such as Session
Initiation Protocol (SIP), ISDN.A SCCP client uses
TCP/IP to communicate with one or more Call
Manager applications in a cluster. It uses the Real-
time Transport Protocol (RTP) over UDP-transport
for the bearer traffic (real-time audio stream).
Configuration of IP Phones
First you need to set the following topology ip phones / analog phones but connect phones to
power one by one after finishing configuration:

7. Next you will need to configure your switch with the following commands:
 Switch(config)#interface range fa0/1 – 6
 Switch(config-if-range)#switchport mode access
 Switch(config-if-range)#switchport voice vlan 1

Then we need to configure our router to provide IP address to IP phones and set the calling
numbers for phones , we will use CME call manager express embedded with router IOS itself.
 Router(config)#interface fa 0/0
 Router(config-if)#ip add 10.0.0.1 255.0.0.0
 Router(config-if)#no shut
 Router(config-if)#exit
 Router(config)#ip dhcp pool HCL
 Router(dhcp-config)#network 10.0.0.0 255.0.0.0
 Router(dhcp-config)#default-router 10.0.0.1
 Router(dhcp-config)#option 150 ip 10.0.0.1
 Router(dhcp-config)#exit
 Router(config)#telephony-service
 Router(config-telephony)#max-dn 10 ( max numbers on directory)
 Router(config-telephony)#max-ephones 10 (max number of phones)
 Router(config-telephony)#ip source-address 10.0.0.1 port 2000
 Router(config-telephony)#auto assign 1 to 10
 Router(config)#ephone-dn 1 (phone number 1)
 Router(config-ephone-dn)#number 100 (phone calling number)
 Router(config)#ephone-dn 2
 Router(config-ephone-dn)#number 101
 Router(config)#ephone-dn 3
 Router(config-ephone-dn)#number 102
 Router(config)#ephone-dn 4
 Router(config-ephone-dn)#number 103
 Router(config)#ephone-dn5
 Router(config-ephone-dn)#number 104

8. Chapter 8. Future Enhancement
Voice over Internet Protocol (VoIP) is one of the hottest and most hyped technologies in the
communications industry. Businesses and consumers are already taking advantage of the cost
savings and new features of making calls over a converged voice-data network, and the logical
next step is to take those advantages to the wireless world. The most widely publicized benefit
of VoIP is the ability to save costs on long distance charges and to network multiple offices
together. Businesses that have a data connection between their offices can utilize VoIP
technology to bypass long distance networks and provide more efficient communications
between offices. In a traditional setting, someone would have to dial the phone number to a
branch office, possibly paying a long distance charge for the call, wait for a receptionist or
automated system to answer and then become connected to the party they’re trying to reach.
Using VoIP, a person can simply dial an extension number and be connected immediately to a
party in another office, whether across town or around the world avoiding costly long distance
charges.
A second benefit is in the design of many telephone systems, often called IP based
systems. Rather than traditional phone systems with their own wiring infrastructure, IP based
systems use a data network infrastructure. This convergence of voice and data into a single
platform has tremendous advantages in simplifying the administration of the communications
network. Plus, IP utilizes data infrastructure that most likely already exists in many companies.
A third benefit is the ability to have remote phones with a single telephone number. For
example, an employee could work out of their home in New York, utilizing a phone number
with a California area code. This enables corporations to truly take advantage of having a
virtual office and or remote agents working out of a variety of location