IRJET-Identifying Disaster Area using Wireless Technology

International Research Journal of Engineering and Technology (IRJET) e-ISSN: 2395-0056
Volume: 05 Issue: 06 | June -2018 www.irjet.net p-ISSN: 2395-0072
© 2018, IRJET | Impact Factor value: 6.171 | ISO 9001:2008 Certified Journal | Page 18
Identifying Disaster Area using Wireless Technology
Aruna.P1, Viji.D2
1ME student, Dept of Electronics & Communication Engineering, Excel College of Engineering and Technology,
Tamil Nadu, India
2Assistant Professor, Dept of Electronics & Communication Engineering, Excel College of Engineering and
Technology, Tamil Nadu, India
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Abstract - The new technology has the ability to transmit a
voice over Internet protocol process networks by using an
Asterisk PBX. The purpose of this research is to design and
implement a telephony program thatusesexistingWIFIin p2p
(Peer-to- Peer) or WLAN (Wireless Local Area Network) as a
means of communication between mobile phones andexisting
intercom systems at no cost sothatcommunicationcanalsobe
established on disaster area using excisting WIFI. Theasterisk
software will use a correlation between currentaddress books
available in mobile phones to convert phone numbers into IP
addresses. Voice over Internet Protocol (VoIP)isusedforvoice
and data communication applications. VoIPapplicationshave
acceptable quality and relatively low cost when compared
with landline and cellular communication. Because of the
bandwidth efficiency and low costs, businesses are migrating
from traditional copper-wire telephone systems to VoIP
systems. Main objective of the paper is to build a common IP-
PBX system for traditional analog telephone system, digital
softphone and dedicated VOIP phones using Asterisk server.
Communication is established among various nodes using the
Soft-routing. In order to connect the wireless nodes to this
network, a wireless access point is created, through which the
smart phones are connected to this VOIP network.
Key Words: VOIP, Asterisk, WIFI, IP-PBX, WLAN
1. INTRODUCTION
VoIP is a form of communication that allows you to make
phone calls over a broadband internet connection insteadof
typical analog telephone lines. Basic VoIP access usually
allows you to call others who are also receivingcallsoverthe
internet. Interconnected VoIP services also allow you to
make and receive calls to and from traditional landline
numbers, usually for a service fee. [8] Some VoIP services
require a computer or a dedicated VoIP phone, while others
allow you to use your landline phone to place VoIP calls
through a special adapter. The operating system with
Asterisk is installed in Raspberry Pi. The Raspberry Pi is a
series of credit card–sized single-board computers
developed in the United Kingdom by the Raspberry Pi
Foundation with the intention of promoting the teaching of
basic computer science in schools and developing countries.
Asterisk supports audio protocols such as SIP which is
Session Initiation Protocol used for the audio
communication. This package consists of several features
such as Voicemail, Call Waiting, Caller ID, and Call Transfer
and so on.
In many years ago an old telephone system was improved to
a new substitute known as Private Branch Exchange (PBX).
PBX system performs communication tasks suchasinbound
calls and outbound calls. VoIP technology was developed in
1995. Some VoIP services need only a regular phone
connection, while others allow you to make telephone calls
using an Internet connection instead. Some VoIP services
may allow you only to call other people using the same
service, but others may allow you to call any telephone
number - including local, long distance, wireless and
international numbers. The support of telephony services
over mobile phone has been used everywhere using
technology such as GSM (Global System for Mobile) and 3rd
Generation mobile telecommunication 3G , but at high cost.
On the other hand, IP telephony try to reduce the cost for
supporting this service over mobile phone, but it is facing
difficulties since the same feature is supported on desktop
and laptop at lower complexity. The challenge is to provide
the same service over mobile phone at no cost, as it hasbeen
described in this project. Two approaches are suggested in
this paper to meet the objective of having free telephony
services over mobile phones. These are the use of WIFI
technology over AP (Access Point) and WIFI over p2p(Peer-
to-Peer).
1.1 Asterisk
Asterisk is a software implementation of a telephone
privatebranch exchange(PBX);itallowsattachedtelephones
to make calls to one another, and to connect to other
telephone services, such as the public switched telephone
network (PSTN) and Voice over Internet Protocol (VoIP)
services. Its name comes from the asterisk symbol "*". The
Asterisk software includes many features available in
proprietary PBX systems: voice mail, conference calling,
interactive voiceresponse(phonemenus),andautomaticcall
distribution. Users can create new functionality by writing
dial plan scripts in several of Asterisk's own extensions
languages, by adding custom loadable modules written in C,
or by implementing Asterisk Gateway Interface(AGI)
programs using any programming language capable of
communicating via network TCP sockets.
Asterisk supports several standard voice over IP
protocols, including the Session Initiation Protocol (SIP), the
Media Gateway ControlProtocol(MGCP),andH.323.Asterisk
supports most SIP telephones, acting both as registrar and
back-to-back user agent, andcan serve asagatewaybetween
IP phones and thepublicswitchedtelephonenetwork(PSTN)

via T- or E-carrier interfaces or analog FXO cards. The Inter-
Asterisk eXchange (IAX) protocol, RFC 5456, native to
Asterisk, provides efficient trunking of calls among Asterisk
PBXes, in addition to distributing some configuration logic.
Many VoIP service providers support it for call completion
into the PSTN, often because they themselves have deployed
Asterisk or offer it as a hosted application. Some telephones
also support the IAX protocol. [7]
By supportinga variety of traditional and VoIP telephony
services, Asterisk allows deployers to build telephone
systems, or migrate existing systems to new technologies.
Some sites are using Asterisk to replace proprietary PBXes,
others provideadditionalfeatures,suchasvoicemailorvoice
response menus, or virtual call shops, or to reduce cost by
carrying long-distance calls over the Internet (toll bypass).
Various codecs areusedforcompressionanddecompressthe
audio and video signals and are listed in table -1 and table -2
respectively.
Table -1: Audio Codecs used in VoIP
Table -2: Video Codecs used in VoIP
1.2 TDM TO PCM Interface
A TDM to PCM Interface, also called computer–telephone
integration or CTI, is a common name foranytechnologythat
allows interactions on a telephone and a computer to be
integrated or coordinated.Thetermispredominantlyusedto
describe desktop-basedinteractionforhelpingusersbemore
efficient, though itcanalsorefertoserver-basedfunctionality
such as automatic call routing. CTI enables control of the
phone through the PC as shown in figure 1. This allows
computer software such as email programs to connectyouto
the telephone system with just the click of a mouse.
To operate Computer Telephony Integration, the
computer needs the appropriate software to control the
system phone or communicate with the telephone system.
Many PBX manufacturers offer such software solutions for
their devices. For communication between the telephone
system and the computer software, there are standardized
protocols such as CSTA (Computer Supported
TelecommunicationsApplications).Thisprotocoldefinesthe
type and format of the data to be transmitted without
specifying the actual transport layer, thereby allowing CTI
solutions to be implemented by many more manufacturers.
Fig -1: Computer Telephony Integeration
1.3 IP Telephone
IP telephony (Internet Protocol telephony) is a general
term for the technologies that use the Internet Protocol's
packet-switched connections to exchange voice, fax, and
other forms of information that have traditionally been
carried over the dedicated circuit-switched connections of
the public switched telephone network (PSTN). Using the
Internet, calls travel as packets of data on shared lines,
avoiding the tolls of the PSTN. The challenge in IP telephony
is to deliver the voice, fax, or video packets in a dependable
flow to the user. Much of IP telephony focuses on that
challenge. Currently, unlike traditional phone service, IP
telephony service is relativelyunregulatedbygovernment.In
the United States, the Federal Communications Commission
(FCC) regulates phone-to- phone connections, but says they
do not plan to regulate connections between a phone user
and an IP telephony service provider.
VoIP is an organized effort to standardize IP telephony. IP
telephony is an important part of the convergence of
computers, telephones, andtelevisionintoasingleintegrated
information environment. Also see another general term,
computer-telephony integration (CTI), which describes
technologies for using computers to manage telephone calls.
VoIP phones can be simple software-based softphones as in
figure 3 or purpose-built hardware devicesasinfigure2,that
appear much like an ordinary telephone or a cordless phone.
Traditional PSTN phones are used as VoIP phones with
analog telephone adapters (ATA).

Fig -2: IP Phone – Audio Codecs 320HD
Fig -3: Zoiper Softphone for PC and Android
2. PROPOSED WORK
In the early years of telephone lines, calls went through
public switchboards, where operators manually directed
them to the correct receivers and are known as Electronic
Private Branch Exchange (EPBX) as shown in figure 4. As
computers developed, an update to PBX appeared which
enables the PBX to automatically route the calls to the
receiver. This system is known as Electronic Private
Automatic Branch Exchange (EPABX) and is shown in figure
4.
Fig -4: PBX and EPABX system
Finally, the internet came,andwithit,developerscreated
the ability to channel calls through the internet’s data
network and isknown as Voice overInternetProtocol(VoIP).
This sparked a business telephone revolution that would
continue to develop through the 21st century and into the
digital revolution. [8].Instead of physically connecting to the
PBX with copper wiring, phones connect to the PBX over an
office's Local Area Network (LAN), often leveragingthesame
Internet connectivity that your office computers do and the
general schematic of VoIP is shown in figure 5. Generally
there are two types of IP-PBX systems based on the location
of server. One of them is Hosted or cloud-based PBXsystems,
which are the virtual equivalents of the physical equipment
used by traditional business phone systems. Another one is
On-Site IP-PBX, which has the server locally.
Fig -5: Basic VoIP network Layout
2.1 Problem Statement
The purpose of this research is to design and implement
an intercom system that uses existing LAN (Local Area
Network) or WLAN (Wireless Local Area Network) as a
means of communication between mobile phones and
existing intercom systems at no cost. This system will allow
user to make voice and videoconversation.Inadditiontothat
conference calls can also be made at free of cost. Different
security services relevant for VoIP are presented and we
argue that end-to-end authentication and encryption should
be provided by default.
Fig -6: Block Diagram of Proposed IP-PBX system
VoIP applications have acceptable quality and relatively
low cost when compared with landline and cellular
Communication. Becauseofthebandwidthefficiencyandlow
costs, businessesare migrating from traditionalcopper-wire
telephone systems to VoIP systems. Main objective of the
project is to build a common IP-PBX system for traditional
analog telephone system, digital softphone and dedicated
VOIP phones using Asterisk server as shown in figure 6.
Communicationisestablishedamongvariousnodesusingthe

Soft-routing. In order to connect the wireless nodes to this
network, a wireless access point is created, through which
the smart phones are connected to this VOIP network.
2.2 Advantages of Proposed System
i) Proposed system is highly reliable, as it is
operated in LAN and WLAN. In addition to that, it
can be operated using small battery as the power
requirement is very less when compared to the
existing system.
ii) Better sound quality can be obtained by using the
CODECs for lossless compression. And also better
video quality can also be achieved in video calling
by using pass- through CODECs.
iii) Since the raspberry pi is fully customizable and
reprogrammable, the user or organization can
have the complete access and control.
iv) It is not mandatory to have the IP-phones. Instead
of that, smartphones, PCs, and LAPTOPs can be
used as IP-Phones using the softphone software.
3. HARDWARE DESCRIPTION
Hardware components of the proposed work explained
briefly along with their specifications as below:
3.1 RASPBERRY PI 3 - MODEL B
The Raspberry Pi 3 Model B is the third generation
Raspberry Pi as shown in figure 7. This powerful credit-
card sized single board computer can be used for many
applications and supersedes the original Raspberry Pi
Model B+ and Raspberry Pi 2 Model B. [8] Additionally it
adds wireless LAN & Bluetooth connectivity making it the
ideal solution forpowerfulconnecteddesigns.Raspberrypi
3 specifications are as below:
i) 1.2GHz Quad-Core ARM Cortex-A53
ii) Broadcom BCM2387 chipset
iii) 802.11 bgn Wireless LAN and Bluetooth
4.1 (Bluetooth Classic and LE)
iv) 1GB RAM
v) 64 Bit CPU
vi) 10/100 BaseT Ethernet socket
Fig -7: Raspberry pi 3 – Model B
3.3 WIFI Router
A wireless router is a device that performs thefunctions
of a router and also includes the functions of a wireless
access point. It is used to provide access to the Internet
or a private computer network. Depending on the
manufacturer and model, it can function in a wired local
area network, in a wireless-only LAN, or in a mixed wired
and wireless network.
Table -3: 802.11 physical layer standards
3.3. IP-Phone
A VoIP phone or IP phone uses voice over IP technologies
for placing and transmitting telephone calls over an IP
network, such as the Internet, instead of the traditional
public switched telephone network (PSTN).Digital IP-based
telephone service uses control protocols such as the Session
Initiation Protocol (SIP), Skinny Client Control Protocol
(SCCP) or various other proprietary protocol. AudioCodes
320HD IP Phone is based on AudioCodes proprietary High
Definition (HD) voice technology, providing clarity and a
better audio experience in Voice- over-IP(VoIP) calls, which
is shown in figure 8.
The 320HD IP Phone is a fully-featured telephone that
provides voice communication over an IP network, allowing
you to place and receive phone calls, put calls on hold,
transfer calls, make conference calls, and so on
Fig -8: Audio Codec 320HD IP-Phone

4. SOFTWARE DESCRIPTION
4.1 ASTERISK WITH FREEPBX
Asterisk is a software implementation of a telephone
private branch exchange (PBX); it allows attached
telephones to make calls to one another, and to connect to
other telephone services, such as the public switched
telephone network (PSTN)andVoiceoverInternetProtocol
(VoIP) services. Its name comes from the asterisk
symbol "*".
FreePBX is a web-based open source graphical user
interface (GUI) that manages Asterisk, a voice over IP and
telephony server. FreePBX is licensed under the GNU
General Public License version 3. It is a component of the
FreePBX Distro, which is an independently maintained
Linux system derived from the source code of the CentOS
distribution, having Asterisk pre- installed. Figure 9 shows
the FreePBX administrative panel for asterisk.
Fig -9: FREEPBX control panel
4.2 LINUX OS IN ATA
SIP (Session Initiation Protocol) is identified by IETF,
use to create, change and release the session of one or
several participants. These sessions are like Internet,
Multimedia Conference, IP phone or Multimedia
Distribution. The session participants could communicate
by Multicast, Mesh Unicast, or mixtureofthetwo.Figure10
shows the configuration of the analog phone using the SIP
protocol in the gateway web server.
Fig -10: Easy Phone HTTP webserver
5.RESULT AND DISCUSSION
The proposed IP-PBX using the raspberry-pi is tested by
running the system for continuously for 6 hours and traced
the activity of the system for malfunction. In the same
manner the system is tested for 3 days. But the system
operates perfectly on the three days. During those testing,
various wired and wireless nodes are connected to the
server and variouscallsaremadesimultaneouslyfortesting
and the functionality doesn’t change with time.
T=N/fs=64/(20MHz)=3.2μs
From the time period the frequency is 312.5 KHz. Ion
Here 802.11n 20 MHz uses 64 subcarriers. Among them 48
subcarriers are data, 4 are pilots, 11 are zero-padded on
either side of the spectrum, and 1 is the null DC subcarrier.
The zero-padding also reduces interference between
adjacent frequency channels.
The connection Layout of ProposedIP-PBXsystemisshown
below
Fig 11: Connection Layout of Proposed IP-PBX System
By using the configuration shown in figure 11, the video
call is established between the PC and the smartphone by
using the zoiper softphones. When the data rate is below
256Kbps, then there is no jitter in the call. Figure 12 shows
the actual hardware setup.
Fig -12: Hardware set up of Proposed IP-PBX system

6. CONCLUSION
The raspberry pi based intercom system can be installed
easily along with the existing intercom or LAN or WLAN
systems. This type conversion enables the user to use the
smartphone as a node of intercom system with free of cost.
Due to low cost, small and medium scale industries and
college and school campus can afford this system with less
investment. Then by installing this system in Public Wi-Fi,
free calling facility can be provided to the Public.
REFERENCES
[1] Ashwini S.Gawarle (2017), “Design a Free Voice
Calling System Using Raspberry Pi”, International
Journal of Research in Engineering and Applied
Sciences (IJREAS).
[2] Chunhui Yuan and Hongli Zhao (2016),
“Implementing VoIP VoiceCommunicationSystem
based on Soft-switch Technology”, 2016
International Conference on Cyber-Enabled
Distributed Computingand Knowledge Discovery.
[3] Hussein Al-Saadawi and Asaf Varol (2017), “Voice
over IP Forensic Approaches: A Review”, 978-1-
5090-5835-8/17/$31.00 ©2017 IEEE.
[4] Nikhil A.karotiya, Nikhil P.Wyawahare and
Dr.S.L.Haridas (2016), “Review paper on Point to
Point Communication with the use of Power over
Ethernet based on VOIP system on Asterisk”,
InternationalConferenceonAdvancesinElectrical,
Electronics, Information, Communication andBio-
Informatics (AEEICB16).
[5] Pramila.B.Bamnote (2016), “Design and
implementation of wifi based intercom system
using Arm11”, International Research Journal of
Engineering and Technology (IRJET).
[6] Rahul C. Vaidya and Prof. S.S. Kulkarni (2012),
“Voice over IP Mobile Telephony Using WIFI”,
International Journal of Scientific & Engineering
Research Volume 3, Issue 12, December-2012.
[7] RyotaNishimura,KojiSugioka,DaisukeYamamoto,
Takahiro Uchiya, and Ichi Takumi (2014), “AVoIP-
based Voice Interaction System for a Virtual
Telephone Operator Using Video Calls”, 2014IEEE
3rd Global Conference on Consumer Electronics
(GCCE).
[8] Serge Fabrice Mbianda Ngongang, Navid Tadayon
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Feasibility Analysis”, 2016 Advances in Wireless
and Optical Communications (RTUWO).

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