International Journal of Information Technology (IJIT) – Volume 6 Issue 2, Mar - Apr 2020
RESEARCH ARTICLE OPEN ACCESS
Enhancing Social Media Experience through Voice over Internet
Protocol (VOIP) Using Asterisk and PHP
Arnold Mashud Abukari [1], Akhmad Daniel Sembiring [2]
Computer Science Department [1], Tamale Technical University, Ghana
Vitraining [2], Indonesia
ABSTRACT
The demands of people for Information Technology related services are very high and this needs to be catered for. In view of
this, Our solution which is in the form of a service is developed to contribute to the ever dynamism of the social networking
sector of Information Technology. The solution was developed by adopting the Agile methodology. This methodology is
customer centered and considers the continual inputs of the customer as a vital key in streamlining the process to produce a
better product/service.
Keywords:- VOIP
their messages on their various products or personal messages.
I. INTRODUCTION
To this end, our proposed solution seeks to achieve
The use of Information Technology is one of the fastest developing a system that can deliver voice messages on
developing sectors of any economy. This has given rise to so facebook and thereby enhancing social networking sites.
many innovations to improve and make human life very easy. The goal of this paper is to design and implement a facebook
Notably among the improvements is the emergence of Voice service using voice over internet protocol (VoIP) that will
Over Internet Protocol which enables users to communicate allow users to dial and record their voices or sound onto their
voice messages through the TCP/IP (Transmission Control facebook walls. We designed our proposed solution using
Program/Internet Protocol). Asterisks application and PHP whiles using a softphone which
Information Technology has also succeeded in bringing is only accessible through the use of a configured softphone or
people with varied backgrounds and different geographical IP Phone within a reachable network to test the proposed
locations together through the emergence and enhancements system.
of social media sites and social networking. These social Agile methodology is an approach to the project
networking sites have contributed towards breaking social management/Software development which help to respond to
barriers, improving businesses and serves as a media for the unpredictability of building software through incremental,
entertainments and reliable and efficient communication. iterative work cadences, known as sprints. This methodology
Facebook has distinguished itself in achieving the set was developed to deal with situation where the waterfall
objectives of a social networking site. It is against this model fails. The biggest drawback of waterfall model is that it
background that this Voice Over Internet Protocol (VoIP) assumes that every requirement of the project can be
service called is developed and documented in this paper. identified before any design or coding occurs.
Our proposed Solution is intended to enhance and bring Agile Methodology is the ability of a project to sense an
dynamism of posting messages on facebook. Businesses and environmental change and respond efficiently and effectively
individuals can post voice-based or sound-based messages on to that change.
their facebook walls. This will enable interested parties to This paper adopted the Agile Methodology in developing our
understand the nature of their businesses or to authenticate proposed solution due to it’s flexibility and growing wide use.
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International Journal of Information Technology (IJIT) – Volume 6 Issue 2, Mar - Apr 2020
Facebook users and is not exclusive to T-Mobile customers
(Kaplan and Haenlein, 2010).
SOFTPHONE
A softphone is a software program for making telephone calls
over the Internet using a general purpose computer, rather
than using dedicated hardware. Often a softphone is designed
to behave like a traditional telephone, sometimes appearing as
an image of a phone, with a display panel and buttons with
which the user can interact. A softphone is usually used with a
headset connected to the sound card of the PC, or with a USB
phone.
Figure 1: Agile methodology ASTERISK OPEN PBX
Content Management System using CakePHP was designed to
Asterisk is an open source framework for building
get the required data and information from interested users.
communications applications. Asterisk turns an ordinary
These information will be used to authenticate the facebook
computer into a communications server. Asterisk powers IP
user before messages could be posted to his facebook wall.
PBX systems, VoIP gateways, conference servers and other
The Content Management System (CMS) runs a database for
custom solutions. It is used by small businesses, large
storing user information.
businesses, call centers, carriers and government agencies,
worldwide. Asterisk is free and open source.
II. EXISTING SYSTEMS ON VOIP
FACEBOOK
Asterisk is a framework for building multi-protocol real-time
communications applications and solutions. Asterisk is to
T-Mobile on April 19, 2011 introduced Bobsled, a new brand realtime voice and video applications as what Apache is to
aimed at bridging traditional telecommunications and Internet- web applications: the underlying platform. Asterisk abstracts
based voice and data services to enable people to stay the complexities of communications protocols and
connected in a simple and cost-effective way. The first technologies, allowing you to concentrate on creating
product available under the new Bobsled by T-Mobile brand innovative products and solutions.
is the Bobsled application for Facebook, which provides
Facebook’s more than 500 million users worldwide with free,
one-touch calling to their Facebook friends from a personal
III. OUR PROPOSED SYSTEM
computer and through the social platform’s chat window. To be able to access the VoIP facebook post service the user
needs to register with his facebook details through a network.
In addition to making live voice calls across the globe, users
The following process shows how to successfully register a
will also be able to send voice messages to their friends either
person for the service:
privately or via their “walls.” The Bobsled application for
Facebook is available today as a free download for all
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International Journal of Information Technology (IJIT) – Volume 6 Issue 2, Mar - Apr 2020
The user visits the CMS with URL
http://192.168.1.100/agi.recorded/users/login
User enters facebook details, four (4) digit PIN, email
and SIP extension number
User details are stored onto the database
The system validate your facebook details with
facebook
User account is created
Figure 4: An IVR flowchart on how to use the VoIP facebook
post service.
The figure above is the flow chart for using the twitty service.
A user dials 6000 to access twitty’s IVR (interactive voice
response). The IVR guides the user to record and post voice
message on facebook.
SYSTEM IMPLEMENTATION
The experiment set up for this project is implemented in one
of two ways; local and global setup.
Figure 2: User details information form on CMS
Figure 3: how users are stored on the CMS
SENDING A VOIP POST ON FACEBOOK
Figure 5: Local setup
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International Journal of Information Technology (IJIT) – Volume 6 Issue 2, Mar - Apr 2020
callerid=211
[201]
type=friend
host=dynamic
[203]
type=friend
host=dynamic
[204]
type=friend
host=dynamic
[205]
type=friend
host=dynamic
[206]
type=friend
host=dynamic
EXTENTIONS CONFIGURATIONS (extentions.conf)
Figure 6: Global Setup The dialplan and extentions were configured at the
extentions.conf file of the Asterisk system. This file is located
SESSION INITIATION PROTOCOL
at the /etc/asterisk directory. The below configurations was
CONFIGURATION (sip.conf)
done on the extensions.conf file:
The Session Initiation Protocol (SIP) is an IETF (Internet
[default]
Engineering Task Force)-defined signaling protocol widely exten =>2000,1,AGI(curr.php)
exten => 6000,1,AGI(cobajuga.php)
used for controlling communication sessions such as voice
exten => 6000,1,AGI(voiceupdateprompt.php)
and video calls over Internet Protocol (IP). The protocol can exten => 1533,1,AGI(tesUpdate.php)
exten => 1534,1,AGI(tesUpdate.php)
be used for creating, modifying and terminating two-party
In the above configurations, [default] indicates the context
(unicast) or multiparty (multicast) sessions. Sessions may
name having the respective extention numbers and dialplans.
consist of one or several media streams.
The pointers to the files to be accessed are also indicated
The sip.conf of the asterisk is where the protocols were
in .php file extensions as well as the Asterisk Gateway
configured to be able to communicate with the IP Phones or
Interface (AGI). The voiceupdateprompt.php file was used for
the softphones. The sip.conf can be found at /etc/asterisk
my final presentation.
directory of the Asterisk server. Below are the configured
ASTERISK COMMAND LINE INTERFACE (CLI)
parameters for sip.conf:
The command Line Interface of asterisk monitors the
[200]
activities of all configurations on asterisk. To access the
type=friend
host=dynamic Asterisk command line interface, the linux command
“/etc/init.d/asterisk start” to start asterisk was sent.
[211]
username=211 A successful registration of the softphone or IP Phone is
secret=1234
shown on the Command Line Interface (CLI). The figure
context=default
type=friend below indicates a successful registration as indicated by the
host=dynamic
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International Journal of Information Technology (IJIT) – Volume 6 Issue 2, Mar - Apr 2020
softphone in its “ready” state and as detected by the Asterisk
CLI:
Figure 9: SIP Account configuration
The command line interface (CLI) will be shown on the linux
system whiles the softphone status changes to “ready” as
indicated below:
Figure 7: Command Line Interface
CONFIGURING THE SOFTPHONE (X-lite 3.0)
After installing x-lite 3.0, the softphone was configured to
communicate with the asterisk server using the “SIP account
settings” at the show menu of the softphone.
Figure 10: softphone successfully connected to asterisk server
If Asterisk was already running before changing the sip.conf,
then reload the dialplan and SIP channel with the following
two commands:
*CLI> dialplan reload
*CLI> sip reload
THE PROPOSED SYSTEM’S CMS
Figure 11: CMS Login interface
Figure 8: Softphone configuration
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International Journal of Information Technology (IJIT) – Volume 6 Issue 2, Mar - Apr 2020
voiceupdateprompt.php: [agi_calleridname] => 211
…
voiceupdateprompt.php: [agi_dnid] => 6000
…
voiceupdateprompt.php: [agi_context] => default
voiceupdateprompt.php: [agi_extension] => 6000
…
voiceupdateprompt.php:
-- Playing 'voicefb/2mono' (escape_digits=)
(sample_offset 0)
Figure 12: Registered users on CMS
-- Playing 'voicefb/1mono' (escape_digits=)
(sample_offset 0)
IV. RESULTS AND ANALYSIS -- <SIP/211-0000000f> Playing 'beep.ulaw' (language 'en')
The extension number configured to access the twitty service …
is 6000. A user dials the number and establishes a call. A -- Playing 'voicefb/3mono' (escape_digits=) (sample_offset 0)
preprogrammed voice prompt guides the user to complete the -- <SIP/211-0000000f> Playing 'beep.ulaw' (language 'en')
process. The figure below indicates the status of the softphone -- Playing 'voicefb/7mono' (escape_digits=) (sample_offset
when a call has been established. 0)
-- Playing '/var/www/messages/211_20111121190350.mp3
(escape_digits=) (sample_offset 0)
…
[Nov 10 18:05:35] WARNING[3257]: chan_sip.c:3859
__sip_autodestruct: Autodestruct on dialog
'f766b336e1707e10ZDI0ZDE3ZTM4MzAzMDQwZGYwMGM
1NzRiMTUxYjQyNGE.' with owner in place (Method: BYE)
[Nov 10 18:06:35] NOTICE[3257]: chan_sip.c:24173
handle_request_subscribe: Received SIP subscribe for peer
Figure 13: Call established without mailbox: 211
After the call is established, the CLI displays all activities
regarding the established call and all voice processing After posting your voice message on facebook, the softphone
application in the configured context (default) as shown in the is then changed into the “hung up” state as indicated below:
outputs below.
== Using SIP RTP CoS mark 5
-- Executing [6000@default:1] AGI("SIP/211-0000000f",
"voiceupdateprompt.php") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-
bin/voiceupdateprompt.php
…
voiceupdateprompt.php: [agi_callerid] => 211
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International Journal of Information Technology (IJIT) – Volume 6 Issue 2, Mar - Apr 2020
V. CONCLUSION AND RECOMMENDATION
Our proposed solution enhances the facebook platform by
offering alternative and dynamic way of posting voice
messages on facebook user’s wall through an Interactive
Voice Response (IVR) application.
Social networking has significantly influenced human
interactions and thus open to, and in need of more innovation
Figure 14: Hang up state and growth. In view of this, our proposed solution has
211_20111121190350.mp3 contributed to the growing dynamism of the social networking
landscape.
It is however recommended that:
Developers should regularly monitor the facebook
API’s to identify changes made by facebook to
enable them modify their codes to always keep the
service active.
More security measures should be instituted to
protect users information from unauthorized access.
Telecommunication companies should install
Asterisk E1 cards to enable users to call using their
own telephone lines/mobile phones
Studies to enhance this service should be conducted
to make it very user friendly and accessible.
Other social networking platforms should be
included to enable participation of all social media
users.
Value Added Service (VAS) Companies should
improve the service to make it a profit making
service of the telecommunication industry.
ACKNOWLEDGMENT
We wish to acknowledge the contributions and support of
Mr. Emmanuel Egyei Mensah and Pearl Esenam Dumenu all
of Quantum Group Ltd in Ghana.
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Figure 15: Posts on facebook (page 61)
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International Journal of Information Technology (IJIT) – Volume 6 Issue 2, Mar - Apr 2020
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