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This report is for final year project to complete degree in Computer Science. It emphasis on Applications of Computer Sciences. It was supervised by Dr. Abhisri Yashwant at Bengal Engineering and Science University. Its main points are: Video, Conferencing, Asterisk, PBX, Client, Server, Application, Internet, Remote, Agent
Typology: Study Guides, Projects, Research
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The objective of this document is to stipulate the so far progress of the project “CC ANYWHERE”. This document comprises the bird view of distinguished modules with the overall architecture. The document covers installation & configuration of different modules of Asterisk PBX and the bottlenecks faced during installation.
The CC ANYWHERE™ is an innovation in call centre suite that incorporates the notion to allow workforce access the call centre from anywhere. Introducing the concept of web based call centre suite, the call centre won’t be merely contained in one building, but would allow administrators and call agents to interact with the system from anywhere on the internet using VoIP. As long as they have an internet connection and headphones and microphones, the agents can log into the system from anywhere in the world and they would be able to receive calls, customer information, campaign details and scripts using the web based call centre suite. Supervisors and Administrators can easily monitor their remote agents as effectively as if they were in the same building, by listening to recordings, viewing their real-time performance results, and sending messages to them individually or broadcasting to the entire group. Video conferencing feature of CC ANYWHERE adds the functionality of video conferencing in asterisk, which is not available in asterisk so far.
Here is the simple architecture of CC ANYWHERE.
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There are four main modules of CC ANYWHERE™. The best part of CC ANYWHERE™ modules is that they have a high level of cohesion and less coupling. The modules can perform their functionalities independently, and when integrated with each other would result in a complete solution of call center that has never been achieved so far.
1.3.1 Video conferencing application
Video conferencing application will be like a plug-in for asterisk to provide the functionality of video conferencing which is not yet been developed in asterisk. This module will take the video & audio data by collaborating with other modules of Asterisk and provide the conferencing feature. Video conferencing interface endows with the functionality to initiate a video conference among the employees of call center. A manager or supervisor can add different employees in a conference and they have the privileges to accept any other person to join the ongoing conference. The Video conference application resides within the Asterisk as shown in the diagram.
Asterisk ( PBX)
Asterisk PBX
Video Conferencing application
Server Application Internet
Client Application
Remote Agent
Figure 1 CCANYWHERE Overall Diagram
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1.3.3 SoftVideoPhone Application
The module is responsible for providing the image of a phone on computer screen where agent can interact using the buttons. The SoftVideoPhone application connects with the Sever application after authentication. It takes the data from user, and route it to the internet and also gets the inbox data and make the data available to the terminals.
SoftVideoPhone Application
Call Agent
Text Audio Video
In BOX
Internet Figure 4 CCANYWHERE client application overview diagram
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1.3.4 CRM Interfaces
This module is intended to provide all the customer data/history/log on the internet. Using the CRM interfaces the call agents and Administrators would be able to access customer records and agent records using a browser. It connects with the Asterisk database, and CRM integrated in Asterisk and gets data, to be displayed in the form of charts and other sophisticated ways, so that the analysis of customer records can be made available on fingertips.
VoIP Voice over Internet Protocol DML Dynamic Module Loader SIP Session Initiation Protocol CTI Computer Telephony Integration API Application Programming Interface IT Information Technology PSTN Public Switching Telephone Networks TDM Time Division Multiplexing PBX Private Branch Exchange IAX Inter-Asterisk Exchange TCP Transmission Control Protocol RTP Real Time Protocol
BRIs Basic Rate Interfaces PRIs Primary Rate Interfaces IVR Interactive Voice Response SVP Soft Video Phone Table 1 Acronyms and abbreviations
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The building process can only be done by having the root permissions. Write the following commands in terminal/console in order to navigate to the respective directory and then building the source. -- cd /usr/src/libpri-*
After the installation of libpri and zaptel in the respective order, Asterisk can be installed on the system. For this navigate to the directory where the extraction of asterisk archive was made by giving the command cd /usr/src/asterisk-.*
Now the commands should be given in exactly the specified order.
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This command may be skipped by fresh users, but it’s better to check it as it also has the information about inter dependences of different modules and drivers. If a required driver is not yet installed, the modules that are associated with it would have a double * to indicate the dependency with the driver.
This segment contains the detail about the work accomplished until now.
In order to configure & customize Asterisk Software based PBX to be used with CC ANYWHERE, changing and modification is made in the configuration files of different modules of Asterisk.
Figure 5 Screen shot of menuconfig command
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for other client if he/she is not available or busy at that moment and he/she can listen to those messages later. Manager can view the information the logs of any client from the CLI interface of Asterisk.
Audio conferencing feature is provided by Asterisk only if Digium® hardware is used. Zaptel drivers make use of the hardware to provide mixing engine for conferencing and timing source as Asterisk does not provide the timing itself. This problem is resolved by using the system clock as a timing source. Modifications are made in Zaptel driver files to point the internal clock to serve as a timing source. Successful audio conference is set up between the clients without using Digium hardware.
Asterisk provides the video call feature. In order to enable video in asterisk the sip.conf configuration file of asterisk is modified. Under the general section of sip.conf the following lines are added.
[general] videosupport=yes Then for each sip user we would have to add the supported codec. It is noted that only SIP and IAX2 channels provide the video call support. Video call feature is quite infancy in Asterisk. The following issues concerning video call, are observed.
Asterisk has question regarding video codec negotiation Advanced video attributes such as profile/level (H.263+, MPEG4, H.264), bandwidth, standard annexes, frame rate and image size are not exchanged and negotiated by Asterisk. Asterisk has no capability to playback classical video formats such as .mov, MP4 or 3gp files. Asterisk does not provide any video transcoding capabilities.
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In order to make use of the video call feature of Asterisk for video conferencing, the issues will be resolved by customizing and modifying the available algorithms for video codec negotiation, and transcoding for asterisk. The basic architecture of video conferencing application is given below.
Asterisk Application API
Asterisk Gateway Interface (AGI) Asterisk Management Interface (AMI) Paging Directory Dialing Voicemail Video Conferencing Application
Asterisk Channel API SIP IAX
Asterisk File Format API
Ph263p .h .h
Asterisk Codec Translator API
H.263p Speex
Codec translator
Application Launcher
Switching core
Scheduler And I/O Manager
CDR Core
Figure 6 Video Conferencing basic Architecture