Desain dan Implementasi Sistem Contac center berbasis Asterisk Server

ABSTRAKSI: Perkembangan teknologi telekomunikasi yang cukup pesat mendorong kebutuhan untuk berkomunikasi mendapatkan suatu informasi melalui telepon semakin tinggi.Di sisi lain, call center yang biasanya digunakan di perkantoran atau universitas untuk memudahkan mendapatkan informasi, masih menggunakan komunikasi circuit based.Padahal dalam dunia komunikasi global saat ini, trend komunikasi mulai bergeser dari jaringan circuit atau PSTN ke model komunikasi melalui IP atau yang biasa disebut VoIP (Voice over IP). Komunikasi yang popular saat ini adalah komunikasi yang berbasis web, dimana web mulai diterapkan untuk menjadi call center yang komunikasinya melalui IP, sehingga fungsi dari call center yang hanya melayani panggilan menjadi contact center yang memiliki fitur panggilan, chat dan FAQ. Untuk bisa mensupport data suara digital yang beroperasi di jalur internet protokol dibutuhkan suatu hardware yang disebut IP PBX yang saat ini fungsi dari IP PBX sudah bisa digantikan oleh software yang disebut Asterisk . Pada tugas Akhir ini dirancang dan direalisasikan suatu contact center system yang memiliki fitur call audio, call audio video, chat dan Frequently Asked Question yang bertujuan untuk memudahkan pengguna mendapatkan suatu informasi dalam berbagai cara. Fitur call baik audio maupun audio video menggunakan konsep VoIP dengan protokol SIP dengan memanfaatkan Asterisk server yang mempunyai fungsi-fungsi PBX. Dari hasil pengujian yang dilakukan, didapatkan hasil dari call audio yaitu one way delay dengan rata-rata 68.977 ms, jitter dengan rata-rata 13.753 ms, dan packet loss dengan rata-rata 4.8 %. Nilai MOS yang didapatkan dari panggilan audio adalah sebesar 4.328 dan untuk panggilan video adalah sebesar 2.04.Kualitas panggilan audio termasuk ke dalam kategori Bagus kerena memiliki nilai MOS 4.328 dari 5, sedangkan kualitas panggilan video bisa dikatakan tidak terlalu bagus karena memiliki nilai MOS 2.04 dari 5.KATA KUNCI: VoIP, Asterisk, Contact Center, SIPABSTRACT: The development of telecommunications technology rapidly making the need to communicate to get information over the phone higher. On the other hand, call center are typically used in the office or university to make it easier to get information, still using circuit based communications.Whereas, in the current world of global communication, the trend began to shift from circuit or PSTN network to the model communication via IP or commonly known as VoIP (Voice over IP). Popular communication in this era is web based communication, where the web began to be applied to be a call center which are the communications using IP, so the function of the call center that serves only call can transform into the contact center which has free calls, chat and FAQ. To support digital voice data lines operating in the internet protocol requires a hardware called IP PBX which is the function of the IP PBX can be replaced by software called Asterisk. The result of this final project is a contact center system that has features call audio, audio-video calls, chat and Frequently Asked Questions te goals to allow users to get the information in various ways. Call features both audio and audio-video using the concept of VoIP with SIP protocol by utilizing Asterisk servers that have a PBX functions. From the results of tests performed, the results obtained from the audio call is one way delay by an average of 68 977 ms, with an average jitter 13 753 ms, and packet loss by an average of 4.8%. MOS values obtained from an audio call is for 4328 and for video calls is at 2:04.The quality of audio calls can be categorized into“Good” because it has a MOS value 4.328 of 5, while the quality of video calls can be said not too good because it has MOS value 2.04 of 5.KEYWORD: VoIP, Asterisk, Contact Center, SI

An intelligent IP-based call center with fault tolerance design.

Leung Cheung-chi.Thesis (M.Phil.)--Chinese University of Hong Kong, 2001.Includes bibliographical references (leaves 76-78).Abstracts in English and Chinese.Chapter 1 --- INTRODUCTION --- p.1Chapter 1.1 --- Background --- p.1Chapter 1.2 --- Objective --- p.2Chapter 1.3 --- Overview of the Thesis --- p.3Chapter 2 --- APPLICATION OF VOIP IN CALL CENTER --- p.6Chapter 2.1 --- An Intelligent IP-based Call Center Model --- p.6Chapter 2.1.1 --- Major Components --- p.7Chapter a) --- VoIP Gateways --- p.7Chapter b) --- Automatic Call Distributor (ACD) --- p.8Chapter c) --- Operators --- p.8Chapter d) --- Monitoring Tool --- p.9Chapter 2.1.2 --- Major Functions --- p.9Chapter 2.2 --- Experimental Study of an IP-to-IP Call Center - VoIP Application in Education --- p.10Chapter 2.2.1 --- Architecture --- p.11Chapter 2.2.2 --- Voice Connection Server --- p.12Chapter 2.2.3 --- Call Establishment --- p.14Chapter 2.2.4 --- A Preliminary Implementation --- p.14Chapter 3 --- THE ACD AND ITS SOFTWARE STRUCTURE --- p.17Chapter 3.1 --- Three-Layer Software Structure --- p.17Chapter 3.1.1 --- Network Infrastructure Layer --- p.18Chapter 3.1.2 --- Call Management Layer --- p.18Chapter 3.1.3 --- Application Layer --- p.19Chapter 3.1.4 --- Interoperation Between Layers --- p.19Chapter 3.2 --- Advantages of Adopting this Software Structure --- p.20Chapter 3.3 --- Functional Overview of the ACD --- p.21Chapter 3.3.1 --- Call Establishment --- p.21Chapter 3.3.2 --- Call Waiting --- p.23Chapter 3.3.3 --- Call Forwarding --- p.25Chapter 3.3.4 --- Routing Mechanism in the ACD --- p.26Chapter a) --- "Queues, Operator Groups and Operators" --- p.26Chapter b) --- Priority Based Call Routing --- p.28Chapter c) --- Routing of New Incoming Calls --- p.29Chapter d) --- Assigning Calls in Waiting Queues to Operators --- p.32Chapter 4 --- IMPLEMENTATION OF THE ACD --- p.34Chapter 4.1 --- Requirements in implementing the ACD --- p.34Chapter 4.1.1 --- Asynchronous Method Call --- p.34Chapter 4.1.2 --- Transaction Planning --- p.36Chapter 4.1.3 --- Failure Handling --- p.37Chapter 4.2 --- Available Technologies --- p.38Chapter 4.2.1 --- Enterprise JavaBean (EJB) --- p.38Chapter a) --- Entity Bean --- p.40Chapter b) --- Session Bean --- p.40Chapter c) --- Usage of Session Beans and Entity Beans --- p.41Chapter 4.2.2 --- COM+ --- p.42Chapter 4.2.3 --- EJB vs COM+ --- p.43Chapter 4.3 --- Implementation --- p.47Chapter 4.3.1 --- Mapping the EJB model to the Implementation of the ACD --- p.47Chapter 4.3.2 --- Design of Entity Beans --- p.49Chapter 4.3.3 --- Design of Session Beans --- p.51Chapter 4.3.4 --- Asynchronous Method Call --- p.53Chapter 4.3.5 --- Transaction Planning --- p.55Chapter 4.3.6 --- Failure Handling --- p.57Chapter a) --- Failure Handling for VoIP gateways --- p.58Chapter b) --- Failure Handling in the ACD --- p.60Chapter 5 --- AN EXPERIMENT --- p.64Chapter 5.1 --- Experiment on the Call Center Prototype --- p.64Chapter 5.1.1 --- Setup of the Experiment --- p.64Chapter 5.1.2 --- Experimental Results --- p.66Chapter a) --- Startup Time for Different Components --- p.66Chapter b) --- Possessing Time for Different Requests --- p.67Chapter 5.2 --- Observations --- p.69Chapter 5.2.1 --- Observations on Experimental Results --- p.69Chapter 5.2.2 --- Advantages and Disadvantages of Using EJB --- p.70Chapter 6 --- CONCLUSIONS --- p.72BIBLIOGRAPHY --- p.7

A comprehensive VoIP system with PSTN connectivity.

Yuen Ka-nang.Thesis (M.Phil.)--Chinese University of Hong Kong, 2001.Includes bibliographical references (leaves 133-135).Abstracts in English and Chinese.Abstract --- p.iAcknowledgement --- p.iiiChapter 1. --- INTRODUCTION --- p.1Chapter 1.1. --- Background --- p.1Chapter 1.2. --- Objectives --- p.1Chapter 1.3. --- Overview of Thesis --- p.2Chapter 2. --- NETWORK ASPECT OF THE VOIP TECHNOLOGY --- p.3Chapter 2.1. --- VoIP Overview --- p.3Chapter 2.2. --- Elements in VoIP --- p.3Chapter 2.2.1. --- Call Setup --- p.3Chapter 2.2.2. --- Media Capture/Playback --- p.4Chapter 2.2.3. --- Media Encoding/Decoding --- p.4Chapter 2.2.4. --- Media Transportation --- p.5Chapter 2.3. --- Performance Factors Affecting VoIP --- p.6Chapter 2.3.1. --- Network Bandwidth --- p.6Chapter 2.3.2. --- Latency --- p.6Chapter 2.3.3. --- Packet Loss --- p.7Chapter 2.3.4. --- Voice Quality --- p.7Chapter 2.3.5. --- Quality of Service (QoS) --- p.7Chapter 2.4. --- Different Requirements of Intranet VoIP and Internet VoIP --- p.8Chapter 2.4.1. --- Packet Loss/Delay/Jitter --- p.8Chapter 2.4.2. --- Interoperability --- p.9Chapter 2.4.3. --- Available Bandwidth --- p.9Chapter 2.4.4. --- Security Requirement --- p.10Chapter 2.5. --- Some Feasibility Investigations --- p.10Chapter 2.5.1. --- Bandwidth Calculation --- p.10Chapter 2.5.2. --- Simulation --- p.12Chapter 2.5.3. --- Conclusion --- p.17Chapter 2.5.4. --- Simulation Restrictions --- p.17Chapter 3. --- SOFTWARE ASPECT OF THE VOIP TECHNOLOGY --- p.19Chapter 3.1. --- VoIP Client in JMF --- p.19Chapter 3.1.1. --- Architecture --- p.20Chapter 3.1.2. --- Incoming Voice Stream Handling --- p.23Chapter 3.1.3. --- Outgoing Voice Stream Handling --- p.23Chapter 3.1.4. --- Relation between Incoming/Outgoing Voice Stream Handling --- p.23Chapter 3.1.5. --- Areas for Further Improvement --- p.25Chapter 3.2. --- Capture/Playback Enhanced VoIP Client --- p.26Chapter 3.2.1. --- Architecture --- p.27Chapter 3.2.2. --- Native Voice Playback Mechanism --- p.29Chapter 3.2.3. --- Native Voice Capturing Mechanism --- p.31Chapter 3.3. --- Win32 C++ VoIP Client --- p.31Chapter 3.3.1. --- Objectives --- p.32Chapter 3.3.2. --- Architecture --- p.33Chapter 3.3.3. --- Problems and Solutions in Implementation --- p.37Chapter 3.4. --- Win32 DirectSound C++ VoIP Client --- p.38Chapter 3.4.1. --- Architecture --- p.39Chapter 3.4.2. --- DirectSound Voice Playback Mechanism --- p.40Chapter 3.4.3. --- DirectSound Voice Capturing Mechanism --- p.44Chapter 3.5. --- Testing VoIP Clients --- p.45Chapter 3.5.1. --- Setup of Experiment --- p.45Chapter 3.5.2. --- Experiment Results --- p.47Chapter 3.5.3. --- Experiment Conclusion --- p.48Chapter 3.6. --- Real-time Voice Stream Mixing Server --- p.48Chapter 3.6.1. --- Structure Overview --- p.48Chapter 3.6.2. --- Experiment --- p.53Chapter 3.6.3. --- Conclusion --- p.54Chapter 4. --- EXPERIMENTAL STUDIES --- p.55Chapter 4.1. --- Pure IP-side VoIP-based Call Center ´ؤ VoIP in Education --- p.55Chapter 4.1.1. --- Architecture --- p.55Chapter 4.1.2. --- Client Structure --- p.56Chapter 4.1.3. --- Client Applet User Interface --- p.58Chapter 4.1.4. --- Observations --- p.63Chapter 4.2. --- A Simple PBX Experiment --- p.63Chapter 4.2.1. --- Structural Overview --- p.63Chapter 4.2.2. --- PSTN Gateway Server Program --- p.64Chapter 4.2.3. --- Problems and Solutions in Implementation --- p.66Chapter 4.2.4. --- Experiment 1 --- p.66Chapter 4.2.5. --- Experiment 2 --- p.68Chapter 5. --- A COMPREHENSIVE VOIP PROJECT 一 GRADUATE SECOND PHONE (GSP) --- p.72Chapter 5.1. --- Overview --- p.72Chapter 5.1.1. --- Background --- p.72Chapter 5.1.2. --- Architecture --- p.76Chapter 5.1.3. --- Technologies Used --- p.78Chapter 5.1.4. --- Major Functions --- p.80Chapter 5.2. --- Client --- p.84Chapter 5.2.1. --- Structure Overview --- p.85Chapter 5.2.2. --- Connection Procedure --- p.89Chapter 5.2.3. --- User Interface --- p.91Chapter 5.2.4. --- Observations --- p.92Chapter 5.3. --- Gateway --- p.94Chapter 5.3.1. --- Structure Overview --- p.94Chapter 5.3.2. --- Connection Procedure --- p.97Chapter 5.3.3. --- Caller ID Simulator --- p.97Chapter 5.3.4. --- Observations --- p.98Chapter 5.4. --- Server --- p.101Chapter 5.4.1. --- Structure Overview --- p.101Chapter 5.5. --- Details of Major Functions --- p.103Chapter 5.5.1. --- Secure Local Voice Message Box --- p.104Chapter 5.5.2. --- Call Distribution --- p.106Chapter 5.5.3. --- Call Forward --- p.112Chapter 5.5.4. --- Call Transfer --- p.115Chapter 5.6. --- Experiments --- p.116Chapter 5.6.1. --- Secure Local Voice Message Box --- p.117Chapter 5.6.2. --- Call Distribution --- p.118Chapter 5.6.3. --- Call Forward --- p.121Chapter 5.6.4. --- Call Transfer --- p.122Chapter 5.6.5. --- Dial Out --- p.124Chapter 5.7. --- Observations --- p.125Chapter 5.8. --- Outlook --- p.126Chapter 5.9. --- Alternatives --- p.127Chapter 5.9.1. --- Netmeeting --- p.127Chapter 5.9.2. --- OpenH323 --- p.128Chapter 6. --- CONCLUSIONS --- p.129Bibliography --- p.13

What’s the Hang Up? The Future of VoIP Regulation and Taxation in New Hampshire

Alice in Austria wishes to call her friend Bob in Boston, using a Boston area code to avoid charges for an international call. Using VoIP, Alice may initiate her call from any location in Austria where she may find Internet access. Once Alice connects to the Internet, she can transmit her call with the aid of a VoIP service provider, such as Skype. In order to hear and communicate with Bob, Alice can rely on a microphone and a headset that she can plug into her computer. Through VoIP, not only may Alice carry on a telephone conversation, but most service providers also allow her to record conversations and manage other information, such as voice mail. The rise of Voice over Internet Protocol (“VoIP”) services “means nothing less than the death of the traditional telephone business,” as the ability to make free calls over a high-speed Internet connection in the future “undermines the existing pricing model for telephony.” This disruptive, convergent technology is blurring the boundary between Internet services and telephone services because VoIP functions like the traditional telephone system, but travels as ones and zeros through a broadband Internet connection. As a result, the Federal Communications Commission (“FCC”) has questioned whether to classify VoIP as an information service, generally free from FCC regulation under the Telecommunications Act of 1996, or as a telecommunication service, subject to a comprehensive regulatory regime and common carrier obligations. This note discusses why most VoIP services, with the exception of phone-to-phone Internet Protocol (“IP”) telephony, should be classified as information services and, as such, should remain free from state taxation – focusing specifically on the taxation in New Hampshire. Part II focuses on the technology of VoIP and how it differs from traditional telephony. Part III discusses the distinction between information and telecommunication services in the Telecommunications Act of 1996, whether VoIP may qualify as Internet access in light of the Internet Tax Freedom Act (“ITFA”) of 1998, and the federal regulation of VoIP. Finally, Part IV addresses the debate over taxation of VoIP in New Hampshire and discusses why VoIP services should not yet be taxed by the New Hampshire Department of Revenue Administration in light of federal law and the best interests of local businesses and consumers

Why the Government Should Not Regulate Internet Telephony?

The Federal Communications Commission has requested comments on the regulation of voice telephone services delivered over the Internet, dubbed "VoIP" or Voice over Internet Protocol. This paper examines whether there is a need to regulate VoIP. We conclude that there is no economic rationale for regulating VoIP and that consumers will likely be worse off if VoIP is regulated. Furthermore, the emergence of new technologies, such as VoIP, is rapidly eroding the rationale for continuing to regulate local telephone services.

Detecting and Mitigating Denial-of-Service Attacks on Voice over IP Networks

Voice over IP (VoIP) is more susceptible to Denial of Service attacks than traditional data traffic, due to the former's low tolerance to delay and jitter. We describe the design of our VoIP Vulnerability Assessment Tool (VVAT) with which we demonstrate vulnerabilities to DoS attacks inherent in many of the popular VoIP applications available today. In our threat model we assume an adversary who is not a network administrator, nor has direct control of the channel and key VoIP elements. His aim is to degrade his victim's QoS without giving away his presence by making his attack look like a normal network degradation. Even black-boxed, applications like Skype that use proprietary protocols show poor performance under specially crafted DoS attacks to its media stream. Finally we show how securing Skype relays not only preserves many of its useful features such as seamless traversal of firewalls but also protects its users from DoS attacks such as recording of conversations and disruption of voice quality. We also present our experiences using virtualization to protect VoIP applications from 'insider attacks'. Our contribution is two fold we: 1) Outline a threat model for VoIP, incorporating our attack models in an open-source network simulator/emulator allowing VoIP vendors to check their software for vulnerabilities in a controlled environment before releasing it. 2) We present two promising approaches for protecting the confidentiality, availability and authentication of VoIP Services

Denial of Service in Voice Over IP Networks

In this paper we investigate denial of service (DoS) vulnerabilities in Voice over IP (VoIP) systems, focusing on the ITU-T H.323 family of protocols. We provide a simple characterisation of DoS attacks that allows us to readily identify DoS issues in H.323 protocols. We also discuss network layer DoS vulnerabilities that affect VoIP systems. A number of improvements and further research directions are proposed