Msc Thesis Thesis Title: Designing and Optimization of VOIP PBX Infrastructure by Naveed Younas Rana Student ID

Total Page:16

File Type:pdf, Size:1020Kb

Msc Thesis Thesis Title: Designing and Optimization of VOIP PBX Infrastructure by Naveed Younas Rana Student ID MSc Thesis Thesis Title: Designing and optimization of VOIP PBX infrastructure By Naveed Younas Rana Student ID: 1133670 Department of computer science and technology University of Bedfordshire Supervisor: Dr. Ali Mansour 1 Abstract In the recent decade, communication has stirred from the old wired medium such as public switched telephone network (PSTN) to the Internet. Present, Voice over Internet Protocol (VoIP) Technology used for communication on internet by means of packet switching technique. Several years ago, an internet protocol (IP) based organism was launched, which is known as Private Branch Exchange "PBX", as a substitute of common PSTN systems. For free communication, probably you must have to be pleased with starting of domestic calls. Although, fairly in few cases, VoIP services can considerably condense our periodical phone bills. For instance, if someone makes frequent global phone calls, VoIP talk service is the actual savings treat which cannot achieve by using regular switched phone. VoIP talk services strength help to trim down your phone bills if you deal with a lot of long-distance (international) and as well as domestic phone calls. However, with the VoIP success, threats and challenges also stay behind. In this dissertation, by penetration testing one will know that how to find network vulnerabilities how to attack them to exploit the network for unhealthy activities and also will know about some security techniques to secure a network. And the results will be achieved by penetration testing will indicate of proven of artefact and would be helpful to enhance the level of network security to build a more secure network in future. 2 Author consent form Author’s name Naveed Younas Rana Title of Thesis Designing and optimization of VOIP PBX infrastructure Degree MSc Computer Networking I have read and fully understand the rules and regulations of the University of the Bedfordshire concerning final thesis submission. I AM AS FOLLOW: I am the Author of the following thesis. Work demonstration is originally designed and optimised I do accept University of Bedfordshire Policy concerning final thesis submission. AUTHOR’S SIGNATURE: Naveed Younas Rana AUTHOR STUDENT ID: date 1133670 23-05-2013 3 Acknowledgement I would like to be appreciative for help and support of the very kind people around me, especially I would like to thankful to my supervisor Dr. Ali Mansour for his greater guidance and for technical support as well in this project. Thanks for him to make it possible for successful completion of this project. He has offered me explicit support throughout. Report writing and network designing were a difficult challenges but I have done my job with his help successfully. His given feedback made me able to review my progress in writing dissertation. His support has been invaluable for me on academic level and I am extremely thankful for this. Secondly, I would like to be thankful to Faisal Fayyaz Qurashi and be appreciative for his support and help in designing the network and overall throughout this project. 4 Abbreviations Following are the important abbreviations: VoIP Voice over Internet Protocol PBX Private Branch Exchange QoS Quality of Service PSTN Public Switched Telephone Network SIP Session Initiation Protocol TCP Transmission Control Protocol IPSec Internet Protocol Security FTP File Transport Protocol NAT Network Address Translation RTPR Real time Transport Protocol ISP Internet Service Provider MPLS Multiprotocol Label Switching TCP Transmission Control Protocol UDP User Datagram Protocol HTTP Hypertext Transfer Protocol 5 Table of Contents Abstract Author concern form Acknowledgement Abbreviations 1 Introduction .................................................................................................................................... 9 1.1 Aim and objectives .................................................................................................................. 9 1.2 Problem definitions ................................................................................................................. 9 1.3 Methodology ......................................................................................................................... 10 1.3.1 Project requirements .................................................................................................... 10 1.3.2 Scenario ......................................................................................................................... 10 1.3.3 Expected Results ........................................................................................................... 10 1.4 Organizing documentation ................................................................................................... 11 1.5 Summary ............................................................................................................................... 11 2 Literature Review .......................................................................................................................... 12 2.1 VoIP ....................................................................................................................................... 12 2.1.1 History ........................................................................................................................... 13 2.1.2 Features ........................................................................................................................ 13 2.1.2.1 Infrastructure ............................................................................................................ 13 2.1.2.2 Free calling ................................................................................................................ 14 2.1.2.3 Quality ....................................................................................................................... 14 2.1.2.4 Flexibility ................................................................................................................... 14 2.1.3 VoIP standards .............................................................................................................. 14 2.1.4 VoIP requirements ........................................................................................................ 15 2.2 IP PBX .................................................................................................................................... 15 2.2.1 Background ................................................................................................................... 16 2.2.2 Architecture .................................................................................................................. 17 2.2.2.1 IP ............................................................................................................................... 17 2.2.2.2 Hybrid IP .................................................................................................................... 17 2.2.2.3 Linux .......................................................................................................................... 18 2.2.2.4 Windows ................................................................................................................... 18 2.2.2.5 Real-time ................................................................................................................... 18 6 2.2.3 Functionalities ............................................................................................................... 18 2.3 QoS ........................................................................................................................................ 19 2.3.1 Parameters .................................................................................................................... 19 2.3.1.1 Packet loss ................................................................................................................. 19 2.3.1.2 Jitter .......................................................................................................................... 19 2.3.1.3 Latency ...................................................................................................................... 19 2.3.2 Importance of QoS ........................................................................................................ 19 2.3.3 Integrated services ........................................................................................................ 20 2.3.4 MPLS traffic engineering ............................................................................................... 20 2.4 Security ................................................................................................................................. 21 2.4.1 Challenges and vulnerabilities ...................................................................................... 21 2.4.1.1 Denial of service attacks (DoS): ................................................................................. 22 2.4.1.2 Access attack: ............................................................................................................ 22 2.4.1.3 Modification attack: .................................................................................................. 22 2.4.1.4 Vishing: ...................................................................................................................... 22 2.4.1.5 Toll fraud: .................................................................................................................
Recommended publications
  • On IP Networking Over Tactical Links
    On IP Networking over Tactical Links Claude Bilodeau The work described in this document was sponsored by the Department of National Defence under Work Unit 5co. Defence R&D Canada √ Ottawa TECHNICAL REPORT DRDC Ottawa TR 2003-099 Communications Research Centre CRC-RP-2003-008 August 2003 On IP networking over tactical links Claude Bilodeau Communications Research Centre The work described in this document was sponsored by the Department of National Defence under Work Unit 5co. Defence R&D Canada - Ottawa Technical Report DRDC Ottawa TR 2003-099 Communications Research Centre CRC RP-2003-008 August 2003 © Her Majesty the Queen as represented by the Minister of National Defence, 2003 © Sa majesté la reine, représentée par le ministre de la Défense nationale, 2003 Abstract This report presents a cross section or potpourri of the numerous issues that surround the tech- nical development of military IP networking over disadvantaged network links. In the first sec- tion, multi-media services are discussed with regard to three aspects: applications, operational characteristics and service models. The second section focuses on subnetworks and bearers; mainly impairments caused by characteristics of the wireless environment. An overview of the Iris tactical bearers is provided as an example of a tactical IP environment. The last section looks at how IP can integrate these two elements i.e. multi-media services and impaired sub- network links. These three sections are unified by a common theme, quality of service, which runs in the background of the discussions. Résumé Ce rapport présente une coupe transversale ou pot-pourri de questions reliées au développe- ment technique des réseaux militaires IP pour des liaisons défavorisées.
    [Show full text]
  • TR 102 633 V1.1.1 (2008-10) Technical Report
    ETSI TR 102 633 V1.1.1 (2008-10) Technical Report Corporate Networks (NGCN); Next Generation Corporate Networks (NGCN) - General 2 ETSI TR 102 633 V1.1.1 (2008-10) Reference DTR/ECMA-00352 Keywords IP, SIP ETSI 650 Route des Lucioles F-06921 Sophia Antipolis Cedex - FRANCE Tel.: +33 4 92 94 42 00 Fax: +33 4 93 65 47 16 Siret N° 348 623 562 00017 - NAF 742 C Association à but non lucratif enregistrée à la Sous-Préfecture de Grasse (06) N° 7803/88 Important notice Individual copies of the present document can be downloaded from: http://www.etsi.org The present document may be made available in more than one electronic version or in print. In any case of existing or perceived difference in contents between such versions, the reference version is the Portable Document Format (PDF). In case of dispute, the reference shall be the printing on ETSI printers of the PDF version kept on a specific network drive within ETSI Secretariat. Users of the present document should be aware that the document may be subject to revision or change of status. Information on the current status of this and other ETSI documents is available at http://portal.etsi.org/tb/status/status.asp If you find errors in the present document, please send your comment to one of the following services: http://portal.etsi.org/chaircor/ETSI_support.asp Copyright Notification No part may be reproduced except as authorized by written permission. The copyright and the foregoing restriction extend to reproduction in all media. © European Telecommunications Standards Institute 2008.
    [Show full text]
  • The People Who Invented the Internet Source: Wikipedia's History of the Internet
    The People Who Invented the Internet Source: Wikipedia's History of the Internet PDF generated using the open source mwlib toolkit. See http://code.pediapress.com/ for more information. PDF generated at: Sat, 22 Sep 2012 02:49:54 UTC Contents Articles History of the Internet 1 Barry Appelman 26 Paul Baran 28 Vint Cerf 33 Danny Cohen (engineer) 41 David D. Clark 44 Steve Crocker 45 Donald Davies 47 Douglas Engelbart 49 Charles M. Herzfeld 56 Internet Engineering Task Force 58 Bob Kahn 61 Peter T. Kirstein 65 Leonard Kleinrock 66 John Klensin 70 J. C. R. Licklider 71 Jon Postel 77 Louis Pouzin 80 Lawrence Roberts (scientist) 81 John Romkey 84 Ivan Sutherland 85 Robert Taylor (computer scientist) 89 Ray Tomlinson 92 Oleg Vishnepolsky 94 Phil Zimmermann 96 References Article Sources and Contributors 99 Image Sources, Licenses and Contributors 102 Article Licenses License 103 History of the Internet 1 History of the Internet The history of the Internet began with the development of electronic computers in the 1950s. This began with point-to-point communication between mainframe computers and terminals, expanded to point-to-point connections between computers and then early research into packet switching. Packet switched networks such as ARPANET, Mark I at NPL in the UK, CYCLADES, Merit Network, Tymnet, and Telenet, were developed in the late 1960s and early 1970s using a variety of protocols. The ARPANET in particular led to the development of protocols for internetworking, where multiple separate networks could be joined together into a network of networks. In 1982 the Internet Protocol Suite (TCP/IP) was standardized and the concept of a world-wide network of fully interconnected TCP/IP networks called the Internet was introduced.
    [Show full text]
  • Ipnext 700™ NGN IP-PBX High-Performance Next Generation IP-PBX Solution
    IPNext 700™ NGN IP-PBX High-performance Next Generation IP-PBX Solution AddPac Technology 2006, Sales and Marketing www.addpac.com V2oIP Internetworking Solution AddPac VoIP Media Gateway SIP Proxy Server (Multiple E1/T1) WAN, Internet RAS AddPac PSTN VPMS (ATM, FR, IP Network) Note Book Phone AddPac LAN AddPac Embedded VoIP Dial-up Gateway GateKeeper AddPac ATM Router Phone FAX AddPac AddPac FAX FAX VoIP Gateway AddPac AddPac VoIP Router (1~2 Ports) VPN Gateway Metro Ethernet AddPac Phone AddPac Switch ATM Metro Phone Ethernet Switch AddPac WAN Router FAX FAX Multi-service FAX Router LAN Phone AddPac FAX Phone Phone VoIP Gateway LAN PBX (4~8 Ports) AddPac Phone FoIP Broadcasting AddPac VPN Gateway System AddPac AddPac LAN Phone Phone VBMS VPN Gateway AddPac AddPac Speaker VoIP Gateway Phone Note Book AddPac VoIP Gateway (4~8 Ports) (Digital E1/T1) Phone VoIP Gateway (1~2 Ports) Amp. Phone Phone AddPac AddPac AddPac AddPac FAX Voice Broadcasting AddPac Voice Broadcasting L2 Ethernet Switch L2 Ethernet Switch System L2 Ethernet Switch FAX Terminal FAX AddPac AddPac Phone IP PBX IP Phone PC PC AddPac Phone VoIP Gateway AddPac (4 Ports) LAN LAN Call Manager FAX Phone AddPac AddPac FAX Voice Broadcasting AddPac IP Phone FAX Speaker Terminal AddPac IP Video Codec NGN VoIP Gateway AddPac AddPac (8~16 Ports) Video Gateway Amp. NGN VoIP Gateway Phone (16~64 Ports) Speaker PC Speaker FAX Phone AddPac Amp. IP Phone Amp. Video AddPac Video Speaker Phone Input Video AddPac IP Phone Phone Display Video Phone Phone Input IP Phone Phone Display Amp.
    [Show full text]
  • Positron Will Empower Your Business with Its Feature Rich, Easy to Use IP PBX Phone System
    The IP PBX System for Today & Tomorrow Positron will empower your business with its feature rich, easy to use IP PBX phone system Affordable, Compatible, Powerful, Reliable POSITRON Telecommunication Systems The product for the future of communications... available today! The G-Series IP PBX Positron is Committed to Making Your Business More Powerful Business Phone Systems with its Unied IP PBX Phone System Save time and money with a system that’s easy to setup and deploy Positron Telecom’s G-Series full function IP PBX business phone systems address the demands of fully Globalize your network integrated unified communication solutions. Supporting standards based open protocols allow enterprises to easily move away from legacy, proprietary systems and move to modern VoIP based so that you can be communications that provide long term benefits. reached wherever Bridge the gap between your legacy system and VoIP. The following tools were developed to you are save time and effort during the initial installation and ongoing support of the product: · Phones can be automatically configured · User data is imported with one click · Secure remote access capability · Intuitive Web configuration screens in multiple languages Get voice, data and video in one unique solution, with unsurpassed voice quality and robust echo cancellation, eliminating echo, noise and distortion. Positron’s IP PBX phone system grows with your business, without worries of outgrowing it. The G-Series product family caters to your company’s needs, and determines the right product based on the size of your business. The IP PBX phone system comes fully enabled with no setup fees and integrates traditional telephony networks and VoIP in a single, incredibly easy-to-use box.
    [Show full text]
  • SIP Phones Explained by Gary Audin March 12, 2014
    White Paper SIP Phones Explained By Gary Audin March 12, 2014 What is a SIP Phone? Manufacturers, vendors and service providers describe the Session Initiation Protocol (SIP) as if it were the foremost technical solution for Voice over IP (VoIP). It does have numerous benefits; SIP brings enhanced VoIP and powers other technologies such as SIP phones and SIP trunks. Understanding SIP and the uses for business allows users to make the greatest gains. First Came the IP Phone With the introduction of the proprietary IP PBX came the proprietary IP phone. An IP phone is designed to communicate over an Ethernet LAN network connection. It is fully interoperable on private networks as well as the Internet. The limitation is that IP PBX specific IP phones use their own unique proprietary signaling protocols. Therefore, users must buy the IP phones from the IP PBX vendor. Almost no other IP phones would work on the proprietary IP PBX. The creation of the SIP standard helped change the IP phone landscape. SIP is a standard signaling protocol, a commonly implemented standard for VoIP. It is also applied to a wide range of devices beyond VoIP including video, instant messaging (IM), and other forms of media. The development of SIP also brought a level of standardization for IP phones. The SIP standard opened up the IP phone market to a wide range of phones at competitive prices. Eventually IP PBX vendors had to support SIP phones to meet the demand. The SIP phone has become the dominant choice for hosted and cloud-based services for customers who choose not to own an IP PBX.
    [Show full text]
  • Linux and Asterisk Dotelephony U
    LINUX JOURNAL ASTERISK | TRIXBOX | ADHEARSION | VOIP | SELINUX | SIP | NAT VOIP An MPD-Based Audio Appliance ™ Asterisk Since 1994: The Original Magazine of the Linux Community | Trixbox | Adhe Linux and Asterisk a rsion | DoTe lephony VoIP | >> ASTERISK FOR SELinux CONVENIENT PHONE CALLS | SIP >> WIRESHARK SNIFFS VOIP PROBLEMS | NAT | >> VOIP THROUGH NAT Digit a l Music >> EMBEDDED ASTERISK | >> ADHEARSION Wiresh WITH ASTERISK a rk >> COMMUNIGATE PRO AND VOIP MARCH 2007 | ISSUE 155 M www.linuxjournal.com A R C USA $5.00 H CAN $6.50 2007 I S S U E 155 U|xaHBEIGy03102ozXv+:. PLUS: The D Language Today, Dan configured a switch in London, rebooted servers in Sydney, and watched his team score the winning goal in St. Louis. With Avocent data center solutions, the world can finally revolve around you. Avocent puts secure access and control right at your finger tips – from multi-platform servers to network routers, your local data center to branch offices, across the hall or around the globe. Let others roll crash carts to troubleshoot – with Avocent, trouble is on ice. To learn more, visit us at www.avocent.com/ice to download Data Center Control: Guidelines to Achieve Centralized Management whitepaper or call 866.277.1924 for a demo today. Avocent, the Avocent logo and The Power of Being There are registered trademarks of Avocent Corporation. All other trademarks or company names are trademarks or registered trademarks of their respective companies. Copyright © 2006 Avocent Corporation. MARCH 2007 CONTENTS Issue 155 ILLUSTRATION ©ISTOCKPHOTO.COM/STEFAN WEHRMANN ©ISTOCKPHOTO.COM/STEFAN ILLUSTRATION FEATURES 50 Time-Zone Processing with Asterisk, Part I 70 Expose VoIP Problems with Wireshark Hello, this is your unwanted wake-up call.
    [Show full text]
  • The Transition to an All-IP Network: a Primer on the Architectural Components of IP Interconnection
    nrri The Transition to an All-IP Network: A Primer on the Architectural Components of IP Interconnection Joseph Gillan and David Malfara May 2012 NRRI 12–05 © 2012 National Regulatory Research Institute About the Authors Joseph Gillan is a consulting economist who primarily advises non-incumbent competitors in the telecommunications industry. David Malfara is president of the ETC Group, an engineering and business consulting firm. Disclaimer This paper solely represents the opinions of Mr. Gillan and Mr. Malfara and not the opinions of NRRI. Online Access This paper can be found online at the following URL: http://communities.nrri.org/documents/317330/7821a20b-b136-44ee-bee0-8cd5331c7c0b ii Executive Summary There is no question that the “public switched telephone network” (PSTN) is transitioning to a network architecture based on packet technology and the use of the Internet Protocol (IP) suite of protocols. A critical step in this transition is the establishment of IP-to-IP interconnection arrangements permitting the exchange of voice- over-IP (VoIP) traffic in a manner that preserves the quality of service that consumers and businesses expect. This paper provides a basic primer on how voice service is provided by IP-networks and discusses the central elements of IP-to-IP interconnection. The ascendancy of IP networks is the architectural response to the growth of data communications, in particular the Internet. Although some voice services rely on the public Internet for transmission (a method commonly known as “over-the-top” VoIP), the Internet does not prioritize one packet over another; as a result, quality cannot be guaranteed for these services.
    [Show full text]
  • Past, Present and Future of IP Telephony
    International Conference on Communication Theory, Reliability, and Quality of Service Past, Present and Future of IP Telephony Vasco N. G. J. Soares1,3, Paulo A. C. S. Neves1,3, and Joel J. P. C. Rodrigues2,3 1 Superior School of Technology, Polytechnic Institute of Castelo Branco, C. Branco, Portugal 2Department of Informatics, University of Beira Interior, Covilhã, Portugal 3Institute of Telecommunications, Networks and Multimedia Group, Portugal [email protected]; [email protected]; [email protected] Abstract instance, to seamless use of both voice and data messaging. Since the late 90's IP telephony, commonly referred VoIP is basically the means to grab audio and video to as Voice over IP (VoIP), has been presented as a in digital form, divide it in small chunks that can be revolution on communications enabling the possibility transferred through the network as packets. After, it to converge historically separated voice and data reassembles the chunks on the other side in a networks, reducing costs, and integrating voice, data convenient way so that people having a phone and video on applications. This paper presents a study conversation have the idea of a circuit switching over the standard VoIP protocols H.323, Session ordinary call, as shown in Figure 1. Initiation Protocol (SIP), Media Gateway Control VoIP presents a shift on corporate voice Protocol (MGCP), and H.248/Megaco. Given the fact communications where the traditional Private Branch that H.323 and SIP are more widespread than the eXchange (PBX) based systems were used to provide others, we focus our study on them. For each of these internal cost-free communications and sharing of protocols we describe and discuss its main external telephone lines.
    [Show full text]
  • Network Working Group J. Postel Request for Comments: 790 ISI September 1981 Obsoletes Rfcs
    Network Working Group J. Postel Request for Comments: 790 ISI September 1981 Obsoletes RFCs: 776, 770, 762, 758, 755, 750, 739, 604, 503, 433, 349 Obsoletes IENs: 127, 117, 93 ASSIGNED NUMBERS This Network Working Group Request for Comments documents the currently assigned values from several series of numbers used in network protocol implementations. This RFC will be updated periodically, and in any case current information can be obtained from Jon Postel. The assignment of numbers is also handled by Jon. If you are developing a protocol or application that will require the use of a link, socket, port, protocol, or network number please contact Jon to receive a number assignment. Jon Postel USC - Information Sciences Institute 4676 Admiralty Way Marina del Rey, California 90291 phone: (213) 822-1511 ARPANET mail: POSTEL@ISIF Most of the protocols mentioned here are documented in the RFC series of notes. The more prominent and more generally used are documented in the Protocol Handbook [17] prepared by the Network Information Center (NIC). Some of the items listed are undocumented. In all cases the name and mailbox of the responsible individual is indicated. In the lists that follow, a bracketed entry, e.g., [17,iii], at the right hand margin of the page indicates a reference for the listed protocol, where the number cites the document and the "iii" cites the person. Postel [Page 1] RFC 790 September 1981 Assigned Numbers Network Numbers ASSIGNED NETWORK NUMBERS This list of network numbers is used in the internet address [33]. The Internet Protocol (IP) uses a 32 bit address and divides that address into a network part and a "rest" or local address part.
    [Show full text]
  • IP Telephony Applicability in Cloud Computing Aplicabilidad De Telefon´Ia IP En La Computacion´ En La Nube
    JOURNAL OF SCIENCE AND RESEARCH: REVISTA CIENCIA E INVESTIGACION,´ E-ISSN: 2528-8083, VOL. 3, CITT2017, PP. 128-133 128 IP Telephony Applicability in Cloud Computing Aplicabilidad de telefon´ıa IP en la computacion´ en la nube Francisco Palacios1,*, Mitchell Vasquez´ Bermudez´ 1,2,y, Fausto Orozco1,z, and Diana Espinoza Villon´ 1,⊗ 1Universidad de Guayaquil, Ecuador 2Universidad Agraria del Ecuador ffrancisco.palacioso;mitchell.vasquezb;fausto.orozcol;[email protected] [email protected] Received: August 15, 2017 — Accepted: September 15, 2017 How to cite: Palacios, F., Vasquez´ Bermudez,´ M., Orozco, F., & Espinoza Villon,´ D. (2018). Aplicabilidad de telefon´ıa IP en la computacion´ en la nube. Journal of Science and Research: Revista Ciencia e Investigacion,´ 3(CITT2017), 128-133. https://doi.org/10.26910/ issn.2528-8083vol3issCITT2017.2018pp128-133 Abstract—This paper carries out a research related to the applicability of VoIP over Cloud Computing to guarantee service stability and elasticity of the organizations. In this paper, Elastix is used as an open source software that allows the management and control of a Private Branch Exchange (PBX); and for developing, it is used the services given Amazon Web Services due to their leadership and experience in cloud computing providing security, scalability, backup service and feasibility for the users. Keywords—VoIP, Cloud Computing, Elastix, Amazon Web Service. Resume—Este trabajo lleva a cabo una investigacion´ relacionada con la aplicabilidad de VoIP sobre Cloud Computing para garantizar la estabilidad del servicio y la elasticidad de las organizaciones. En este documento, Elastix se utiliza como un software de codigo´ abierto que permite gestion´ y control de una central telefonica´ privada (PBX); y para el desarrollo, se utilizan los servicios prestados a Amazon Web Services debido a su liderazgo y experiencia en computacion´ en la nube que brinda seguridad, escalabilidad y servicio de respaldo y viabilidad para los usuarios.
    [Show full text]
  • Distributed IP-PBX
    Distributed IP-PBX Tele-convergence of IP-PBX / PSTN / FAX / legacy PABX And Distributed network approach with area isolation AL FARUQ IBNA NAZIM Deputy Manager – Corporate Solution Link3 Technologies Ltd. www.link3.net Contributor Acknowledgement Ahmed Sobhan – Link3 Technologies Ltd. Adnan Howlader AGENDA • Background • Application and Appliances • Architecture & Benefits • Case Study • Key Findings BACKGROUND BACKGROUND : The Beginning An organization with a headquarter located centrally which had 15+ zonal areas and had more than 150+ branches located remotely with zones. What do they have and practice: . They have a central internet connectivity. They had zones connected with E1. They had a central IP-PBX soft switch. They had application systems, mailing etc. running centrally. Had PABX for inter-telecommunication and PSTN dropped to call out and have calls in for all places. BACKGROUND : Realization . Having communication through Datacom. Having independent system for zones to be operational even if data link unavailable . Having PSTN to be trunked to remote locations. Having PABX lines to be trunked to few locations. So they asked for 1. MUX for all locations. 2. Routers to integrate with data & MUX. 3. PABX unit for independent dialer. 4. Microwave setups for connectivity. BACKGROUND : Realization Complexity !!! Expensive !!! Mess !!! Burden !!! ……. BACKGROUND : Realization • Packet switching. • Layer-3 network for connectivity nodes. • Single box solution. • Integrated communication equipment. • Complexity minimization. BACKGROUND : Telecommunication • A huge mix of evolved technologies. • Divided in circuit and packet switched network. Visualization is from the Opte Project of the various routes through a portion of the Internet. BACKGROUND : Circuit Vs Packet switching BACKGROUND : PABX, PSTN & IP-PBX • PBX: Switchboard operator managed system using cord circuits.
    [Show full text]