DOCSLIB.ORG

Voice Over Internet Protocol (VOIP): Overview, Direction and Challenges 1 U

Total Page：16

File Type：pdf, Size：1020Kb

Download full-text PDF Read full-text

Load more

View metadata, citation and similar papers at core.ac.uk brought to you by CORE provided by International Institute for Science, Technology and Education (IISTE): E-Journals Journal of Information Engineering and Applications www.iiste.org ISSN 2224-5782 (print) ISSN 2225-0506 (online) Vol.3, No.4, 2013 Voice over Internet Protocol (VOIP): Overview, Direction And Challenges 1 U. R. ALO and 2 NWEKE HENRY FIRDAY Department of Computer Science Ebonyi State University Abakaliki, Nigeria 1Email:- [email protected] 2Email: [email protected] ABSTRACT Voice will remain a fundamental communication media that cuts across people of all walks of life. It is therefore important to make it cheap and affordable. To be reliable and affordable over the common Public Switched Telephone Network, change is therefore inevitable to keep abreast with the global technological change. It is on this basis that this paper tends to critically review this new technology VoIP, x-raying the different types. It further more discusses in detail the VoIP system, VoIP protocols, and a comparison of different VoIP protocols. The compression algorithm used to save network bandwidth in VoIP, advantages of VoIP and problems associated with VoIP implementation were also critically examined. It equally discussed the trend in VoIP security and Quality of Service challenges. It concludes by reiterating the need for a cheap, reliable and affordable means of communication that would not only maximize cost but keep abreast with the global technological change. Keywords: Voice over Internet Protocol (VoIP), Public Switched Telephone Network (PSTN), Session Initiation Protocol (SIP), multipoint control unit 1. Introduction Voice over Internet Protocol (VoIP) is a technology that makes it possible for users to make telephone calls over the internet or intranet networks. The technology does not use the traditional Public Switched Telephone Network (PSTN); instead calls are made over an internet protocol data network. VoIP has great benefits of increased saving, high quality voice and video streaming and several other added value services. Examples of VoIP software are: Skype, Google talks and windows live messenger (Di Wu, 2002). 2. Overview of VoIP VoIP stand for Voice over Internet Protocol. VoIP enables us to compress and convert voice signal to digital signal and transmit it through Internet Protocol (IP)-enabled network like Internet, Ethernet and Wireless LAN. VoIP uses Internet protocol to manage voice packet over Internet protocol (IP) network. Types of VoIP Service There are different types of VoIP based on the infrastructures employed by the owner of the network. Listed are some popular services used in VoIP. Computer to Computer Computer to Computer service provides Internet telephony free using the same softphone software such as Skype, Instant Messaging, AOL etc. It is a software-based VoIP service and both the caller and the receiver must be using their computer in order to place calls. The following requirement must be met to use computer to computer VoIP service: softphone software, a sound card and good Internet service. With computer to computer VoIP service, the user may not be able to call either landline or mobile phones, and also the recipient must be online in order to call him/her. Computer to phone and vice versa This is a software-based and hardware-based service. Softphone software is used to route the call to an Internet protocol and hands off to a conventional telephone network . To use the service, one needs to subscribe and be charged at a low rate. Examples include Skype, MSN and Google Talk that provides the service to enable their customers to call landline from their computer. Computer to phone requirements are Internet-enabled phone and computer, VoIP service subscription, modem and Analog Terminal Adapter to convert the call signal to digital signal and also back to analog signal. Computer to phone does not allow emergency call users and needs to have a computer connected to the internet. Phone to Phone 18 Journal of Information Engineering and Applications www.iiste.org ISSN 2224-5782 (print) ISSN 2225-0506 (online) Vol.3, No.4, 2013 This is a hardware-based service that allows the caller and receiver to call each other using the Internet. Many telephone companies use this to handle long distance calls. VoIP convert the audio sound into data packets and transfer these packets over the Internet. It allows emergence calls and does not need PSTN for initiation and termination of calls. 2.1 VoIP system Voice over Internet Protocol (VoIP) system is made up of End User Equipments, Network component, call processor, Gateway and Server. The End User Equipment is the part mainly used to access the VoIP system and communicates with other End point (Greg Tucker , 2004). End User equipments initiate and maintain the signals needed to establish calls over the IP network. It also helps to convert voice to data packet, voice and video communication, monitoring and network surveillance capability. They are connected to the network with cable and maybe wireless devices. Examples of end user equipment are IP-enabled phones, multimedia computers and VoIP enabled workstations installed with softphone . The Network Component includes routers, switches, cables, firewalls and wireless access point. There are many existing institution’s Internet Protocol networks that will enable the installation of our VoIP system. The network must be strong enough to support the VoIP system because the impact of IP network is higher than the data network. Gateway is an optional component of VoIP network that helps to route calls, handle call connection, and manage terminal and Multipoint control unit (MCU) (Mona & Nirmala, 2002). It is responsible for providing address translation, bandwidth control, and access control and helps to improve the quality of service and security of VoIP network. Gateway serves as an interface between public switched telephone networks (PSTN) and VoIP network. Gateway is made up of Signalling Gateway (GS), Media Gateway (MG), Media Controller and multipoint control unit (MCU). Signalling Gateway helps to manage signal traffic while the media gateway manages media signal. The media gateway controller manages traffic between signalling gateway and media gateway while multipoint control unit is the part that establishes conference calls between three or more people for converged voice, video and data conference. It provides audio or combination of audio, video and data conference depending on the strength of participating endpoint (Greg, 2004). Call processor is software running on end user equipment operating system that helps in call set up, call monitoring, user authorisation, signal coordination and bandwidth control. It translates a phone number to IP address. Figure 1 shows the components of VoIP system. Figure 1 Typical topology of VoIP network 19 Journal of Information Engineering and Applications www.iiste.org ISSN 2224-5782 (print) ISSN 2225-0506 (online) Vol.3, No.4, 2013 The server enables the establishment of call and support for other features in the system. The Session initiation server allows the user to forward calls to different location in the VoIP network. 2.2 VoIP Protocols and Standard There are many protocols and standards for VoIP network (H.323, SIP, MGCP, RTSP, RSVP, SDP etc) but only the most commonly used ones are discussed. H.323 protocol H.323 protocol was developed by International Telecommunication Union and Telephony (ITU-T) in 1997 based on Real-time Transport Control protocol (RTCP) for sending voice, video and data over IP-based network (Rakesh, 2000). H.323 provides multimedia conferencing on Local Area Network (LAN) and brings together point to point communication and multipoint conferences. It was widely adopted because it is reliable and easy to maintain. The components include terminals, Gateway, Gatekeeper and Multipoint control unit (MCU). Terminal is the end point that provides real-time communication for the VoIP network. Gateway is the interface between IP network and the PSTN to another H.323 gateway. It provides Internet translation between the different terminals. Gatekeeper is the most vital component of the H.323 protocol. It acts as the central point for all calls and provides services for end point registrations. The functions of Gatekeeper are: • Translate alias address to transport address • Deny or grant access based on call authorisation, source and destination address. • Call signalling with end point terminals • Control the number of terminals permitted at a time in H.323 protocol (bandwidth management) • Maintain the list of ongoing call H.323 calls to determine the busy terminals for bandwidth management. • Rejection of call from terminal due to authorisation failure in the use of H.225 signalling (Call authorisation). Another important component of H.323 protocol is the multipoint control unit (MCU). It acts as a bridge that enables two or more terminals and Gateway to participate in a multipoint conferencing. MCU is made up of Multipoint controller (MC) and Multipoint processor. Multipoint controller determines the capabilities of the network terminal using H.245 protocol stack but does not perform multiplexing of audio, video and data. Multipoint processor is responsible for multiplexing of media stream . H.323 consists of a number of protocol suites. The protocol suites and functions are listed: • H.245 provides capabilities for channel usage, advertisement, establishment and
Recommended publications
  • Application Note IPTV Services

    Application Note IPTV Services

    Application Note Contents Title Managing IPTV Performance Overview.......................................................... 1 Series IP Video Performance Management IPTV Services .................................................. 1 Date February 2008 Factors Affecting the Performance of IPTV.... 2 Video Impairments .......................................... 4 Video Performance Metrics ................................ 5 IPTV, Internet TV, and Video on Demand provide exciting new revenue opportunities IPTV Performance Management ........................ 6 for service providers. This Application Note Summary .......................................................... 8 describes some of the typical issues and problems affecting IPTV service quality, and introduces a cost-effective approach to service quality monitoring. IPTV Services make it publicly available. IPTV offers exciting new opportunities for service • Video on Demand (VOD) is a service that providers to introduce integrated voice, video, and provides access to movies or other video data services over broadband. A number of video content on demand. This could be part service types can be delivered over IP: of an IPTV service or could be a service offered independently over the Internet. • “IPTV” is generally used to refer to a closed Video over IP service with a broad range In some cases, IPTV services are not seen as of content delivered by a service provider. directly competing with existing cable or satellite Some definitions of IPTV would suggest service but may provide some
  • Title: Communicating with Light: from Telephony to Cell Phones Revision

    Title: Communicating with Light: from Telephony to Cell Phones Revision

    Title: Communicating with Light: From Telephony to Cell Phones Revision: February 1, 2006 Authors: Jim Overhiser, Luat Vuong Appropriate Physics, Grades 9-12 Level: Abstract: This series of six station activities introduces the physics of transmitting "voice" information using electromagnetic signals or light. Students explore how light can be modulated to encode voice information using a simple version of Bell's original photophone. They observe the decrease of the intensity of open-air signals by increasing the distance between source and receiver, and learn the advantage of using materials with different indices of refraction to manipulate and guide light signals. Finally, students are introduced to the concept of bandwidth by using two different wavelengths of light to send two signals at the same time. Special Kit available on loan from CIPT lending library. Equipment: Time Required: Two 80-minute periods NY Standards 4.1b Energy may be converted among mechanical, electromagnetic, Met: nuclear, and thermal forms 4.1j Energy may be stored in electric or magnetic fields. This energy may be transferred through conductors or space and may be converted to other forms of energy. 4.3b Waves carry energy and information without transferring mass. This energy may be carried by pulses or periodic waves. 4.3i When a wave moves from one medium into another, the waves may refract due a change in speed. The angle of refraction depends on the angle of incidence and the property of the medium. 4.3h When a wave strikes a boundary between two media, reflection, transmission, and absorption occur. A transmitted wave may be refracted.
  • Softphone - Cisco IP Communicator FAQ

    Softphone - Cisco IP Communicator FAQ

    Softphone - Cisco IP Communicator FAQ Getting Started Q. How do I order a Softphone instead of a Desk Phone? You can select a Softphone by the following process; Log a request via the Raise a Service Request button on the bottom of the page get a desk phone to get a Softphone: The form you need to fill in is under – Phones, Faxes, Mobiles and Accessories-- >Desk Phones and Accessories A. Fill in all the required details on the form, to select a Softphone, check the “Phone 1” Box and select the Softphone option from the drop down menu in “New Handset you require” As a softphone requires a headset, please select an option, the standard wired headset will be supplied at no charge, if you require a wireless headset, select this option and we will contact you do discuss the additional costs If you require a Softphone as well as a desk phone, this can be provided but an additional cost of $150 is required to cover the costs of the additional licencing. Q. What is Cisco IP Communicator? Cisco IP Communicator is a desktop application that turns your computer into a full- A. featured Cisco Unified IP Phone, allowing you to place, receive, and otherwise handle calls from any location where you can connect to the corporate network. For example, if you are on a business trip, you can use your IP Communicator to receive calls and check voice messages while you are online. Or, if you are working from home, co-workers can reach you by dialing your work number.
  • IP Telephony Fundamentals: What You Need to Know to “Go to Market”

    IP Telephony Fundamentals: What You Need to Know to “Go to Market”

    IP Telephony Fundamentals: What You Need to Know to “Go To Market” Ken Agress Senior Consultant PlanNet Consulting What Will Be Covered • What is Voice over IP? • VoIP Technology Basics • How Do I Know if We’re Ready? • What “Real” Cost Savings Should I Expect? • Putting it All Together • Conclusion, Q&A 2 What is Voice Over IP? • The Simple Answer – It’s your “traditional” voice services transported across a common IP infrastructure. • The Real Answer – It’s the convergence of numerous protocols, components, and requirements that must be balanced to provide a quality voice experience. 3 Recognize the Reality of IP Telephony • IP is the catalyst for convergence of technology and organizations • There are few plan templates for convergence projects • Everybody seems to have a strong opinion • Requires an educational investment in the technology (learning curve) – Requires an up-front investment in the technology that can be leveraged for subsequent deployments • Surveys indicate deployment is usually more difficult than anticipated • Most implementations are event driven (that means there is a broader plan) 4 IP Telephony vs. VoIP • Voice over IP – A broad technology that encompasses many, many facets. • IP Telephony – What you’re going to implement to actually deliver services across your network – Focuses more on features than possibilities – Narrows focus to specific implementations and requirements – Sets appropriate context for discussions 5 Why Does Convergence Matter? • Converged networks provide a means to simplify support structures and staffing. • Converged networks create new opportunities for a “richer” communications environment – Improved Unified Messaging – Unified Communications – The Promise of Video • Converged networks provide methods to reduce costs (if you do things right) 6 The Basics – TDM (vs.
  • SERVIZIO VOIP Dal 2005 Ci Siamo Specializzati Nella Fornitura Di

    SERVIZIO VOIP Dal 2005 Ci Siamo Specializzati Nella Fornitura Di

    SERVIZIO VOIP Dal 2005 ci siamo specializzati nella fornitura di servizi VOIP (Voice Over IP) di elevata qualità, con un’offerta tecnologicamente evoluta, posizionandoci in un segmento di mercato di nicchia. Abbiamo così realizzato un servizio basato su una tecnologia allo stato dell'arte che permette di sostituire le linee telefoniche tradizionali con linee VOIP, garantendone la stessa affidabilità e qualità . guarda il video di presentazione PLUS - Casi di successo documentati (rif. nostro sito Web www.timenet.it – sezione Case History) - Portabilità di ogni tipo di linea , comprese Selezioni Passanti (GNR) su BRI e PRI, dai tutti i principali Operatori , con programmazione temporale concordata con il Cliente della data del passaggio delle linee. - Servizio di backup : è sempre garantita la raggiungibilità del Cliente - Videochiamate punto – punto tra due numerazioni VOIP. - Servizio di deviazione di chiamata e voicemail . - Azzeramento dei costi fissi (canoni linee telefoniche). - Tariffe chiare, senza scatto alla risposta, conteggiate per gli effettivi secondi di conversazione. - Nuove numerazioni geografiche singole o GNR. - Numero di chiamate simultaneo illimitato anche con un solo numero VOIP. - Compatibilità testata con le maggiori piattaforme hardware (centralini e gateway): Avaya, AAstra, Samsung, Siemens, Asterisk, Patton, Audiocodes, Draytek, Linksys, etc… APPROFONDIMENTI CHE TIPO DI LINEE POSSIAMO PORTARE - Linee analogiche (POTS) e ISDN numero principale più numerazioni aggiuntive - Linee ISDN e PRI GNR (Selezione Passante) 10, 100, 1000 numeri - Linee in ULL e native VOIP di altri Operatori DA QUALI OPERATORI POSSIAMO EFFETTUARE LA NUMBER PORTABILITY Possiamo effettuare Number Portability da tutti i principali Operatori: Telecom Italia (anche numerazioni VoIP Alice Business Voce), BT-Albacom, Eutelia, Fastweb, Vodafone e Wind .
  • What Is the Impact of Mobile Telephony on Economic Growth?

    What Is the Impact of Mobile Telephony on Economic Growth?

    What is the impact of mobile telephony on economic growth? A Report for the GSM Association November 2012 Contents Foreword 1 The impact of mobile telephony on economic growth: key findings 2 What is the impact of mobile telephony on economic growth? 3 Appendix A 3G penetration and economic growth 11 Appendix B Mobile data usage and economic growth 16 Appendix C Mobile telephony and productivity in developing markets 20 Important Notice from Deloitte This report (the “Report”) has been prepared by Deloitte LLP (“Deloitte”) for the GSM Association (‘GSMA’) in accordance with the contract with them dated July 1st 2011 plus two change orders dated October 3rd 2011 and March 26th 2012 (“the Contract”) and on the basis of the scope and limitations set out below. The Report has been prepared solely for the purposes of assessing the impact of mobile services on GDP growth and productivity, as set out in the Contract. It should not be used for any other purpose or in any other context, and Deloitte accepts no responsibility for its use in either regard. The Report is provided exclusively for the GSMA’s use under the terms of the Contract. No party other than the GSMA is entitled to rely on the Report for any purpose whatsoever and Deloitte accepts no responsibility or liability or duty of care to any party other than the GSMA in respect of the Report or any of its contents. As set out in the Contract, the scope of our work has been limited by the time, information and explanations made available to us.
  • Avaya IP Softphone Release 4 Version 1 Online Reference

    Avaya IP Softphone Release 4 Version 1 Online Reference

    Avaya IP Softphone Release 4 Version 1 Online Reference September 2002 About This Guide This guide presents the information contained in the Avaya IP Softphone R4 Version 1 online help. 1 About Avaya IP Softphone Introduction Avaya IP Softphone is a collection of computer telephony integration (CTI) applications that enables you to control telephone calls (both incoming and outgoing) directly from your PC. Avaya IP Softphone enables you to log into your company's server remotely and make and receive telephone calls from the telephone extension. Avaya IP Softphone increases your telecommunications capabilities by providing the following applications: • Avaya IP Softphone Avaya IP Softphone is an application that enables you to control your telephone calls (both incoming and outgoing) directly from your PC. • Avaya iClarity IP Audio Avaya iClarity IP Audio is an application that performs the following functions: o Enables you to log into the server. You must be logged into the server before you can use Avaya IP Softphone to control your telephone calls. o Handles the voice communications when you use Avaya IP Softphone in the Road Warrior configuration (voice over IP). When you make or receive a call with Avaya IP Softphone, Avaya iClarity IP Audio enables you to speak to and hear the other party via a headset connected to your PC or the PC’s microphone and speakers. Avaya IP Softphone Configurations Avaya IP Softphone supports the following three configurations: • Road Warrior Configuration (Voice over IP) The Road Warrior configuration enables travelers to use the full feature set of your company's telephone system from temporary remote locations anywhere in the world (such as a hotel room).
  • Voip Primer Voice Over Internet Protocol

    Voip Primer Voice Over Internet Protocol

    VoIP Primer Voice over Internet Protocol WHY THE NET MAY REPLACE MA BELL: A GUIDE FOR STATE AGING SERVICE SYSTEMS NATIONAL AGING INFORMATION AND REFERRAL SUPPORT CENTER NATIONAL ASSOCIATION OF STATE UNITS ON AGING ½ WASHINGTON, DC VoIP Primer Voice over Internet Protocol Why the Net May Replace Ma Bell : A GUIDE FOR STATE AGING SERVICE SYSTEMS SEPTEMBER 2004 NATIONAL AGING INFORMATION AND REFERRAL SUPPORT CENTER NATIONAL ASSOCIATION OF STATE UNITS ON AGING ½ WASHINGTON, DC This publication is supported in part by grant No. 90-AM-2746 from the Administration on Aging, U.S. Department of Health and Human Services. Grantees undertaking projects under government sponsorship are encouraged to express freely their findings and conclusions. Points of view or opinions therefore for not necessarily reflect official Administration on Aging policy. Table of Contents 1 Introduction 3 VoIP: A Primer 5 How does VoIP Work? 9 Is It Worthwhile? Why Switch? 13 Before You Jump on the Bandwagon 15 Telephones Then and Now 19 Glossary 23 Selected References Introduction “WHO COULD have foreseen what the telephone bells have done to ring out the old ways and to ring in the new; to ring out delay and isolation and to ring in the efficiency and friendliness of a truly united people?" —Herbert N. Casson, The History of the Telephone Fully Illustrated, 1910 nternet Voice, also known as Voice over Internet Protocol (VoIP) or IP Itelephony, allows people to make telephone calls anywhere in the world using a high speed Internet connected computer as a phone. To receive or make a call, VoIP callers simply need to load special software on their computers or use a special computer adapter.
  • VOICE OVER INTERNET PROTOCOL (Voip)

    VOICE OVER INTERNET PROTOCOL (Voip)

    S. HRG. 108–1027 VOICE OVER INTERNET PROTOCOL (VoIP) HEARING BEFORE THE COMMITTEE ON COMMERCE, SCIENCE, AND TRANSPORTATION UNITED STATES SENATE ONE HUNDRED EIGHTH CONGRESS SECOND SESSION FEBRUARY 24, 2004 Printed for the use of the Committee on Commerce, Science, and Transportation ( U.S. GOVERNMENT PUBLISHING OFFICE 22–462 PDF WASHINGTON : 2016 For sale by the Superintendent of Documents, U.S. Government Publishing Office Internet: bookstore.gpo.gov Phone: toll free (866) 512–1800; DC area (202) 512–1800 Fax: (202) 512–2104 Mail: Stop IDCC, Washington, DC 20402–0001 VerDate Nov 24 2008 14:00 Dec 07, 2016 Jkt 075679 PO 00000 Frm 00001 Fmt 5011 Sfmt 5011 S:\GPO\DOCS\22462.TXT JACKIE SENATE COMMITTEE ON COMMERCE, SCIENCE, AND TRANSPORTATION ONE HUNDRED EIGHTH CONGRESS SECOND SESSION JOHN MCCAIN, Arizona, Chairman TED STEVENS, Alaska ERNEST F. HOLLINGS, South Carolina, CONRAD BURNS, Montana Ranking TRENT LOTT, Mississippi DANIEL K. INOUYE, Hawaii KAY BAILEY HUTCHISON, Texas JOHN D. ROCKEFELLER IV, West Virginia OLYMPIA J. SNOWE, Maine JOHN F. KERRY, Massachusetts SAM BROWNBACK, Kansas JOHN B. BREAUX, Louisiana GORDON H. SMITH, Oregon BYRON L. DORGAN, North Dakota PETER G. FITZGERALD, Illinois RON WYDEN, Oregon JOHN ENSIGN, Nevada BARBARA BOXER, California GEORGE ALLEN, Virginia BILL NELSON, Florida JOHN E. SUNUNU, New Hampshire MARIA CANTWELL, Washington FRANK R. LAUTENBERG, New Jersey JEANNE BUMPUS, Republican Staff Director and General Counsel ROBERT W. CHAMBERLIN, Republican Chief Counsel KEVIN D. KAYES, Democratic Staff Director and Chief Counsel GREGG ELIAS, Democratic General Counsel (II) VerDate Nov 24 2008 14:00 Dec 07, 2016 Jkt 075679 PO 00000 Frm 00002 Fmt 5904 Sfmt 5904 S:\GPO\DOCS\22462.TXT JACKIE C O N T E N T S Page Hearing held on February 24, 2004 ......................................................................
  • The Comcast Softphone User Guide for Microsoft Windows ®

    The Comcast Softphone User Guide for Microsoft Windows ®

    THE COMCAST SOFTPHONE USER GUIDE FOR MICROSOFT WINDOWS ® THE COMCAST SOFTPHONE USER GUIDE FOR MICROSOFT WINDOWS® 1 CONTENTS Introduction ��������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������3 System Requirements and Devices ������������������������������������������������������������������������������������������������������������������������������������������������������������������3 Login ���������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������4 Emergency Calling �������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������7 Basics on Using Your Comcast Softphone �����������������������������������������������������������������������������������������������������������������������������������������������������8 Placing A Call ���������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������� 11 Answering a Call ���������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������������12
  • Integration of Voice Over IP Services Over Ipv6 Networks

    Integration of Voice Over IP Services Over Ipv6 Networks

    View metadata, citation and similar papers at core.ac.uk brought to you by CORE provided by UCL Discovery Kirstein, P; Lambrinos, L; (2007) Integrating Voice over IP Services in IPv4 and IPv6 Networks. In: Proceedings of the International Multi-Conference on Computing in the Global Information Technology. IEEE ARTICLE Integrating Voice over IP services in IPv4 and IPv6 networks Lambros Lambrinos Dept.of Communication and Internet studies Cyprus University of Technology Limassol 3603, Cyprus [email protected] Peter Kirstein Department of Computer Science University College London (UCL) Gower Street, London, WC1E 6BT, UK [email protected] Abstract The transition from the current IPv4 network addressing model to the next generation IPv6 network addressing model has always been deemed as a lengthy process. As the deployment of IPv6 infrastructures evolves, interaction with the existing IPv4 networks is unavoidable. This interaction may occur at the network level (simple data pass-through) or at the application level (between IPv6 and IPv4 endpoints). In this paper we concentrate on application level interoperability and select an increasingly popular application: Voice Over IP (VoIP). The use of VoIP technologies is becoming more widespread as end-users and network providers recognise the potential benefits that the usage of the application provides. It is important to provide an environment where IPv4 and IPv6 networks and applications / devices can interoperate in the context of VoIP. This environment is based on an architecture that integrates all the components together (whether they support IPv4 or IPv6 or both) thereby providing ubiquitous access to IP telephony services based on the SIP signaling protocol.
  • VOIP-108 G2 / Connect Ipx8 – Intercom-Over-IP Solution

    VOIP-108 G2 / Connect Ipx8 – Intercom-Over-IP Solution

    VOIP-108 G2 / Connect IPx8 – Intercom-over-IP Solution Intelligent and seamless interfacing to the outside world is the key to success in many intercom applications. Connect IPx8 is a specialised Audio-over-IP interface to connect your intercom system to IP-based networks. The VOIP-108 G2 Client Card and the Connect IPx8 are Connect IPx8 8-Port Panel Interface designed to transport quality audio with low bandwidth The Connect IPx8 is a 19”/1RU unit which converts eight over digital IP-based networks. Designed for mission- AES or analogue signals into IP data and vice versa. The critical applications like the broadcast industries the device is available in three versions, offering different devices provide several failsafe options, which ensure interface options on the rear of the unit. The CAT5 and audio is available even under extreme circumstances. COAX versions are for connecting panels and other AES signals, while the AIO version is perfect for the connection There are three major applications for the devices: of 4-wires and other analogue sources. The Connect IPx8 • multi-port matrix-to-matrix connections (Trunking) can flawlessly connect up to eight standard Artist 1000, • multi-port matrix-to-control panel connections 2000 or 3000 series control panels with full functionality (Remote panels) to an Artist matrix via an IP-network. • multi-channel distribution of audio lines Easy Configuration VOIP-108 G2 8-Port Matrix Client Card The VOIP-108 G2 client card and the Connect IPx8 The new VOIP-108 G2 card converts eight Artist matrix panel interface can be configured directly in the Director ports into a compressed IP-stream and vice versa.