Before now, all telephony services are based on the circuit-switched networks, known as the Public Switched Telephone Networks (PSTN). But nowadays, a new trend has emerged that provides telephony services over internet protocol networks, referred to as Voice over IP (VoIP) or IP telephony. There are a few driving force and motivational factors that are behind the VoIP network - the cost savings, the voice and data applications' integration etc are the crux of the innovation. The cost saving is achieved mainly by reduced cost of making a long distance call, maintenance, deployment, management etc., while the integrated applications of voice and data are teleconferencing, integrated voice mail and email, the call distribution intelligence and automation. Therefor, with this era of innovative business practices in information technology that involves voice and data networks integration, both transmitted over the internet and implemented on a local area network (LAN).This book is undoubtedly beneficial to Small and Medium sized businesses for convergence of data and voice networks in their existing data network to accommodate the VoIP solution in the same LAN.
An IP phone uses Voice over IP technologies allowing telephone calls to be made over an IP network such as the internet instead of the ordinary PSTN system. Calls can traverse the Internet, or a private IP Network such as that of a company. The phones use control protocols such as Session Initiation Protocol, Skinny Client Control Protocol or one of various proprietary protocols such as that used by Skype. IP phones can be simple software-based Softphones or purpose-built hardware devices that appear much like an ordinary telephone or a cordless phone. Ordinary PSTN phones are used as IP phones with analog telephony adapters (ATA).It may have many features an analog phone doesn't support, such as e-mail-like IDs for contacts that may be easier to remember than names or phone numbers.
Traditional telephony last about 100 years in use as the basic voice communication because of its reliability that satisfy the normal needs. Packet switching network, especially the Internet, which rapidly spread , attract more and more applications, because of its flexibility and efficiency. One of the most important applications is VoIP, which transmit voice in the same method of transmitting data, taking into account that voice is a real time application. Once we can transmit the voice over IP network, we do not need to route calls using the expensive and large central switches of PSTN, since we can route the calls in the same manner that we route data in IP network. This is the task of the soft switch, soft switch may seem to be the counterpart of central office or PBX of PSTN network. Finally, this project outlines the overall system of VoIP communication, and then show how to implement a whole telephone system based on IP protocols. The implementation of this project will improve the telephony service and management in (IUG).
High Quality Content by WIKIPEDIA articles! H.323 is an umbrella Recommendation from the ITU Telecommunication Standardization Sector (ITU-T) that defines the protocols to provide audio-visual communication sessions on any packet network. The H.323 standard addresses call signaling and control, multimedia transport and control, and bandwidth control for point-to-point and multi-point conferences. It is widely implemented by voice and videoconferencing equipment manufacturers, is used within various Internet real-time applications such as GnuGK and NetMeeting and is widely deployed worldwide by service providers and enterprises for both voice and video services over Internet Protocol (IP) networks. It is a part of the ITU-T H.32x series of protocols, which also address multimedia communications over Integrated Services Digital Network (ISDN), Public Switched Telephone Network (PSTN) or Signaling System 7 (SS7), and 3G mobile networks. H.323 Call Signaling is based on the ITU-T Recommendation Q.931 protocol and is suited for transmitting calls across networks using a mixture of IP, PSTN, ISDN, and QSIG over ISDN.
Voice over IP telephony is gaining popularity quickly, and is often replacing traditional PSTN connectivity due to its low cost and easy implementation. In moving from a legacy PSTN system to a newer technology such as VoIP, security is often overlooked by vendors and users. This can present a significant problem in the corporate environment, where security in terms of both privacy and service is a key requirement. For residential users, credential theft can result in a significant annoyance, such as toll fraud or identity theft. This book focuses on attacks that compromise privacy, and details methods of detecting and mitigating these attacks. IPSec is used to ensure that signalling and media traffic are unable to be intercepted. These techniques should help administrators of networks carrying voice traffic to implement necessary measures to ensure that security and privacy remain intact. They should be particularly useful in corporate environments, where sensitive information needs to remain inaccessible to anyone other than its intended recipients.
Compared to traditional (PSTN) voice networks, a Voice over Internet Protocol network is a convergence of a signaling network and a data network using Internet Protocol (IP). The use of shared media by VoIP systems opens the door to some uncertainty as to the source of a call. While in the traditional voice networks one has to tap into a specific circuit to eavesdrop, in an IP network any equipment connected to the target LAN can identify, store and playback the VoIP packets that traverse that LAN. An unprotected, unauthenticated IP network makes VoIP susceptible to hostile use, such as call hijacking, connection tear down, denial of service, or sending computer viruses over the network. In this work, we perform a series of attacks against a VoIP application, and prove that they succeed with nothing more than a couple of identity tokens captured from the network traffic as prerequisites. We then design an Intrusion Detection System implementing a gradual attack-response procedure, destined to inform and protect the End-Users of the Application Under Test.
High Quality Content by WIKIPEDIA articles! Asterisk is a software implementation of a telephone private branch exchange (PBX) originally created in 1999 by Mark Spencer of Digium. Like any PBX, it allows attached telephones to make calls to one another, and to connect to other telephone services including the public switched telephone network (PSTN) and Voice over Internet Protocol (VoIP) services. Its name comes from the asterisk symbol. Asterisk is released under a dual license model, using the GNU General Public License (GPL) as a free software license and a proprietary software license to permit licensees to distribute proprietary, unpublished system components.
Existing wireless networks enable different important applications over IP based network such as public internet and Voice over Internet protocol is one of the important applications which have become a possible alternative to public switched telephone network (PSTN). This project investigates one of the time sensitive data applications on WiMAX i.e. VoIP. I carried on WiMAX because WiMAX networks provide advance features and protocols to support the Quality of service (QoS). This project also indicates about the impact of load and mobility on the voice calls. This book provides details about the Voice Over Internet Protocol (VoIP), Which includes types of VoIP calls, VoIP System and Components, VoIP Protocols, VoIP codecs and VoIP Quality of Service (QoS). It also provides detailed study of WiMAX including WiMAX Protocol Layer and WiMAX Quality Of Service(QoS)
Voice Over IP steht für Sprachübertragung mit Hilfedes Internet Protokolls und wird als bedeutendertechnischer Fortschritt in derTelekommunikationsbranche angesehen. Es wird nichtmehr nur das IP-Netz im Vordergrund betrachtet,sondern dessen Konvergenz zu anderen Netzen. Wieerfolgt das Zusammenspiel zwischen zweiunterschiedlichen Netzen, speziell dem PSTN undIP-Netzwerk? Wie verhalten sich die den Netzenzugrunde liegenden Protokolle ISUP und SIPzueinander? Die Autorin Marlies Andreeff untersuchtdas Umsetzen der einzelnen ISUP-Nachrichten und ihrerParameter in die SIP/SDP-basierten Nachrichten undzieht als Basis die Empfehlung Q.1912.5 von der ITU-Theran. Zusätzlich beschreibt die Autorin dieunterschiedlichen Protokollwelten SIP und ISUP inihren Grundlagen und Besonderheiten sowie die für dasInterworking wichtigen Kommunikationselemente. DieSignalisierungs- abläufe werden in Sequenzdiagrammendargestellt. Um das Interworking dem Leser aufanschauliche Weise nahe zu bringen hat die AutorinPraxisbeispiele integriert. Dieses Buchrichtet sich an Studenten der Informations- undKommunikationstechnik sowie an alle die sich für dasNext Generation Network interessieren.