Telecommunications technologies are evolving at a rapid pace. The old Public Switched Telephone Network (PSTN) is being replaced with wireless and voice over IP (VoIP) systems. This requires the service providers to offer their services on competitive prices, on one hand, and to ensure the interoperability of their services over hetero- geneous networks on the other. Added to this is the challenge of keeping up with the expectations of the clientele as regards quality of service (QoS). Thus to enable the successful deployment and functioning of a telecommunications network, it is equally important to estimate the speech quality as it may be perceived by the humans. The goal of this research was to derive superior non-intrusive speech quality esti- mation models. Model superiority was sought in a multi-objective sense: 1) enhance- ment of prediction accuracy of the derived models as compared to the previous ones. 2) model simplicity or parsimony was desired as it may enhance the computational efficiency. In this research this is achieved by employing a novel approach based on Genetic Programming (GP).
Please note that the content of this book primarily consists of articles available from Wikipedia or other free sources online. Yate is a telephony engine, a softswitch with PBX capabilities, originally created in 2004 by Paul Chitescu of NullTeam. It is released under a GNU General Public License with an exception for linking with OpenH323 and PWlib (licensed under MPL). Started as a free/open source softswitch for connecting PSTN with VoIP, support for other telephony technologies like conference, PBX, IVR has been added later on. During the years Yate has evolved into a Unified Communications server providing advanced integration with instant messaging, video and fax. Historically the basic function of Yate is the softswitch, connecting together call legs either to external devices or internal server resources. The logic for routing calls uses messages and can be implemented in configuration files, database queries or external programs. Call signalling is possible using PSTN protocols like ISDN or SS7, either directly attached over a TDM interface card (T1, E1, BRI), analog interface or by a remote Signaling gateway. A variety of VoIP protocols can be used without the need of special hardware: SIP, H.323, IAX2 or Jingle
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).
Amplitude-shift keying (ASK) is a form of modulation that represents digital data as variations in the amplitude of a carrier wave. The amplitude of an analog carrier signal varies in accordance with the bit stream (modulating signal), keeping frequency and phase constant. The level of amplitude can be used to represent binary logic 0s and 1s. We can think of a carrier signal as an ON or OFF switch. In the modulated signal, logic 0 is represented by the absence of a carrier, thus giving OFF/ON keying operation and hence the name given. Like AM, ASK is also linear and sensitive to atmospheric noise, distortions, propagation conditions on different routes in PSTN, etc. Both ASK modulation and demodulation processes are relatively inexpensive. The ASK technique is also commonly used to transmit digital data over optical fiber. For LED transmitters, binary 1 is represented by a short pulse of light and binary 0 by the absence of light. Laser transmitters normally have a fixed "bias" current that causes the device to emit a low light level. This low level represents binary 0, while a higher-amplitude lightwave represents binary 1.
Voice over IP (VoIP) and Internet Multimedia Subsystem technologies (IMS) are rapidly being adopted by consumers, enterprises, governments and militaries. These technologies offer higher flexibility and more features than traditional telephony (PSTN) infrastructures, as well as the potential for lower cost through equipment consolidation and, for the consumer market, new business models. However, VoIP systems also represent a higher complexity in terms of architecture, protocols and implementation, with a corresponding increase in the potential for misuse.In this book, the authors examine the current state of affairs on VoIP security through a survey of 221 known/disclosed security vulnerabilities in bug-tracking databases. We complement this with a comprehensive survey of the state of the art in VoIP security research that covers 245 papers. Juxtaposing our findings, we identify current areas of risk and deficiencies in research focus. This book should serve as a starting point for understanding the threats and risks in a rapidly evolving set of technologies that are seeing increasing deployment and use. An additional goal is to gain a better understanding of the security landscape with respect to VoIP toward directing future research in this and other similar emerging technologies.
Seven Deadliest Unified Communications Attacks provides a comprehensive coverage of the seven most dangerous hacks and exploits specific to Unified Communications (UC) and lays out the anatomy of these attacks including how to make your system more secure. You will discover the best ways to defend against these vicious hacks with step-by-step instruction and learn techniques to make your computer and network impenetrable.The book describes the intersection of the various communication technologies that make up UC, including Voice over IP (VoIP), instant message (IM), and other collaboration technologies. There are seven chapters that focus on the following: attacks against the UC ecosystem and UC endpoints, eavesdropping and modification attacks, control channel attacks, attacks on Session Initiation Protocol (SIP) trunks and public switched telephone network (PSTN) interconnection, attacks on identity, and attacks against distributed systems. Each chapter begins with an introduction to the threat along with some examples of the problem. This is followed by discussions of the anatomy, dangers, and future outlook of the threat as well as specific strategies on how to defend systems against the threat. The discussions of each threat are also organized around the themes of confidentiality, integrity, and availability.This book will be of interest to information security professionals of all levels as well as recreational hackers.Knowledge is power, find out about the most dominant attacks currently waging war on computers and networks globallyDiscover the best ways to defend against these vicious attacks, step-by-step instruction shows you howInstitute countermeasures, don't be caught defenseless again, and learn techniques to make your computer and network impenetrable
- Install, configure, deploy, secure, and maintain Asterisk - Build a fully-featured telephony system and create a dial plan that suits your needs - Learn from example configurations for different requirement - Implement 3rd party applications which will help you manage Asterisk through a web interface Asterisk is a powerful and flexible open source framework for building feature-rich telephony systems. As a Private Branch Exchange (PBX) which connects one or more telephones, and usually connects to one or more telephone lines, Asterisk offers very advanced features, including extension-to-extension calls, queues, ring groups, line trunking, call distribution, call detail rerecords, and call recording. This book will show you how to build a telephony system for your home or business using this open source application. 'Asterisk 1.6' takes you step-by-step through the process of installing and configuring Asterisk. It covers everything from establishing your deployment plan to creating a fully functional PBX solution. Through this book you will learn how to connect employees from all over the world as well as streamline your callers through Auto Attendants (IVR) and Ring Groups. This book is all you need to understand and use Asterisk to build the telephony system that meets your need. You will learn how to use the many features that Asterisk provides you with. It presents example configurations for using Asterisk in three different scenarios: for small and home offices, small businesses, and Hosted PBX. Over the course of ten chapters, this book introduces you to topics as diverse as Public Switched Telephony Network (PSTN), Voice over IP Connections (SIP / IAX), DAHDI, libpri, through to advanced call distribution, automated attendants, FreePBX, and asterCRM. With an engaging style and excellent way of presenting information, this book makes a complicated subject very easy to understand. An easy introduction to using and configuring Asterisk to build feature-rich telephony systems for small, medium and large businesses What you will learn from this book : - Install, configure, and deploy Asterisk to build a fully-featured telephony system - Install and use FreePBX - Connect your Asterisk server with your phone service (via PSTN, SIP, etc) as well as learn to deploy some basic PBX features such as queues, voicemail and music on hold - Determine how calls are routed through the Asterisk server by creating a dialplan - Create extensions, distribute calls in an orderly fashion using queues, and present callers with a greeting using automated attendants (IVR) - Install and learn how to monitor, record, and capture detailed call logs - Install and use asterCRM (customer relation management solution) to streamline your business operations - Gain knowledge of security precautions, network deployment recommendations as well as maintenance tips such as backups and preparing disaster recovery plans for keeping the Asterisk system running smooth and secure Who this book is written for This book is aimed at anyone who is interested in building a powerful telephony system using the free and open source application, Asterisk, without spending many thousands of dollars buying a commercial and often less flexible system. This book is suitable for the novice and those new to Asterisk and telephony. Telephony or Linux experience will be helpful, but not required.
- 16, 24 or 32 FXS or FXO—Simultaneous voice or fax calls on all ports. Advanced local call switching.- Full SIP and T.38 support—Supports the complete range of industry standard VoIP: SIP, H.323, T.38 fax, fax and modem bypass, DTMF relay. Codecs G.729, G.723 etc.- Outstanding Interoperability—Interoperable for voice and T.38 fax with leading SIP service providers, soft-switch vendors and Asterisk IP-PBXThe SmartNode 4300 VoIP Gateway provides 12 to 32 analog FXS or FXO interfaces to connect phones, fax or PSTN trunk lines to your IP-based communications (IP PBX, UC systems and SIP Trunks).Like every SmartNode, the SN4300 supports every industry-standard CODEC to deliver toll-quality voice on every call. The Unified Communications Agent™ (UCA) provides any-to-any multi-path switching (simultaneous SIP, H.323, ISDN, and POTS calls with routing and conversion between TDM/PSTN and IP/Ethernet networks—plus T.38 and SuperG3 FAX). VoIP-over-VPN with voice encryption provides secure voice and data via IPsec with AES/DES strong encryption and automated keying via Internet Key Exchange (IKE). In addition, advanced call-router functionality includes least-cost call routing with flexible dialed-number plan support.The SmartNode survivability suite provides PSTN fallback to ensure business continuity in case the IP network fails. In addition, SmartNode delivers a smooth transition to VoIP with strong number portability support accepting incoming calls from the PSTN throughout the VoIP service provider's number porting process.Preserve investments in legacy phone equipment while taking the next steps toward unified communications with the SN4300 VoIP Gateway. Providing 12 to 32 FXS or FXO interfaces and one 10/100/1000 Ethernet, the SN4300 delivers a reliable, cost-effective solution for the Enterprise.
A complete and systematic treatment of signal processing for VoIP voice and fax This book presents a consolidated view and basic approach to signal processing for VoIP voice and fax solutions. It provides readers with complete coverage of the topic, from how things work in voice and fax modules, to signal processing aspects, implementation, and testing. Beginning with an overview of VoIP infrastructure, interfaces, and signals, the book systematically covers: * Voice compression * Packet loss concealment techniques * DTMF detection, generation, and rejection * Wideband voice modules operation * VoIP Voice-Network bit rate calculations * VoIP voice testing * Fax over IP and modem over IP * Country deviations of PSTN mapped to VoIP * VoIP on different processors and architectures * Generic VAD-CNG for waveform codecs * Echo cancellation * Caller ID features in VoIP * Packetization--RTP, RTCP, and jitter buffer * Clock sources for VoIP applications * Fax operation on PSTN, modulations, and fax messages * Fax over IP payload formats and bit rate calculations * Voice packets jitter with large data packets * VoIP voice quality Over 100 questions and answers on voice and more than seventy questions and answers on fax are provided at the back of the book to reinforce the topics covered throughout the text. Additionally, several clarification, interpretation, and discussion sections are included in selected chapters to aide in readers' comprehension. VoIP Voice and Fax Signal Processing is an indispensable resource for professional electrical engineers, voice and fax solution developers, product and deployment support teams, quality assurance and test engineers, and computer engineers. It also serves as a valuable textbook for graduate-level students in electrical engineering and computer engineering courses.