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)
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).
Wireless local area networks (WLANs) have traditionally been used to transport only data, but are now being used to carry voice traffic as well as provided new combined voice and data services. Voice over WLANs also offers more flexibility than wired networks (changes to a WLAN don't require changes to installed wiring) and greater capacity than wired networks. This book provides a solid overview of voice over WLANs/VoIP (voice over internet protocol) technology, including voice coding, packet loss, delay and jitter, and echo control. It shows how to combine both WLAN and VoIP technology to create effective voice over WLAN systems. Gives complete details on integrating voice and data services on WLANs, including wide area networks. Explores quality of service (QoS) and security issues. Step-by-step descriptions of how to plan and implement voice over WLAN networks.
This book presents and discusses current IPTV technology. Internet Protocol television (IPTV) is a system through which internet television services are delivered using architecture and networking methods of the Internet Protocol Suite over a packet-switched network infrastructure. IPTV services may be classified into three main groups: live television, time-shifted programming, and video on demand. Generally, IPTV works on the TV with a set-top box that accesses channels, subscription services, on demand and other interactive multimedia services over a secure, end-to-end operator managed broadband IP data network with desired QoS to the public with a broadband Internet connection. This book discusses the IPTV architecture, network technologies, implementation of IPTV and different types of STBs that are in use with special reference to New Zealand. The standards, different business models, hardware and software of IPTV are also described.
This book contains a robust and efficient dynamic technique for load balancing in WiMAX. WiMAX is one of the most recent broadband wireless technology that based on IEEE 802.16 standard and its amendments. This technique makes sure of balancing among all base stations in the whole network. It also take into account the quality of service parameters for real-time applications such as voice over Internet Protocol and video conferencing.
Speaker identity plays an important role in human communication. In addition to the linguistic content, speech utterances contain acoustic information of the speaker characteristics. This book describes voice conversion, a technique that aims at changing the voice of one speaker (a source speaker) into the voice of another specific speaker (a target speaker) without changing the linguistic information. The relationship between the source and target speaker characteristics is learned from the training data. Voice conversion can be used in various applications and fields: text-to-speech systems, dubbing, speech-to-speech translation, games, voice restoration, voice pathology, etc. The book provides up-to-date information for readers interested in voice conversion. It gives an overview of the field and gives insights for the future in the field.
Speech coding is one of the fundamental concepts for the accommodation of number of users in a communication channel. Many algorithms are used for speech coding. A variant of Codebook-Excited Linear Prediction (CELP) speech coding is widely used in wireless telephone communication (most of CDMA networks uses this type of vocoder), and has been accepted as the international standard for transmission of speech over telephone networks. This project explains what CELP speech coding is, why it is necessary for wireless communication, and describes unique design of a CELP speech coder. The CELP coder has been highly prosperous in practice and has spawned a series of standardized coders based on the same fundamental principles. This project implements the voice coder of a fixed-codebook which consists of 256 white-noise samples and takes seven seconds to encode one second of human speech. The speech coder produces an intelligible representation of the speech signal, which is very close to the original voice. The speech encoder and decoder are simulated and the output of CELP encoder is transmitted to the CELP decoder lying in different PC.
The payment industry moved from magnetic stripe cards to the EMV protocol over the past few years. This change in technology led to a drastic reduction in fraud through electronic payments. In the UK alone, losses in the high street have been reduced by 67% from ?218.8m in 2004 to ?72.1m in 2009. On the other hand, losses due to fraud have increased over the Internet. The majority of Internet payment protocols do not take advantage of the security provided by EMV protocol and the smart card technology. Most of the transactions are protected by a simple username and a password. Furthermore, customers have no assurance that their credit card details are well protected by merchants processing and storing such details. This book proposes a new Internet payment protocol that leverages the security provided by the EMV protocol and applies it to the Internet commerce context. By using mobile devices with Near Field Communication technology support together with contactless credit cards, a customer can pay for goods bought over the Internet with the same level of security found in store.
Various standards organizations have considered signaling for voice and video over Internet Protocol(IP) from different approaches. The Inter-Asterisk eXchange (IAX) protocol is used as the promising VoIP protocol by the service provider because of its simplicity and NAT friendliness. Meanwhile, the Real time SWitching (RSW) has the ability to combine voice and video services. Incidentally, these two heterogeneous clients pose considerable problems for users who have to choose between two solutions offering different advantages and disadvantages. While RSW is being used in many areas, IAX is being deployed in many VoIP services. Hence, RSW interoperability and coexistence with IAX is considered very important to support new deployments that could RSW as an alternative packet telephony signaling protocol.
Software cost estimation is the process of predicting the effort required to develop a software system. Many estimation models have been proposed over the last 30 years. Models may be classified into 2 major categories: algorithmic and non-algorithmic. Each has its own strengths and weaknesses. A key factor in selecting a cost estimation model is the accuracy of its estimates. Unfortunately, despite the large body of experience with estimation models, the accuracy of these models is not satisfactory. The work includes comment on the performance of the estimation models and description of several newer approaches to cost estimation. A CBR based efficient search technique has been introduced so that can help to obtain the best result.
The Dinka people live in the recently born state of South Sudan. Their language, counting over two millions of speakers, has been studied with growing interest for its rich prosodic system, featuring phonological distinctions in tone, vowel length and voice quality. Dinka songs play an important role as social aggregators and as carriers of private and collective memory. In this book the author studies for the first time the production and perception of the voice quality distinction in songs. Starting from the results already yielded by the acoustic analysis of Dinka speech data (Luanyjang dialect), acoustic measurements are applied to song data in order to determine if and how the phonological distinction between modal and breathy voice quality is realized in songs. The perceptual dimension is assessed with a perception test, where listeners are asked to distinguish between the two types of phonation. The introductory chapters and the up-to-date extensive bibliography make this book not only a good starting point to know more about the definition and measurement of voice quality, but also as a quick and easy-to-read introduction to the fascinating complexity of Dinka.
With the recent advances in speech signal processing techniques, the need to detect the presence of speech accurately in the incoming signal under different noise environments has become a major concern of the industry. The separation of speech segment from the non-speech segment in an audio signal is achieved using a Voice Activity Detectors (VAD). VAD’s are a class signal processing methods that detects the presence or absence of speech in short segments of audio signal. A VAD has a pivotal role as a preprocessing block in wide range of speech applications. An integrated VAD in speech communication system, improves channel capacity, reduces co-channel interference and power consumption in portable electronic devices in cellular radio systems and allows simultaneous voice and data applications in multimedia communications. In slowly varying non-stationary environments where speech is corrupted by noise, a VAD is used to learn noise characteristics and estimate the noise spectrum. Furthermore, the output from the VAD is helpful in improving the performance of the speech recognition systems which applies a technique called non-speech frame dropping (FD) to reduce the insertion error