The EMA system: a CTI based e-mail alerting service

The integration of Internet services and telephony services is a new area for the development of telecommunications services. One example is an e-mail alerting service that uses the telephony network for e-mail notification. The EMA system is a computer telephony integration (CTI) application that checks a user's mailbox on the mail server and informs him/her over the phone when new e-mail arrives, eliminating the need for permanent Internet connection. The EMA system has a Web-based interface, enabling the user to configure service parameters. The EMA system is developed as a distributed and concurrent application. It consists of seven modules: the console, Web interface, Web handler, controller, voice machine, database, and mail checker, using communication solutions based on component object model (COM) technology. This article describes the structure of the EMA system, its implementation, and advantages for users     

An introduction to TSAPI and network telephony

Discusses the integration of the telephone into the desktop work environment. This objective can be achieved in a number of different ways. The approach described uses the client-server model. Network telephony is a service that provides the capability to establish, answer, route, and terminate telephone calls under the control of applications on either desktop computers or servers resident in the network. It is accessed by applications via a standard programming interface (telephony services application programming interface, TSAPI) and utilizes a centralized server-based interface with the private branch exchange (PBX) to access the switching and telephone control services that the PBX provides. Creating a logical control link between the application software on the desktop computer and the telephone on the desk eliminates the need to physically connect the telephone to the desktop computer. The author distinguishes between the application programming interface and the network telephony service provider. The author assumes the services are provided by NTS R2.2 release and a PBX-based switching service. However, the author uses the term PBX to include switching services provided by key telephone or hybrid systems, PC-based telephony cards, or other appropriate technologies     

一种基于AMBE-2000的语音系统的设计与实现

介绍一种基于AMBE-2000的高质量语音系统,该系统使用DVSI公司开发的AMBE-2000语音芯片,可以在低速率下保持声音自然,语音清晰,并且在选择语音速率和误码率上提供了很高的灵活性。此外,它还具有低成本,低功耗等优点。因此,该系统具有广泛的应用前景,可以应用于诸如安全通信、话音多路传输、卫星通信、多媒体应用、蜂窝电话等多种语音处理的场合。     

Ein Multi-Agentensystem für die modellbasierte Echtzeit-Fehlerdiagnose

Real-time AI systems, among them multiagent systems, are gaining importance in complex technical applications. This paper presents VEX, an ambitious multiagent system for model-based real-time fault diagnosis in modular production systems. The architecture comprises several process agents, a simulation agent and a user agent. We discuss the most interesting characteristics of this real-time AI architecture. As implementation examples we present the multi-DSP core of the simulation agent, the client-server communication protocol for the process agents and the user agent’s graphical interface.     

Echo cancellation strategy using chaotic modulated speech

A novel speech coding strategy that can be used for telephony channels and that allows existing echo cancellers to approach their fundamental operating limit is reported. A chaotic-based modulation regime which uses logistic mapping is exploited to whiten the speech PSD. By using the described strategy, an echo attenuation of 50.3 dB was achieved after only 2000 iterations (250 ms) as opposed to the conventional 23.4 dB over the same interval. Furthermore, the model misadjustment was -17.8 and -82.6 dB after 250 ms of uncoded and chaotic coded speech, respectively     

PACS: personal access communications systems an alternativetechnology for PCS

The personal access communications system (PACS) is an American National Standards Institute common air interface standard developed for the 1.9 GHz PCS band in the United States. The PACS uses frequency division duplexing technology and is optimized to support low-mobility pedestrian outdoor usage and wireless local loop applications in a medium-range environment. PACS-Unlicensed B (PACS-UB) is a version of PACS using time division duplexing. The PACS-UB has been optimized for private, indoor wireless PBX applications and cordless telephony. Both modes of operation are supported using the same portable hardware and the same signaling protocol     

Radio interfaces make the difference in 3G cellular systems

Third-generation cellular telephony is on its way-not, unfortunately, as a single worldwide system, but as three incompatible ones. The main difference between the three lies in their choice of radio interface technology. This fact is crucial for several reasons, since the radio interface determines not only the fundamental capacity of a mobile radio network, but also how it deals with such issues as interference, multipath distortion, and handing off calls from one base station to another as users move around. Consequently, as might be expected, the choice of radio interface has a dramatic effect on the complexity of the system and its cost. Also, global travellers will need more than one phone with which to communicate, at least until trimode phones reach the market. To understand what is being developed, and why, the author begins with one of the stated goals of third-generation (3G) systems, namely to support variable user data rates as high as 2 Mb/s. In one way or another, all three approaches provide for adaptive bandwidth-on-demand. Two of the systems use wideband code-division multiple access (WCDMA) for the radio interface. The other uses two variations of time-division multiple access (TDMA)     

Prediction of Speech Quality Based on Resilient Backpropagation Artificial Neural Network

The paper presents a system for monitoring and assessment the speech quality in the IP telephony infrastructures using modular probes. The probes are placed at key nodes in the network where aggregating packet loss data. The system dynamically measures speech quality and results are collected on a central server. For data analysis we applied four-state Markov model for modeling the impact of network impairments on speech quality, afterwards, the resilient back propagation (Rprop) algorithm was used to train a neural network. Information about the speech quality are displayed in the form of automatically generated graphs and tables. The proposed solution has been tested with selected codecs and further generalizes the already presented concepts of the speech quality estimation in the IP environment.

     

High-quality coding of telephone speech and wideband audio

Digital speech technology is reviewed, with the emphasis on applications demanding high-quality reproduction of the speech signal. Examples of such applications are network telephony, ISDN terminals for audio teleconferencing, and systems for the storage of audio signals, which include the important subclass of wideband speech. Depending on the application, the bandwidth of input speech can vary from about 3 kHz to nearly 20 kHz. Coding for digital telephony at 4 and 8 kb/s, network quality coding at 16 kb/s, and coding for audio at 7 and 20 kHz are examined. Future directions in the field are discussed with respect to anticipated technology applications and the algorithms needed to support these technologies     

Design considerations for computer-telephony applicationprogramming interfaces and related components

Discusses computer telephony integration (CTI) architecture, including the application programming interface (API) and underlying operating system components and the benefits they can bring to all segments of the computer telephone industry. Although CTI APIs have existed in various forms for more than two decades, the work of crafting the ideal interface continues. As operating systems and telecommunications network technology evolves, so do the APIs and system components that bring together the computing and communications worlds. The author presents some of the considerations that have gone into the design of the Windows telephony application programming interface (TAPI) and which will continue to guide the further evolution of the interface     

Handoff with DSP Support: Enabling Seamless Voice Communications across Heterogeneous Telephony Systems on Dual-Mode Mobile Devices

In this paper we investigate the problem of voice communications across heterogeneous telephony systems on dual-mode (WiFi and GSM) mobile devices. Since GSM is a circuit-switched telephony system, existing solutions that are based on packet-switched network protocols cannot be used. We show in this paper that an enabling technology for seamless voice communications across circuit-switched and packet-switched telephony systems is the support of digital signal processing (DSP) techniques during handoffs. To substantiate our argument, we start with a framework based on the Session Initiation Protocol (SIP) for vertical handoffs on dual-mode mobile devices. We then identify the key obstacle in achieving seamless handoffs across circuit-switched and packet-switched systems, and explain why DSP support is necessary in this context. We propose a solution that incorporates time alignment and time scaling algorithms during handoffs for supporting seamless voice communications across heterogeneous telephony systems. We conduct testbed experiments using a GSM-WiFi dual-mode notebook and evaluate the quality of speech when the call is migrated from WiFi to GSM networks. Evaluation results show that such a cross-disciplinary solution involving signal processing and networking can effectively support seamless voice communications across heterogeneous telephony systems.     

Transformation of whispering voice to pseudo-real voice for unvoiced telephony and communication aid for voice-handicapped persons

For voice handicapped people, an easy to use voicing aid device is wanted. In mobile telephony, so-called non-speaking speech communication is an expected solution for essential privacy as well as for acoustic nuisance prevention. The study introduced here intends to cover both issues, introducing a system where the whispering (non-speaking voice or talk without vocal fold activation) signal is converted to pseudo-real voice signal, which is to be sent to, or heard by, the other party. The study also includes validation tests with multiple volunteers for its output legibility. Unlike general concept of speech regeneration being inclined to signal recognition or decomposition to text followed by electronic reading (voicing), our system converts it almost directly without recognition or decomposition steps. The processing is based on repetitive playback of short time autocorrelation, conducted by synthetic pitch pulse. A real time software pipeline process is under development.     

The realization of integrated services in a digital distributed star local network

The BIDS system will go on trial in early 1990 and it will be a unique system for combining broadband, telephony and data services. The present design uses analogue and digital transmission techniques, but a design for a completely digital system may be required. A study of future requirements had to be undertaken to ensure that the system would adequately serve the customer of the year 2000 and onwards. A design for a digital secondary link is presented employing a modular optical dual-wavelength diplex system.     

Speech and Video Telephony Quality Characterization and Prediction of Live Contemporary Mobile Communication Networks

Quality of Service (QoS) characterization and prediction is of utmost importance in contemporary operating cellular communications networks. Measurements data of speech and video telephony have been collected using modern experimental equipment. More specifically, key performance indicators of radio, speech and video quality are evoked. The objective of our study is to critically investigate the performance of speech and video telephony at live cellular networks correlating significant QoS parameters from radio and the service side. Simple non-linear regression models are also proposed for speech and video quality prediction. Finally, the paper represents the splendid positive influences of the continuous performance evaluation for the optimization of the mobile networks. There are also briefly given guidelines for mobile networks benchmarking.     

VTOA/VolP/N-ISDN Telephony Gateway

This article presents the architecture and implementation of a telephony gateway for interworking between N- ISDN, ATM and IP telephony. In this way, interworking is achieved both within private networks and with the PSTN, address translation being performed according to both the vtoa (atm interface) and H .323 (ip interface) specifications. The gateway implementation is based on a PC, presenting a cost- effective alternative to the equipment currently available on the market. Moreover, its highly modular software architecture allows new telephony interfaces to be easily added.     

An analog/digital interface for cellular telephony

This paper describes a mixed-signal ASIC for dual-mode (analog/digital) cellular telephony applications. It consists of two transmit and two receive channels corresponding to the I and Q channels of a quadrature phase-shift keying (QPSK) modulation system. It also includes three 8 b DAC's for control purposes, as well as a bandgap voltage reference and bus interface circuitry. The chip is part of a four-chip implementation of an IS-54 dual mode telephone. The chip was implemented in a 0.8 μm n-well double-metal CMOS process and uses a 5 V power supply. The die area of the chip was 23 mm² and the average power consumption was 125 mW     

Internet phone — changing the telephony paradigm?

This paper provides an overview of the history, current status, and underlying technology of Internet telephony — the transport of speech information over a packet-switched connectionless network. The economics of Internet telephony and a number of possible system architectures for interconnection with the switched telephone network are examined. The paper concludes with a look at the quality of service issues raised by Internet telephony.     

True number portability and advanced call screening in a SIP-basedIP telephony system

Custom local area signaling service features offered in the PSTN have certain limitations due to the closed nature of PSTN network signaling. The adoption of telephony over IP (IP telephony) will enable a new paradigm of services and features that are not possible to implement in today's PSTN. This is especially the case for services that make use of personal, trusted information, which can be provided by a user's personal digital assistant. We demonstrate how personal information can be coupled with an IP telephony service to provide user-customized call handling by the network. In particular, we describe a demonstration architecture that includes Ethernet-attached phones running SIP, with an interface to synchronize with PDAs that supply personal information. The proposed architecture is quite flexible; it can support enhanced versions of the current PSTN and private branch exchange services, in addition to many new features and services. We describe true number portability and advanced call screening as examples of new services in a hybrid PSTN/IP telephony environment     

Performance of Video Telephony Services in UMTS using Live Measurements and Network Emulation

While speech services in mobile communication systems are investigated quite well, video telephony services are relatively novel in this sector. Because of the market penetration of camera equipped mobile phones the video telephony service is expected to become a widely used service. In this article an introduction of the protocols used for video telephony in UMTS is given. Concepts for the performance evaluation in terms of video (PEVQ) and audio quality (PESQ) are presented and utilized. Evaluations are performed by both live measurements and network emulation. The results show that there is quite potential for improvements related to video telephony in UMTS in terms of video quality and channel setup time. Finally, an improved radio bearer configuration is provided which aims at a better integration of video telephony services into the UMTS architecture.     

Agent-Based Approach to Mass-Oriented Production Planning: Case Study

This paper discusses the potential of multiagent planning techniques in the production-planning domain, with special focus on mass-oriented production. The research presented in the paper has been centered around ExPlanTech-a specific implementation of a production-planning multiagent system. Suitability of ExPlanTech for mass-oriented and project-driven manufacturing is also discussed in the paper. Applicability of multiagent concepts is demonstrated on a multiagent planning architecture and production-planning case study at a Skoda Auto Engine Plant