Programmable Conference Server

Conferencing services for Internet telephony and multimedia can be enhanced by the integration of other Internet services, such as instant messaging, presence notification, directory lookups, location sensing, email and web. These services require a service programming architecture that can easily incorporate new Internet services into the existing conferencing functionalities, such as voice-enabled conference control. W3C has defined the Call Control eXtensible Markup Language (CCXML), along with its VoiceXML, for telephony call control services in a point-to-point call. However, it cannot handle other Internet service events such as presence enabled conferences. In this paper, we propose an architecture combining VoiceXML with our Language for End System Services (LESS) and the Common Gateway Interface (CGI) for multi-party conference service programming that integrates existing Internet services. VoiceXML provides the voice interface to LESS and CGI scripts. Our architecture enables many novel services such as conference setup based on participant location and presence status. We give some examples of the new services and describe our on-going implementation

Real time speech translator

This document is written to report on my work on the Real Time Voice Translator Project, the project I carried out as my final thesis project during the academic year 2007‐2008. During this period I have been working in the Research and Development Center (RDC) for Mobile Applications, a department of the Czech Technical University (CTU) in Prague. In the RDC I was a member of the Automatic Call Center Project (ACC Project) team, and within it, I was assigned to carry out the Real Time Voice Translator Project. The Automatic Call Center Project (ACC Project), now renamed to Voice2Web Project, is a project carried out by the Research and Development Center. The RDC is a department inside the Electro Technical Faculty of the CTU that carries out Research and Development projects regrding the Information Technologies (IT). Some of its partners are IBM, Vodafone and Ericson, who the RDC is doing projects for.  The ACC Project began on 2007 and its aim is to develop Voice Applications, within the IBM and RDC agreement, using IBM Voice Technologies and whatever open standards or open source software. IBM is an ACC Project partner and provides financing for it. It also provides hardware and software licenses to the ACC Project and gives us support. The members of the ACC Project are developing several Voice Applications at the same time, all them following the ACC Project purposes.   Although this document is focused on the Real Time Voice Translator Project, it will also explain in the introduction some aspects of the ACC Project. This is because the Real Time Voice Translator Project has a lot of points in common with it and it is worth, to understand it well, understand some points of the ACC Project as well. 

Read my Feed – from RSS to SIP

Abstract: RRS web feeds are read by a synthetic voice, trough a SIP VoIP call, offering eye-free access to huge amount of classified textual content available on the Web. DTMF browsing allows to choose in between different RSS providers, and to cycle trough RRS titles, until the desired full article is selected, and read. The main content in the page is located by explicit parsing for known feeds, or by heuristic reasoning for new feeds and pages, which can be directly accessed by passing their URI as a SIP address parameter. Scalability is attained by caching of network and audio data instances, and expressivity improved by generation of SSML markup on the basis of the original HTML and CSS code. The whole system is made of a collection of Open Source components and public W3C standards, and the use of Festival for Speech Synthesis makes the service available for any supported language

Analysis of NGN service creation technologies

Network Operators can see next Generation Networks (NGN) as new revenue stream, thanks to the potential they could have in increasing the service offering. Therefore it’s important to understand how proposed technologies and solutions in NGN market can enable, flexible and easy service creation 3. This paper presents the result of the investigation of Eurescom P1109 project [1] in the area of advanced technologies that enable the introduction of new services in NGNs [5]. These technologies are evaluated with respect to some key evaluation criteria and then a comparison is provided

Technologies and Guidelines for Service Creation in NGN

Network Operators can see Next Generation Networks (NGN) as new revenue stream, thanks to the potential they could have in increasing the service offering. Therefore it is important to understand how proposed technologies and solutions in NGN market can enable, flexible and easy service creation. This paper presents the result of the investigation of Eurescom P1109 project in the area of advanced technologies that enable the introduction of new services in NGNs. These technologies are evaluated with respect to some key evaluation criteria and then a comparison is provided

Next Generation Networks: the service offering standpoint

Next Generation Networks (NGN) is the application of Internet, IP and IT solutions to Telecom Services, including (but not only) the integration and sometimes the substitution of circuit switching with packet switching either for trunking or for access. In this paper, we will present the objectives and results of the Eurescom Project P1109 [1]. The overall goal is to support this view, in evaluating solutions for NGNs from a service-offering standpoint and understanding the wider effects of introducing NGNs in terms of the inter-operability and functionality of NGN products

Unified Messaging Systems: an Evolutionary Overview

Over the last decade, the widespread demand and use of the internet has changed the direction of the telecommunications industry as it was recognised that the internet could be used as an inexpensive way to handle not only data but also voice communications. This convergence of traditional voice and data technologies towards an IP-based open architecture has been paralleled by a convergence of the internet and mobile communications. As a result of these convergences, unified messaging has emerged as a technically viable service. Integrated messaging services that offer partial unification of different message types are already in the marketplace. This paper asks what unified messaging means and examines underlying architectural developments that are likely to shape the unified messaging applications of the future

Constructing a low-cost, open-source, VoiceXML

Voice-enabled applications, applications that interact with a user via an audio channel, are used extensively today. Their use is growing as speech related technologies improve, as speech is one of the most natural methods of interaction. They can provide customer support as IVRs, can be used as an assistive technology, or can become an aural interface to the Internet. Given that the telephone is used extensively throughout the globe, the number of potential users of voice-enabled applications is very high. VoiceXML is a popular, open, high-level, standard means of creating voice-enabled applications which was designed to bring the benefits of web based development to services. While VoiceXML is an ideal language for creating these applications, VoiceXML gateways, the hardware and software responsible for interpreting VoiceXML applications and interfacing with the PSTN, are still expensive and so there is a need for a low-cost gateway. Asterisk, and open-source, TDM/VoIP telephony platform, can be used as a low-cost PSTN interface. This thesis investigates adding a VoiceXML service to Asterisk, creating a low-cost VoiceXML prototype gateway which is able to render voice-enabled applications. Following the Component-Based Software Engineering (CBSE) paradigm, the VoiceXML gateway is divided into a set of components which are sourced from the open-source community, and integrated to create the gateway. The browser requires a VoiceXML interpreter (OpenVXI), a Text-To-Speech engine (Festival) and a speech recognition engine (Sphinx 4). The integration of the components results in a low-cost, open-source VoiceXML gateway. System tests show that the integration of the components was successful, and that the system can handle concurrent calls. A fully compliant version of the gateway can be used in the real world to render voice-enabled applications at a low cost.KMBT_363Adobe Acrobat 9.55 Paper Capture Plug-i

VOICE BASED FOR BANKING SYSTEM

The trouble with traditional banking system service resulted difficulties, latency and low quality of service, not suitable for disable people and require extra manpower to perform simple bank activities. The goal of this project is to build a voice recognition based system which specifies on the banking activities element and specializes in using voice as a medium to run bank activities via telephony network system. Three fundamental objectives were addressed in the study. First, to develop two-way interactive program of banking system, which use voice as importantmechanism to receive instruction and response to user. Second, it support to first objective which to develop such a user friendly andhighsecurity voice banking system which requires the user first logs on to the system by furnishing the assigned customer identification number and personal identification number before user proceed for further actions. And therefore, there must have a strong database structure development of the application in the voice banking system that purposely to maintain the integrity of the data stored and responds to authorized user only. For third objective, is to determine the best programming in order to implement in telephony network system. There is a study and architecture on how voice can be accepted, manipulated and generated by using combination two types of programming which are Cold Fusion and VoiceXML, which is goes to the third objective. The functions of this system is proved and demanded by user as it provides such convenience and easy services with just use voice to transmit the instruction. Hence, this strategy will grab large number of customers and simultaneously will generate huge profit too to the bank institution that applies this system. It is hoping that, by developing this system it will be a platform for next developer to host the system and can be use a large number of customers simultaneously and efficiently. Keyword: Voice based, telephony, combination of programming, architectur