Advanced concept of voice communication server on embedded platform

The paper deals with a design of an embedded Voice communication server which was developed within the scope of the BESIP project (Bright Embedded Solution for IP Telephony). The project brings a modular architecture with additional functionality such as a speech quality monitoring and a protection against security threats.The speech quality assessment is carried out in a simplified computational E-model and we implemented our proposal into the BESIP as an optional component. In the security module. We applied a standard approach to the intrusion detection and protection and in addition to the mentioned modules we come up with an idea of unified configuration based on the NETCONF protocol. We implemented ntegrated the complex support of NETCONF configuration protoco into OpenWRT and our modifications were accepted by OpenWRT community. The paper describes the inidvidual modules, their features and entire BESIP concept.Scopus892b23322

Advanced solution of SIP communication server with a new approach to management

The paper deals with the development of an advanced solution for a SIP communication server. Works were carried out within the scope of Bright Embedded Solution for IP Telephony (BESIP) project. The output of the project should be a modular architecture with additional functionality such as speech quality monitoring and security of IP telephony. We sought to unify the configuration of individual components based on the NETCONF protocol. In order to be able to implement the idea into OpenWrt, we had to integrate a complex support for the NETCONF configuration protocol. Our modifications of OpenWrt in respect of NETCONF were accepted by the OpenWrt community and have been included in OpenWrt/Trunk branch. The paper explains and describes the whole concept of the BESIP project and its individual modules.Web of Science59454954

The data extraction using distributed crawler inside multi-agent system

The paper discusses the use of web crawler technology. We created an application based on standard web crawler. Our application is determined for data extraction. Primarily, the application was designed to extract data using keywords from a social network Twitter. First, we created a standard crawler, which went through a predefined list of URLs and gradually download page content of each of the URLs. Page content was then parsed and important text and metadata were stored in a database. Recently, the application was modified in to the form of the multi-agent system. The system was developed in the C# language, which is used to create web applications and sites etc. Obtained data was evaluated graphically. The system was created within Indect project at the VSB-Technical University of Ostrava

A methodology for measuring voice quality using PESQ and interactive voice response in the GSM channel designed by OpenBTS

This article discusses a methodology for rating the quality of mobile calls. Majority telecommunications service from the perspective of the whole world is using mobile telephony networks. One of the problems affecting this service and its quality are landscape barriers, which prevent the spread signal. Price and complex construction of classic BTS does not allow their dense distribution. In such cases, one solution is to use OpenBTS technology. Design of OpenBTS is more available, so it can be applied to much more places and more complex points. Purpose of this measurement is a model for effective stations deployment, due to shape and distribution of local barriers that reduce signal power, and thus the quality of speech. GSM access point for our mobile terminals is OpenBTS USRP N210 station. The PESQ method for evaluating of speech quality is compared with the subjective evaluation, which provides Asterisk PBX with IVR call back. Measurement method was taken into account the call quality depending on terminal position. The measured results and its processing bring knowledge to use this technology for more complicated locations with degraded signal level and increases the quality of voice services in telecommunications

Interactive VoiceXML module into SIP-based warning distribution system

This article discusses the use of the Voice Extensible Markup Language (VoiceXML, VXML) to create a complex voice menu in danger alert communication system. The system was created as a part of research at Department of Telecommunications at the VSB - Technical University of Ostrava. Creating a voice menu provides end-users more information about the impending danger as well as instructions on how to behave in a given situation. If users receive a pre-recorded warning message in the form of a phone call, it will provide a telephone number on which they can obtain more information. In order to achieve the desired functionality, we had to use open-source PBX Asterisk, the VoiceGlue package which features both the VoiceXML interpreter and the Text-to-Speech (TTS) moduleScopus14934433

IP telephony based danger alert communication system and its implementation

The paper deals with development of a web application allowing to deliver pre-recorded voice messages by using SIP generator. The developed application is a part of complex system, which has been evolved in Dpt. of Telecommunications, Technical University of Ostrava for last three years. Our intent is focused on disaster management, the message, which should be delivered within specified time span, is typed in the application and text-to-speech module ensures its transormation to a speech format, after that a particular scenario or warned area is selected and a target group is automatically unloaded. For this purpose, we have defined XML format for delivery of phone numbers which are located in the target area and these numbers are obtained from mobile BTS's (Base transmission stations). The advantage of this aproach lies in controlled delivery and finally the list of unanswered calls is exported and these users are informed via SMS. In the paper, the core of the danger alert system is described including algorithms of the voice message delivery

Hráňová posuvná výztuž pro pluhové poruby ve středně mohutných slojích

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Aplikace plastových optických vláken v automobilovém průmyslu

Import 26/09/2007Prezenční454 - Katedra elektroniky a telekomunikační technik

Frame Relay Technology

Tato diplomová práce nastiňuje oblast datové paketové komunikace, přičemž je zaměřena na technologii Frame Relay z hlediska základních vlastností, principu a funkčnosti Frame Relay sítě a architektury Frame Relay protokolu. Úkolem této práce je navrhnout a realizovat Frame Relay síť v laboratorních podmínkách za pomocí směrovačů od firmy Cisco a implementovat do navrhnuté sítě nástroje pro řízení provozu, jejichž nastavení bude ověřeno pomocí generátoru provozu. Samotná Frame Relay síť je tvořena zařízeními nakonfigurovanými jako Frame Relay přepínače a směrovače. Jako nástroje pro řízení provozu byly implementovány pro jednotlivé virtuální okruhy garantovaná datová přenosová rychlost označovaná jako CIR, datové množství Bc s garantovaným přenosem Frame Relay spojení a signalizace stavů přetížení na rozhraních Frame Relay přepínačů.This thesis outlines the area of data packet communication with focus on Frame Relay technology in term of basic features, principle and functionality of Frame Relay network and architecture Frame Relay protocol. The task of this work is to design and realize the Frame Relay network in laboratory conditions using routers from Cisco company and implement instruments for traffic management into designed network, the settings will be verified through the traffic generator. The Frame Relay network consists of device configured as Frame Relay switches and routers. As a tool for traffic management were implemented for each virtual circuit guaranteed data transfer rate, known as CIR, Bc quantities of data with a guaranteed transfer Frame Relay connection and signalization status congestion on the Frame Relay switches.Prezenční454 - Katedra telekomunikační technikyvýborn