Karel Tomala

Monitoring the quality of speech in the communication system BESIP

This paper deals with the monitoring and evaluation of speech quality between two endpoints which has been developed in the BESIP project (Bright Embedded Solution for IP Telephony). The BESIP is a simple and effective solution for SIP communication server enabling an easy integration into a computer network based on open-source solutions. This device serves to provide IP telephony for smaller workplaces. The main reason for the development and implementation of a module for monitoring into the BESIP system was the ability to evaluate speech quality. The principle of this module is based on an ability to assess an estimated speech quality at the users side without the necessity to compare original and degraded signals. The speech quality evaluation is performed in accordance with ITU-T G.107.

Embedded multiplatform SIP server solution

This paper describes BESIP (Bright Embedded Solution for IP Telephony) project. It was developed by LipTel team supported by CESNET. Primary goal was to implement multiplatform embedded SIP communication server with unified configuration interface. The paper explains and describes the whole concept and individual modules, acquaints with the current state and with the future intents. SIP server is based on OpenWRT project core and, there is Asterisk and Kamailio inside as SIP engines. Kamailio was selected for security and reliability, Asterisk for PBX functions. Next to this, there is Web frontend for managing device, Kamailio and Asterisk. Last, but very important part, is NETCONF server which is part of image for unified management.

Automatic analysis of attack data from distributed honeypot network

There are many ways of getting real data about malicious activity in a network. One of them relies on masquerading monitoring servers as a production one. These servers are called honeypots and data about attacks on them brings us valuable information about actual attacks and techniques used by hackers. The article describes distributed topology of honeypots, which was developed with a strong orientation on monitoring of IP telephony traffic. IP telephony servers can be easily exposed to various types of attacks, and without protection, this situation can lead to loss of money and other unpleasant consequences. Using a distributed topology with honeypots placed in different geological locations and networks provides more valuable and independent results. With automatic system of gathering information from all honeypots, it is possible to work with all information on one centralized point. Communication between honeypots and centralized data store use secure SSH tunnels and server communicates only with authorized honeypots. The centralized server also automatically analyses data from each honeypot. Results of this analysis and also other statistical data about malicious activity are simply accessible through a built-in web server. All statistical and analysis reports serve as information basis for an algorithm which classifies different types of used VoIP attacks. The web interface then brings a tool for quick comparison and evaluation of actual attacks in all monitored networks. The article describes both, the honeypots nodes in distributed architecture, which monitor suspicious activity, and also methods and algorithms used on the server side for analysis of gathered data.

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) module.

Interactive video audio system and its performance evaluation

The paper discusses the Interactive Audio Video System - IVAS which has been developed for INDECT project. The INDECT project aims at developing the tools for enhancing the security of citizens and protecting the confidentiality of recorded and stored information and is a part of the 7 FP EU. The presented paper is a result of applied research and is a contribution into the portal which integrates the interactive video and audio system (IVAS). In the paper, we describe the function of the IVAS system in the INDECT portal, the methodology of the performance tests for calls and registrations, as well as an application that was implemented for this purpose. The aim was to obtain data to determine how much is able to IVAS system in its current implementation to handle calls and how many smartphones can at one moment up to the register. This information is important to optimize the load and is a crucial factor in the design of computing resources for IVAS.

QoS of Triple Play Services in LTE Networks

This paper deals with studying the effects of performance LTE network throughput for data traffic. Utilisation rate of networks has a significant impact on the quality parameters of triple play services. LTE network was simulated by the software module additionally implemented in a development environment MATLAB. Throughput model has been obtained using this simulation that was used to test the QoS parameters for voice/video. Voice and video data streams using different codecs are transmitted for the obtained throughput. The measured qualitatively QoS parameters determine the resulting quality services from the perspective of end user perception.

IP TELEPHONY BASED DANGER ALERT COMMUNICATION SYSTEM AND ITS IMPLEMENTATION

This article discusses a danger alert system created as a part of the research project at Department of Telecommunications of Technical University of Ostrava. The aim of the system is to distribute pre-recorded voice messages in order to alert the called party in danger. This article describes individual technologies, which the application uses for its operation as well as issues relating to hardware requirements and transfer line bandwidth load. The article also describes new algorithms, which had to be developed in order to ensure the reliability of the system. 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 transformation 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 BTSs (Base transmission stations). The benefit of such communication compared to others, is the fact, that it uses a phone call and, therefore, it is possible to get feedback who accepted the message and to improve efficiency of alert system. Finally, the list of unanswered calls is exported and these users can be informed via SMS.

Security solution against denial of service attacks in BESIP system

This article deals about embedded SIP communication server with an easy integration into the computer network based on open source solutions and its effective defense against the most frequent attack in the present - Denial of Service. The article contains brief introduction into the Bright Embedded Solution for IP Telephony – BESIP and describes the most common types of DoS attacks, which are applied on SIP elements of the VoIP infrastructure including the results of defensive mechanism that has been designed.

QoS of multimedia services in 4G networks

This paper deals with quality of service (QoS) for triple play, mainly focused on voice and video. For studying changes in quality parameters at different network utilization, was it necessary to simulate LTE networks. For this purpose, the Matlab was chosen, in which we focused on the throughput. We simulated technology in particular scenarios, we changed the frequency and time division duplex, SNR, while we also simulated the behavior of different channels, such as EPA (Extended Pedestrian A Model), EVA (Extended Vehicular A model), ETU (Extended Typical Urban model) and HST (High Speed Train condition). By simulating different scenarios in Matlab, we obtained the required model of throughput and we could determine how it will affect the quality of triple-play services in LTE.

Interactive multimedia module into danger alert communication system

This paper discusses the transmitting video calls and the use of the Voice Extensible Markup Language (VoiceXML, VXML) to create a complex multimedia 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 multimedia 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, both audio and video. 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) module and FFmpeg for streaming video content.