Filip Rezac

Analysis of the Public Channel of Quantum Key Distribution Link

Quantum key distribution (QKD) relies on the laws of physics to establish a symmetric binary key between remote parties. A QKD link involves the realization of a quantum channel for the transmission of quantum key material encoded in certain photon properties, as well as a public channel for verification of the exchanged key material. This paper deals with the mutual dependence of these channels and analyzes the impact of performance of both channels on the overall key material establishment process. This paper presents measurement data obtained under laboratory conditions as well as the results obtained by establishing a virtual QKD link. Despite the common beliefs that increased quantum bit error rate implies a larger amount of traffic on the public channel, our measurements prove the opposite. The obtained data clearly show that the public channel has a major impact on the overall performance of the QKD link.

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.

Malicious traffic monitoring and its evaluation in VoIP infrastructure

The paper deals with the need to enhance security of the VoIP infrastructure. There are several ways to achieve an enhancement in security. We opted for the honeypot which can provide us information about attackers behaviour. We will examine in particular a VoIP honeypot referred to as Artemisa. It is one of many existing honeypots tailored for IP telephony. The paper describes its function and application within a real IP telephony infrastructure. The aim of another tested honeypot is to gather data about the attacks while simulating a SSH server. The gathered information could be crucial for further improvements of the existing security mechanism in our VoIP network.

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.

Monitoring of Speech Quality in Full-Mesh Networks

The quality of speech in network environment is a growing concern of all the companies and institutions, that migrated or are planning to migrate their voice communications to IP based technologies. This trend together with the increasing amount of Internet traffic in general poses a great challenge for network administration to be able to configure and manage the network so it can carry the multimedia traffic without the excessive delays, which degrade the quality of speech as it is perceived by the end users. Continuous monitoring of key nodes of internal infrastructure as well as external interconnections is the possible way to increase the quality of multimedia transmissions for most users, because it allows the network administrators to be informed as soon as the problem rises and consequently to change the network routing and queuing policies to bypass the connection experiencing the quality issues. The tool for this type of monitoring and its architecture are described in detail in this paper.

Modular Test System for Determination of Protection against Attacks in VoIP Networks

A VoIP (Voice over Internet Protocol) soft switch is essential for every IP telephony infrastructure. Frequently, it provides other services besides those related to VoIP. Very often, these switches are the target of external attackers. This article describes the system designed to test whether the targeted VoIP soft switch is sufficiently secured and protected against possible real threats. The system tests the switches for resistance to the most prevailing attacks. Based on whether the penetration is successful or not, the system assesses their overall security. The article describes applications and algorithms used by the system. The conclusion includes recommendations and principles to ensure efficient protection of a VoIP soft switch against existing threats. The system is designed as an open-source web application, allowing for access irrespective of the operation system platform, and can be extended with additional testing modules.

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.

VoIP call quality assesment based on RPROP neural networks

The modelling of the network effects on the quality of speech in the Voice over IP networks is the main focus of this paper. The main purpose of the ideas presented here is to achieve high-precision estimation of the speech quality in the environments where the classical approaches of speech quality determination fail. To achieve this high precision a modular neural network model is used to map the effects of a packet loss on the speech quality based on the PESQ reference. To incorporate the temporal effects E-model is partially utilized as well. This way a universal tool capable of harnessing the information about the speech quality for stress testing and monitoring of the local infrastructure has been developed enabling the telephony infrastructure administrators to evaluate the performance and stability of the systems in their hands. Moreover, a high-performance simulation environment has been developed as well to ensure sufficient amount of measurement data for the statistical analysis.

Utilizing the Neural Networks for Speech Quality Estimation Based on the Network Characteristics

The paper deals with an issue of the speech quality estimation in Voice over IP technology under packet loss. Packet loss is a major problem for real-time Internet applications, 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. The general and RFC3611-compliant solution, which allows for quick and precise speech quality estimation without the need to analyze or model the voice signal carried by the RTP (Real-time Transport Protocol) packets, is the contribution of this paper. The proposed solution is tested on G.711 A-law and further generalizes the already presented concepts of the speech quality estimation in the IP environment. The proposed approach of speech quality assessment belongs to non-intrusive methods and is based on the back-propagation neural networks.

Modelling speech and video quality in networks based on internet protocol

A development of the tool meant to model and estimate the qualitative parameters of the speech and video transmission is the main aim of this paper. The influence of queuing techniques of individual network elements on the quality of service is explored and a model to predict this influence is created based on this information. By using this model the network administrators can successfully plan the network infrastructure with the emphasis on the communication quality, which is so important these days. The results of the model are presented using the standardised scales of mean opinion score and peak signal-to-noise ratio, which makes the results easy to understand. To complete the picture the data transmissions are explored as well making the presented tool a universal mean to predict quality of service parameters for all kinds of services.