Jan Rozhon

Prediction model of QoS for Triple play services

The aim of this work is to design an application utilizing the QoS prediction model for Triple play services, the input parameters of which are used QoS policies and network state. In order to analyze the efficiency of individual policies designed for serving the packet queues, it was necessary to perform practical measurements in an overloaded network and monitor the total network delay, Jitter and packet loss. Results of the measurements after the conformance check were used for purposes of creating QoS prediction models for each type of service (voice, video and data).

Development of a speech quality monitoring tool based on ITU-T P.862

This paper deals with new proposal of a speech quality monitoring tool based on the perceptual evaluation of speech quality in accordance with ITU-T P.862 and its application in a research project investigating an impact of meteorological conditions on the speech quality in a GSM network. Every time we make a call using the GSM network the speech quality differs. This is mainly caused by the numerous factors influencing the wireless communication between the users cell phone and the GSM base station such as weather, users current speed or location. The latter two aspects have been thoroughly measured and discussed, therefore almost anyone can determine with ease what the attenuation of the signal is for example in the building or how the signal is distorted due to the switching between base stations caused by the users movement. On the other hand the correlation between speech quality in the GSM network and the current weather situation has not yet been measured nor estimated. This paper tries to deal with this gap and presents one of the possible solutions to perform speech quality measurement in the GSM network using open source Asterisk PBX.

SIP Back to Back User Benchmarking

In the branch of performance testing of SIP based infrastructure no standardized methodology has been adopted yet. This is why we decided to develop a methodology suitable for testing and evaluating SIP server’s performance. This paper should open a discussion about defining the parameters and methodology for benchmarking SIP based infrastructure, because it is the keystone in modern NGN voice communications. The methodology presented in this paper focuses on one particular variant of the SIP server – the Back to Back User Agent (B2BUA), as it is one of the most commonly used solution in the VoIP. The method is enhanced by the definition of the performance rating factor, which allows for comparing the performance of the different B2BUAs platform independently. This factor utilizes one of the features of the B2BUA – the fact that media during the call are passing through the SIP server. The whole method is described in this paper and confirmed experimentally as well.

Impact of packet size variation in overlay quantum key distribution network

Quantum Key Distribution (QKD), based on the laws of physics rather than the computational complexity of mathematical problems, provides an information-theoretically secure way of establishing symmetrical binary keys between two geographically distant users. This paper is oriented to the practical realization of QKD public channels which are usually implemented as overlay point-to-point connections. We address the problem of minimizing the key material consumption by changing packet overhead. Our results show that the efficiency of communication in overlay QKD networks may increase when packets of larger sizes are used. However, this tuning directly affects the performance of overall communication. We evaluated this approach using an overlay network module which was implemented in the NS-3 simulator. The obtained results can be used for other overlay networks as well.

SIP Registration Burst Load Test

The paper deals with performance testing a SIP infrastructure. While the working methodology for measuring performance and effectiveness of the SIP B2BUA and SIP Proxy has recently been introduced, the more complex application of this methodology is still missing. This paper tries to fill this gap and improve the methodology further to better fit into the planned modular design of the testing platform, which is being designed in the VoIP laboratory of VSB – Technical University of Ostrava. By separating registrations from calls, we were able to measure both cases without the need of extensive postprocessing of data to ensure the data in one case is not affected by the ones from the other case. Furthermore the security vulnerability of the SIP protocol has been harnessed to allow measuring software for performing both registrations and calls together but in individual processes, which builds the basis for planned and already mentioned modular design of the platform. In this paper the results from separate registration stress test analysis will be presented as one of the examples of usage of the mentioned methodology.

Network performance QoS estimation

This paper deals with network performance simulation and quality of Triple-play services in IP networks. Based on our designed analytical model, the application has been created by means of which it is possible to predict qualitative parameters for QoS in IP networks, with regard to policies applied for packet processing by routers and with regard to the level of total network utilization. The contribution of this paper lies in designing a new model capable of predicting the quality of Triple-play services respecting queuing policy and transmission speed of interfaces on routers in packet networks.

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.

Predictive model for determining the quality of a call

In this paper the predictive model for speech quality estimation is described. This model allows its user to gain the information about the speech quality in VoIP networks without the need of performing the actual call and the consecutive time consuming sound file evaluation. This rapidly increases usability of the speech quality measurement especially in high load networks, where the actual processing of all calls is rendered difficult or even impossible. This model can reach its results that are highly conformant with the PESQ algorithm only based on the network state parameters that are easily obtainable by the commonly used software tools. Experiments were carried out to investigate whether different languages (English, Czech) have an effect on perceived voice quality for the same network conditions and the language factor was incorporated directly into the model.

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.