Jan Rozhon

VSB - Technical University of Ostrava · Department of Telecommunications

Every attempt to access to the Internet through a web browser, email sent, VPN connection, VoIP call, instant message or other use of telecommunications systems involves cryptographic techniques. The most commonly applied technique is asymmetric cryptography, which is generally executed in the background without the user even being aware. It establ...

This article deals with quality of service (QoS) in internet protocol (IP) telephony by applying software-defined networking (SDN) tools. The authors develop a new design that deterministically classifies real-time protocol (RTP) streams based on data found in session initiation protocol (SIP) using SIP proxy as a mediator, and the concept making t...

A QKD network can be considered an add-on technology to a standard communication network that provides IT-secure cryptographic keys as a service. As a result, security challenges resulting in the suspension of functional work must be addressed. This study analyzes a Denial of Service (DoS) attack on the Key Management System (KMS), one of the criti...

This paper aims to describe and evaluate the proposed calibration model based on a neural network for post-processing of two essential meteorological parameters, namely near-surface air temperature (2 m) and 24 h accumulated precipitation. The main idea behind this work is to improve short-term (up to 3 days) forecasts delivered by a global numeric...

Recent trends show a growing interest in VoIP services and indicate that guaranteeing security in VoIP services and preventing hacker communities from attacking telecommunication solutions is a challenging task. Spam over Internet Telephony (SPIT) is a type of attack which is a significant detriment to the user’s experience. A number of techniques...

With the increased number of Software-Defined Networking (SDN) installations, the data centers of large service providers are becoming more and more agile in terms of network performance efficiency and flexibility. While SDN is an active and obvious trend in a modern data center design, the implications and possibilities it carries for effective an...

The technologies of the Internet of Things (IoT) have an increasing influence on our daily lives. The expansion of the IoT is associated with the growing number of IoT devices that are connected to the Internet. As the number of connected devices grows, the demand for speed and data volume is also greater. While most IoT network technologies use cl...

In recent years, noticeable progress has been made in the development of quantum equipment, reflected through the number of successful demonstrations of Quantum Key Distribution (QKD) technology. Although they showcase the great achievements of QKD, many practical difficulties still need to be resolved. Inspired by the significant similarity betwee...

This paper proposes a convolutional neural network (CNN) based encoder model to compress and code speech signal directly from raw input speech. Although the model can synthesize wideband speech by implicit bandwidth extension, narrowband is preferred for IP telephony and telecommunications purposes. The model takes time domain speech samples as inp...

In recent years, noticeable progress has been made in the development of quantum equipment, reflected through the number of successful demonstrations of Quantum Key Distribution (QKD) technology. Although they showcase the great achievements of QKD, many practical difficulties still need to be resolved. Inspired by the significant similarity betwee...

This paper describes different approaches for the face authentication from the features and classification abilities point of view. Authors compare two types of features - Histogram of Oriented Gradients (HOG) and Local Binary Patterns (LBP) including their combination. These parameters are classified using Multilayer Neural Network (MLNN) and Supp...

The call quality is an important issue of the modern voice over IP (VoIP) networks. This is given by their inherent “best effort” nature and the fact that any packet can be delayed or lost along the transmission path. To ensure sufficient call quality from the customer’s point of view a multitude of approaches can be adopted, i.e. dynamic change of...

The paper presents a system for monitoring and assessment the speech quality in the IP telephony infrastructures using modular probes. The probes are placed at key nodes in the network where aggregating packet loss data. The system dynamically measures speech quality and results are collected on a central server. For data analysis we applied four-s...

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 usuall...

This article discusses the impact of multilayer neural network parameters for speaker identification. The main task of speaker identification is to find a specific person in the known set of speakers. It means that the voice of an unknown speaker (wanted person) belongs to a group of reference speakers from the voice database. One of the requests w...

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 t...

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 paper deals with modelling network effects on the quality of speech. The packet loss modelling is based on the fourstate markov chain, afterwards, the resilient back-propagation (Rprop) algorithm is applied to train a neural network. The proposed solution allows for quick and precise speech quality estimation without the need to analyze the voi...

In this paper, we present a software tool capable of predicting the final quality of triple play services by using the most common assessment metrics. The quality of speech and video in network environment is a growing concern of all the internet service providers to carry the multimedia traffic without the excessive delays and losses, which degrad...

The paper presents a system for monitoring the speech quality in the IP telephony infrastructures using modular probes that are placed at key nodes in the network. The system is developed mainly within research of the Czech educational and scientific network, but the tool can generally be used in any IP telephone traffic. The system is capable of u...

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 configur...

The paper deals with presentation of the IVAS system within the 7FP EU 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. It is a part of the Seventh Framework Programme of European Union. We participate in INDECT portal and th...

The paper deals with performance measurement and assessment of SIP based network. The complex methodology for SIP benchmarking and performance testing based on IETF draft versions has been developed by our team so that the results coming from the measurement specified in this methodology could be compared platform independently. Our earliest implem...

This paper deals with a benchmarking of SIP infrastructure and improves the methodology of SIP performance evaluation further to better fit into the design of the SIP testing platform, which is being designed in the VSB - Technical University of Ostrava. By separating registrations from calls, we were able to measure both cases without the need of...

The paper deals with a speech quality monitoring tool which we have developed in accordance with PESQ (Perceptual Evaluation of Speech Quality) and is automatically running and calculating the MOS (Mean Opinion Score). Results are stored into database and used in a research project investigating how meteorological conditions influence the speech qu...

This article deals with the impact of virtualization techniques on interactive delay-sensitive applications running in real-time, particularly IP telephony. Many institutions, organizations and home users often adopt the virtualized solutions for their safety, ease of administration and backup. Virtualization, which was chiefly the prerogative of c...

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 q...

The paper deals with an end-to-end delay model in IP network and the proposed model is focused on time sensitive applications based on Real Time Protocol, especially IP telephony. In order to predict overall delay of datagrams in IP network we need to express a contribution of all individual components influencing the delivery of signal on the tran...

In the field of performance metrics and measurements of SIP (Session Initiation Protocol) Proxy and B2BUA (Back-to-Back User Agent) no standardized methodology has been presented yet. This gap results in a problematic determination of a hardware, the performance of which would be cost-effective and sufficient for the running the SIP Server in a giv...

The paper deals with SW PBX Asterisk and with features supporting video calls but our primary object is not switching video calls but their automatic recording and replaying. Asterisk is open source telephone exchange which brings capability for audio and video calls but does not allow recording and playback of video files. We have implemented thes...

This paper deals with a testing method suitable for SIP infrastructure. The performance testing is an issue of research and no standardized methodology has been adopted yet. We present the main ideas of the methodology that allows for testing the keystone of SIP based infrastructure – the SIP Server – in both SIP Proxy and B2BUA (Back to Back User...

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,...

This paper deals with a testing method suitable for SIP infrastructure. The performance testing is an issue of research and no standardized methodology has been adopted yet. We present the main ideas of our method that have been verified on open source software PBX Asterisk, as a representative of the well-known SIP solution. To be able to compare...

This paper deals with techniques of measuring and assessment of the voice transmitted in IP networks and describes design of quality measurement, which can be used for Cisco Gateways. Cisco gateways send Calculated Planning Impairment Factor in every CDR (Call Detail Record). Our design is based on collection of CDR's, their storing into SQL d...

This paper deals with techniques of measuring and assessment of the voice transmitted in IP networks and describes design of quality measurement, which can be used for Cisco Gateways. Cisco gateways send Calculated Planning Impairment Factor in every CDR (Call Detail Record). Our design is based on collection of CDR's, their storing into SQL d...

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 th...