Miroslav Voznak

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.

Effective packet loss estimation on VoIP jitter buffer

The paper deals with an influence of network jitter on effective packet loss in dejitter buffer. We analyze behavior of jitter buffers with and without packet reordering capability and quantify the additional packet loss caused by packets dropped in buffer on top of the measured network packet loss. We propose substitution of packet loss parameterP pl in ITU-T E-Model by effective packet loss P plef incorporating network jitter, a jitter buffer size and a packet size as additional input parameters for E-Model.

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.

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.

Network Performance QoS Prediction

This paper deals with QoS prediction of triple play services in IP networks. Based on our proposed model, speech or video quality can be calculated with regard to policies applied for packet processing by routers and to the level of total network utilization. This new simulating model was implemented in SW tool which enables networkers to predict objective QoS parameters of triple play services and to help them in network design. The contribution of this paper lies in designing a new model capable of predicting the quality of Triple-play services in networks based on IP.

Imperfect channel state information of AF and DF energy harvesting cooperative networks

Wireless information and powered transfer networks (WIPT) has recently been implemented in 5th generation wireless networks. In this paper, we consider half-duplex relaying system in which the energy constrained relay node collects energy via radio frequency (RF) signals from the surrounding resources. Regarding energy harvesting protocol, we propose power time switching-based relaying (PTSR) architecture for both amplify-and-forward (AF) and decode-and-forward (DF). Especially, we reveal the analytical expressions of achievable throughput, ergodic capacity and energy-efficient in case of imperfect channel state information (CSI) for both AF and DF network. Through numerical analysis, we analyse the throughput performance, energy-efficient and ergodic capacity for different parameters, including power splitting ratio and energy harvesting time. Moreover, we also depict the performance comparison between AF and DF network with perfect and imperfect CSI. The results in numerical analysis reveal that the result of AF relaying network is less significant than DF relaying network in the various scenarios.

Pattern Recognition Methods and Features Selection for Speech Emotion Recognition System.

The impact of the classification method and features selection for the speech emotion recognition accuracy is discussed in this paper. Selecting the correct parameters in combination with the classifier is an important part of reducing the complexity of system computing. This step is necessary especially for systems that will be deployed in real-time applications. The reason for the development and improvement of speech emotion recognition systems is wide usability in nowadays automatic voice controlled systems. Berlin database of emotional recordings was used in this experiment. Classification accuracy of artificial neural networks, k-nearest neighbours, and Gaussian mixture model is measured considering the selection of prosodic, spectral, and voice quality features. The purpose was to find an optimal combination of methods and group of features for stress detection in human speech. The research contribution lies in the design of the speech emotion recognition system due to its accuracy and efficiency.

Mobility prediction in wireless cellular networks for the optimization of Call Admission Control schemes

In last years there have been a growing interest for cellular networking, due to the availability of different mobile applications and the explosion of mobile devices utilization. If managed in an adequate way, cellular systems with mobile hosts can offer excellent performance and satisfactory QoS levels. In order to avoid service degradations, it is very important to decide whether a new connection can be accepted into the system, with the main aim of maximizing bandwidth utilization while avoiding quality degradations, with more emphasis for non-tolerant applications. Our proposed idea shows how a statistical approach can enhance system performance, without considering a particular prediction scheme (based on Markov theory, neural networks, data mining, Holt-Winters or similars): the proposal has been integrated with a threshold-based statistical bandwidth multiplexing scheme in order to propose the In-Advance Multiplexing Call Admission Control (IAM-CAC) scheme for cellular networks. Simulation campaigns have shown that the performance of the proposed idea in terms of admitted flows, and bandwidth utilization are really acceptable.

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.