In this paper, the state-of-the-art laboratory environment is presented that is aimed for experimental verification of the earlier obtained analytical OFDM error floor model but now in concrete LTE FDD downlink channel conditions, which are for this purpose hardware and software simulated in the Communications Systems Laboratory of the University of Dubrovnik. At this point, the very preliminary verification of the earlier derived error floor formula is reported as the test results achieved by means of the industry-standard simulation tool closely match the ones coming out of the earlier model-specific basic Monte-Carlo simulations.

In this paper, we propose a model for estimating the error floor in a small-time-dispersion environment - typically indoor, where both channel and overall OFDM symbol are represented stochastically. The developed novel model for the error floor prediction involves modified common channel time dispersion parameters as well as the ones characterizing the OFDM signal. The validity of the model was confirmed by the results of the corresponding Monte-Carlo simulations.

This paper addresses the problem of locating a single source from noisy received signal-strength (RSS) measurements in wireless sensor networks (WSNs). To overcome the non-convexity of the maximum likelihood (ML) optimization problem, we provide an efficient convex relaxation that is based on the second order cone programming (SOCP), for both cases of known and unknown source transmit power, and we use a simple iterative procedure to solve the problem when the transmit power and the path loss exponent (PLE) are simultaneously unknown. Simulation results demonstrate that the new approach outperforms the existing ones in terms of the estimation accuracy, while in terms of the complexity, it represents a good balance when compared to the existing approaches.

A novel algorithm for transmit beamforming to single cochannel multicast group is presented in this paper. We consider the max-min fairness (MMF) based beamforming problem where the maximization of the smallest receiver signal-to-noise ratio (SNR) over the secondary users subject to constraints on the transmit power and interference caused to the primary users. It is shown that this problem, which is nonconvex NP-hard, can be approximated by a convex second-order cone programming (SOCP) problem. Then, an iterative algorithm which successively improves the SOCP approximation is presented. Simulation results show the superior performance of the proposed approach, together with a reduced computational complexity, as compared to the state-of-the-art approach.

This paper provides the results of practical performance benchmark analysis of the state-of-the-art procedures for restoration and protection of Ethernet traffic that is transported across SDH networks, namely: SNCP, STP/RSTP and LCAS, taking into account specific parameters from the point of view of both network operator and network service user, such as: switchover time, utilization of network resources, and finally, simplicity, costs and ownership (user or operator) of the implementation.

Networks protocols are implemented in combinations of software, firmware, and hardware on each end of a connection. Introduction of multiservice networks demands more intelligent control over network usage and more efficient application development practices that enable achieving Quality–of-Service (QoS) goals. As applications over the network increase, so does the need to diagnose performance at the application level, knowing how traffic patterns are affected in terms of guaranteed bandwidth, delay and reliability. In this chapter, the concept for testing multiservice networks in real-life situations during stationary time intervals, by using expert-system-based protocol analysis, is described. The architecture of hardware and software needed for data acquisition and testing, that, as a working example, was conducted on a major network with live traffic, is proposed, as well as the appropriate algorithm for estimating standard QoS parameters from the measured data that mainly included decoding with precise time-stamps and expert-system comments, resulting from the appropriate processing of the network data.

IP multimedia subsystem (IMS) is the base for convergence of mobile and fixed networks as it offers possibilities to create open infrastructure of services based on IP protocol, which will enable simpler development of a variety of multimedia services. In this paper, we present testing of a voice call performance over various wireless networks, especially UMTS, using Open IMS Core system, and analyze the functionality of selected scenarios for voice calls, comparing the so obtained results with the performance of the commercial IMS system that provides built-in enhanced QoS and other functionalities, as well. We also analyze how coding rate affects the quality of voice calls in our test systems, presenting the measurements’ results for relevant QoS parameters and stationary scenario of the observed system. The results show that the open source solution for the IMS system, while having the QoS functionalities and the server with lower performance than those of the commercial solution, still achieves excellent performance, especially for the narrowband codec, while for the wideband speech codec, the performance remains somewhat lower than the one obtained for the commercial solution. All observed solutions exhibited good results when used for speech transmission, except the GPRS and the EDGE network when used with the wideband codec.

In this paper, we propose the concept and the means for practical testing of TCP congestion window. With this respect, we present the architecture of the hardware and expert-system-based distributed protocol analysis that we used for practical data acquisition and testing, which we conducted on a major network with live traffic (Electronic Financial Transactions data transfer), as well as the appropriate algorithm for estimating the actual congestion window size from the measured data — mainly decoding with precise time-stamps (100 ns resolution locally and 1 μs with GPS clock distribution) and expert-system-processed results of the accordingly filtered data from the special-hardware-based capture buffer. We used statistical analysis for evaluation whether the data belonged to the specific (in this case assumed normal) cumulative distribution function, or whether two data sets exhibited the same statistical distribution — the condition-sine-qua-non for a stable interval. Finally, as the measured data-based congestion window values exhibited fitting, with satisfactory statistical significance, to the truncated normal distribution, we applied the appropriate model for estimation of the relevant parameters of the congestion window distribution: its mean value and variance.

For in-service assessing of voice call quality on a data network, measurement methods must have knowledge about network impairments such as delay, jitter and datagram loss. The E-model, finally defined by the ITU-T Recommendation G.107, was developed with such characteristics, providing a single quality-of-service rating value, derived from the relevant impairments, and directly mapped to classical mean-opinion-score values, still requested in most practical situations. In this paper, we propose a practical approach to in-service quality-of-service assessment of simulated VoIP calls on a corporate network that includes WAN (Frame Relay) links as well. We address typical issues with this respect, through investigating real capabilities of corporate WAN infrastructure for supporting VoIP services in such environment.

In this paper, a solution is proposed for testing TCP congestion window process in a real-life network situation during stationary time intervals. With this respect, the architecture of hardware and expert-system-based distributed protocol analysis is presented that we used for data acquisition and testing, conducted on a major network with live traffic (Electronic Financial Transactions data transfer), as well as the appropriate algorithm for estimating the actual congestion window size from the measured data that mainly included decoding with precise time-stamps (100ns resolution locally and 1ms with GPS clock distribution) and expert-system comments, resulting from the appropriate processing of the network data, accordingly filtered prior to arriving to the special-hardware-based capture buffer. In addition, the paper presents the statistical analysis model that we developed for the evaluation whether the data belonged to the specific (in this case, normal) cumulative distribution function, or whether two data sets exhibit the same statistical distribution - the conditio sine qua non for a TCP-stable interval. Having identified such stationary intervals, it was found that the measured-data-based congestion window values exhibited very good fitting (with satisfactory statistical significance) to the truncated normal distribution. Finally, an appropriate model was developed and applied, for estimating the relevant parameters of the congestion window distribution: its mean value and the variance.

This paper studies how end-to-end application peiformance(of Electronic Financial Transaction traffic, in particular)depends on the actual protocol stacks, operating systemsand network transmission rates. With this respect, the respectivesimulation tests of peiformance of TCP and UDP protocolsrunning on various operating systems, ranging from Windows,Sun Solmis, to Linux have been implemented, and thedifferences in peiformance addressed focusing on throughputand response time.

In this paper, we present an analytical model aimed to evaluate effects of very slow Nakagami/Ricean fading, shadowing and additive Gaussian noise on the mutual interdependence between successive errors in NCMFSK, MCPSK and MCQAM systems. We present the closed form solutions for transition probabilities and show that strong amplitude correlation within several successive signaling intervals, due to fast signal fluctuations, causes mutually dependent error occurrences in these intervals. We demonstrate how the transition probabilities are sensitive to both fading and shadowing severity. Apparently, this problem is even more significant in presence of deep fluctuation of shadowing. This finally points to importance of analyzing such error dependency in presence of a very slow fading, which actually represents the scenarios where correlation coefficient of amplitude fluctuations on successive signaling intervals is equal to 1. The results obtained in this paper can be useful in the design of burst error control in fading/shadowing channel.

In this paper, a novel analytical approach to developing the probability of outage in shadowed Nakagami-m and Ricean fading channel is presented, where instead of lognormal probability density function (pdf), gamma pdf is used to describe shadowing, providing mathematical framework for deriving performance metrics such as the probability of outage. The validity of the obtained analytical expressions was checked by comparing the results of the according calculations with the results of related Monte Carlo simulations, with very good matching.