Estimation and Control across Analog Erasure Channels

The automatic assessment of perceived quality of VoIP communications requires a subtle characterization of packet loss processes. This is more critical when delivery channels offer a time-varying performance, such as mobile and wireless “best effort” IP channels. Typically, the characterization of packet loss processes is done using theoretical models calibrated at runtime. At the end of a monitoring interval, a set of metrics is extracted and used by parameter-based quality models to estimate degradation caused losses. We design a new packet loss model that explicitly differentiates loss instances as a function of their perceptual effects. Moreover, our packet loss model considers the context where packet loss instances happen inside a given loss pattern. Furthermore, it enables simultaneously capturing and characterizing in a precise way long- and short-term behavior of packet loss processes. The characterization metrics extracted from the model can be used to build reliable parameter-based quality assessment models.

Technical reports indicate that wireless and mobile devices will account for 71% of all IP traffic by 2022, an increase of 19% over four years. This increase is related to advances in wireless data communication technologies. Wireless networks have become one of the most important ways to connect devices to the Internet, therein improving productivity and encouraging information sharing. IEEE 802.11, known as Wi-Fi, has become the main standard for wireless local area networks. The most important metrics for measuring the quality of Wi-Fi are delay, jitter, and packet loss. Packet loss occurs when one or more packets fail to reach their destination and can occur for a variety of reasons. Packet loss influences the user's perceived quality of applications over Wi-Fi networks, mainly multimedia and real-time applications. The availability of accurate models for packet loss in Wi-Fi networks enables the development of more efficient methods for performance analysis and network design, as well as better computational simulations. Modeling packet loss in such networks presents a major challenge because packets may be lost for many different reasons, including signal attenuation, noise, multipathing, signal refraction, thermal noise, competition for media access and buffer issues. In this paper, we provide an overview of the causes of packet loss and a comprehensive survey of the available models for packet loss in Wi-Fi networks. The potential benefits of the survey are: (i) the systematic presentation of available packet loss models for Wi-Fi networks, their parameters, and respective packet loss rate evaluation, (ii) comparison of models considering validation scenarios and input parameters, and (iii) description of open issues and future research directions. We hope that our analysis will help researchers understand the most important characteristics of the packet loss process in Wi-Fi networks and the strengths and weaknesses of the main packet loss models.

The streaming of real-time audio/video data is very challenging because of the time-varying and unreliable wireless channels, video content characteristics, limited bandwidth, dynamic topology, heterogeneous and distributed environment and high packet loss rate because of wireless interference and channel fading. Because of such open issues present in wireless networks, it is very difficult to satisfy the requirements of audio/video streaming applications such as low delay, low packet loss, jitter control etc. In such scenarios of opportunistic networks where real time data is being streamed from server node to client node, Network Coding and its variants can be used by such nodes to meet different requirements. Network coding changes the role of such multimedia nodes from store and forward to encode data packets. Encoding process includes mathematical operations on data packets. This survey paper has considered number of speculative and practical approaches and cases where network coding or its variant applied either fully or partially on multimedia traffic with the aim to improve performance and to provide protection against packet losses. This review has mainly focused on the opportunity of performance enhancement of MPEG-4 traffic over Wireless Network using Random Linear Network Coding (RLNC) with Multi Generation Mixing (MGM). Using Multi Generation Mixing, packets of greater importance has got more protection, less loss, more reconstruction and recovery of real time data.

With the development of Internet computing techniques, continuous data streams from remote sites are commonly used in scientific and commercial applications. Correspondingly, there is increasing demand of assuring the integrity and authenticity of received data streams. Existing strategies of assuring data integrity and authenticity mainly use message authentication codes (MAC) generated on data blocks and transfer the MAC to the receiver for authentication through either out of band communication or in band communication. Transferring the MAC via out of band communication inevitably introduces communication overhead and additional complexity to synchronize the out of band communication with the data communication. Transferring the MAC via in band channel can be achieved by either appending the MAC to the original data or embedding the MAC into the original data, which would either incur communication overhead or change the original data. It would be desirable to be able to authenticate the stream data without any communication overhead and changing the original data at the same time. To deal with data packet or block loss, many of existing stream data authentication schemes rely on hash chaining, the current usage of which results in uncertainty in authenticating the subsequent data blocks once the first data packet or block loss is detected. In this paper, we propose a novel application layer authentication strategy called DaTA. This authentication scheme requires no change to the original data and causes no additional communication overhead. In addition, it can continue authenticating the rest of data stream even if some data loss has been detected. Our analysis shows that our authentication scheme is robust against packet loss and network jitter. We have implemented a prototype system to evaluate its performance. Our empirical results show that our proposed scheme is efficient and practical under various network conditions

Discrete time steps in digital computer analysis of aquifers containing pumped wells

PurposeThe purpose of this paper is to present some numerical methods based on different time stepping and space discretization methods for the Allen‐Cahn equation with non‐periodic boundary conditions.Design/methodology/approachIn space the equation is discretized by the Chebyshev spectral method, while in time the exponential time differencing fourth‐order Runge‐Kutta (ETDRK4) and implicit‐explicit scheme are used. Also, for comparison the finite difference scheme in both space and time is used.FindingsIt is found that the use of implicit‐explicit scheme allows use of a large time‐step, since an explicit method has less order of accuracy as compared to implicit‐explicit method. In time‐stepping the proposed ETDRK4 does not behave well for this special kind of partial differential equation.Originality/valueThe paper presents some numerical methods based on different time stepping and space discretization methods for the Allen‐Cahn equation with non‐periodic boundary conditions.

The advent of network coding promises to change many aspects of networking. Network coding moves away from the classical approach of networking, which treats networks as akin to physical transportation systems. We overview some of the main features of network coding that are most relevant to wireless networks. In particular, we discuss the fact that random distributed network coding is asymptotically optimal for wireless networks with and without packet erasures. These results are extremely general and allow packet loss correlation, such as may occur in fading wireless channels. The coded network lends itself, for multicast connections, to a cost optimization which not only outperforms traditional routing tree-based approaches, but also lends itself to a distributed implementation and to a dynamic implementation when changing conditions, such as mobility, arise. We illustrate the performance of such optimization methods for energy efficiency in wireless networks and propose some new directions for research in the area.

This paper provides heuristic methods for obtaining a burning number, which is a graph parameter measuring the speed of the spread of alarm, information, or contagion. For discrete time steps, the heuristics determine which nodes (centers, hubs, vertices, users) should be alarmed (in other words, burned) and in which order, when afterwards each alarmed node alarms its neighbors in the network at the next time step. The goal is to minimize the number of discrete time steps (i.e., time) it takes for the alarm to reach the entire network, so that all the nodes in the networks are alarmed. The burning number is the minimum number of time steps (i.e., number of centers in a time sequence alarmed “from outside”) the process must take. Since the problem is NP complete, its solution for larger networks or graphs has to use heuristics. The heuristics proposed here were tested on a wide range of networks. The complexity of the heuristics ranges in correspondence to the quality of their solution, but all the proposed methods provided a significantly better solution than the competing heuristic.

Rapid growth and increasing requirements for service quality, reliability, and efficiency have made traffic evaluation an essential consideration in design and operation of data transmission networks. In this paper we analyse performance measures of data transmission network servicing batch data flows. The article provides analysis of the one channel queuing system transmitting data packet flow general distributed. In this article we introduce queuing system with finite and infinite buffer capacity. In this article we analyse such data packets transmission performance param- eters: average time spent by data packet in system, data packet delay variation, data packet loss. The data packet arrival rate and channel data packet transmission rate are independent of system state. A key mechanism of the proposed traffic model is that packet time spent in the system depends on channel utilization level, packet flow type and size of transmit- ted data. The proposed simulation model can be used when packet interarrival time follows such distributions: Pareto, Poisson, uniform, normal distribution. Simulation results of performance measures are defined by mathematical mod- els. Considering simulation results recommendations were proposed to improve performance measures for batch data flow transmission over data networks.

In recent years, wavelet transforms have been successfully applied to image and video compression. In a distributed, multimedia computing environment, compressed video and image files need to be transmitted to users for real-time replay. Because the network is subject to packet losses, the image packetization scheme should take packet losses into account in their design. In this paper, we present an error concealment algorithm for wavelet-based compressed images over packet-switched networks. Two different packetization schemes, called the intrablock-oriented and interblock-oriented schemes, in conjunction with wavelet coding, are presented. In our approach, adjacent wavelet coefficients are scattered into different packets, so that any packet contains encoded wavelet coefficients at different resolution layers. When a packet is lost due to network congestion or transmission errors, a lost wavelet coefficient is estimated by the receiver through a linear interpolation of its adjacent ones. Our approach is evaluated under two different packet loss models with various packet loss probabilities through simulations driven by actual video sequences.© (1996) COPYRIGHT SPIE--The International Society for Optical Engineering. Downloading of the abstract is permitted for personal use only.

Errors introduced by a wireless medium are more frequent and profound than contemporary wired media. Some of these errors, which are not corrected by the physical layer, result in Medium Access Control (MAC) layer bit errors and packet losses. Design of wireless protocols and applications can benefit substantially from a thorough understanding of these MAC layer impairments. This paper evaluates and proposes Markov-based stochastic chains to model the 802.11b MAC-to-MAC channel behavior for both bit errors and packet losses. We introduce an Entropy Normalized Kullback-Leibler measure to evaluate the performance of existing and new bit error and packet loss models. Based on the proposed measure, and contrary to recent results for mobile networks, we demonstrate that the traditional two-state Markov chain provides a very suitable model for the 802.11b MAC-to-MAC packet loss process. However, this simple model is not adequate for bit errors observed at the MAC layer of wireless local area networks. Consequently, we evaluate three other Markov-based chains for modeling these errors: full-state, hidden, and hierarchical Markov chains. Among these chains, we illustrate that the full-state Markov bit error model, evaluated under a wide range of order values, renders the best performance.

Discrete- and continuous-time approaches are frequently used to model the role of heterogeneity on dynamical interacting agents on the top of complex networks. While, on the one hand, one does not expect drastic differences between these approaches, and the choice is usually based on one's expertise or methodological convenience, on the other hand, a detailed analysis of the differences is necessary to guide the proper choice of one or another approach. We tackle this problem by investigating both discrete- and continuous-time mean-field theories for the susceptible-infected-susceptible (SIS) epidemic model on random networks with power-law degree distributions. We compare the discrete epidemic link equations(ELE) and continuous pair quenched mean-field (PQMF) theories with the corresponding stochastic simulations, both theories that reckon pairwise interactions explicitly. We show that ELE converges to the PQMF theory when the time step goes to zero. We performed an epidemic localization analysis considering the inverse participation ratio (IPR). Both theories present the same localization dependence on network degree exponent γ: for γ<5/2 the epidemics are localized on the maximum k-core of networks with a vanishing IPR in the infinite-size limit while, for γ>5/2, the localization happens on hubs that do not form a densely connected set and leads to a finite value of the IPR. However, the IPR and epidemic threshold of ELE depend on the time-step discretization such that a larger time step leads to more localized epidemics. A remarkable difference between discrete- and continuous-time approaches is revealed in the epidemic prevalence near the epidemic threshold, in which the discrete-time stochastic simulations indicate a mean-field critical exponent θ=1 instead of the value θ=1/(3-γ) obtained rigorously and verified numerically for the continuous-time SIS on the same networks.

Information technology is developing rapidly, especially in the field of computer science, telecommunications and information systems. In the early days of telecommunications, voice networks could not be combined with data networks. However, more recently it became possible to integrate data, audio and video services onto a single network, known as a multimedia network. The capabilities of the Internet are also rapidly increasing, and now, in addition to data packets (email, web browsing) the Internet transfers audio and video packets. Voice over Internet Protocol (VoIP) is a multimedia service. This technology can be used to transmit audio, video, and data packets. Consequently, it is becoming the most preferred communications technology. VoIP will eventually replace the use of traditional telephony – Public Switched Telephone Networks (PSTN), although the switchover process is challenging. A chief concern it that, while PSTN transmits audio packets over a closed network, VoIP sends audio packets using an open network – the Internet. In PSTN, eavesdroppers must have direct access to physical network to obtain information from communications. Whereas with VoIP, eavesdroppers can monitor data packets from they are connected to the Internet. Hence, security and privacy are important considerations when switching to VoIP systems. One solution is anonymous systems, which can be used to implement security and privacy protocols. However, this typically reduces the Quality of Service (QoS) of the VoIP. This empirical research study examines VoIP performance in an anonymous network – The Onion Routing (Tor). Two scenarios are implemented using the real Tor network to investigate three QoS metrics for VoIP: latency, jitter and packet loss. As recommended, latency in VoIP should be less than 400 ms, jitter should be less than 50 ms and packet loss should not be more than 5%. In addition, the research calculates the probability of attackers in the two scenarios implemented and evaluate Tor network forecasting. Experiments were conducted in reference to two scenarios. The first scenario is VoIP calls routed through a Tor network with three Tor relays (default Tor). And the second scenario is VoIP calls routed through a Tor network with two Tor relays. Experiments were performed in three periods; December 2012, July 2013, and October 2013. During each experimental time period, a hundred calls were captured for each scenario. Experimental results show that the QoS of VoIP over the two Tor relays was better than the VoIP over the three Tor relays. However, a probability of attackers calculation found that the VoIP with three Tor relays returned better anonymity than that with two Tor relays. The best results were returned over the experimental period in July 2013, when the acceptable calls using two Tor relays reached 64 calls at 5% packet loss. Meanwhile, the worst experimental results were returned in October 2013, when acceptable calls numbered 11 using three Tor relays at 1% packet loss. At the end of 2013, the actual data for relay numbers and bandwidth approached forecast result with time series analysis. Meanwhile, the numbers of Tor users differed from the Tor users forecast. In August 2013, Tor users increased dramatically; this is attributed to the fact that BotNet attack was using the Tor network to attack their target leading to a fivefold increase.

Attempts to simulate piano tones by summing sine waves of inharmonic frequenices have not been totally successful in the past. The physical understanding of what happens to a string inside a piano has never been fully modeled in all detail. A program has been developed to synthesize a piano sound by creating a model of vibrating string. The model is formed by using a differential equation to describe the motion of the string. The terms in the equation account for both the inharmonicity due to the string's stiffness and the decay rates for each harmonic due to the string's internal damping. The model is obtained by solving the differential equation in discrete time steps using fast Fourier transforms in the spatial domain. The waveform of the string is then determined and can be sampled at each step in time.

The Distributed Coordination Function (DCF) is the fundamental access method defined in the IEEE 802.11 Standard for Wireless Local Area Networks (WLANs). With this standard, the Access Point (AP) and the mobile stations consume a significant amount of energy to contend for access to the shared wireless channel. In order to improve the energy efficiency of WLANs, we investigate in this paper a simple and backwards compatible mechanism, called Bi-Directional DCF (BD-DCF), that enables bidirectional communications between wireless stations with a single channel access invocation. The key idea is to let the AP or any mobile station transmit a data packet together with the acknowledgement upon the successful reception of a data packet. This approach reduces the communication overhead, the channel contention, and better balances uplink and downlink transmission opportunities. We evaluate the performance of the proposed BD-DCF by means of computer-based simulation considering different traffic loads, degrees of traffic symmetry, data packet sizes, and data rates. The results presented in this paper show that the BD-DCF protocol can improve the energy efficiency of DCF up to 50%.