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This paper delves into the output feedback control of systems with dual multi‐memoryless channels, where the data packets transmitted from the sensor to the controller (S/C) and from the controller to the actuator (C/A) are suffered from two different multi‐parallel Markovian fading channels. Two new multi‐state Markovian chains are introduced, by which the multiple fading channels system is transformed into a new jumping parameter system. Then a stochastic maximum principle is given along with an optimal state feedback controller for the case that the state is known. When the state is unknown, a Markov jump linear estimator is designed by using the completing square method. In this case, an output feedback controller is obtained by substituting the real system state with its estimation. Meanwhile, the separation principle is shown. Finally, the validity of these methods is verified by simulation examples. [ABSTRACT FROM AUTHOR]

Thanks to their flexibility in research, prototyping, and development, Uncrewed Aerial Vehicles (UAV) are seeing increasing use in agriculture, healthcare, logistics, and other industries. With the increase in the use of UAV in civilian sectors, comes an increase in the possibility of errors in telemetry connections. Issues with the telemetry connection may be catastrophic and cause accidents and crashes and may also adversely affect the performance of the subsystems connected to the telemetry. In this study, a low-cost software-based redundant telemetry link system for UAV is proposed. The proposed system minimises the received packet loss using packet deduplication and is compared with a dynamic link switching method. In the proposed system, both fault tolerance and robustness are introduced into the UAV telemetry link. The analysis is performed using the ArduPilot UAV Simulation environment as a producer and a Golang implementation of the proposed methods. [ABSTRACT FROM AUTHOR]

The purpose of this study is to evaluate and compare how well different routing protocols perform in terms of load sharing, link failover, and overall network performance. Wireshark was used for packet-level analysis, VMWare was used for virtualization, GNS3 was used for network simulation, and Iperf3 was used to measure network performance parameters. Convergence time, packet losses, network jitter, and network delay are the parameters that were selected for assessment. To examine the behaviors of Open Shortest Path First (OSPF) and Enhanced Interior Gateway Routing Protocol (EIGRP) routing protocols in a variety of network settings, a simulated network environment incorporating both protocols along with Border Gateway Protocol (BGP) is created for the research. The setup for the experiment entails simulating different network conditions, such as fluctuating traffic loads and connection failures, to track how the protocols function in dynamic situations. The efficiency metrics for OSPF and EIGRP with BGP are measured and evaluated using the data generated by Wireshark and Iperf3. The results of this research show that EIGRP has a better failover convergence time and packet loss percentage as compared to the OSPF. [ABSTRACT FROM AUTHOR]

In today's digital world, no one and nothing is safe from potential cyberattacks. There is also no 100% protection from such attacks. Therefore, it is advisable to carry out various studies related to the effects of the different cyberattacks on the performance of the specific devices under attack. In this work, a study was carried out to determine how individual TCP DoS attacks affect the parameters of VoIP (Voice over IP) voice and video streams. For the purpose of this work, a model of a simple IP network has been created using the GNS3 IP network-modeling platform. The VoIP platform used was Asterisk Free PBX. Tools from Kali Linux were used to implement the individual TCP DoS attacks; IP-network-monitoring tools and round-trip-delay-measurement tools were also used. The proposed study is applicable to multiple VoIP platforms wherein voice and video traffic are passed/processed by the VoIP server. From the obtained results, it was found that Asterisk Free PBX is very well secured against TCP DoS attacks, which do not affect the platform performance or the parameters of the voice and video streams. The values of the observed parameters, such as jitter, packet loss, round-trip delay, etc., are very far from the maximum allowable values. We also observed a low load on the CPU and RAM of the system during the whole study. [ABSTRACT FROM AUTHOR]

The advent of Network Function Virtualization (NFV) and Service Function Chains (SFCs) unleashes the power of dynamic creation of network services using Virtual Network Functions (VNFs). This is of great interest to network operators since poor service quality and resource wastage can potentially hurt their revenue in the long term. However, the study shows with a set of test-bed experiments that packet loss at certain positions (i.e., different VNFs) in an SFC can cause various degrees of resource wastage and performance degradation because of repeated upstream processing and transmission of retransmitted packets.To overcome this challenge, this study focuses on resource scheduling and deployment of SFCs while considering packet loss positions. This study developed a novel SFC packet dropping cost model and formulated an SFC scheduling problem that aims to minimize overall packet dropping cost as a Mixed-Integer Linear Programming (MILP) and proved that it is NP-hard. In this study, Palos is proposed as an efficient scheme in exploiting the functional characteristics of VNFs and their positions in SFCs for scheduling resources and deployment to optimize packet dropping cost. Extensive experiment results show that Palos can achieve up to 42.73% improvement on packet dropping cost and up to 33.03% reduction on average SFC latency when compared with two other state-of-the-art schemes.

This paper presents an evaluation of the latest MPEG-5 Part 2 Low Complexity Enhancement Video Coding (LCEVC) for video streaming applications using best effort protocols. LCEVC is a new video standard by MPEG, which enhances any base codec through an additional low bitrate stream, improving both video compression efficiency and and transmission. However, there is an interplay between packetization, packet loss visibility, choice of codec and video quality, which implies that prior studies with other codecs may be not as relevant. The contributions of this paper is, therefore in twofold: It evaluates the compression performance of LCEVC and then the impact of packet loss on its video quality when compared to H.264 and HEVC.The results from this evaluation suggest that, regarding compression, LCEVC outperformed its base codecs, overall in terms average encoding bitrate savings when using the constant rate factor (CRF) rate control. For example at a CRF of 19, the average encoding bitrate was reduced by 18.7% and 15.8% when compared with the base H.264 and HEVC codecs respectively. Furthermore, LCEVC produced better visual quality across the packet loss range compared to its base codecs and the quality only started to decrease once packet loss exceeded 0.8-1%, and decreases at a slower pace compared to its equivalent base codecs. This suggests that the LCEVC enhancement layer also provides error concealment. The results presented in this paper will be of interest to those considering the LCEVC standard and expected video quality in error-prone environments [ABSTRACT FROM AUTHOR]

Voice over Internet Protocol (VoIP) is an interactive telecommunication technology that suggests shared, cooperative, and quality transactions in the form of communications over the internet. VoIP normally differs from traditional and conventional circuit-based networks in the form of a low rate because all the communication takes place through the internet. The user has to pay only for an internet connection. Real-time voice data transmission is very difficult when compared with ordinary text data transmission. A main challenge in telecom engineering is maintaining quality. The VoIP transmission faces a lot of inconvenience. It suffers from packet loss, low quality, jitter, delay, and on-time delivery. These parameters influence and degrade performance. The codec plays a major role in VoIP transmission. This paper reports on the fundamental functionalities of VoIP, including coder and decoder operations for quality output, broadcast technology, VoIP connection setups, advantages, applications, and emerging trends in VoIP. The performance of various codecs was analyzed to identify the most suitable codec for voice transmission. [ABSTRACT FROM AUTHOR]

In the context of an increasingly advanced era, Internet of Things (IoT) technology has emerged as a significant innovation across a range of fields. One of the most rapidly developing Internet of Things (IoT) technologies is the Long Range Wide Area Network (LoRaWAN). LoRaWAN is capable of long-distance communication while simultaneously consuming minimal power. In this study, we analyze the coverage of the LoRaWAN network in transmitting data with Cikumpa river water objects, with a 100–600 meters distance between the transmitter (TX) and receiver (RX). This study assesses the RSSI network quality, LoRaWAN SNR, and LoRaWAN network QoS quality concerning throughput, delay, jitter, and packet loss parameters. The testing results demonstrated that the LoRaWAN network coverage reaches a maximum distance of 600 meters. Researchers conducted the testing in the Cikumpa River area. They then analyzed the RSSI and SNR test results in the morning, afternoon, and evening. The results of the RSSI test and calculations demonstrate that as the distance between the transmitter and receiver increases, the RSSI value decreases. The RSSI testing conducted in the morning exhibited a range of -99 dBm to -121 dBm, with the SNR values spanning from -3.25 dB to 8.75 dB. The results of the daytime RSSI tests ranged from -104 dBm to -124 dBm, with the corresponding SNR values ranging from -8.50 dB to 9.00 dB. The RSSI test results for the afternoon period exhibited a range of -96 dBm to -120 dBm, while the SNR demonstrated a range of -7.25 dB to 9.00 dB. In addition, the quality of service (QoS) can be considered stable based on the results of the RSSI and SNR for each test. During the testing process, conducted at distances between 100 and 600 meters, there was no packet loss when data transmission occurred. This research demonstrates the potential for utilizing LoRaWAN technology to monitor a desired object remotely.

Summary: Mobility management allows the networks to track down a mobile user's location for sending datagrams and to retain the connection when the user changes its location points repeatedly. Whenever a node switches between one base station and another, the connectivity is maintained through a process called handoff management. For real‐time traffic implementations in wireless network communications, handover performance is essential. A mobile node is blocked from transmitting and receiving a datagram for a longer time as the handover delay increases. Unacceptable handover delays are common in real‐time communication services. To guarantee smooth connectivity and continuous provision of services, effective handoff methods are required. The handoff efficiency of network‐based distributed mobility management (DMM) is studied extensively in this article, along with a comparison to PMIPv6. We specifically investigate how handover delay, session recovery, and packet loss are influenced by cell radius. From the analytical and simulation findings this paper concludes that network‐based DMM performs better than PMIPv6. [ABSTRACT FROM AUTHOR]

This paper addresses bandwidth limitations resulting from Denial-of-Service (DoS) attacks on Artificial Intelligence of Things (AIOT) systems, with a specific focus on adverse network conditions. First, to mitigate the impact of DoS attacks on system bandwidth, a novel model predictive control combined with a dynamic time-varying quantization interval adjustment technique is designed for the encoder–decoder architecture of AIOT systems. Second, the network state is modeled to represent a Markov chain under suboptimal network conditions. Furthermore, to guarantee the stability of AIOT systems under random packet loss, a Kalman filter algorithm is applied to precisely estimate the system state. By leveraging the Lyapunov stability theory, the maximum tolerable probability of random packet loss is determined, thereby enhancing the system's resilient operation. Simulation results validate the effectiveness of the proposed method in dealing with DoS attacks and adverse network conditions. [ABSTRACT FROM AUTHOR]

It is challenging to enable fast channel switching and packet-loss repair in low-delay live video distribution without negatively influencing the steady-state viewing performance. For example, regularly breaking the inter-frame dependency by introducing intra-predicted keyframes enables random access, but is costly in terms of rate-distortion performance. For this reason, the keyframe-injection method minimizes the impact by sending a compression-efficient normal video stream to all end-users. As accompaniment, a companion stream that solely consists of keyframes is sporadically used for only those users that switch channels or experience packet loss. This paper describes the requirements to implement keyframe injection in three video coding standard generations (H.264/AVC, H.265/HEVC, and H.266/VVC). We evaluated the impact that keyframe injection has on the quality of the video in terms of a decrease in VMAF, PSNR and SSIM. We demonstrate that the quality reduction caused by keyframe insertion is generally low, meaning that keyframe injection typically is imperceptible. However, drift-error artifacts become perceptible over time for rare outliers. Moreover, we pinpointed the cause of this worst-case artifact type to be halfpel interpolation. As a solution, codecs can disable subpel motion estimation, and future standards could design their filters more carefully. Lastly, it should be noted that keyframe injection will only be applied sporadically when users require a random access or experience packet loss, and only to those users. Most interestingly, all other users receive a compression-efficient stream wherein the inter-frame dependency is not artificially broken at regular short intervals. As such, our proposed solution makes low-latency video distribution efficient and viable in multiple coding standards. [ABSTRACT FROM AUTHOR]

Wireless networks play a pivotal role in various domains, including industrial automation, autonomous vehicles, robotics, and mobile sensor networks. This research investigates the critical issue of packet loss in modern wireless networks and aims to identify the conditions within a network's environment that lead to such losses. We propose a packet status prediction model for data packets that travel through a wireless network based on the IEEE 802.15.4 standard and are exposed to five different types of interference in a controlled experimentation environment. The proposed model focuses on the packetization process and its impact on network robustness. This study explores the challenges posed by packet loss, particularly in the context of interference, and puts forth the hypothesis that specific environmental conditions are linked to packet loss occurrences. The contribution of this work lies in advancing our understanding of the conditions leading to packet loss in wireless networks. Data are retrieved with a single CC2531 USB Dongle Packet Sniffer, whose pieces of information on packets become the features of each packet from which the classifier model will gather the training data with the aim of predicting whether a packet will unsuccessfully arrive at its destination. We found that interference causes more packet loss than that caused by various devices using a WiFi communication protocol simultaneously. In addition, we found that the most important predictors are network strength and packet size; low network strength tends to lead to more packet loss, especially for larger packets. This study contributes to the ongoing efforts to predict and mitigate packet loss, emphasizing the need for adaptive models in dynamic wireless environments. [ABSTRACT FROM AUTHOR]

Copyright of Przegląd Elektrotechniczny is the property of Przeglad Elektrotechniczny and its content may not be copied or emailed to multiple sites or posted to a listserv without the copyright holder's express written permission. However, users may print, download, or email articles for individual use. This abstract may be abridged. No warranty is given about the accuracy of the copy. Users should refer to the original published version of the material for the full abstract. (Copyright applies to all Abstracts.)

The deployment of resource-constrained and densely distributed Internet of Things (IoT) devices poses significant challenges for 5G communication systems due to the increased likelihood of inter-tier interference. This interference can degrade network performance and hinder the transmission of data in a reliable and efficient manner. Using an incremental Radial Basis Function (RBF) technique, this paper proposes a novel approach for cross-tier interference mitigation in 5G communication among resource-constrained dense IoT networks. Utilizing the incremental RBF method to model and optimize interference patterns in resource-constrained dense IoT networks is the primary innovation of our approach. In contrast to conventional interference mitigation techniques, which view interference as a static phenomenon, our method adapts to the dynamic nature of IoT networks by incrementally updating the RBF model. This enables precise modeling of the various interference scenarios and real-time modification of interference mitigation parameters. Utilizing the spatial distribution of IoT devices, this approach improves interference mitigation. The proposed method intelligently allocates resources and optimizes interference mitigation parameters based on the location and density of IoT devices. This adaptive resource allocation improves network capacity, reliability, and overall system performance by maximizing the utilization of available resources while minimizing interference. We demonstrate the effectiveness of the incremental RBF-based approach in mitigating cross-tier interference in resource-constrained dense IoT networks within the 5G ecosystem through extensive experiments and simulations. Our findings indicate substantial improvements in communication performance, including increased throughput, decreased packet loss, and decreased latency.

This paper presents a dataset comprising 700 video sequences encoded in the two most popular video formats (codecs) of today, H.264 and H.265 (HEVC). Six reference sequences were encoded under different quality profiles, including several bitrates and resolutions, and were affected by various packet loss rates. Subsequently, the image quality of encoded video sequences was assessed by subjective, as well as objective, evaluation. Therefore, the enclosed spreadsheet contains results of both assessment approaches in a form of MOS (Mean Opinion Score) delivered by the absolute category ranking (ACR) procedure, SSIM (Structural Similarity Index Measure) and VMAF (Video Multimethod Assessment Fusion). All assessments are available for each test sequence. This allows a comprehensive evaluation of coding efficiency under different test scenarios without the necessity of real observers or a secure laboratory environment, as recommended by the ITU (International Telecommunication Union). As there is currently no standardized mapping function between the results of subjective and objective methods, this dataset can also be used to design and verify experimental machine learning algorithms that contribute to solving the relevant research issues.

This paper addresses the challenging problem of achieving consensus in multi-agent systems (MASs) under the simultaneous influence of communication delays and packet loss. We introduce a novel switched system model that differentiates between delays caused by network-induced temporal factors and those resulting from packet loss, offering a more precise and less conservative analysis compared to traditional unified delay models. A key contribution of this paper is the design of a low-gain controller, developed using a parametric algebraic Riccati equation, which enhances system robustness without requiring recalibration when the number of agents varies. We also employ Lyapunov stability theory to establish constraints that ensure consensus is maintained despite the intermittent and transient nature of packet loss. Theoretical analysis confirms that our enhanced time-delay model effectively reduces system conservatism. The efficacy of the proposed approach is validated through numerical simulations, demonstrating its capability to achieve consensus in MASs even in the presence of dynamic delays and packet dropouts.

In this paper, the weighted group consensus for a kind of discrete heterogeneous multi-agent systems (HMASs) with packet loss in cooperative-competitive networks based on self-adaptive controller is studied. Based on self-adaptive controller, packet loss and cooperative-competitive relation, a novel control protocols have been designed for this system without satisfying the in-degree balance of the vertex. Some sufficient conditions have been obtained for the weighted group consensus of this kind of HMASs, by using graph theory, matrix analysis and complex frequency method. Based on weighted parameters, control parameters, packet loss rate and cooperative-competitive relation, the upper bound of the input time delay can be calculated. Finally, some simulation experiments are listed to show the effectiveness of the derived results.

Software-defined networking (SDN) is an emerging technology that addresses the limitations of legacy networks when handling the growing traffic demands. Legacy networks face issues like heavy traffic loads, congestion and delays. SDN offers a solution to these challenges. However, an efficient routing algorithm is required to transmit data packets through the network. While there are existing routing algorithms, there remains a need for a highly efficient one that minimizes packet loss and optimizes link utilization. One commonly used routing for mitigating issues, such as packet loss, delay and bandwidth allocation, is multipath routing. The inherent flexibility of SDN, which provides a comprehensive overview of network status, empowers routing algorithms to generate efficient routing paths. We introduce the enhanced dynamic multipath routing (EDMR) algorithm in this context. This algorithm dynamically generates and updates routing paths within the network's switches. Its dynamic nature significantly reduces packet loss and minimizes end-to-end delays. [ABSTRACT FROM AUTHOR]

Focusing on the network congestion and packet loss in the networked systems, a recursive quantization-based event triggered observer (RQETO) is proposed in this paper. The RQETO is composed of the local observer and the remote observer. The local observer reduces the amount of transmitted measured output by incorporating the event triggered strategy with the improved quantization mechanism. The improved quantization mechanism is based on the recursive algorithm, which can constrain the quantization error within the maximum allowable quantization error. The remote observer compensates the impact of packet loss by Bernoulli distribution model and hold-input strategy after receiving the data transmitted by the local observer. Through the local observer and the remote observer, network congestion is adequately alleviated, and the impact of packet loss is compensated while obtaining the accurate state estimation. The stability of the RQETO is proved by Lyapunov method, and the effectiveness of the RQETO is demonstrated on brushless direct current motor and Net-Con PC104 experimental platform. [ABSTRACT FROM AUTHOR]

In this research paper, we investigate the problem of remote state estimation for nonlinear discrete systems. Specifically, we focus on scenarios where event-triggered sensor schedules are utilized and where packet drops occur between the sensor and the estimator. In the sensor scheduler, the SOD mechanism is proposed to decrease the amount of data transmitted from the sensor to a remote estimator and the phenomena of packet drops modeled with random variables obeying the Bernoulli distribution. As a consequence of packet drops, the assumption of Gaussianity no longer holds at the estimator side. By fully considering the non-linearity and non-Gaussianity of the dynamic system, this paper develops an event-trigger particle filter algorithm to relieve the communication burden and achieve an appropriate estimation accuracy. First, we derive an explicit expression for the likelihood function when an event trigger occurs and the possible occurrence of packet dropout is taken into consideration. Then, using a special form of sequential Monte–Carlo algorithm, the posterior distribution is approximated and the corresponding minimum mean-squared error is derived. By contrasting the error covariance matrix with the posterior Cramér–Rao lower bound, the estimator's performance is assessed. An illustrative numerical example shows the effectiveness of the proposed design. [Display omitted] • Limited bandwidth networked systems require sensors scheduling algorithm design. • Event-based particle filter is designed in conjunction with scheduling algorithms and packet dropout. • With Cramér–Rao lower bound the proposed scheduling algorithms improve clearly the filter performance. • Innovation scheduling allows robustness against packet dropouts. [ABSTRACT FROM AUTHOR]

Nowadays, cellular applications rule the digital world with their betterment. However, security is the primary concern for a better communication range during the communication process. If the messages are hacked, data overhead and collisions occur. So, the present research work has aimed to design the novel Buffalo-based Autoencoder Security Framework (BbASF) developed in the Orthogonal-Frequency-Division- Multiplexing (OFDM) channel. Consequently, the function of the designed model is checked with the Denial of Service (DoS)-CICIDS dataset. The planned model is tested in the python environment. After that, the communication parameters were validated and compared with other schemes. The presented approach has earned the finest outcome in all performance assessments than the compared models. Furthermore, it has reduced the packet loss to the desired level, 2.2e-6. In addition, the presented novel BbASF is good at forecasting malicious behavior; it has earned the best attack detection score of 99.6%. Hence, the present model efficiently predicts malicious features and enriches communication facilities. [ABSTRACT FROM AUTHOR]

Copyright of Computer Measurement & Control is the property of Magazine Agency of Computer Measurement & Control and its content may not be copied or emailed to multiple sites or posted to a listserv without the copyright holder's express written permission. However, users may print, download, or email articles for individual use. This abstract may be abridged. No warranty is given about the accuracy of the copy. Users should refer to the original published version of the material for the full abstract. (Copyright applies to all Abstracts.)

One of the major concerns for service providers and application developer are Quality of experience (QoE), where high traffic congestion on the Internet leads to the degradation of video quality. However, the effectiveness of video transmission is minimized due to the network based on packet loss, bandwidth, and delay. Because of bandwidth limitations, the videos transmitted are obtained in low quality. Meanwhile, various outcomes such as reduction in throughput, re-buffering, or mosaic are determined in packet loss which validated the video streaming obtained in reliable or unreliable mode. Therefore this paper proposes an Improved Fuzzy Weighted queueing based Crossover Fire Hawk (IFW-CFH) algorithm for effective real-time video transmission. The objective of the IFW-CFH approach is to reduce the delay, packet loss, and bandwidth to enhance the video quality via two key mechanisms namely congestion control mechanism as well as packet scheduling mechanism. During the generation of encoded video frames, the packaged packets to the local buffer are transmitted by the scheduler using our proposed IFW-CFH algorithm. Finally, the experimentation is conducted and the results show that the proposed method minimized transmission delay, packet loss, and bandwidth by 13.8% for effective real-time video transmission compared to the existing methods. [JJCIT 2023; 9(3.000): 207-219]

Nowadays, computer networks are used in education, sports, business, transportation, manufacturing, and more. Thus, it makes the network more complex and harder to monitor, especially in terms of performance. The implementation of network monitoring software presents a viable solution for resolving this issue. It is because network monitoring software can monitor behaviours, provide alertness for any failing component of a computer network, and measure performance such as CPU usage, bandwidth, throughput, response time, and packet loss. However, network monitoring software has common problems, such as being hard to set up and making it hard to find the best network monitoring that fits the requirements. Thus, this study aims to identify network monitoring software that is easy to configure and analyse the performance of network monitoring software in terms of response time and packet loss. The study comprises the implementation of open-source network monitoring software, including PRTG, OpManager, Zabbix, and LibreNMS. The two experiments were conducted by launching TCP attacks for response time and UDP attacks for packet loss. Both attacks consisted of four scenarios: 15 threads, 30 threads, 45 threads, and 60 threads. As a result, PRTG offers a simpler installation procedure and easier configuration options, with a straightforward and attractive Graphical User Interface (GUI) that allows for all configurations to be conveniently performed directly through the interface. For monitoring websites, users need to add the device that was targeted by the website and choose an HTTP sensor for response time and PING for packet loss. The evaluation was based on the lowest average response time and consistency in the detection of packet loss. Consequently, Zabbix demonstrates the highest level of responsiveness about web services, which have the lowest average response time and consistently exhibit the lowest occurrence of packet loss.

WAMPAC (Wide Area Monitoring Protection and Control) applications are becoming crucial for granting a stable operation of the electricity transmission grid. These systems use a set of sensors distributed between different electrical substations to gather real-time measurements from the field. These sensors are called Phasor Measurement Units (PMUs). Using the gathered data, different monitoring, protection, and control algorithms are run in a Phasor Data Concentrator (PDC) located in a central location. These algorithms close the loop via the generation of remedial commands, which are sent back to the field level with stringent delay, security, and reliability requirements. GOOSE (Generic Object Oriented Substation Events) protocol, defined by IEC 61850 (IEC stands for International Electrotechnical Commission), is used for that aim and also considers the option of sending these commands over IP networks (this option is called Routed-GOOSE). The present article proposes two alternatives for the tunneling of GOOSE frames over IP. Both options allow the decoupling of the transmission and the security aspects, thus increasing flexibility and allowing for easier deployment. The first option, called VX-GOOSE, is a combination of standard protocols, allowing the sending of these frames over UDP/IP tunnels. The tests that have been carried out demonstrate that, under certain network conditions, the transmission of GOOSE frames over UDP may fail, and in some extreme cases, even a whole burst of GOOSEs could be lost. This may have very bad consequences for a distributed electrical system. It should be noted that this limitation affects both VX-GOOSE and Routed-GOOSE. To overcome these limitations, the second option, called Simplemux blast mode, includes a novel mechanism that provides delivery guarantees and a reduced delay, with the counterpart of a certain degree of redundancy. As shown in the experiments, the incurred delays can be significantly reduced when remote locations are connected via unreliable networks, whereas the bandwidth increase caused by redundancy can be kept at reasonable levels. Finally, it should be remarked that although GOOSE is a relevant example use case, this approach can be applied in other fields where flows require very low delay and delivery guarantees. [ABSTRACT FROM AUTHOR]

Providing uninterrupted wireless connectivity to vehicles has posed a significant challenge in the context of Vehicle–to–Infrastructure (V2I) communications especially in heterogeneous network environments. This limitation primarily arises due to the constrained infrastructure of vehicular networks. Solutions like Wireless Access for Vehicular Environment (WAVE) protocol and On–Board Units (OBUs) have aimed to expand network capacity and enhance communication between vehicles and infrastructure. However, significant network resources are spent on selecting appropriate target networks during handover in heterogeneous networks. To address issues like incorrect handover network selection and handover delays, researchers have proposed various multi–criteria decision making (MCDM) techniques to choose the optimal candidate cell within HetNet environment. Nevertheless, a majority of these solutions do not account for variations in criteria weights during handoff decisions, which impacts the efficient utilization of resources in V2I communications. To mitigate the problem of incorrect network selection during handover, this study introduces a V2I Multicriteria based Handover Algorithm (mV2I–MHA), considering network parameters such as cost, bandwidth, packet loss ratio, and packet latency. The algorithm’s performance was evaluated using MATLAB R2020a. The obtained results were compared with an existing algorithm where it demonstrated a significant improvement of 27.6% in throughput and a reduction in packet loss rates. [ABSTRACT FROM AUTHOR]

Typical communication and detection issues in a nonideal information environment, such as sensing failure, communication and sensing delay, and packet loss, further aggravate the adverse impacts of human-driven vehicle (HV) uncertainty on a mixed vehicle platoon. To guarantee the performance of the mixed vehicle platoon featuring HVs and connected and automated vehicles under the nonideal information environment, this article proposes a platoon control strategy integrating a combined longitudinal and lateral control and message recovery. Specifically, by building the dataset associated with HV behaviors, a long short-term memory (LSTM) predictor is established to recover the problematic HV messages (i.e., position, velocity, and heading) caused by the nonideal information environment. Furthermore, based on the boundary of the HV states, an evaluation and correction (EC) method is presented to suppress the adverse impacts of prediction failures. Then, a combined longitudinal and lateral controller cooperating with the LSTM predictor and EC method is developed to enhance the stability and safety of the mixed vehicle platoon under the nonideal information environment. In a theoretical analysis, the relatedness between the asymptotic stability and string stability of the platoon and predictor accuracy is strictly proved. Finally, comparative experiments verify the effectiveness of the proposed control strategy by employing driver-in-the-loop simulations. [ABSTRACT FROM AUTHOR]

One of the major concerns for service providers and application developers is the Quality of Experience (QoE), where high traffic congestion on the Internet leads to the degradation of video quality. However, the effectiveness of video transmission is minimized due to the network based on packet loss, bandwidth and delay. Because of bandwidth limitations, the videos transmitted are obtained in low quality. Meanwhile, various outcomes, such as reduction in throughput, re-buffering or mosaic, are determined in packet loss which validated the video streaming obtained in reliable or unreliable modes. Therefore, this paper proposes an Improved Fuzzy Weighted queueingbased Crossover Fire Hawk (IFW-CFH) algorithm for effective real-time video transmission. The objective of the IFW-CFH approach is to reduce delay, packet loss and bandwidth to enhance the video quality via two key mechanisms; namely, congestion control mechanism as well as packet scheduling mechanism. During the generation of encoded video frames, the packaged packets to the local buffer are transmitted by the scheduler using our proposed IFW-CFH algorithm. Finally, experimentation is conducted and the results show that the proposed method minimized transmission delay, packet loss and bandwidth by 13.8% for effective real-time video transmission compared to the existing methods. [ABSTRACT FROM AUTHOR]

Wireless sensor networks (WSNs) are designed to monitor their surroundings, an application that has been leveraged significantly by advances in the Internet of Things (IoT). However, the susceptibility of wireless systems to errors and malicious attacks remains a challenge. Identity deception, particularly, has emerged as a significant issue, exacerbated by the redundancy of multi-hop routing, leading to potentially damaging attacks on routing protocols. Routing, the method employed in WSNs to disseminate data to base stations, has recently seen the incorporation of trust mechanisms to enhance security and foster cooperation among nodes. Routing decisions are made based on the anticipated trustworthiness of individual nodes. Considering the vulnerability of WSNs to various attacks, secure routing is of paramount importance. In this research, we propose a MultiLevel Trust Evaluation Model using Replicated Auditor Node (MLTEM-RAN) for secure route detection. This model aims at maximizing the packet delivery rate. It takes into account information about each relay node along the path, including the trust value and the current condition of each node. The trust value of a node is defined as the attack probability of the node, which is based on historical behaviors. The node's status, on the other hand, is a composite measure that considers both the node's remaining energy and its distance to the sink node. When compared with existing models, the proposed MLTEM-RAN model demonstrates superior performance in terms of packet delivery rate. This study thus represents a significant step forward in the development of secure and efficient routing strategies for WSNs. [ABSTRACT FROM AUTHOR]

From the last decades different types of network schemes are pitched to enhance the user performance. Software Defined Networks (SDN) is also considered as important factor for different network schemes and its proper administration or management. Due to major deployment in today's networking era SDN are further sub divided in to commercial and opensource controllers. Commercial and open-source controllers are utilized in different type of businesses. According to our knowledge considerable amount of literature is available on these controllers but did not provide or analyse performance of these controllers on different network parameters. This paper evaluates and compares the performance of two well-known SDN open-source controllers POX and RYU with two performance assessments. The first assessment is the implementation of optimal path by using Dijkstra's algorithm from source to destination. Second assessment is the creation of a custom topology in our desired tool (MiniNet emulator). Then, the performance in terms of QoS parameters such as Jitter, throughput, packet loss, and packet delivery ratio are computed by two end hosts in each network. After the assessments, the performance of POX are optimal as compare to the RYU and best suited to be deployed in any scenario. [ABSTRACT FROM AUTHOR]

Wireless Mobile Ad Hoc Networks (MANETs) present topologies where nodes act as both hosts and routes, necessitating efficient routing protocols to handle frequent route updates and transmission congestion. This paper proposes a novel routing algorithm, Fuzzy Controller Ad hoc On-demand Distance Vector (FC-AODV), to address these challenges. The FC-AODV algorithm is designed to improve packet delivery ratio, throughput, end-to-end delay, packet loss, remaining energy, and routing overhead compared to the traditional AODVprotocol. Through extensive simulations using NS 2, the proposed algorithm demonstrates superior performance across varying network scenarios. Results indicate an average increase of 12.4% in packet delivery ratio, 10.27% in throughput, and a decrease of 5.37% in end-to-end delay compared to AODV. Additionally, FC-AODV reduces packet loss by 5.57%, increases remaining energy by 6.4%, and decreases routing overhead by 6.37%. Future research directions include applying the FC-AODV concept to satellite networks, exploring the integration of shortest path algorithms, and evaluating FC-AODV in real-time network environments. [ABSTRACT FROM AUTHOR]

The distributed filtering problem is studied for a multi-rate uniform sampling linear discrete system over sensor networks, where packet losses may occur during data exchange between nodes. The multi-rate system is transformed into a single-rate system by using a virtual measurement method. At each sensor node, the estimates received from neighbour nodes are fused by matrix or scalar weighting according to their different estimation accuracy. A distributed recursive filter with the Kalman consensus filter structure is devised based on the fused estimate, where the filtering gain and consensus gain are solved by locally minimising an upper bound of the filtering error covariance. The obtained gains and the upper bound of the covariance depend on a group of free positive scalar parameters. The scalar parameters are optimised by minimising the upper bound of the covariance. It is proved that the upper bound of the covariance of the filter with optimal gains and parameters is smaller than the covariance of the local filter based on sensor measurements. It avoids calculating cross-covariance matrices between nodes and has small computational burden. The exponential boundedness in the mean square of the filtering error is analysed. A target tracking system verifies the effectiveness of the algorithm. [ABSTRACT FROM AUTHOR]

Science networks and their hosted applications require large and frequent data transfers, but these transfers are subject to network performance degradation, including queuing delays and packet drops. However, well known network dynamics along with limited instrumentation access complicate the creation of an accurate method that predicts different performance aspects of data transfers. In this study, we develop a lightweight machine learning tool to predict end-to-end packet retransmission in science flows of arbitrary size. We also identify the minimum set of necessary path and host measurements needed as input features in our predictor in order to achieve high accuracy. In our evaluation process our predictor demonstrated low training times and was able to provide accurate estimates (97%–99%) for packet retransmissions of data transfers of arbitrary sizes. The results also manifest that the our solution was able to predict retransmit behavior reasonably well (66%) even for previously unseen data if training and testing datasets had similar statistics.

Science networks and their hosted applications require large and frequent data transfers, but these transfers are subject to network performance degradation, including queuing delays and packet drops. However, well known network dynamics along with limited instrumentation access complicate the creation of an accurate method that predicts different performance aspects of data transfers. In this study, we develop a lightweight machine learning tool to predict end-to-end packet retransmission in science flows of arbitrary size. We also identify the minimum set of necessary path and host measurements needed as input features in our predictor in order to achieve high accuracy. In our evaluation process our predictor demonstrated low training times and was able to provide accurate estimates (97%–99%) for packet retransmissions of data transfers of arbitrary sizes. The results also manifest that the our solution was able to predict retransmit behavior reasonably well (66%) even for previously unseen data if training and testing datasets had similar statistics.

Fertilization plays a very large role in crop yields if done in the right way, effective and efficient fertilization will be achieved if the soil fertility conditions are known in advance. However, the problem is that farmer fertilization is still carried out by means of estimates based on land area without estimating the factor of soil needs.In this study the system uses a Wireless Sensor Network where there are 2 nodes installed on the land to monitor the measurement of soil moisture, soil pH, and nutrients using soil moisture sensors, soil pH sensors, and NPK sensors. From these sensors, it will be integrated through the Arduino Uno microcontroller which will later be connected to the LoRa system to send measurement information data on the land to the raspberry pi.The communication system between nodes is designed with a peer to per topology. The results of this study are the use of appropriate fertilizers and then can restore soil conditions to ideal soil conditions. From the experimental results, it is found that the peer to peer communication system can work well.