DMP Packet Scheduling For Wireless Sensor Network

Communication is the most important thing in the success of an activity. Communication that occurs in a computer network does not only process data packet transfers between computers, but computer technology is currently able to transfer data in the form of sound frequencies that are processed into data packets and sent between computers on the internet network. The development of internet technology is not only using for getting information, but also that is widely used. One of them is audio call or VoIP (Voice over Internet Protocol), this technology has been used by several messenger service providers, such as Yahoo Messenger, Goggle Talk, Skype, and others. Voice over Internet Protocol is a technology that makes internet media able to conduct voice communication remotely directly. Analog sound signals, like what you hear when communicating on the telephone are converted to digital data and sent over the network in the form of data packets in real time. Asterisk is a softswitch to operate the proxies, with session initiation protocol (SIP) based. The aim is to build an Asterisk-VoIP based server, so that it can be developed in future research as needed. The methodology in outline consists of two lines, namely literature study and PPDIOO (Plan, Prepare, Design, Implement, Operate, Optimize). It has been implemented in the Banten Province Communication and Information Service, which has been previously established by the internet network. So that VoIP here is functioned for maximizing it.

Packet collision is the primary source of unnecessary energy usage in wireless sensor networks (WSNs). To address this issue, the advanced packet scheduling algorithm is presented. In real-time WSNs, some packet scheduling techniques can also affect and delay data transmission. Efficient packet scheduling is vital for minimizing energy consumption and data transmission delay in WSNs. It is crucial because it guarantees the fair and timely distribution of various kinds of data packets according to their priorities. Node will directly transmit the packet to its destination based on its priority. However, current approaches like first come first serve (FCFS) schedulers often result in significant latency, hindering timely data delivery to base stations (BS). To address this, we propose a novel dynamic multi-level round search mayfly optimizer-based priority packet scheduling technique (MRSMOPS). This technique utilizes a traffic-aware-weighted queue (TAWQ) model to schedule tasks based on priority, optimizing transmission slots for flowing packets. Our objective is to reduce overall transmission completion time while adhering to delay limits and energy budgets. By integrating the newly developed MRSMOPS method, the efficiency of packet scheduling is enhanced in WSNs. The proposed algorithm has demonstrated significant improvements in performance metrics, achieving a 15% reduction in packet delay and a 10% enhancement in energy efficiency compared to existing methods.

The importance of cloud computing grows staggeringly by the day as newer technology is introduced resulting in greater connectivity across systems worldwide inculcating higher demand for resources supplied by cloud providers. Every datacentre responsible for providing these resources consumes a hefty amount of power and cloud service providers have to bear the monetary costs that come with it. Recent studies show that energy consumption of resources at utilizations over 70% grows non-linearly and hence it is best to avoid resources reaching high levels of utilization. In this paper, we introduce EPTS (Energy-saving Pre-emptive Task Scheduling) that utilizes the 0-70% interval of utilization of resources and evenly distributes the energy consumption of tasks by pre-empting tasks consuming high amounts of power by less power-demanding tasks. These pre-empted tasks are rescheduled into the virtual machines where they may result in lower utilization rates, hence increasing energy conservation. We compare EPTS with MaxUtil and FCFS (First Come First Serve) via thorough experimentation on different synthetic datasets and show that EPTS outperforms MaxUtil and FCFS by approximately 50% in terms of energy consumption.

Chapter 14 - Scheduling packets

Structural monitoring of critical bridge structures can greatly benefit from the use of wireless sensor networks (WSNs), however energy harvesting for the operation of the network remains a challenge in this setting. While solar and wind power are possible and credible solutions to energy generation, the need for positioning sensor nodes in shaded and sheltered locations, e.g., under a bridge deck, is also often precluding their adoption in real-world deployments. In some scenarios vibration energy harvesting has been shown as an effective solution, instead.This paper presents a multihop vibration energy-harvesting WSN system for bridge applications. The system relies on an ultra-low power wireless sensor node, driven by a novel vibration based energy-harvesting technology. We use a receiver-initiated routing protocol to enable energy-efficient and reliable connectivity between nodes with different energy charging capabilities. By combining real vibration data with an experimentally validated model of the vibration energy harvester, a hardware model, and the COOJA simulator, we develop a framework to conduct realistic and repeatable experiments to evaluate the system before on-site deployment.Simulation results show that the system is able to maintain energy neutral operation, preserving energy with careful management of sleep and communication times. We also validate the system through a laboratory experiment on real hardware against real vibration data collected from a bridge. Besides providing general guidelines and considerations for the development of vibration energy-harvesting systems for bridge applications, this work highlights the limitations of the energy budget made available by traffic-induced vibrations, which clearly shrink the applicability of vibration energy-harvesting technology for WSNs to low traffic applications.

The problem of data packet dropout and transmission delays induced by communication channel in networked control systems (NCSs) in both continuous-time case and discrete-time case is studied in this paper. We model the NCSs with data packet dropout and delays as ordinary linear systems with input delays and this enables us to apply the rich theory of delay systems to analysis and design of such NCSs. For the continuous-time case, our technique is based on Lyapunov-Razumikhin function method. For the discrete-time case, we use the Lyapunov-Krasovskii based method. Attention is focused on the design of memoryless state feedback controllers that guarantee stability of the closed-loop systems. We present the admissible bounds of data packet loss and of delays in terms of linear matrix inequalities. Numerical examples illustrate the effectiveness of the proposed approach.

Motivated by the emerging standards for indoor pico-cellular wireless systems, such as the Bluetooth, we propose and study the scheduling policies for master driven time division duplex (TDD) wireless networks. In these networks, the frequency band is divided into time slots, and each end (i.e., master or slave) takes turns in using the time slots. In Bluetooth, a slave transmits packets in the reverse slot only after the master polls the slave in a forward slot (by sending data to it). The conventional scheduling policies such as round robin do not perform well in these systems as they are not suited to the tight coupling of the uplink-downlink. We propose new scheduling policies, (i) the priority scheme, and, (ii) the K-fairness scheme that utilize the state at the master and slaves to schedule the TDD slots effectively. Active slaves are differentiated based upon the binary information (i.e., the presence or absence of packets in a slave queue) about the master-slave queue pairs. The priority scheme achieves high throughput as compared to the packet-by-packet generalized processor sharing (PGPS) based policies while guaranteeing a minimal service to each active slave while the K-fairness policy is characterized by a tight fairness bound as well as high system throughput. We then extend these policies for scheduling variable size data in the presence of voice. Further, since Bluetooth supports variable size data packets (1, 3 or 5 slots) on the same connections, the segmentation and reassembly (SAR) can significantly impact scheduling of data packets by varying packet size distribution. We propose an intelligent SAR policy (ISAR) and compare it with the naive random-SAR in which the data packet sizes (i.e., 1, 3 or 5) are assigned probabilistically. ISAR adapts MAC packet size at the master and slave queues depending on the data arrival rates at both the queues.

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.

A close look at many existing systems reveals their two- or multi-layer nature, where a number of coexisting networks interact and depend on each other. For instance, in the Internet, any application-level graph (such as a peer-to-peer network) is mapped on the underlying IP network that, in turn, is mapped on a mesh of optical fibers. This layered view sheds new light on the tolerance to errors and attacks of many complex systems. What is observed at a single layer does not necessarily reflect well the state of the entire system. On the contrary, a tiny, seemingly harmless disruption of one layer, may destroy a substantial or essential part of another layer, thus making the whole system useless in practice. In this thesis we consider such two-layer systems. We model them by two graphs at two different layers, where the upper-layer (or logical) graph is mapped onto the lower-layer (physical) graph. Our main goals are the following. First, we study the robustness to failures of existing large-scale two-layer systems. This brings us some valuable insights into the problem, e.g., by identifying common weak points in such systems. Fortunately, these two-layer problems can often be effectively alleviated by a careful system design. Therefore, our second major goal is to propose new designs that increase the robustness of two-layer systems. This thesis is organized in three main parts, where we focus on different examples and aspects of the two-layer system. In the first part, we turn our attention to the existing large-scale two-layer systems, such as peer-to-peer networks, railway networks and the human brain. Our main goal is to study the vulnerability of these systems to random errors and targeted attacks. Our simulations show that (i) two-layer systems are much more vulnerable to errors and attacks than they appear from a single layer perspective, and (ii) attacks are much more harmful than errors, especially when the logical topology is heterogeneous. These results hold across all studied systems. A natural next step consists in improving the failure robustness of two-layer systems. In particular, in the second part of this thesis, we consider the IP/WDM optical networks, where an IP backbone network is mapped on a mesh of optical fibers. The problem lies in designing a survivable mapping, such that no single physical failure disconnects the logical topology. This is an NP-complete problem. We introduce a new concept of piecewise survivability, which makes the problem much easier in practice. This leads us to an efficient and scalable algorithm called SMART, which finds a survivable mapping much faster (often by orders of magnitude) than the other approaches proposed to date. Moreover, the formal analysis of SMART allows us to prove that a given survivable mapping does or does not exist. Finally, this approach helps us to find vulnerable areas in the system, and to effectively reinforce them, e.g., by adding new links. In the third part of this thesis, we shift our attention one layer higher, to the application-over-IP setting. In particular, we consider the design of Application-Level Multicast (ALM) for interactive applications, where a single source sends a delay-constrained data stream to a number of destinations. Interactive ALM should (i) respect stringent delay requirements, and (ii) proactively protect the system against overlay node failures and against (iii) the packet losses at the IP layer. We propose a two-layer-aware approach to this problem. First, we prove that the average packet loss rate observed at the destinations can be effectively approximated by a purely topological metric that, in turn, drops with the amount of IP-level and overlay-level path diversity available in the system. Therefore, we propose a framework that accommodates and generalizes various techniques to increase the path diversity in the system. Within this framework we optimize the structure of ALM. As a result, we reduce the effective loss rate of real Internet topologies by typically 30%-70%, compared to the state of the art. Finally, in addition to the three main parts of the thesis, we also present a set of results inspired by the study of ALM systems, but not directly related to the 'two-layer' paradigm (and thus moved to the Appendix). In particular, we consider a transmission of a delay-sensitive data stream from a single source to a single destination, where the data packets are protected by a Forward Error Correction (FEC) code and sent over multiple paths. We show that the performance of such a scheme can often be further improved. Our key observation is that the propagation times on the available paths often significantly differ, typically by 10-100ms. We propose to exploit these differences by appropriate packet scheduling, which results in a two- to five-fold improvement (reduction) in the effective loss rate.

This paper presents a comparison of the forward link performance of the HDR and 1XTREME systems. Both systems have been proposed as possible evolutionary paths for the 1.25 MHz cdma2000 system, and have been designed to enable high data rate packet transmission. Link-level performance is evaluated by chip-level time-domain simulation of the forward data traffic channel in the presence of additive white Gaussian noise, and is then used in a system-level simulator modeling a network of 19 three-sector cells. The system-level simulator accounts for slow and fast fading and two types of multipath channel profiles, as described by the pedestrian A and vehicular B ITU-R channel models. The packet scheduling is performed using a proportionally fair (PF) scheduler. The systems use bit rate adaptation and hybrid ARQ to achieve 1% packet error rate (PER). All users in the system are assumed to always have data packets available for transmission. System performance is evaluated for an embedded sector in terms of average sector throughput as a function of the number of users, distributions of data rates, and individual throughputs over a set of users. Results show the benefits of multiuser diversity exploited by the PF scheduler, and present throughput and fairness comparisons for the two systems.

Being more economical in comparison to a wired network, Wireless Sensor Network (WSN) could be a good candidate for multimedia applications. Transmission is the most energy-consuming process and decisive factor for determining the lifetime of a WSN. Having an attribute of scarce energy resources and inclusion of multimedia traffic in WSN, render the development of an energy-efficient and collision avoiding routing scheme for WSN. MAC protocols based on CSMA/CA are also prone to collisions that result in a repetitive call to a back-off algorithm. In CSMA, the inclusion of a collision avoidance scheme increases the overhead and results in an underutilization of the channel and large delay incurred by the data packets. In this paper, a novel packet scheduling strategy (PSSN) is presented to diminish the chances of collisions along a single path in a shared wireless medium. The PSSN is a query-driven approach and works without RTS and CTS control signals. Each packet is scheduled after a fixed interval that is decided by the PSSN keeping in view the objective of collision reduction along a single path. To validate the proposed scheme, we simulated it in Network-Simulator-2 and compared it with well known built-in AODV and AOMDV protocols. The proposed protocol outperforms the above two with respect to energy efficiency and delays while maintaining a better throughput level.

Limited computational resources in Wireless Sensor Networks (WSNs) considered a key challenge in sensor environment. With such constraints on the network topology, scheduling real-time and non-real-time data packet could not perform as expected as on other type of the wireless environment, eventually, could have an impact on the network performance in terms of power degradation and average latency. This paper presents the base packet scheduling structure alongside the classification of packet scheduling algorithms and examples of most well-known algorithms for each class of scheduling scheme. Additionally, the authors highlight some weak points in each presented example in order to give a deep understanding of packet scheduling classes and algorithms.

The average queue delay of optical packet increases when the packets block in the virtual output queue header in the asynchronous optical packet switching. We analyze the character of variable length Internet packet and propose the preemptive short packets priority (PSPP) algorithm for reducing the total packet queue time in the queue. In the PSPP algorithm, the short packet can preempt the transmission time of the long packet and can be served first. The analysis and the simulation showed the PSPP algorithm can make the average queue time of the short packet decrease to zero almost and can decrease the total packet average queue time largely when the traffic load was middle and low. The PSPP could guarantee the low average queue delay for the real -time TCP traffic.

With the applications of Internet of Things (IoT) for smart cities, the real-time performance for a large number of network packets is facing serious challenge. Thus, how to improve the emergency response has become a critical issue. However, traditional packet scheduling algorithms cannot meet the requirements of the large-scale IoT system for smart cities. To address this shortcoming, this paper proposes EARS, an efficient data-emergency-aware packet scheduling scheme for smart cities. EARS describes the packet emergency information with the packet priority and deadline. Each source node informs the destination node of the packet emergency information before sending the packets. The destination node determines the packet scheduling sequence and processing sequence according to emergency information. Moreover, this paper compares EARS with a first-come, first-served, multilevel queue algorithm and a dynamic multilevel priority packet scheduling algorithm. Simulation results show that EARS outperforms these previous scheduling algorithms in terms of packet loss rate, average packet waiting time, and average packet end-to-end delay.

In order to build a mobile ad hoc network, only some battery powered nodes are required to be deployed; no additional infrastructure is necessary. Routing, scheduling etc. algorithms for this kind of networks have to be designed in such a manner that battery power in nodes can be saved, that is, algorithms should be energy efficient. In most cases, an external priority is assigned to data packets which we have to take care. Along with that, in WEEP, we consider residual lifetime of source and destination nodes to estimate whether the current packet can be sent to destination. Moreover, if possibility of completion of a communication session is less, then more weight is given to data packets belonging to those sessions. The present article aims at reducing the number of route discovery sessions through energy efficient scheduling policy. Therefore, packets travelling through paths consisting of delicate links, are blessed with more priority than those travelling through strong wireless links. Simulation results show that performance improvement produced by WEEP is very significant compared to other protocols used to scheduling in ad hoc networks. As far as the authors know, WEEP is a unique scheduling algorithm in ad hoc network that considers expected residual lifetime of destination along with performance and fragility of a path.