Exploiting slack time for just-in-time scheduling in wireless sensor networks

We consider the problem of real-time data collection in wireless sensor networks, in which data need to be delivered to one or more sinks within end-to-end deadlines. To enhance performance with respect to end-to-end deadline miss ratio, existing approaches schedule packets by prioritizing them based on per-packet deadlines and other factors such as the distance to the sink. However, important factors affecting the end-to-end performance such as queuing delays and buffer overruns have largely been ignored in the existing real-time schemes. Packet prioritization by itself cannot assist with these issues, and may in fact, exacerbate them for real-time data collection, since many high priority packets may simultaneously contend for the constrained network resources. In sensor networks, where the channel bandwidth and buffer space are often quite limited, these issues can dramatically impact real-time performance. Based on this observation, we propose Just-in-Time Scheduling (JiTS) strategies where packets are judiciously delayed within their slack time to reduce contention and load balance the use of the network buffers. We explore several policies for delaying data packets at different intermediate nodes considering potential contention. In addition, we also show that the routing protocol has a significant impact on real-time performance. In particular, shortest path routing leads to considerably better performance than geographic forwarding, which is often used for real-time data transmission in wireless sensor networks. Using an extensive simulation study, we demonstrate that JiTS can significantly improve the deadline miss ratio and packet drop ratio compared to two state-of-the-art approaches for real-time packet delivery for sensor networks (RAP and SPEED) under various scenarios. Notably, JiTS requires neither lower layer (e.g., MAC layer) support nor synchronization among the sensor nodes.     

Source rate encoding mechanisms for binary congestion control

This paper focus on congestion control for best-effort packet-switching networks, where congested routers use 1 bit per packet to communicate with sources. Sources adapt their rates according to the sequence of bits received. Routers do not keep per-flow information but perform selective marking based on the source rate value inserted in each packet. We propose a new strategy for source rate encoding in forward packets, directly applicable to existing network protocols (e.g. IP). The scheme supports differentiated classes with respect to rate allocation. We test, by simulation, this encoding mechanism as well as the performance of the router and source algorithms.     

A versatile model for predicting the performance of deflection-routing networks

Deflection routing can be used in networks whose stations have the same number of input and output links. Fixed-length packets arrive synchronously on the station's input links at the beginning output link that offers the shortest path to its destination. Since the number of packet buffers at each output link is finite, the simultaneous contention of two packets for the last buffer of the common output link must be resolved by “deflecting” one of the packets according to a specified criterion (e.g. at random, by destination proximity, or by packet age). Deflection routing can therefore be used with as few as one packet buffer per output link.

The potentially unbounded number of routes that a given packet can take makes analyzing the performance of such networks difficult. Using independence assumptions, we develop an efficient, high-fidelity performance model of deflection routing that allows us to estimated the mean end-to-end packet delay in a network that has any given two-connected topology, a single packet buffer at each output port, and an arbitrary traffic matrix.     

Performance analysis of some random access systems having an intelligent central node

We consider a computer communication network in which users communicate with each other via a central node. The available channel is divided into the up- and downlink subchannels. Users contend for the uplink according to the pure ALOHA protocol. The central node is capable of recognizing erroneous packets, rejecting duplicate packets, and storing error-free packets. Given that the total available channel capacity is the limiting resource, the system throughput and the average packet delay are derived. Aiming at the improvement of system performance, we found the channel capacity divisions that maximize the throughput and minimize the average packet delay (numerically). In order to obtain the above-mentioned performance characteristics, we have assumed the infinite users' population model.     

Multiple-access insights from bounds on sensor localization

In ad hoc location-aware sensor networks, unlocalized sensors can estimate their locations based on the triangulation of range measurements from location-aware sensors or “anchors”, whose locations are known or estimated a priori. In this paper, we investigate the relationship between multiple-access design parameters and the accuracy of location estimation in ad hoc sensor networks. Bounds on the average localization accuracy in a packet-based sensor network with time-of-arrival (TOA) based distance estimation are used to investigate the connection between the average effective throughput of packets and the average localization accuracy. On this basis, we (i) develop an analytical framework that allows us to analyze the relationship between network parameters and the average localization accuracy obtained in ad hoc sensor networks with a spread-spectrum physical layer, and (ii) show that, for such networks, minimizing the average localization error is equivalent to maximizing the average effective throughput. We further demonstrate that the developed framework allows the extraction of the optimal network parameters that maximize localization accuracy. The trends in the localization accuracy with respect to the network parameters observed through analysis are then validated via simulation studies. Finally, some aspects of the modeling of location-aware sensor networks that warrant investigation and require more sophisticated modeling strategies are listed.     

Packet scheduling with joint design of MIMO and network coding

In this paper, we propose a joint design of MIMO technique and network coding (MIMO-NC) and apply it to improve the performance of wireless networks. We consider a system in which the packet exchange among multiple wireless users is forwarded by a relay node. In order to enjoy the benefit of MIMO-NC, all the nodes in the network are mounted with two antennas and the relay node possess the coding capability. For the cross traffic flows among any four users, the relay node not only can receive packets simultaneously from two compatible users in the uplink (users-to-relay node), but also can mix up distinct packets for four destined users into two coded packets and concurrently send them out in the same downlink (relay node-to-users), so that the information content is significantly increased in each transmission. We formalize the problem of finding a schedule to forward the buffered data of all the users in minimum number of transmissions in such a system as a problem of finding a maximum matching in a graph. We also provide an analytical model on maximum throughput and optimal energy efficiency, which explicitly measures the performance gain of the MIMO-NC enhancement. Our analytical and simulation results demonstrate that system performance can be greatly improved by the efficient utilization of MIMO and network coding opportunities.     

Mean waiting time evaluation of packet switches for centrally controlled PBX''s

The performance analysis of packet switched networks usually does not account for the loss of transmission bandwidth caused by the switch processing time. In wide area networks, where transmission channels are of the order of 64 kbps or less, this is a valid approach since switch processing time is negligible compared to average transmission times for packets. However, in Local Area Networks (LANs), the transmission channels are usually of the order of 1 mbps or greater and thus when evaluating the performance of a LAN that is based on a packet switch, it is necessary to consider the effect of the switch processing time which is now comparable to that of the average packet's transmission time. This paper investigates the effect of this processing time on network performance, and an exhaustive cyclic queueing algorithm to minimise this effect on overall network performance is discussed. An approximate analytical technique for evaluating the performance of such a LAN is derived. This technique makes it possible to do interactive calculations which can be used in the provisioning and configuring of such networks.     

基于最近社交圈的社交时延容忍网络路由策略

无稳定拓扑使时延容忍网络(delay tolerant networks, DTN)路由协议主要通过增加冗余数据包副本提高路由性能.社交网络是DTN的一种典型应用场景,但由于其网络规模相对较大,当网络负载高时,通常的DTN路由不能有效控制数据包副本的数量,从而产生大量丢包导致性能下降.借鉴MANET网络中利用分簇结构控制网络冗余路由数据包的思想,通过分析社交网络中节点的移动模型,定义了在社交关系的约束下,聚合移动规律相近的节点构成最近社交圈的节点簇组成策略.提出了一种基于该分簇结构的分为簇外喷射、簇间转发和簇内传染3个阶段的社交时延网络路由协议.实验证明,这种基于最近社交圈分簇结构的路由能有效地控制冗余数据包副本的产生,并在高网络负载的情况下仍然能够达到较好的性能.     

Delay Constraint Error Control Protocol for Real-Time Video Communication

Real-time video communication over wireless channels is subject to information loss since wireless links are error-prone and susceptible to noise. Popular wireless link-layer protocols, such as retransmission (ARQ) based 802.11 and hybrid ARQ methods provide some level of reliability while largely ignoring the latency issue which is critical for real-time applications. Therefore, they suffer from low throughput (under high-error rates) and large waiting-times leading to serious degradation of video playback quality. In this paper, we develop an analytical framework for video communication which captures the behavior of real-time video traffic at the wireless link-layer while taking into consideration both reliability and latency conditions. Using this framework, we introduce a delay constraint packet embedded error control (DC-PEEC) protocol for wireless link-layer. DC-PEEC ensures reliable and rapid delivery of video packets by employing various channel codes to minimize fluctuations in throughput and provide timely arrival of video. In addition to theoretically analyzing DC-PEEC, the performance of the proposed scheme is analyzed by simulating real-time video communication over “real” channel traces collected on 802.11b WLANs using H.264/AVC JM14.0 video codec. The experimental results demonstrate performance gains of 5–10 dB for different real-time video scenarios.      

Tabu marking scheme to speedup IP traceback

The IP traceback is an important mechanism in defending against distributed denial-of-service (DDoS) attacks. In this paper, we propose a probabilistic packet marking (PPM) scheme, Tabu Marking Scheme (TMS), to speedup IP traceback. The key idea of “tabu mark” is that, a router still marks packets probabilistically, but regards a packet marked by an upstream router as a tabu and does not mark it again. We study the impact of the traffic aggregation on the convergence behavior of PPM schemes. Furthermore we derive a new analytical result on the partial coupon collection problem, which is a powerful tool applicable for computing the mean convergence time of any PPM scheme. Our study shows that the idea of “tabu mark” not only helps a PPM scheme that allows overwriting to reduce the convergence time under a DDoS attack, but also ensures the authentication of the routers’ markings.     

基于多核平台的高速网络流量实时捕获方法

随着互联网上应用的丰富和网络带宽的增长,带来的安全问题也与日剧增,除了传统的垃圾邮件、病毒传播、DDoS攻击外,还出现了新型的隐蔽性强的攻击方式.网络探针工具是一种部署在局域网出口处的旁路设备,能够收集当前进出网关的全部流量并进行分析,而网络探针工具中最重要的模块就是数据包的捕获.传统的Linux网络协议栈在捕获数据包时有诸多性能瓶颈,无法满足高速网络环境的要求.介绍了基于零拷贝、多核并行化等技术的多种新型的数据包捕获引擎,并基于Intel DPDK平台设计并实现了一个可扩展的数据包捕获系统,它能够利用接收端扩展(receiver-side scaling, RSS)技术实现多核并行化的数据包捕获、模块化的上层处理流程.除此之外,还讨论了更有效、更公平的将数据包分发到不同的接收队列所应使用的Hash函数.经过初步的实验验证,该系统能够实现接近线速的收包并且多个CPU核心间实现负载均衡.     

Modelling and performance analysis of multi-hop ad hoc networks

Mobile ad hoc networks are becoming very attractive and useful in many kinds of communication and networking applications. Due to the advantage of numerical analysis, analytical modelling formalisms, such as stochastic Petri nets, queuing networks and stochastic process algebra have been widely used for performance analysis of communication systems. To the best of our knowledge, there is no previous analytical study that analyses the performance of multi-hop ad hoc networks, where mobile nodes move according to a random mobility model in terms of the end-to-end delay and throughput. This work presents a novel analytical framework developed using stochastic reward nets for modelling and analysis of multi-hop ad hoc networks, based on the IEEE 802.11 DCF MAC protocol, where mobile nodes move according to the random waypoint mobility model. The proposed framework is used to analyse the performance of multi-hop ad hoc networks as a function of network parameters such as the transmission range, carrier sensing range, interference range, number of nodes, network area size, packet size, and packet generation rate. The proposed framework is organized into several models to break up the complexity of modelling the complete network, and make it easier to analyse each model as required. The framework is based on the idea of decomposition and fixed point iteration of stochastic reward nets. The proposed models are validated using extensive simulations.     

Multi-Sensor Estimation for Unreliable Wireless Networks with Contention-Based Protocols

The state estimation plays an irreplaceable role in many real applications since it lays the foundation for decision-making and control. This paper studies the multi-sensor estimation problem for a contention-based unreliable wireless network. At each time step, no more than one sensor can communicate with the base station due to the potential contention and collision. In addition, data packets may be lost during transmission since wireless channels are unreliable. A novel packet arrival model is proposed which simultaneously takes into account the above two issues. Two scenarios of wireless sensor networks (WSNs) are considered: the sensors transmit the raw measurements directly and the sensors send the local estimation instead. Based on the obtained packet arrival model, necessary and sufficient stability conditions of the estimation at the base station side are provided for both network scenarios. In particular, all offered stability conditions are expressed by simple inequalities in terms of the packet arrival rates and the spectral radius of the system matrix. Their relationships with existing related results are also discussed. Finally, the proposed results are demonstrated by simulation examples and an environment monitoring prototype system.     

A multi-path mechanism for reliable VoIP transmission over wireless networks

With the advances in audio encoding standards and wireless access networks, voice over IP (VoIP) is becoming quite popular. However, real-time voice data over lossy networks (such as WLAN and UMTS), still posses several challenging problems because of the adverse effects caused by complex network dynamics. One approach to provide QoS for VoIP applications over the wireless networks is to use multiple paths to deliver VoIP data destined for a particular receiver. This paper introduced cmpSCTP, a transport layer solution for concurrent multi-path transfer that modifies the standard stream control transmission protocol (SCTP). The cmpSCTP aims at exploiting SCTP’s multi-homing capability by selecting several best paths among multiple available network interfaces to improve data transfer rate to the same multi-homed device. Through the use of path monitoring and packet allotment techniques, cmpSCTP tries to transmit an amount of packets corresponding to the path’s ability. At the same time, cmpSCTP updates the transmission strategy based on the real-time information of all of paths. Using cmpSCTP’s flexible path management capability, we may switch the flow between multiple paths automatically to realize seamless path handover. The theoretical analysis evaluated the model of cmpSCTP and formulated optimal traffic fragmentation of VoIP data. Extensive simulations under different scenarios using OPNET verified that cmpSCTP can effectively enhance VoIP transmission efficiency and highlighted the superiority of cmpSCTP against the other SCTP’s extension implementations under performance indexes such as throughput, handover latency, packet delay, and packet loss.     

Traceback in wireless sensor networks with packet marking and logging

In a hostile environment, sensor nodes may be compromised and then be used to launch various attacks. One severe attack is false data injection which is becoming a serious threat to wireless sensor networks. An attacker uses the compromised node to flood the network and exhaust network resources by injecting a large number of bogus packets. In this paper, we study how to locate the attack node using a framework of packet marking and packet logging. We propose a combined packet marking and logging scheme for traceback (CPMLT). In CPMLT, one packet can be marked by up to M nodes, each node marks a packet with certain probability. When one packet is marked by M nodes, the next marking node will log this packet. Through combining packet marking and logging, we can reconstruct the entire attack path to locate the attack node by collecting enough packets. In our simulation, CPMLT achieves fast traceback with little logging overhead.     

A routing framework for load balancing of bandwidth sensitive traffic in differentiated service networks

The Differentiated Service (DiffServ) network model has been defined as a scalable framework for providing Quality of Service to applications. In this model, traffic is classified into several service classes with different priorities inside queues of IP routers. The premium service class has the highest priority. Due to the high priority of premium traffic, the global network behaviour against this service class, including routing and scheduling of premium packets, may impose significant influences on traffic of other classes. These negative influences, which could degrade the performance of low-priority classes with respect to some important metrics such as the packet loss probability and the packet delay, are often called the inter-class effects. To reduce the inter-class effects, the premium-class routing algorithm must be carefully selected such that (1) it works correctly (i.e., without loop) under the hop-by-hop routing paradigm; and (2) the congestion resulted from the traffic of premium class over the network becomes minimum. In this paper, we first introduce a novel routing framework, named compatible routing, that guarantees loop-freedom in the context of hop-by-hop routing model. Then, upon this framework, we propose two multipath architectures for load balancing of high-priority traffic on DiffServ networks. Our extensive simulations clearly demonstrate that the proposed methods distribute the premium bandwidth requirements more efficiently over the whole network and perform better than the existing algorithms, especially in the case of complex and highly loaded networks.     

The impact of local policies on the quality of packet routing in paths,trees, and rings

We consider the packet routing problem in store-and-forward networks whose topologies are either paths, trees, or rings. We are interested by the quality of the solution produced, with respect to a global optimal solution, if each link uses a (fixed) local policy to schedule the packets which go through it. The quality of the derived solutions is measured using the worst case analysis for two global optimality criteria, namely the maximum arrival date of a packet at its destination (or makespan) and the average arrival date of the packets at their destinations. We consider the setting where n packets, each one having a size (or length) and a destination, are released from the same source. In the case of rings, there exist two paths between the source and a destination. Each packet is owned by a user which chooses a path to its destination. We assume that users are rational: knowing the local policy used by the links and the state of the network, a user chooses the path which minimizes the arrival date of its packet at its destination. We are then interested by the quality of the Nash equilibria obtained.     

Fine-Granularity Transmission Distortion Modeling for Video Packet Scheduling Over Mesh Networks

Packet scheduling is a critical component in multi-session video streaming over mesh networks. Different video packets have different levels of contribution to the overall video presentation quality at the receiver side. In this work, we develop a fine-granularity transmission distortion model for the encoder to predict the quality degradation of decoded videos caused by lost video packets. Based on this packet-level transmission distortion model, we propose a content-and-deadline-aware scheduling (CDAS) scheme for multi-session video streaming over multi-hop mesh networks, where content priority, queuing delays, and dynamic network transmission conditions are jointly considered for each video packet. Our extensive experimental results demonstrate that the proposed transmission distortion model and the CDAS scheme significantly improve the performance of multi-session video streaming over mesh networks.