无线Mesh网络中跨层协作的路由策略

网络编码可以提高无线Mesh网络的吞吐量,但是网络编码在无线Mesh网络中实际应用获得最大网络利用率是需要解决的问题。提出一种多路径策略,能够通过将网络编码和TCP进行最大化融合提高网络的利用率。网络编码被加入到现有的网络系统,通过解决速率控制问题和分组调度问题,调整源节点的数据编码分块,降低数据包重传的次数,提高网络的吞吐量。     

基于二进制网络编码的无线广播重传方法

为了提高无线广播网络中数据包的传输效率,提出了一种新的基于二进制网络编码的高效无线广播重传方法(WBRBNC).法基于数据包分布矩阵(PDM),在广播重传过程中使用一种新的数据包算法选取编码包,对这些丢包采用二进制网络编码方法进行编码,然后再进行广播,使一次广播重传可以使多个具有不同丢包的接收机受益.分析和仿真表明,该方法能有效保证接收节点的编码可解性,使重传次数显著减少,更接近于理论值,从而大大提高了传输效率.另外,该方法计算开销小,很适合应用在卫星广播网和无线传感器网等资源受限的系统中.     

WiNoC中EF-ACK容错无线接口设计

无线片上网络中的无线信道面临着严重的可靠性挑战,无线路由器的容错设计对整个片上网络的传输效率有着较大的影响.本文提出一种EF-ACK容错无线接口设计,将多条确认信息配置在一个数据包内,通过无线信道传递确认信息数据包;在无线接口处设立重传数据缓冲区,以更高效的方式确认数据以及控制错误数据包的重传;另外,提出了基于网络状态的编解码控制,在网络情况较差时用BCH编码的方式提高数据的鲁棒性.实验表明,本文方案使用了较小的额外面积和功耗开销,高效地完成了对于数据的无线确认反馈,且在错误率较高时,可以保证网络中较低的网络延迟和较高的饱和吞吐量,大大提高了网络的性能.     

基于随机网络编码的无线广播重传方案及性能分析

针对自动重复重传(ARQ)机制在无线广播系统中吞吐量性能不佳的缺陷,提出一种基于随机网络编码的广播重传方案RNC-ARQ.对于广播节点,采用随机线性码对所有丢失包进行编码组合重传.对于接收节点,当接收的编码包累积到一定数量后可通过解码操作恢复出原始数据.该方案可有效减少重传次数,改善无线广播的吞吐量性能.基于Gilbert-Elliott模型描述的突发错误信道,建立了信道状态和节点接收处理流程合并的多状态马尔可夫模型,并以此为基础推导了RNC-ARQ方案的吞吐量闭合解.最后,使用NS-2模拟器评估RNC-ARQ方案的性能,结果表明在突发差错信道下,基于随机网络编码重传方案的吞吐量优于传统的选择承传ARQ方案和基于异或编码的重传方案.     

一种改善TCP性能的链路层重传方案

有线网络中TCP拥塞控制机制是建立在网络丢包的基础之上的,所以该机制不能适应无线网络中高误码率造成的无线链路丢包的情况。无线链路层重传技术是改善网络性能因无线信道误码率较高而下降的一项重要措施。文中研究了WCDMA无线网络中链路层重传技术对无线TCP数据传输的影响,比较两种重传方案,通过OPNET仿真技术对其进行仿真比较,得出其中一种更有效的改善TCP传输性能的方案。     

中继协作无线网络中不完美反馈下基于网络编码的重传方案

无线广播网络传输过程中,目的节点反馈信息丢失或部分丢失导致发送节点不能了解目的节点的真实接收状态.为提高不完美反馈下无线网络的重传效率,本文提出中继协作无线网络中不完美反馈下基于网络编码的重传方案.本方案基于部分可观察马尔科夫决策过程对不完美反馈下的重传过程进行建模.发送节点根据系统观测状态和最大置信度更新系统估计状态,根据数据包发送顺序,优先选择最早丢失且能够恢复最多丢包的编码包重传.目的节点缓存不可解编码包以提升编解码机会.重传过程中源节点关注目的节点请求包需求,相同情况优先选择传输可靠性较高的中继节点,以提升传输有效性.仿真结果表明,在不完美反馈下相对于传统方案,本方案可有效提高重传效率.     

无线网络中基于机会网络编码的加权广播重传

在无线广播网链路状态不同和丢包率高的条件下,基于机会网络编码的数据分发策略面临传输效率低和计算复杂度高的问题。针对这一问题,该文提出一种新的基于机会网络编码的加权广播重传(Weighted Opportunistic Network Coding Retransmission, WONCR)方案。该方案通过构建加权数据包分布矩阵(Weighted Packet Distribution Matrix, WPDM),在重传过程中采用新的调度算法进行编码数据包的选取,并将选取的数据包进行XOR编码后再重传。机会仿真结果表明,WONCR方案提高了传输效率,且计算开销较小,实现了无线广播网中高效、可靠的数据分发。     

一种基于机会式网络编码的高效广播重传方法

为了提高无线广播网络中数据传输的效率,该文提出了一种新颖的基于机会式网络编码的重传方法。将机会式网络编码技术应用于丢包的重传,并采用高效的丢包组合策略生成重传包。根据网络终端的丢包情况,首先创建丢包的哈希表,再根据哈希表快速选择满足一定编码条件的丢包以生成重传数据包,从而在提高重传性能的同时,有效地降低了重传方法的复杂度。仿真结果表明该方法相比已有算法能有效地减少重传次数,并提高重传包发送和接收的效率。     

一种新的基于虚拟队列的无线多播网络编码调度策略

网络编码由于其传输效率高的特性,近年来在无线多播网络中得到广泛的应用。针对无线多播网络中丢包自动重传效率低的问题,该文提出一种新的基于虚拟队列中数据包到达时间的编码调度策略(CSAT)。在CSAT策略中,为了提高编码效率,采用虚拟队列来存放初始以及未被所有接收者接收到的数据包。考虑到队列的稳定性,CSAT策略按照一定的比率从主次队列选择发送;在次队列发送数据包时,结合了编码和非编码两种方式,根据数据包到达队列的先后,选取能够使较多数据包参与编码的方式发送。仿真结果表明,该文所提的CSAT编码调度策略在有效提高了数据包传输效率的同时,提高了网络的吞吐量并降低了平均等待时延。     

一种应用于Ad Hoc无线局域网的随机接人协议

本文分析了应用于Ad Hoe无线局域网的BSAC(Buffered Slotted ALOHA CDMA)随机接入协议，并且首次提出了采用Markov(马尔可夫)链方法的分析模型．此模型应用两个Markov链模型，一个表示节点中的M／M／I／k排队模型，另一个表示网络中活动节点数量．两个Markov链模型通过节点空闲概率相互联系．在此模型基础上，本文详细分析了扩频增益、队列长度与最大允许重传次数等输入参数对网络吞吐量、平均延迟与丢包概率等性能尺度的影响，推导得出了BSAC协议的吞吐量极限．另外，本文还引入多数据包接收技术一延迟捕获技术，该项技术可以有效降低数据包冲突概率，相对于没有采用延迟捕获技术的BSAC协议，平均提高吞吐量29．1％，最大吞吐量提高20．8％，使网络性能接近于理论极限．     

Detecting spurious timeouts in wireless cellular networks using DS‐Agent

Large and sudden variations in packet transmission delays are often unavoidable in wireless networks. Such large delays, refer to as delay spikes (DSs), are likely to exceed several times the typical network round‐trip‐time figures, which can cause TCP spurious timeouts. The spurious timeouts lead to unnecessary retransmissions and reduction of the TCP sender's transmission rate, and degradation of TCP throughput. In this paper we propose a new scheme called DS‐Agent. The spurious timeout is detected by a DS‐Agent and thus TCP sender can response to this spurious timeout accordingly. The simulation results show the better performance of DS‐Agent scheme compared with F‐RTO and TCP Reno in the presence of DSs which is caused by mobility. Copyright © 2006 John Wiley & Sons, Ltd.     

TCP Performance in IEEE 802.11-Based Ad Hoc Networks with Multiple Wireless Lossy Links

We propose a packet-level model to investigate the impact of channel error on the transmission control protocol (TCP) performance over IEEE-802.11-based multihop wireless networks. A Markov renewal approach is used to analyze the behavior of TCP Reno and TCP Impatient NewReno. Compared to previous work, our main contributions are listed as follows: 1) modeling multiple lossy links, 2) investigating the interactions among TCP, Internet Protocol (IP), and media access control (MAC) protocol layers, specifically the impact of 802.11 MAC protocol and dynamic source routing (DSR) protocol on TCP throughput performance, 3) considering the spatial reuse property of the wireless channel, the model takes into account the different proportions between the interference range and transmission range, and 4) adopting more accurate and realistic analysis to the fast recovery process and showing the dependency of throughput and the risk of experiencing successive fast retransmits and timeouts on the packet error probability. The analytical results are validated against simulation results by using GloMoSim. The results show that the impact of the channel error is reduced significantly due to the packet retransmissions on a per-hop basis and a small bandwidth delay product of ad hoc networks. The TCP throughput always deteriorates less than ~ 10 percent, with a packet error rate ranging from 0 to 0.1. Our model also provides a theoretical basis for designing an optimum long retry limit for IEEE 802.11 in ad hoc networks.     

TCP Veno: TCP enhancement for transmission over wireless access networks

Wireless access networks in the form of wireless local area networks, home networks, and cellular networks are becoming an integral part of the Internet. Unlike wired networks, random packet loss due to bit errors is not negligible in wireless networks, and this causes significant performance degradation of transmission control protocol (TCP). We propose and study a novel end-to-end congestion control mechanism called TCP Veno that is simple and effective for dealing with random packet loss. A key ingredient of Veno is that it monitors the network congestion level and uses that information to decide whether packet losses are likely to be due to congestion or random bit errors. Specifically: (1) it refines the multiplicative decrease algorithm of TCP Reno-the most widely deployed TCP version in practice-by adjusting the slow-start threshold according to the perceived network congestion level rather than a fixed drop factor and (2) it refines the linear increase algorithm so that the connection can stay longer in an operating region in which the network bandwidth is fully utilized. Based on extensive network testbed experiments and live Internet measurements, we show that Veno can achieve significant throughput improvements without adversely affecting other concurrent TCP connections, including other concurrent Reno connections. In typical wireless access networks with 1% random packet loss rate, throughput improvement of up to 80% can be demonstrated. A salient feature of Veno is that it modifies only the sender-side protocol of Reno without changing the receiver-side protocol stack.     

TCP with delayed ack for wireless networks

This paper studies the TCP performance with delayed ack in wireless networks (including ad hoc and WLANs) which use IEEE 802.11 MAC protocol as the underlying medium access control. Our analysis and simulations show that TCP throughput does not always benefit from an unrestricted delay policy. In fact, for a given topology and flow pattern, there exists an optimal delay window size at the receiver that produces best TCP throughput. If the window is set too small, the receiver generates too many acks and causes channel contention; on the other hand, if the window is set too high, the bursty transmission at the sender triggered by large cumulative acks will induce interference and packet losses, thus degrading the throughout. In wireless networks, packet losses are also related to the length of TCP path; when traveling through a longer path, a packet is more likely to suffer interference. Therefore, path length is an important factor to consider when choosing appropriate delay window sizes. In this paper, we first propose an adaptive delayed ack mechanism which is suitable for ad hoc networks, then we propose a more general adaptive delayed ack scheme for ad hoc and hybrid networks. The simulation results show that our schemes can effectively improve TCP throughput by up to 25% in static networks, and provide more significant gain in mobile networks. The proposed schemes are simple and easy to deploy. The real testbed experiments are also presented to verify our approaches. Furthermore, a simple and effective receiver-side probe and detection is proposed to improve friendliness between the standard TCP and our proposed TCP with adaptive delayed ack.     

Novel Fastest Retransmission and Rate Control Schemes for Improving TCP Performance in Wireless Ad Hoc Networks

A number of different authors have considered the problem of performance degradation of transmission control protocol (TCP) in wireless ad hoc networks. We herein show that pauses in packet transmission due to packet losses are the fundamental cause of performance degradation of TCP in wireless ad hoc networks. To minimize the duration of packet transmission pauses, we propose a fast retransmission scheme for improving TCP performance in consideration of the inter-operability of previously deployed TCPs in wireless ad hoc networks. We also propose an additional rate control scheme for TCPs to reduce the probability of packet contention. Using OPNET and NS2 simulations, we show that our proposed schemes can provide a much better performance than conventional TCPs.     

End-to-end optimized TCP-friendly rate control for real-time video streaming over wireless multi-hop networks

Rate control is an important issue in video streaming applications. The most popular rate control scheme over wired networks is TCP-Friendly Rate Control (TFRC), which is designed to provide optimal transport service for unicast multimedia delivery based on the TCP Reno’s throughput equation. It assumes perfect link quality, treating network congestion as the only reason for packet losses. Therefore, when used in wireless environment, it suffers significant performance degradation because of packet losses arising from time-varying link quality. Most current research focuses on enhancing the TFRC protocol itself, ignoring the tightly coupled relation between the transport layer and other network layers. In this paper, we propose a new approach to address this problem, integrating TFRC with the application layer and the physical layer to form a holistic design for real-time video streaming over wireless multi-hop networks. The proposed approach can achieve the best user-perceived video quality by jointly optimizing system parameters residing in different network layers, including real-time video coding parameters at the application layer, packet sending rate at the transport layer, and modulation and coding scheme at the physical layer. The problem is formulated and solved as to find the optimal combination of parameters to minimize the end-to-end expected video distortion constrained by a given video playback delay, or to minimize the video playback delay constrained by a given end-to-end video distortion. Experimental results have validated 2–4 dB PSNR performance gain of the proposed approach in wireless multi-hop networks by using H.264/AVC and NS-2.     

A novel cross-layer strategy of video transmission to solve performance anomaly in multi-rate IEEE 802.11 networks

When wireless hosts use different rates to transmit data in IEEE 802.11 networks, it will take on the state of performance anomaly which will severely decrease the throughputs of all the higher rate hosts. Hence, it is bad for video service transmission. Considering that video is very sensitive to packet delivery delay but can tolerate some packet losses, we propose a novel cross-layer scheme which takes these two characteristics into consideration. Firstly, the maximum number of retransmissions for a video Medium Access Control (MAC) frame is computed in MAC layer according to video frame rate requirement of application layer and current access delay of MAC layer. Secondly, within the margin of the tolerant Packet Loss Rate (PLR) of application layer, several video MAC frames are allowed to drop so that we can adaptively select the transmission rate as high as possible for the rest of video MAC frames in terms of current channel quality and the maximum number of retransmissions. Experiment results show that the proposed method can reduce the delay and jitter of video service and improve the throughputs of fast hosts. Therefore, it increases the quality of reconstructed video to a certain extent and relieves the performance anomaly of network effectively.     

An energy‐aware transmission mechanism for WiFi‐based mobile devices handling upload TCP traffic

This paper presents an energy‐aware transmission mechanism that improves the throughput and reduces the energy consumption of mobile devices in wired‐cum‐wireless TCP networks. The proposed mechanism places an agent at the base station, which identifies the cause of packet losses in the underlying network. When the mobile device acts as a TCP source, it adjusts the size of the congestion window adaptively according to the cause of packet losses with the aids of the agent in order to improve the transmission performance. In addition, the proposed mechanism lets the communication device to stay in sleep mode after completing the transmission in order to reduce the energy consumption. As a result, the cooperation between the mobile device and the agent improves the transmission performance as well as the energy efficiency greatly. To evaluate the performance of the proposed mechanism, we analyzed the effect of TCP on the communication device for mobile devices and present a power model. With extensive simulations based on the power model, we demonstrate that the proposed mechanism significantly improves the transmission performance, and reduces the energy consumption over a wide range of both wired and wireless packet losses. Copyright © 2008 John Wiley & Sons, Ltd.     

Joint Transmission Rate Control and Opportunistic Network Coding Over Wireless Networks

Existing opportunistic network coding architectures (e.g., COPE) rely on pseudobroadcast to deliver a coded packet to multiple receivers in a single transmission. Only the primary receiver acknowledges the reception by MAC-layer acknowledgements (synchronous ACKs) and the other receivers receive the coded packet by overhearing and acknowledge the reception by asynchronous ACKs, which are usually piggybacked in outgoing data packets. In realistic wireless networks, this mechanism may cause unnecessary retransmissions if asynchronous ACKs are dropped due to packet losses or arrive late and thus compromise the throughput gain brought by network coding. In this paper, we propose a framework of joint rate control and code selection (ORC) to address this issue, aiming at improving the performance gain of opportunistic network coding in wireless networks. The framework of ORC consists of two mechanisms: (1) Rate control: the optimal transmission rate for coded packets is selected by formulating the rate control process as a Finite Horizon Markov Decision Process. (2) Code selection: based on the results of rate selection, the packet combination for forming the coded packet is determined. Numerical results show that ORC can substantially improve the performance gain of opportunistic network coding compared with COPE.     

TCP-Jersey for wireless IP communications

Improving the performance of the transmission control protocol (TCP) in wireless Internet protocol (IP) communications has been an active research area. The performance degradation of TCP in wireless and wired-wireless hybrid networks is mainly due to its lack of the ability to differentiate the packet losses caused by network congestions from the losses caused by wireless link errors. In this paper, we propose a new TCP scheme, called TCP-Jersey, which is capable of distinguishing the wireless packet losses from the congestion packet losses, and reacting accordingly. TCP-Jersey consists of two key components, the available bandwidth estimation (ABE) algorithm and the congestion warning (CW) router configuration. ABE is a TCP sender side addition that continuously estimates the bandwidth available to the connection and guides the sender to adjust its transmission rate when the network becomes congested. CW is a configuration of network routers such that routers alert end stations by marking all packets when there is a sign of an incipient congestion. The marking of packets by the CW configured routers helps the sender of the TCP connection to effectively differentiate packet losses caused by network congestion from those caused by wireless link errors. This paper describes the design of TCP-Jersey, and presents results from experiments using the NS-2 network simulator. Results from simulations show that in a congestion free network with 1% of random wireless packet loss rate, TCP-Jersey achieves 17% and 85% improvements in goodput over TCP-Westwood and TCP-Reno, respectively; in a congested network where TCP flow competes with VoIP flows, with 1% of random wireless packet loss rate, TCP-Jersey achieves 9% and 76% improvements in goodput over TCP-Westwood and TCP-Reno, respectively. Our experiments of multiple TCP flows show that TCP-Jersey maintains the fair and friendly behavior with respect to other TCP flows.