一种环境感知的无线Mesh网络自适应QoS路径选择算法

本文针对如何改善无线多跳Mesh网络的服务质量,满足无线多媒体业务对数据传输的带宽、时延、抖动的要求等问题,研究了一种基于无线信道状态和链路质量统计的MAC层最大重传次数的自适应调整算法。该算法通过对无线Mesh网络的无线信道环境的动态感知,利用分层判断法区分无线分组丢失的主要原因是无线差错还是网络拥塞导致,实时调整MAC层的最佳重传次数,降低无线网络中的分组冲突概率。基于链路状态信息的统计和最大重传策略,提出了一种启发式的基于环境感知的QoS路由优化机制HEAOR。该算法通过动态感知底层链路状态信息,利用灰色关联分析法自适应选择最优路径,在不增加系统复杂度的基础上,减少链路误判概率,提高传输效率。NS2仿真结果表明,HEAOR算法能有效减少重路由次数,降低链路失效概率,提高网络的平均吞吐率。本文提出的方法不仅能够优化MAC层的重传,而且通过发现跨层设计的优化参数实现对路径的优化选择。      

A Cross-Layer Routing Scheme Using Adaptive Retransmission Strategy for Wireless Mesh Networks

In this paper, we develop a link quality-based adaptive adjustment mechanism of the MAC maximum retransmission count to reduce collision probability of wireless Mesh networks. Based on statistics acquired in the link layer and the retransmission strategy, a multi-metric cross-layer on-demand routing scheme is proposed for wireless Mesh networks. The proposed scheme uses information such as available link bandwidth, node residual load rate and transmission efficiency of a path adequately to cross-layer routing. The network layer can adaptively select an optimal path to deliver packets based on the acquired statistics of the MAC layer. Extensive simulation results demonstrate that the proposed scheme can reduce link failure probability, improve network throughput, and decrease the end-to-end delay effectively.     

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.     

Distributed adaptive rate system for wireless packet networks

A distributed adaptive rate system for wireless packet networks is proposed. Compared to the centralized adaptive rate system that needs to know the packet retransmission probability to maximize the throughput, this system need not know the packet retransmission probability in advance and can achieve the optimal adaptive rate system that maximizes the throughput for wireless packet networks.     

D2D网络中基于立即可解网络编码的时延最小化重传方案

该文针对D2D无线网络中多终端并发协作重传冲突避免问题,提出一种基于立即可解网络编码的时延最小化重传方案。在重传阶段,充分利用D2D无线网络终端协作传输数据的优势,结合各终端数据包接收状态,综合考虑时延的影响因素,选取单次重传时延增量较小的数据包生成编码包,最小化重传时延。同时,构建终端冲突图,在图中搜索极大独立集,根据各终端的编码包权重值,选择最大加权独立集中的终端作为并发协作重传终端,从而降低重传次数。仿真结果表明,所提方案能够进一步改善D2D无线网络的重传效率。     

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.     

一种新的基于多节点网络编码的无线广播重传算法

提出了一种基于网络编码的无线网络广播重传算法。该算法按照包丢失分布概率的特点生成新的重传序列,采用多节点的网络编码方法进行丢失包组合实现重传。通过数学分析和仿真证明,该算法能保证接收节点的编码可解性,同时重传次数可达到局部最优性;与传统重传方法相比,该算法可以有效地减少信息包的平均传输次数,从而提高传输效率。     

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。     

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。     

Optimized Congestion Management Protocol for Healthcare Wireless Sensor Networks

The application of Wireless Sensor Networks (WSNs) in healthcare is dominant and fast growing. In healthcare WSN applications (HWSNs) such as medical emergencies, the network may encounter an unpredictable load which leads to congestion. Congestion problem which is common in any data network including WSN, leads to packet loss, increasing end-to-end delay and excessive energy consumption due to retransmission. In modern wireless biomedical sensor networks, increasing these two parameters for the packets that carry EKG signals may even result in the death of the patient. Furthermore, when congestion occurs, because of the packet loss, packet retransmission increases accordingly. The retransmission directly affects the lifetime of the nodes. In this paper, an Optimized Congestion management protocol is proposed for HWSNs when the patients are stationary. This protocol consists of two stages. In the first stage, a novel Active Queue Management (AQM) scheme is proposed to avoid congestion and provide quality of service (QoS). This scheme uses separate virtual queues on a single physical queue to store the input packets from each child node based on importance and priority of the source’s traffic. If the incoming packet is accepted, in the second stage, three mechanisms are used to control congestion. The proposed protocol detects congestion by a three-state machine and virtual queue status; it adjusts the child’s sending rate by an optimization function. We compare our proposed protocol with CCF, PCCP and backpressure algorithms using the OPNET simulator. Simulation results show that the proposed protocol is more efficient than CCF, PCCP and backpressure algorithms in terms of packet loss, energy efficiency, end-to-end delay and fairness.     

Downlink media streaming with wireless fountain coding in wireline‐cum‐WiFi networks

This paper addresses the problem of streaming packetized media data in a combined wireline/802.11 network. Since the wireless channel is normally the bottleneck for media streaming in such a network, we propose that wireless fountain coding (WFC) be used over the wireless downlink in order to efficiently utilize the wireless bandwidth and exploit the broadcast nature of the channel. Forward error correction (FEC) is also used to combat errors at the application‐layer. We analytically obtain the moment generating function (MGF) for the wireless link‐layer delay incurred by WFC. With the MGF, the expected value of this wireless link‐layer delay is found and used by the access point (AP), who has no knowledge of the buffer contents of wireless receivers, to make a coding‐based decision. We then derive the end‐to‐end packet loss/late probability based on the MGF. We develop an integrated ns‐3/EvalVid simulator to evaluate our proposed system and compare it with the traditional 802.11e scheme which is without WFC capability but equipped with application‐ and link‐layer retransmission mechanisms. Through extensive simulations of video streaming, we show that streaming with WFC is able to support more concurrent video flows compared to the traditional scheme. When the deadlines imposed on video packets are relatively stringent, streaming with WFC also shows superior performance in terms of packet loss/late probability, video distortion, and video frame delay, over the traditional scheme. Copyright © 2011 John Wiley & Sons, Ltd.     

Packet level acknowledgement and Go‐Back‐N protocol performance in infrared wireless LANs

Infrared wireless LANs may employ repetition rate (RR) coding to increase the symbol capture probability at the receiver. This paper examines the effectiveness of RR coding to utilization for infrared LANs using the physical and link layer parameter values proposed in the Advanced Infrared (AIr) protocol standard, which is developed by the Infrared Data Association (IrDA). Infrared LANs employ a Go‐Back‐N (GBN) automatic repeat request (ARQ) retransmission scheme at the Link Control (LC) layer to ensure reliable information transfer. To efficiently implement RR coding, the receiver may return after every DATA packet a suggestion for the suitable RR value to be used by the transmitter and implement a Stop‐and‐Wait (SW) ARQ scheme at the medium access control (MAC) layer. The effectiveness of employing this optional SW ARQ scheme at the MAC layer is discussed. Analytical models for the ARQ retransmission schemes are developed and employed to compare protocol utilization for different link parameter values such as window size, packet length and LC time out periods. This analysis identifies the ARQ protocol that maximizes performance for the specific link quality and the implemented link layer parameters. The effectiveness of the proposed RR coding to LAN utilization for different ARQ scheme implementation is finally explored. This analysis identifies the link quality level at which RR should be adjusted for maximum performance. It is concluded that if the packet error rate is higher than 0.1–0.4 (depending on the implemented ARQ protocol), the receiver should advise the transmitter to double the implemented RR for maximum performance. These error rate values are high and can be effectively estimated by the transmitter based on packet retransmissions. Thus, the usefulness of the receiver indicating to the transmitter to adjust RR is questionable, as the transmitter can effectively implement the suitable RR value based on packet retransmissions. Copyright © 2003 John Wiley & Sons, Ltd.     

一种基于网络编码的重传算法研究

无线网络中,节点发送的数据分组传输失败后,执行重传机制。传统的重传机制ARQ由于在一次重传中只能发送一个丢失的数据分组,因此传输效率比较低。利用网络编码技术和AQR重传机制,我们可以在重传中使用网络编码,广播发送由多个丢失数据分组编码得到的编码分组,从而提高重传效率。本文中我们提出一种将网络编码应用于多个发送方多个接收方(MSMR)无线网络中的算法RMBNC。理论推导和仿真分析验证了我们提出的算法的有效性。     

TCP-friendly flow control of wireless multimedia using ECN marking

In a wireless network packet losses can be caused not only by network congestion but also by unreliable error-prone wireless links. Therefore, flow control schemes which use packet loss as a congestion measure cannot be directly applicable to a wireless network because there is no way to distinguish congestion losses from wireless losses. In this paper, we extend the so-called TCP-friendly flow control scheme, which was originally developed for the flow control of multimedia flows in a wired IP network environment, to a wireless environment. The main idea behind our scheme is that by using explicit congestion notification (ECN) marking in conjunction with random early detection (RED) queue management scheme intelligently, it is possible that not only the degree of network congestion is notified to multimedia sources explicitly in the form of ECN-marked packet probability but also wireless losses are hidden from multimedia sources. We calculate TCP-friendly rate based on ECN-marked packet probability instead of packet loss probability, thereby effectively eliminating the effect of wireless losses in flow control and thus preventing throughput degradation of multimedia flows travelling through wireless links. In addition, we refine the well-known TCP throughput model which establishes TCP-friendliness of multimedia flows in a way that the refined model provides more accurate throughput estimate of a TCP flow particularly when the number of TCP flows sharing a bottleneck link increases. Through extensive simulations, we show that the proposed scheme indeed improves the quality of the delivered video significantly while maintaining TCP-friendliness in a wireless environment for the case of wireless MPEG-4 video.     

Quality-Driven Capacity-Aware Resources Optimization Design for Ubiquitous Wireless Networks

In this paper, we propose a probability-statistical capacity-prediction scheme to provide probabilistic quality-of-service (QoS) guarantees under the high traffic load of IEEE 802.11 wireless multimedia Mesh networks. The proposed scheme perceives the state of wireless link based on the MAC retransmission statistics and calculates the statistical channel capacity especially under the saturated traffic load. Via a cross-layer design approach, the scheme allocates network resource and forwards data packets by taking the interference among flows and the channel capacity into consideration. Extensive experiments have been carried out on the basis of IEEE 802.11 protocols in order to demonstrate the superiority of the proposed scheme over the existing location-based QoS optimization delivery algorithm in terms of retransmission count, successful delivery rate, and end-to-end delay on the condition of time-varying multi-hop wireless links.     

A cross-layer optimized adaptive modulation and coding scheme for transmission of streaming media over wireless links

In this paper, a novel cross-layer Adaptive Modulation and Coding scheme that optimizes the overall packet loss (by both transmission errors and excessive delays) probability under a given arrival process is developed. To this end, an improved Large Deviations approximation for the fraction of packets that suffer from excessive queuing delay is proposed. This approximation is valid for G/G/1 queues with infinite buffers that are driven by stationary arrival and service processes which satisfy certain conditions. Such models can capture the time correlations in the amount of traffic generated by streaming media sources and the time varying service capacity of a wireless link. Through numerical examples, the proposed AMC policy is shown to achieve a significant reduction in the overall packet loss rate compared to previously proposed schemes. This algorithmic performance gain can be translated into a sizeable decrease in the required transmit power or an analogous increase in the rate of the arrival process, subject to a given maximum packet loss rate Quality of Service constraint. Furthermore, the proposed AMC policy can be combined with ARQ in order to achieve an even lower overall packet loss probability.     

A framework of prioritized burst segmentation supporting controlled retransmission in OBS networks

In optical burst switching (OBS) networks, burst contentions in OBS core nodes may cause data loss. To reduce data loss, retransmission scheme has been applied. However, uncontrolled retransmission may increase network load significantly and data loss probability defeating the retransmission purpose. In addition, in a priority traffic existing OBS network, OBS nodes may apply different retransmission mechanisms to priorities bursts for quality-of-service (QoS) support. This study has developed a controlled retransmission scheme for prioritized burst segmentation to support QoS in OBS networks. Unlike previous works in the literature, we have set a different value to retransmission probability at each contention and propose a retransmission analytical model for burst segmentation contention resolution scheme. In addition, we have applied the proposed retransmission scheme to the prioritized burst segmentation for QoS support. We have taken into account the load at each link due to both fresh and retransmitted traffic, and have calculated the path blocking probability and byte loss probability (ByLP) for high-priority and low-priority burst to evaluate network performance. An extensive simulation has been proposed to validate our analytical model.     

Efficient Retransmission Methods in Wireless MAC Protocol for Multicast

Recent studies on reliable wireless multicast have focused on sending acknowledgement packets from all member stations to the source. Although these studies provide methods of improving the reliability, there have not been any studies on retransmission methods for wireless multicast. Multicast packets are retransmitted based on the unicast transmission rule, which retransmits until all members successfully receive the packet. In this paper, an efficient retransmission method is proposed. The retransmission lasts until the target packet delivery ratio of each member is met. Moreover, the contention window size for retransmission is adjusted based on the reception status of the previous transmission. The performance of the proposed wireless multicast is evaluated by extensive simulations.     

A Real Time Transport Scheme for Wireless Multimedia Communications

In wireless communications systems, a mobile station is typically equipped with limited processing capability and buffer space for transmitting and receiving. The radio link is usually found to be noisy and its propagation delay is sometimes non-negligible as compared with the packet transmission delay. And because of the necessity of flow control and packet retransmission upon error, the delay and throughput performance cannot satisfy the need of a particular traffic type, i.e., real-time multimedia. This paper presents a scheme suitable for the above condition, called the Burst-oriented Transfer with Time-bounded Retransmission (BTTR). The present scheme uses a large transmission window for sending/receiving a burst of time-sensitive data and, within this window, another smaller observation window is repeatedly used for error status feedback via the backward channel. There is time limitation on each retransmission such that the burst of data can be received in a timely manner, however, with some degradation on the packet loss rate. An analysis is given in terms of the expectations of delay, throughput, and packet drop rate. A comparison with an error-free link protocol will also be given. The result shows that the proposed scheme can meet the delay and throughput requirement under reasonable packet drop rate.     

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.