多播网络中基于机会网络编码改进的重传方法

针对高链路分组丢失率下HLAR (hash lookup assisted retransmission)等重传方法存在的编码率低、算法性能下降等缺陷,提出了一种改进的基于机会网络编码的广播重传方法.该方法根据接收节点反馈的丢失分组情况,不仅能够通过散列查找快速选择丢失分组组合进行编码重传,并优先重传能让最多接收节点恢复其丢失分组的单个重传分组,而且基于邻域关联充分挖掘编码机会,允许节点从多个重传分组中恢复丢失分组,从而在保持较低算法复杂度的情况下,有效地提高重传性能.仿真结果表明,相对于已有算法,该方法能有效减少重传次数,提高传输效率.     

基于实测流量的网际分组丢失率分析

分组丢失率是评价网络性能的一个重要指标。提出了一种利用完整TCP流计算实测数据中分组丢失率的方法,并基于实测数据对CERNET江苏省网与外部不同网络分区的分组丢失率进行分析。算法给出了基于实测数据的分组丢失率包括首次分组丢失率pfl、整体分组丢失率psl和重传率prp 3个测度。统计分析表明,在绝大多数情况下,pfl比psl小,而prp比pfl和psl明显要高,验证了分组丢失对网络性能的影响以及重传率明显比分组丢失率高。     

分组话音通信中分组丢失对汉语话音清晰度的影响

分组话音通信中分组丢失对汉语话音质量的影响还未有人研究过。本文给出了在一个令牌总线型话音/数据综合局域网上进行的汉语话音清晰度测试结果。测试结果表明,在收方采用重复丢失前一帧的丢失填补策略下,对于16 kbit/s CVSD编码加静默压缩,以50 ms为话音分组长度的分组话音通信,当分组丢失率达到25％时,汉语话音清晰度的下降不超过3％。     

关于VOLTE语音质量分组丢失率高小区整治的研究

阐述4G语音分组丢失对用户感知的影响,通过大数据对语音分组丢失进行四维五域分析定界,聚焦在“单通”、“断续”、“音质“三个影响用户感知的现象,最后根据VoLTE语音质量定界法,对无线侧分组丢失问题这一影响用户感知主要原因进行分析,并列出影响分组丢失最常见因素及优化解决方法。     

ATM接入交换机上行链路性能的仿真研究

本文介绍了一种以太网ＡＴＭ接入交换机的设计方案,并对其上行链路的性能进行了计算机仿真研究,重点研究了负荷与信元丢失率及负荷与队列时延等的关系。针对以太网分组的特点,我们采用了Ｂａｔｃｈ模型进行实际的仿真。利用该仿真结果指导了一种具体交换机的设计。     

一种Ad Hoc网络单径和多径路由的选择算法

主要分析了Ad Hoc网络中单径路由协议和多径路由协议的特点。从路由中断时所传输分组的丢失率角度考虑,提出了一种单径路由和多径路由的路由协议选择算法。该算法考虑了由于网络的动态拓扑导致链路频繁失效时,原有路由中各节点缓存的分组会被丢弃的特征,通过选择合适的路由协议,降低分组的丢失概率,提高网络效率。     

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。      

基于链路层反馈的DYMO性能分析

DYMO协议是一种新兴的无线移动自组织网络路由协议,但DYMO协议草案并没有规定如何实现链路的监测;而有一部分的丢包是由于路由协议对断路反应迟钝造成。针对以上两个问题,文章提出将链路层反馈机制作为DYMO协议链路监测机制,并通过仿真验证了在小流量场景下,链路层反馈机制使DYMO协议大大降低了对断路的感知时延,及早地通知网络中使用该断路的节点,降低了因节点使用失效路由而导致丢失的分组数量。     

突发性链路感知的自适应链路质量估计方法

提出了一种无线传感器网络中突发性链路感知的自适应链路质量估计方法 EasiLQE。节点周期性地发送控制分组,采用长周期主动探测机制探测链路的质量;同时,节点被动侦听接收到分组的RSSI均值,当低于某一阈值时,触发短周期链路质量探测过程。最后节点使用基于误差的滤波器估计链路在未来一段时间内的质量。实验结果表明,EasiLQE既能实时准确地感知突发性链路在短时间内的持续变化,又能平滑短暂的低幅度波动而保持良好的稳定性,同时维持较低的估计开销。     

低轨卫星网络的航点分段路由及业务性能分析

提出一种基于卫星航点的分段路由（waypoint-segmentrouting,WSR）算法，WSR算法以可预测的卫星网络拓扑运动周期为基础，根据卫星节点链路状态确定卫星航点的位置；利用分段路由灵活规划分组传输路径的机制，提前响应网络拓扑变化，计算得到一条不受网络拓扑快照切换影响的传输路径。基于NS-3仿真平台进行仿真实验，设置源节点与目标节点在反向缝同侧与不同侧两种场景，选取优化链路状态路由（optimized link state routing,OLSR）算法和最短路径算法与WSR进行时延抖动与分组丢失率的对比分析。实验证明WSR与OLSR相比，两种场景下最大时延抖动分别降低46 ms与126 ms，分组丢失率分别降低30%和21%，并且能够解决拓扑快照切换导致分组传输路径中断的问题。     

CCI-based link quality estimation mechanism for wireless sensor networks under non-perceived packet loss

This paper proposes a chip correlation indicator (CCI)-based link quality estimation mechanism for wireless sensor networks under non-perceived packet loss. On the basis of analyzing all related factors, it can be concluded that signal-to-noise rate (SNR) is the main factor causing the non-perceived packet loss. In this paper, the relationship model between CCI and non-perceived packet loss rate (NPLR) is established from related models such as SNR versus packet success rate (PSR), CCI versus SNR and CCI-NPLR. Due to the large fluctuating range of the raw CCI, Kalman filter is introduced to do de-noising of the raw CCI. The cubic model and the least squares method are employed to fit the relationship between CCI and SNR. In the experiments, many groups of comparison have been conducted and the results show that the proposed mechanism can achieve more accurate measurement of the non-perceived packet loss than existing approaches. Moreover, it has the advantage of decreasing extra energy consumption caused by sending large number of probe packets.     

An Investigation of Link Quality Assessment for Mobile Multi-hop and Multi-rate Wireless Networks

Wireless ad hoc networks will be an important component in future communication systems. The performance of wireless ad hoc networks can be improved by link quality-aware applications. Wireless link quality is dynamic in nature, especially in mobile scenarios. Therefore, accurate and fast packet delivery ratio estimation is a prerequisite to good performance in mobile, multi-hop and multi-rate wireless ad hoc networks. In this paper, we propose a novel packet delivery ratio estimation method that improves the accuracy and responsiveness of the packet delivery ratio estimation. The proposed link quality estimation components are implemented in a IEEE 802.11b/g test-bed. The experiment results show that the accuracy of the packet delivery ratio estimation can improve up to 50% in mobile scenarios without introducing overhead. We also show the end-to-end performance impact of this improved estimation on route selection using different routing metrics and configurations. The measurement results show that our packet delivery ratio method leads to better route selection in the form of increased end-to-end throughput compared to traditional methods, which respond slowly to the link dynamics.     

Cross-Layer Fast Link Adaptation for MIMO-OFDM Based WLANs

This paper considers the use of cross-layer fast link adaptation (FLA) for WLANs employing a MIMO-OFDM physical layer. A packet error rate (PER)-based FLA technique that, without loss of generality, makes use of the exponential effective SNR mapping (EESM) is proposed. Additionally, an FLA scheme relying on bit error rate (BER) metrics is introduced that simplifies the link adaptation procedure without any significant performance degradation. Results show that both PER- and BER-based FLA techniques optimize the data throughput while satisfying prescribed quality of service constraints. Channel estimation errors have also been considered, revealing the importance of good channel estimators in order for FLA strategies to work satisfactorily.     

A correction of E-model in quality estimation of packetized speech signal

This letter presents correction of E-model for the quality estimation of the packetized speech signal in the case of burst packet loss. The current estimation depends on the method of calculation, i.e. on the duration of speech signal segment, which is to be packetized by one packet. This inconsistency in estimation is described, and the correction of this imperfection is proposed. It is explained that the Packet-loss Robustness Factor is dependent, besides coder, on the duration of speech signal segment, which is packetized by one packet.     

A novel adaptive traffic prediction AQM algorithm

In the Internet, network congestion is becoming an intractable problem. Congestion results in longer delay, drastic jitter and excessive packet losses. As a result, quality of service (QoS) of networks deteriorates, and then the quality of experience (QoE) perceived by end users will not be satisfied. As a powerful supplement of transport layer (i.e. TCP) congestion control, active queue management (AQM) compensates the deficiency of TCP in congestion control. In this paper, a novel adaptive traffic prediction AQM (ATPAQM) algorithm is proposed. ATPAQM operates in two granularities. In coarse granularity, on one hand, it adopts an improved Kalman filtering model to predict traffic; on the other hand, it calculates average packet loss ratio (PLR) every prediction interval. In fine granularity, upon receiving a packet, it regulates packet dropping probability according to the calculated average PLR. Simulation results show that ATPAQM algorithm outperforms other algorithms in queue stability, packet loss ratio and link utilization.     

Voice packet loss: destination versus internodal links

Voice packet loss behavior at both the destination and internodal links in a packet-switched network is investigated. The fractional loss and blocking time periods for both are derived using a bivariate Markov model. The numerical results show that blocking due to the delay constraint at the destination can result in long periods of consecutive packet loss, which seriously degrade voice quality. The authors' work indicates that packets with excessive delay should be discarded at the internodal links, instead of blocking them at the destination. The relation between the internodal link buffer size and end-to-end permissible queueing delay is established     

Link quality prediction model based on Gaussian process regression

Link quality is an important factor of reliable communication and the foundation of upper protocol design for wireless sensor network.Based on this,a link quality prediction model based on Gaussian process regression was proposed.It employed grey correlation algorithm to analyze correlation between link quality parameters and packet receive rate.The mean of the link quality indication and the mean of the signal-to-noise were selected as input parameters so as to reduce the computational complexity.The above parameters and packet receive rate were taken to build Gaussian process regression model with combination of covariance function,so that link quality could be predicted.In the stable and unstable scenarios,the experimental results show that the proposed model has better prediction accuracy than the one of dynamic Bayesian network prediction model.     

Comprehensive performance analysis of a TCP session over a wireless fading link with queueing

A link model-driven approach toward transmission control protocol (TCP) performance over a wireless link is presented. TCP packet loss behavior is derived from an underlying two-state continuous time Markov model. The approach presented here is (to our knowledge) the first that simultaneously considers (1) variability of the round-trip delay due to buffer queueing; (2) independent and nonindependent (bursty) link errors; (3) TCP packet loss due to both buffer overflow and channel errors; and (4) the two modes of TCP packet loss detection (duplicate acknowledgments and timeouts). The analytical results are validated against simulations using the ns-2 simulator for a wide range of parameters; slow and fast fading links; small and large link bandwidth-delay products. For channels with memory, an empirical rule is presented for categorizing the impact of channel dynamics (fading rate) on TCP performance.     

Bayesian inference of network loss and delay characteristics with applications to TCP performance prediction

In large-scale dynamic communication networks, end systems cannot rely on the network itself to cooperate in characterizing its own behavior. This has prompted research activities on methods for inferring internal network behavior based on the external end-to-end network measurements. In particular, knowledge of the link losses and link delays inside the network is important for network management. However, it is impractical to directly measure packet losses or delays at every router. On the other hand, measuring end-to-end (from sources to destinations) losses or delays is relatively easy. We formulate the problems of link and delay estimation in a network based on end-to-end measurements as Bayesian inference problems and develop several Markov chain Monte Carlo (MCMC) algorithms to solve them. We show how these link loss and delay estimates can be used to predict point-to-point transfer control protocol (TCP) throughput in the network. We apply the proposed link loss and delay estimation algorithms, as well as the TCP throughput estimation algorithms, to data generated by the network simulator (ns-2) software and obtain good agreements between the theoretical results and the actual measurements.     

Video quality protection for real time video streams over wireless networks

Received video quality is dependent on the available link rate and the packet loss ratio, which are correlated in a busy network link. Even low packet loss ratios (PLRs) can significantly reduce the video quality. In this paper, a packet level parity Forward Error Correction (FEC) is applied to the video stream in order to reduce the video PLR. A constant gross data rate is assumed, such that adding a FEC leads to a decrease in effective video data rate. The FEC block is truncated at the end of each video VOP, such that there are no inter-VOP dependencies for FEC correction. An algorithm is proposed to optimize the FEC length, based on the Quality of Experience as modelled by the ITU-T R G.1070 standard. It is shown that the optimization algorithm can significantly increase the video quality, without increasing the gross data rate. The algorithm has been evaluated both analytically and through simulations, which confirm the very significant increases in subjective video quality.