一种能处理无线链路拥塞的TCP/ARQ改进算法

采用链路层自动请求重传（ARQ）方法来对分组差错进行恢复，并对传统的TCP／ARQ算法加以改进，加入了在无线链路拥塞情况下的自适应拥塞避免算法。仿真结果表明，这种能处理无线链路拥塞的TCP／ARQ算法对于提高无线网络的吞吐量非常有效。     

面向无线移动网络的TCP三态显式拥塞控制

针对无线移动网络中传统TCP协议的传输性能下降的问题进行了显式拥塞控制的改进研究,提出了三态显式拥塞控制方法.该方法通过修改传统显式拥塞提醒(ECN)的标记机制,向端节点提供空闲、繁忙、拥塞三种状态,使端节点能更灵活地调节自身的发送速率,从而能提供更平稳的数据流.其特点在于除TCP的传统的和性增加积性减少的阶段外,另外...     

TCP拥塞控制算法研究

随着互联网规模的增长，互联网上的用户和应用都在快速的增长，拥塞已经成为一个十分重要的问题。近年来，在拥塞控制领域开展了大量的研究工作。拥塞控制算法可以分为两个主要部分：在端系统上使用的源算法和在网络设备上使用的链路算法。在介绍拥塞控制算法的基本概念后，本文总结了在TCP协议方面拥塞控制算法的研究现状，并分析了进一步的研究方向。     

面向多媒体流的主动分层多播拥塞控制方案

针对现有分层多播拥塞控制方案存在较大的拥塞响应延时,吞吐率抖动剧烈,和不满足TCP友好的问题,给出了一个主动分层多播服务模型,通过引入主动标记分层、可用带宽主动测量以及优先级分层过滤机制,提出了一种新的面向多媒体流的自适应主动分层多播拥塞控制方案AALM。仿真结果表明,该方案能够有效地改进分层多播拥塞控制的性能,具有较快的拥塞响应速度、较好的稳定性和TCP友好特性。     

一种P2P文件共享系统汇聚拥塞控制机制

针对现有P2P文件共享系统采用并发多连接的文件传输方式,过分占用网络带宽资源,导致其它传统互联网业务性能低下的问题,提出了一种P2P文件共享系统汇聚拥塞控制机制(ACCM).ACCM采用应用层网络测量技术感知节点接入网链路拥塞状况,依据网络拥塞状况动态地调整P2P文件共享系统并发文件传输连接窗口,在最大化网络带宽利用率的基础上实现对传统互联网应用的友好性.网络实验结果表明,在网络拥塞发生时,ACCM能够促使P2P文件共享系统并发连接窗口主动退避,实现和传统互联网应用的和平共处;在网络空闲时,ACCM能够促使P2P文件共享系统扩大并发连接窗口,提高网络带宽资源的利用率.     

计算机网络拥塞模型及控制方法的研究

主要对计算机网络的拥塞控制系统进行深入研究。拥塞控制系统作为计算机网络能否正常运行的关键之一,介绍网络拥塞控制算法,并对目前研究较多和应用较广泛的几个网络拥塞控制算法进行讨论,集中介绍TCP拥塞控制算法和几种基于主动队列管理的路由拥塞控制算法。     

基于ECN带预测验证的拥塞控制算法

提出了一种新的拥塞控制源算法FAV(Forecast and Verify),这是一种在接收方测量分组到达的延迟来预测网络中是否将出现拥塞的算法,并结合ECN来验证预测的准确性。通过和TCP Reno、TCP Vegas算法进行比较实验,表明FAV算法能够有效地减小丢失率和链路延迟,同时保持较高的链路利用率,从而使网络中的拥塞得到较好控制。     

网络拥塞控制算法

随着计算机网络的持续快速发展,各种网络应用需求不断涌现,造成网络数据流量的激增。网络拥塞问题变得越来越严重,网络拥塞控制也一直是网络研究的最关键热点问题之一。在本文中,作者着重阐述了TCP／IP拥塞控制中的典型算法,并指出了这些算法的优缺点。     

Google WebRTC网络拥塞控制策略的研究

伴随着近几年人们对网络视频，语音通讯需求的极大增长，国内网络带宽提速的脚步依然很慢。从而使得网路拥塞的现象更加严重。在本文中，作者阐述了当前TCP拥塞控制的典型算法，着重针对GoogleWebRTC技术中的拥塞解决方案进行分析和研究，达到了理论与实际相互结合的目的。     

TCP拥塞控制机制在高速网络中的局限性

本文介绍了TCP端到端的拥塞控制机制和高速网络的特性 ,分析了TCP拥塞控制机制在高速网络中的局限性。     

Emergency Prioritized and Congestion Handling Protocol for Medical Internet of Things

Medical Internet of Things (MIoTs) is a collection of small and energyefficient wireless sensor devices that monitor the patient’s body. The healthcare networks transmit continuous data monitoring for the patients to survive them independently. There are many improvements in MIoTs, but still, there are critical issues that might affect the Quality of Service (QoS) of a network. Congestion handling is one of the critical factors that directly affect the QoS of the network. The congestion in MIoT can cause more energy consumption, delay, and important data loss. If a patient has an emergency, then the life-critical signals must transmit with minimum latency. During emergencies, the MIoTs have to monitor the patients continuously and transmit data (e.g., ECG, BP, heart rate, etc.) with minimum delay. Therefore, there is an efficient technique required that can transmit emergency data of high-risk patients to the medical staff on time with maximum reliability. The main objective of this research is to monitor and transmit the patient’s real-time data efficiently and to prioritize the emergency data. In this paper, Emergency Prioritized and Congestion Handling Protocol for Medical IoTs (EPCP_MIoT) is proposed that efficiently monitors the patients and overcome the congestion by enabling different monitoring modes. Whereas the emergency data transmissions are prioritized and transmit at SIFS time. The proposed technique is implemented and compared with the previous technique, the comparison results show that the proposed technique outperforms the previous techniques in terms of network throughput, end to end delay, energy consumption, and packet loss ratio.     

TCPW BR: A Wireless Congestion Control Scheme Base on RTT

The wireless network is limited by the transmission medium, and the transmission process is subject to large interference and jitter. This jitter can cause sporadic loss and is mistaken for congestion by the congestion control mechanism. The TCP Westwood protocol (referred to as TCPW) is such that it cannot distinguish between congestion loss and wireless jitter loss, which makes the congestion mechanism too sensitive and reduces bandwidth utilization. Based on this, the TCPW protocol is modified based on the estimate of the Round-Trip Time (referred to as RTT) value-called TCPW BR. The algorithm uses the measured smooth RTT value and divides the congestion level according to the weighted average idea to determine congestion loss and wireless jitter loss. The simulation results show that the TCPW BR algorithm enhances the wireless network’s ability to judge congestion and random errors.     

Efficient Routing Protocol with Localization Based Priority & Congestion Control for UWSN

The nodes in the sensor network have a wide range of uses, particularly on under-sea links that are skilled for detecting, handling as well as management. The underwater wireless sensor networks support collecting pollution data, mine survey, oceanographic information collection, aided navigation, strategic surveillance, and collection of ocean samples using detectors that are submerged in water. Localization, congestion routing, and prioritizing the traffic is the major issue in an underwater sensor network. Our scheme differentiates the different types of traffic and gives every type of traffic its requirements which is considered regarding network resource. Minimization of localization error using the proposed angle-based forwarding scheme is explained in this paper. We choose the shortest path to the destination using the fitness function which is calculated based on fault ratio, dispatching of packets, power, and distance among the nodes. This work contemplates congestion conscious forwarding using hard stage and soft stage schemes which reduce the congestion by monitoring the status of the energy and buffer of the nodes and controlling the traffic. The study with the use of the ns3 simulator demonstrated that a given algorithm accomplishes superior performance for loss of packet, delay of latency, and power utilization than the existing algorithms.     

Robust end-to-end loss differentiation scheme for transport control protocol over wired/wireless networks

The authors propose a robust end-to-end loss differentiation scheme to identify the packet losses because of congestion for transport control protocol (TCP) connections over wired/wireless networks. The authors use the measured round trip time (RTT) values to determine whether the cause of packet loss is because of the congestion over wired path or regular bit errors over wireless paths. The classification should be as accurate as possible to achieve high throughput and maximum fairness for the TCP connections sharing the wired/wireless paths. The accuracies of previous schemes in the literature depends on varying network parameters such as RTT, buffer size, amount of cross traffic, wireless loss rate and congestion loss rate. The proposed scheme is robust in that the accuracy remains rather stable under varying network parameters. The basic idea behind the scheme is to set the threshold for the classification to be a function of the minimum RTT and the current sample RTT, so that it may automatically adapt itself to the current congestion level. When the congestion level of the path is estimated to be low, the threshold for a packet loss to be classified as a congestion loss is increased. This avoids unnecessary halving of the congestion window on packet loss because of the regular bit errors over the wireless path and hence improves the TCP throughput. When the congestion level of the path is estimated to be high, the threshold for a packet loss to be classified as the congestion loss not to miss any congestion loss is decreased and hence improves the TCP fairness. In ns 2 simulations, the proposed scheme correctly classifies the congestion losses under varying network parameters whereas the previous schemes show some dependency on subsets of parameters.     

Rate-adaptive video streaming through packet dispersion feedback

The anticipated growth of IPTV makes selection of suitable congestion controllers for video-stream traffic of vital concern. Measurements of packet dispersion at the receiver provide a graded way of estimating congestion, which is particularly suited to video as it does not rely on packet loss. A closed-loop congestion controller, which dynamically adapts the bitstream output of a transcoder or video encoder to a rate less likely to lead to packet loss, is presented. The video congestion controller is based on fuzzy logic with packet dispersion and its rate of change forming the inputs. Compared with TCP emulators such as TCP-friendly rate control (TFRC) and rate adaptation protocol (RAP), which rely on packet loss for real-time congestion control, the fuzzy-logic trained system?s sending rate is significantly smoother when multiple video-bearing sources share a tight link. Using a packet dispersion method similarly results in a fairer allocation of bandwidth than TFRC and RAP. These gains for video traffic are primarily because of better estimation of network congestion through packet dispersion but also result from accurate interpretation by the fuzzy-logic controller.     

一种融合MAC层拥塞通告的混合网络TCP协议

在研究无线网络媒体接入控制(MAC)层拥塞测度的基础上,提出了一种跨层的显式拥塞通告(ECN)机制,即:当数据包中记录的请求发送(RTS)次数超过给定阈值时,通过ECN向传输控制协议(TCP)源端发送拥塞通告,从而启动TCP拥塞控制.这种跨层设计是对有线网络中基于主动队列管理(AQM)的拥塞控制的有效补充,由此可以得到一种与已有的协议无缝连接的混合网络TCP模型.通过在网络模拟器NS2中构造多流无线局域网和多跳无线/有线混合网络,对所提出的方法进行了仿真,实验结果说明该方法能够提高混合网络的性能,并且具备良好的扩展性.     

A Bayesian ridge regression analysis of congestion's impact on urban expressway safety

With the rapid growth of traffic in urban areas, concerns about congestion and traffic safety have been heightened. This study leveraged both Automatic Vehicle Identification (AVI) system and Microwave Vehicle Detection System (MVDS) installed on an expressway in Central Florida to explore how congestion impacts the crash occurrence in urban areas. Multiple congestion measures from the two systems were developed. To ensure more precise estimates of the congestion's effects, the traffic data were aggregated into peak and non-peak hours. Multicollinearity among traffic parameters was examined. The results showed the presence of multicollinearity especially during peak hours. As a response, ridge regression was introduced to cope with this issue. Poisson models with uncorrelated random effects, correlated random effects, and both correlated random effects and random parameters were constructed within the Bayesian framework. It was proven that correlated random effects could significantly enhance model performance. The random parameters model has similar goodness-of-fit compared with the model with only correlated random effects. However, by accounting for the unobserved heterogeneity, more variables were found to be significantly related to crash frequency. The models indicated that congestion increased crash frequency during peak hours while during non-peak hours it was not a major crash contributing factor. Using the random parameter model, the three congestion measures were compared. It was found that all congestion indicators had similar effects while Congestion Index (CI) derived from MVDS data was a better congestion indicator for safety analysis. Also, analyses showed that the segments with higher congestion intensity could not only increase property damage only (PDO) crashes, but also more severe crashes. In addition, the issues regarding the necessity to incorporate specific congestion indicator for congestion's effects on safety and to take care of the multicollinearity between explanatory variables were also discussed. By including a specific congestion indicator, the model performance significantly improved. When comparing models with and without ridge regression, the magnitude of the coefficients was altered in the existence of multicollinearity. These conclusions suggest that the use of appropriate congestion measure and consideration of multicolilnearity among the variables would improve the models and our understanding about the effects of congestion on traffic safety.     

Effective transmission opportunity allocation scheme for real-time variable bit rate traffic flows with different delay bounds

The medium access control of IEEE 802.11e defines a novel coordination function, namely, hybrid coordination function (HCF), which allocates transmission opportunity (TXOP) to stations taking their quality of service (QoS) requirements into account. However, the reference TXOP allocation scheme of HCF controlled channel access, a contention-free channel access function of HCF, is only suitable for constant bit rate traffic. For variable bit rate traffic, packet loss may occur seriously. The authors propose a TXOP allocation scheme to efficiently allocate bandwidth and meet the QoS requirements in terms of both delay bound and packet loss probability. To achieve high bandwidth efficiency, the authors take advantage of not only intra-flow multiplexing gain of traffic flows with large delay bounds, but also inter-flow multiplexing gain of multiple traffic flows with different delay bounds. According to numerical results obtained by computer simulations, the proposed TXOP allocation scheme results in much higher bandwidth efficiency than previous algorithms under the same constraints of delay bounds and packet loss probability.