FCA-ItswTCM:区分服务中一种公平的拥塞自适应标记算法

 本文提出了一种新的滑动窗口标记算法——公平的拥塞自适应标记算法(FCA-ItswTCM).算法近似识别TCP流和UDP流,适度区分标记,规避拥塞控制机制对公平性的影响;细粒度描述拥塞,预测拥塞,以此自适应调节各流注入黄包比例,兼顾网络拥塞状态对公平性的影响.仿真实验表明,与其他几种滑动窗口标记算法相比,FCA-ItswTCM对确保TCP流和UDP流带宽共享的公平性、提高资源利用率及系统稳定性有较好的效果.     

IEEE 802.11 DCF对TCP性能的影响及算法改进

本文讨论了IEEE 802.11 DCF两种方式(RTS/CTS和CSMA/CA)混合工作时TCP的性能,发现高负载时TCP流存在不稳定的问题.基于此提出了RWB(Reserve Window Backoff)算法,该算法从联合优化的角度出发,减轻了WLAN MAC接入机制和TCP拥塞控制之间的不良影响,从而提高了TCP的吞吐量,改善了TCP流的稳定性.文中通过仿真对算法进行了分析和研究.     

吉比特网络CUBIC拥塞控制算法仿真研究

分析了TCP RENO的不足,比较了BIC算法和CUBIC算法,采用响应函数理论比较了三种TCP算法.在吉比特网络环境下对CUBIC算法进行了仿真.仿真研究表明,CUBIC算法能在高速网络环境中有效的提升网络速度,在拥塞丢包严重,网络流量很低时快速恢复网络速度;同时因为拥塞窗口波动太大,高速TCP流的吞吐量变化太大,带宽的利用率很不稳定.     

IEEE 802．11 DCFY对TCP性能的影响及算法改进

本文讨论了IEEE 802．11 DCF两种方式（RTS／CTS和CSMA／CA）混合工作时TCP的性能,发现高负载时TCP流存在不稳定的问题。基于此提出了RWB（Reserve Window Backoff）算法,该算法从联合优化的角度出发,减轻了WLANMAC接入机制和TCP拥塞控制之间的不良影响,从而提高了TCP的吞吐量,改善了TCP流的稳定性。文中通过仿真对算法进行了分析和研究。     

基于显式速率的TCP友好的UDP拥塞控制策略

本文提出了一种基于显式速率的UDP拥塞控制策略：通过源端和网络中的路由器相互配合，使得实时UDP应用能够根据网络的反馈以瓶颈链路的公平带宽为速率发送数据。此种控制策略对TCP应用是友好的，并且提高了网络的吞吐量和利用率。仿真结果表明：基于显式速率的UDP拥塞控制策略与采用TFRC(TCP—Friendly Rate Control)的UDP拥塞控制策略相比，在吞吐量、TCP友好性等方面性能有较大提高。     

区分服务中一种基于聚集流内部公平性的标记算法

在现今异构刚络Internet,区分服务体系结构通过对不同应用和不同要求的数据流部署不同等级的服务,以提供较大粒度的服务质量(Quality of Service,QoS)保证.不同流之间的公平性问题是区分服务中的研究热点,而标记策略是提高公平性的有效方法.目前许多标记算法只考虑了聚集流之间的公平性,而忽略了聚集流内部流之间的公平性.本文针对同一个聚集流中可能包含不同类型的单个流情况(比如存在自适应TCP流和非适应UDP流、不同速率的多媒体UDP流、采用不同TCP协议的数据流、不同分组大小的数据流),提出了一种基于聚集流内部公平性的标记算法(Fair Aggregate Traffic Marker,FATM).论文构造了不同情况下的模拟场景,并给出了实验的相关参数设置.大量模拟实验结果表明:标记算法FATM在保持聚集流之间的公平性和网络吞吐量的基础上,提高聚集流内部单个流之间的公平性.     

数据包大小对TCP流性能的影响

在互联网时代,如何提高网络通信质量一直是研究热点,当前网络通信大部分数据流都采用TCP流。对于TCP流性能的影响因素如丢包、时延已有大量研究,但是数据包大小对其的影响却罕有涉及。另一方面,网络通信的质量很大程度上取决于路由器的性能,而路由器的性能又很大程度上取决于路由器的缓存设置。因此本文从数据包大小这一新的切入点入手,研究在不同的路由器缓存下,其对TCP流性能的影响。论文介绍了TCP协议、路由器缓存、数据包等相关概念,提出了实验方案的网络拓扑模型,选择了NS2平台进行网络仿真。在实验中,我们通过改变数据包大小和路由器缓存容量,得到了对应的实验数据并进行了分析整理,总结出了TCP流吞吐量和丢包率在数据包大小改变时的变化规律,即丢包率随数据包大小的增加呈线性增长,路由器缓存越小丢包率越大;TCP流的吞吐量随数据包大小的增加缓慢变大,路由器缓存达到一定值后不再成为瓶颈因素。     

基于拥塞估计的TCP窗口在线学习方法

针对传统TCP拥塞窗口更新、控制机制所导致的Ad Hoc网络吞吐量下降的问题,利用机器学习算法来改善TCP在Ad Hoc网络中的性能。该方法利用确认帧的时间间隔,通过连续动作集(CALA)算法快速学习并估计当前网络链路中的拥塞状况,从而能够迅速调整TCP拥塞窗口。仿真实验表明:当Ad Hoc网络环境较好时,学习型TCP的吞吐量略优于TCP-Few、TCP-Reno协议,但在环境较差的情况下,学习型TCP的吞吐量远远优于TCP-Few和TCP-Reno协议。     

一种新的分层多播拥塞控制协议

提出了一种新的自适应分层多播拥塞控制方案(ALM)。ALM是发送方与接收方共同驱动、由路由器辅助流量控制的拥塞控制方案,通过把发送方的动态分层和接收方的自适应速率调整有机结合,不仅增强了分层多播的适应能力,提高了系统的吞吐量,而且较好地满足了TCP友好性。仿真实验表明,ALM能有效地利用网络带宽,解决网络带宽的异构性问题,并能通过接收端计算TCP友好速率,使接收端达到与TCP流公平竞争网络资源的目的。     

一种支持分级服务和最小带宽保证的改进TCP算法

本文提出了一种支持分级服务和最小带宽保证的改进TCP算法－DSAS－TCP，通过对TCP算中慢启动阈值和最小拥塞窗口的分级设置来使不同等级连接的平均吞吐量产生分级，并保证用户的最小带宽需求。通过理论分析和仿真实验验证了该算法的有效性。     

A time-dominated TCP congestion control over heterogeneous networks

The traditional transmission control protocol (TCP) suffers from performance problems such as throughput bias against flows with longer packet roundtrip time (RTT), which leads to burst traffic flows producing high packet loss, long delays, and high delay jitter. This paper proposes a TCP congestion control mechanism, TD-TCP, that the sender increases the congestion window according to time rather than receipt of acknowledgement. Since this mechanism spaces out data sent into the network, data are not sent in bursts. In addition, the proposed mechanism reduces throughput bias because all flows increase the congestion window at the same rate regardless of their packet RTT. The implementation of the mechanism affects only the protocol stack at the sender; hence, neither the receiver nor the routers need modifications. The mechanism has been implemented in the Linux platform and tested in conjunction with various TCP variants in real environments. The experimental result shows that the proposed mechanism improves transmission performance, especially in networks with congestion and/or high packet loss rates. Experiments in real commercial wireless networks have also been conducted to support the proposed mechanism's practical use. Copyright © 2008 John Wiley & Sons, Ltd.     

Dynamics of TCP traffic over ATM networks

Investigates the performance of transport control protocol (TCP) connections over ATM networks without ATM-level congestion control and compares it to the performance of TCP over packet-based networks. For simulations of congested networks, the effective throughput of TCP over ATM can be quite low when cells are dropped at the congested ATM switch. The low throughput is due to wasted bandwidth as the congested link transmits cells from “corrupted” packets, i.e., packets in which at least one cell is dropped by the switch. The authors investigate two packet-discard strategies that alleviate the effects of fragmentation. Partial packet discard, in which remaining cells are discarded after one cell has been dropped from a packet, somewhat improves throughput. They introduce early packet discard, a strategy in which the switch drops whole packets prior to buffer overflow. This mechanism prevents fragmentation and restores throughput to maximal levels     

An enhanced congestion avoidance mechanism for TCP Vegas

TCP Vegas detects network congestion in the early stage and successfully prevents periodic packet loss that usually occurs in traditional schemes. It has been demonstrated that TCP Vegas achieves much higher throughput than TCP Reno. However, TCP Vegas cannot prevent unnecessary throughput degradation when congestion occurs in the backward path. In this letter, we propose an enhanced congestion avoidance mechanism for TCP Vegas. By distinguishing whether congestion occurs in the forward path or not, it significantly improves the connection throughput when the backward path is congested.     

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.     

无线网络中TCP友好流媒体传输改进机制

为保持无线网络中多媒体业务对TCP的友好性，提出了一种适用于无线网络的动态自适应的流媒体传输速率调节机制。该机制通过在接收端区分网络拥塞丢包和链路错误随机丢包，准确判断网络的拥塞状况结合接收端缓存区占用程度，自适应实施多级速率调节，实现了TCP流友好性和流媒体服务质量（QoS）的折中。由于准确区分出无线链路误码丢包和动态调整流媒体QoS要求，该机制能维持较高的网络利用率。仿真实验结果显示在连接数为2和32，链路误码率从0到0．1变化时TCP,TFRC和吞吐量幅度下降幅度较大，WTFCC幅度下降相对较慢，最大相差达2M；在网络负载重时，尽管链路误码率较低，WTFCC区分链路错误与拥塞丢包，因此，端到端丢包率高于TCP和TFRC，但整体传输吞吐量也高于两者。归一化吞吐量显示WTFCC对TCP流友好。     

Steady state and transient state behaviours analyses of TCP connections considering interactions between TCP connections and network

The Internet uses a window‐based congestion control mechanism in transmission control protocol (TCP). In the literature, there have been a great number of analytical studies on TCP. Most of those studies have focused on the statistical behaviour of TCP by assuming a constant packet loss probability in the network. However, the packet loss probability, in reality, changes according to the packet transmission rates from TCP connections. Conversely, the window size of a TCP connection is dependent on the packet loss probability in the network. In this paper, we explicitly model the interaction between the congestion control mechanism of TCP and the network as a feedback system. By using this model, we analyse the steady state and the transient state behaviours of TCP. We derive the throughput and the packet loss probability of TCP, and the number of packets queued in the bottleneck router. We then analyse the transient state behaviour using a control theoretic approach, showing the influence of the number of TCP connections and the propagation delay on the transient state behaviour of TCP. Copyright © 2005 John Wiley & Sons, Ltd.     

主动队列管理中的智能分组丢弃新机制

主动队列管理通过网络中间节点有控制的分组丢弃实现了较低的排队延时和较高的有效吞吐量,是TCP端到端拥塞控制近来研究的一个技术热点.已有的大多数算法在判定分组丢弃时大都沿袭了RED的概率丢弃机制,具有一定计算复杂度的随机数生成过程不利于路由器性能的优化.在本文中,我们首先定义了拥塞指数这一新的测度变量来量化描述网络的拥塞状态.接着,利用模糊逻辑设计了一种新的智能分组丢弃机制,离线的合成推理使得分组丢弃的判定仅需要简单的查表操作和比较运算即可完成,为优化路由器的性能提供了便利.数字仿真的结果表明:智能分组丢弃机制的性能优于经典的RED算法,控制队列的能力强,鲁棒性好,稳定工作域大,能很好地抵抗突发性和非弹性业务的干扰,适合工作在瞬息万变的动态网络环境中.     

Promoting the use of end-to-end congestion control in the Internet

This paper considers the potentially negative impacts of an increasing deployment of non-congestion-controlled best-effort traffic on the Internet. These negative impacts range from extreme unfairness against competing TCP traffic to the potential for congestion collapse. To promote the inclusion of end-to-end congestion control in the design of future protocols using best-effort traffic, we argue that router mechanisms are needed to identify and restrict the bandwidth of selected high-bandwidth best-effort flows in times of congestion. The paper discusses several general approaches for identifying those flows suitable for bandwidth regulation. These approaches are to identify a high-bandwidth flow in times of congestion as unresponsive, “not TCP-friendly”, or simply using disproportionate bandwidth. A flow that is not “TCP-friendly” is one whose long-term arrival rate exceeds that of any conformant TCP in the same circumstances. An unresponsive flow is one failing to reduce its offered load at a router in response to an increased packet drop rate, and a disproportionate-bandwidth flow is one that uses considerably more bandwidth than other flows in a time of congestion     

光突发交换网络中的传输控制协议性能

光突发交换(OBS)是IP over WDM核心网络采用的交换技术。在OBS网络中,送往同一边缘节点的IP分组汇聚成传输和交换的基本单元———数据突发(DB),DB丢弃会导致大量IP分组丢失,显著影响传输层的性能。文中分析OBS网络的参数对传输控制协议(TCP)吞吐量和时延的影响。仿真结果表明,DB丢包率越高,TCP性能越差。在低丢包率情况下,随着DB长度的增加,TCP吞吐量和端到端时延明显增加,高丢包率情况下则不明显;随着汇聚周期的增加,TCP吞吐量逐渐下降,端到端时延逐渐增加。     

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.