Vegas plus: improving the service fairness

TCP Vegas is a congestion avoidance scheme designed to prevent the periodic packet loss which occurs in traditional schemes. Since Vegas successfully avoids such packet loss, it achieves much higher throughput than TCP Reno. However, it does not concern the fairness among source-destination pairs with different round-trip times (RTTs). We propose a different mechanism to adjust the window size, this allows TCP to provide much better fairness regardless the large variation of RTTs     

Syndrome: a light‐weight approach to improving TCP performance in mobile wireless networks

It is well known that the performance of TCP deteriorates in a mobile wireless environment. This is due to the fact that although the majority of packet losses are results of transmission errors over the wireless links, TCP senders still take packet loss as an indication of congestion, and adjust their congestion windows according to the additive increase and multiplicative decrease (AIMD) algorithm. As a result, the throughput attained by TCP connections in the wireless environment is much less than it should be. The key problem that leads to the performance degradation is that TCP senders are unable to distinguish whether packet loss is a result of congestion in the wireline network or transmission errors on the wireless links. In this paper, we propose a light‐weight approach, called syndrome, to improving TCP performance in mobile wireless environments. In syndrome, the BS simply counts, for each TCP connection, the number of packets that it relays to the destination host so far, and attaches this number in the TCP header. Based on the combination of the TCP sequence number and the BS‐attached number and a solid theoretical base, the destination host will be able to tell where (on the wireline or wireless networks) packet loss (if any) occurs, and notify TCP senders (via explicit loss notification, ELN) to take appropriate actions. If packet loss is a result of transmission errors on the wireless link, the sender does not have to reduce its congestion window. Syndrome is grounded on a rigorous, analytic foundation, does not require the base station to buffer packets or keep an enormous amount of states, and can be easily incorporated into the current protocol stack as a software patch. Through simulation studies in ns‐2 (UCB, LBNL, VINT network simulator, http://www‐mash.cs.berkeley.edu/ns/ ), we also show that syndrome significantly improves the TCP performance in wireless environments and the performance gain is comparable to the heavy‐weight SNOOP approach (either with local retransmission or with ELN) that requires the base station to buffer, in the worst case, a window worth of packets or states. Copyright © 2001 John Wiley & Sons, Ltd.     

异构网络下一种新的TCP技术

由于传统传输控制协议（TCP）拥塞控制主要是为带宽时延乘积较小和信道误码率很低的有线网络环境而设计的,因此很难适用于异构网络下长的往返时延RTT、较大的误码率、以及大带宽等问题。提出了一种新的TCP-selective技术,通过在同一个服务器上使用不同的TCP版本,与实际的链路质量相匹配,根据链路信息来调用与之相适应的TCP来解决单一TCP性能增强方案所不能解决的问题,仿真结果证明这种新的TCP技术是有效可行的。     

TCP在不可靠网络环境中的性能分析及其服务质量控制

本文对TCP在拥塞丢失和非拥塞丢失同时存在的不可靠网络环境中的性能进行了理论分析,给出了TCP吞吐量,延时以及分组丢失概率的简明表达式.同时针对多个TCP连接共享一条限速链路时服务质量不均衡问题,提出了轮询丢弃策略和惩罚丢弃策略两种服务质量控制策略.仿真结果表明,轮询丢弃策略可以提高限速链路的总吞吐量,而惩罚丢弃策略可以提高TCP连接之间的公平性.     

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.     

Adjusting the TCP Sending Rate and Retransmissions after Retransmission Timeouts Based on One-Way Queuing Delay in Wireless Mesh Network

Wireless Mesh Network (WMN) is regarded as a viable solution to provide broadband Internet access flexibly and cost efficiently. Improving the performance of Transmission Control Protocol (TCP) in WMNs is an active research area in the networking community. The existing solutions proposed for improving the TCP performance has concentrated on differentiating the DATA packet drops in the forward direction induced by both network congestion as well as transmission errors. However, the recent studies show that in WMNs packet drops occur not only in the forward direction but also in the reverse direction particularly due to hidden terminal, hidden capture terminal, link asymmetry etc. The loss of ACK packets in the reverse direction cause frequent retransmission timeouts subject to needless retransmissions and unnecessary slowing down the growth of congestion window, which causes the performance degradation of TCP. In this paper, we introduce a sender side TCP algorithm, called detection of packet loss (DPL), which is capable to distinguish the type of packet drops either DATA or ACKs caused by transmission errors as well as network congestion based on one-way queuing delay and react accordingly. To justify our contributions, we implement DPL in Qualnet simulator and compare its performance against existing TCP solutions via extensive simulations. Our simulation results show that the proposed algorithm can accurately distinguish the type of packet drops whether it is a DATA or ACK caused by transmission error or congestion and can significantly improve the performance under a wide range of scenarios in WMNs.     

一种基于多径负载均衡的无线互联网TCP拥塞控制策略

针对无线互联网中拥塞控制存在的问题,提出了一种支持多径负载均衡技术的TCP拥塞控制策略。该篆略主要包括拥塞检测和拥塞控制两个部分。其中基于带宽估计的拥塞检测方法,能有效区分差错丢包和拥塞丢包;在检测到拥塞发生后,采用基于多径负载均衡技术的拥塞控制策略,并使用基于时延估计的多径分组分配策略,在缓解拥塞的同时避免了多路通信的包乱序问题,有效的提高吞吐量,减小端到端时延。     

一种支持多媒体通信QoS的拥塞控制机制

本文针对Internet传输协议TCP的和式增加积式减少(AIMD)拥塞控制机制不适应多媒体通信,而目前拥塞控制的研究又大多集中在尽量做好(Best-effort)服务上的问题,结合Internet上多媒体通信的特点及其对QoS的要求,提出了一种将多媒体通信服务质量(QoS)控制和基于速率拥塞控制结合起来的拥塞控制的新机制.本文详细地研究了这一机制,并提出了源端多媒体数据流的带宽控制策略、基于动态部分缓存共享(DPBS)的数据包丢失控制方案和接收端计算包丢失率p的方法.最后给出了整个拥塞控制机制的系统结构.     

Estimation of packet losses due to propagation impairments: Application to TCP performance over satellite

The transmission control protocol (TCP) is widely used to provide reliable data transmission due to its congestion and flow control mechanisms that provide reliable error recovery in higher layers. In satellite links, various atmospheric phenomena may lead to high packet loss rate (PLR) degrading the TCP throughput. Modern satellite systems operate at frequencies above 10 GHz, where rainfall is the dominant fading mechanism leading to high bit error ratio and correlated packet losses. In this paper, a mathematical analysis is presented to accurately describe the statistical properties of the packet‐error process in a dynamically varying satellite channel. The proposed method is extended to provide PLR estimations when block forward error correction (FEC) is employed. A new Markov‐based method, based on the previous analysis and adapted to the rain‐faded satellite channel, is also presented for the estimation of TCP SACK throughput and tested against simulation results. Based on the information provided by the packet‐error model, a study between the TCP performance under various FEC schemes and a proposed adaptive FEC scheme has provided indications about the superiority of the proposed model. Copyright © 2007 John Wiley & Sons, Ltd.     

APS-FeW适用于无线网络的TCP增强方案

TCP的控制隐含了一个假设,即分组丢失都是由网络拥塞引起的,而这一点对于无线网络不成立.因此在无线情况下,会引入了冗余控制,使得传输性能下降.浮点窗口算法(FeW)通过限制TCP的分组注入量,减少冗余控制,在此基础上,提出了改进方案自适应包长算法(APS).仿真数据表明,基于FeW的APS算法(APS-FeW)比FeW算法提升了10%～25%的吞吐量.     

MANET中TCP性能改进研究

TCP是为有线网络所设计的,其拥塞控制机制中,假设丢包是由网络拥塞造成的这一结论在MANET中不再适用。MANET中的高信道误码率,路由频繁中断等因素都会造成丢包。TCP错误的将所有丢包事件都当作拥塞处理,造成了传输性能的极大下降。文中首先总结了MANET中导致TCP传输性能下降的主要原因,然后对现有的一些典型TCP改进方案进行了讨论,最后对这些技术方案进行了比较,并指出今后研究的重点。     

一种异构网络TCP拥塞控制算法

针对互联网中端对端带宽、时延和丢包率等的差异性日益加剧,导致TCP传输性能严重退化,该文提出一种链路自适应TCP拥塞控制算法(INVS)。INVS在拥塞避免阶段初期采用基于指数函数的凸窗口增长函数,以提高链路利用率;在窗口增长函数中引入了自适应增长因子实现窗口增长速率与链路状态相匹配;采用了自适应队列门限的丢包区分策略以提高无线环境下TCP的性能。性能分析和评估表明,INVS提高了TCP拥塞控制算法的吞吐量、公平性、链路利用率和RTT公平性。     

A Novel Wireless TCP and its Steady State Throughput Model

1　Introduction　WiththegrowthofwirelessnetworksandtheInter net,thedatatransmissionserviceoverwirelessnet worksbecomesmoreattractive .InthecurrentInternet,TCPiswidelyusedinpopularapplicationslikeTelnet,FTP ,andHTTP .　TCPisareliableconnection oriented protocolthatimplementscongestioncontrolbymeansofaslidingwindowalgorithm .TCPTahoeandReno[1～ 2 ] ,whichmakeuseoftheSlowStart (SS)andCongestionAvoid ance (CA)algorithmstoadjustthewindowsize ,havegotmuchsuccessintheInternet.Inparticular…     

"De-Randomizing" congestion losses to improve TCP performance over wired-wireless networks

Currently, a TCP sender considers all losses as congestion signals and reacts to them by throttling its sending rate. With Internet becoming more heterogeneous with more and more wireless error-prone links, a TCP connection may unduly throttle its sending rate and experience poor performance over paths experiencing random losses unrelated to congestion. The problem of distinguishing congestion losses from random losses is particularly hard when congestion is light: congestion losses themselves appear to be random. The key idea is to "de-randomize" congestion losses. This paper proposes a simple biased queue management scheme that "de-randomizes" congestion losses and enables a TCP receiver to diagnose accurately the cause of a loss and inform the TCP sender to react appropriately. Bounds on the accuracy of distinguishing wireless losses and congestion losses are analytically established and validated through simulations. Congestion losses are identified with an accuracy higher than 95% while wireless losses are identified with an accuracy higher than 75%. A closed form is derived for the achievable improvement by TCP endowed with a discriminator with a given accuracy. Simulations confirm this closed form. TCP-Casablanca, a TCP-Newreno endowed with the proposed discriminator at the receiver, yields through simulations an improvement of more than 100% on paths with low levels of congestion and about 1% random wireless packet loss rates. TCP-Ifrane, a sender-based TCP-Casablanca yields encouraging performance improvement.     

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。     

Delay-based congestion avoidance for TCP

The set of TCP congestion control algorithms associated with TCP-Reno (e.g., slow-start and congestion avoidance) have been crucial to ensuring the stability of the Internet. Algorithms such as TCP-NewReno (which has been deployed) and TCP-Vegas (which has not been deployed) represent incrementally deployable enhancements to TCP as they have been shown to improve a TCP connection's throughput without degrading performance to competing flows. Our research focuses on delay-based congestion avoidance algorithms (DCA), like TCP-Vegas, which attempt to utilize the congestion information contained in packet round-trip time (RTT) samples. Through measurement and simulation, we show evidence suggesting that a single deployment of DCA (i.e., a TCP connection enhanced with a DCA algorithm) is not a viable enhancement to TCP over high-speed paths. We define several performance metrics that quantify the level of correlation between packet loss and RTT. Based on our measurement analysis, we find that, although there is useful congestion information contained within RTT samples, the level of correlation between an increase in RTT and packet loss is not strong enough to allow a TCP-sender to improve throughput reliably. While DCA is able to reduce the packet loss rate experienced by a connection, in its attempts to avoid packet loss, the algorithm reacts unnecessarily to RTT variation that is not associated with packet loss. The result is degraded throughput as compared to a similar flow that does not support DCA.     

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.     

TCP-H：基于三重判决的TCP拥塞控制

在异构无线网络中存在高误码、切换、信号衰落等链路特性,使传统的TCP拥塞控制机制受到了挑战。在不增加开销的情况下,基于丢包检测、RTT时间和ACK返回速率三重判决,提出了一种新的TCP拥塞控制机制（TCP—H）。仿真结果表明,TCP—H增强了对拥塞和随机差错的区分能力,满足公平性要求,改进的最小RTT计算方法解决了在低延迟向高延迟网络切换的时Vegas,Westwood等算法存在的最小RTT更新问题,有效提高了在异构无鲅网络环埔下TCP的性能．     

Improving TCP performance in mobile networks

In this letter, a new transport layer mechanism is proposed to improve the performance of transport control protocol (TCP) in mobile networks. The proposed mechanism is comprised of two parts: a loss classifier (LC) and a congestion window extrapolator (CWE). Based on LC, the cause of packet loss during roaming is determined. If the loss is considered to be caused by congestion in the wireline, the congestion window is halved; otherwise, the packet is considered to be lost in the last hop, the wireless portion, and the sender adjusts the size of the congestion window based on CWE. We conduct simulations to evaluate the performance of the proposed mechanism. The results show that our mechanism significantly improves TCP performance as compared with existing solutions for mobile networks.     

一种参数自适应的主动队列管理算法-自适应BLUE

BLUE算法是一种典型的主动队列管理(Active Queue Management,AQM)算法,研究表明BLUE算法优于RED算法.BLUE算法使用丢包事件和链路空闲事件控制网络拥塞.但由于BLUE算法在参数设置方面存在不足,尤其是当TCP连接数突然剧烈变动时,容易导致队列溢出或空闲的频繁发生.该文引进参数自适应机制,提出了自适应BLUE算法,并借鉴了RED算法的早期拥塞检测机制.NS仿真实验表明该算法能有效保持队列长度的稳定,减少队列溢出或空闲现象的发生,在提高链路利用率的同时降低丢包率.