流量/拥塞控制技术研究

流量/拥塞控制的基本目的是以分布处理的方式有效地控制结点间的数据流,从而避免网络中出现拥塞。拥塞控制相应的控制策略称为拥塞控制算法(协议)。简述了Internet上基于TCP/IP的拥塞控制机制,分析和比较了TCP/IP上具体实现算法的稳定性,讨论了TCP/IP拥塞控制所面临的问题。     

TCP/IP协议的拥塞控制策略

网络拥塞问题会导致网络性能下降,对网络正常运行有很大影响,严重时会导致死锁的产生.文中分析了网络拥塞的产生的原因,并着重讨论了传统端到端的TCP拥塞控制的策略和方法,传统TCP拥塞控制策略提出了改进策略和网络层的IP拥塞控制策略.最后对基于TCP和IP拥塞控制策略进行了对比探讨.     

早期随机拥塞控制模型及其在ATN中的实现

本文通过对传输控制协议/网际协议(TCP/IP)与面向连接传输层协议/无连接网络层协议(COTP/CLNP)中拥塞控制方法的比较分析,基于随机早期检测算法(RED)提出一种新的网络拥塞控制模型—早期随机拥塞控制模型(ERCCM),适用于所有采用绝对门限拥塞控制的网络系统.基于该模型和RED算法提出了一种适用于航空电信网(ATN)的随机早期置位方法(REM),并对该方法在ATN中的实现进行了仿真,仿真结果表明采用随机置位的方法可以解决绝对门限置位的三个主要问题,极大改善了网络的性能.     

对TCP/IP计算机网络拥塞控制的研究

为提升计算机的网络性能,更好地避免拥塞现象的发生,需要对其进行必要的技术控制。鉴于此,对基于TCP/IP协议的网络拥塞控制方法进行分析。在TCP拥塞控制中主要采用TCP Tahoe,TCP Reno,TCP New Reno以及TCP Sack四种方法,其中TCP New Reno对快速恢复算法进行了改进,通过对TCP协议中的Reno进行可视化处理,实行对网络拥塞的有效管理。而IP拥塞控制方法则分为FIFO,FQ和WFQ,RED以及ECN四种类型,通过队列调度管理方式实现了对网络拥塞的有效管理。     

复杂计算机网络中拥塞控制的研究及若干新思路

介绍了在TCP／IP和ATM网络中的一些拥塞控制技术，重点分析了控制理论的一些成果在拥塞控制中的应用，指出目前的拥塞控制技术中的不足，并提出了一些利用复杂控制系统理论与智能控制系统解决拥塞问题的新思路。     

一种适合数据链的TOP拥塞控制策略

传统因特网传输控制协议（TCP）主要针对有线网络设计，无法应对数据链网络中突发干扰、高误码率、拓扑高动态变化引起的拥塞和吞吐率下降等问题。提出一种适合数据链网络的拥塞控制策略，利用主动探测结果来设定拥塞窗口大小，采用反馈机制进行动态拥塞控制，通过网关节点的缓存和转发来实现快速重传。仿真结果表明，与典型拥塞控制机制相比，提出的策略可明显改善数据链网络的传输性能。     

网络中的拥塞控制管理算法分析与比较

随着网络技术的发展,网络拥塞日益严重,如何解决拥塞,充分、高效地利用网络资源,成为当今急需解决的问题.由于Internet上大多数业务都采用TCP协议,因此TCP的拥塞控制机制对控制网络拥塞具有特别重要的意义.本文介绍了TCP基于窗口的拥塞控制策略和目前常用端到端拥塞控制算法,并对它们的性能进行比较.     

基于自适应丢包区分的卫星信道TCP改进

通过卫星扩展TCP/IP网络无论在商业还是军事上具有十分重要的意义。但由于链路的高时延、高误码率和严重的信道衰落,使得TCP/IP应用于卫星链路时,相对于有线链路性能较差,因此将TCP/IP应用于卫星链路时,必须对TCP/IP进行改进。提出了自适应特定队列管理的方法,其基本思想是根据网络事件动态调整模式匹配参数,使得在拥塞和卫星链路误码率增加的情况下,都具有较高的丢包区分准确率。     

TCP在非对称环境中的研究

本文综述了因特网上非对称网络中 TCP/IP拥塞控制的最新工作 ,分析了降低非对称网络性能的原因 ,分类总结了已经提出的提高 TCP/IP性能的各种解决方案 ,进行了详细的分析和比较 ,并对较理想的方案进行了详尽的描述 ,最后指出了各种方案的优缺点     

一种新的多速率多播拥塞控制策略

本文提出了一种新的多速率多播拥塞控制策略,以满足分层多播接收者的可用带宽异构性。这种接收方驱动的拥塞控制策略,能够根据网络情况变化动态地调整分层的数量及层速率,运用最优层速率分配算法来满足接收者的可用带宽异构性,接收者的可用带宽可以用根据TCP友好经验公式计算出。仿真实验表明,该算法在TCP友好性上有良好的性能,同时它可以明显提高系统的吞吐量。     

Low‐loss TCP/IP header compression for wireless networks

Wireless is becoming a popular way to connect mobile computers to the Internet and other networks. The bandwidth of wireless links will probably always be limited due to properties of the physical medium and regulatory limits on the use of frequencies for radio communication. Therefore, it is necessary for network protocols to utilize the available bandwidth efficiently. Headers of IP packets are growing and the bandwidth required for transmitting headers is increasing. With the coming of IPv6 the address size increases from 4 to 16 bytes and the basic IP header increases from 20 to 40 bytes. Moreover, most mobility schemes tunnel packets addressed to mobile hosts by adding an extra IP header or extra routing information, typically increasing the size of TCP/IPv4 headers to 60 bytes and TCP/IPv6 headers to 100 bytes. In this paper, we provide new header compression schemes for UDP/IP and TCP/IP protocols. We show how to reduce the size of UDP/IP headers by an order of magnitude, down to four to five bytes. Our method works over simplex links, lossy links, multi‐access links, and supports multicast communication. We also show how to generalize the most commonly used method for header compression for TCP/IPv4, developed by Jacobson, to IPv6 and multiple IP headers. The resulting scheme unfortunately reduces TCP throughput over lossy links due to unfavorable interaction with TCP's congestion control mechanisms. However, by adding two simple mechanisms the potential gain from header compression can be realized over lossy wireless networks as well as point‐to‐point modem links. This revised version was published online in June 2006 with corrections to the Cover Date.     

Rate Control Algorithm Considering Mobility in Mobile Ad Hoc Networks

For reducing the loss of data packet due to network congestion and mobility, rate control algorithm is effective means for ensuring the fair allocation of network resource among communication flows and alleviating network congestion. The Transport Control Protocol (TCP) was originally designed for wire-line networks, where the links are assumed to be reliable and with fixed capacities. However, the performance of TCP becomes very poor when it is directly used in ad hoc networks. Some optimization-based congestion control schemes have been proposed to improve TCP performance in ad hoc networks, but the mobility of hosts and route change frequently are not considered for designing rate control algorithm. In this paper, we propose rate control algorithms in a more practical way by taking into account link congestion and node movement. Numerical results show that the rate control algorithms can approach the globally optimal value and show the effect of the node mobility on the algorithms.     

TCP在无线通信中存在的问题及解决方案

TCP和IP协议非常简单且可靠,它们的组合决定了目前的大多数通信方式(从有线骨干网到混合网)。TCP协议最初是为有线网络而设计的,目前已成为大多数应用事实上的标准。在有线网络中随机比特差错率可以忽略,拥塞主要由包丢失造成。很多研究都表明未修改的TCP协议在无线环境中的性能很差,因为它无法区分数据包的丢失是由于拥塞还是传输差错造成的。文章分析了TCP在无线IP通信中存在的问题,详细给出了相应的解决方案。     

Congestion Control in Store and Forward Tandem Links

This paper discusses the problem of congestion in message-switched data communication networks. This condition occurs when more traffic enters a network than can reasonably be served. Two types of techniques employed to control congestion in already existing national networks are identified. A queueing model which can be used to analyze and compare these control schemes for a simple tandem link network is developed. Finally, analytic results are presented which describe the behavior of the simple network using the two control schemes, and which should be useful in the design of more general networks as well.     

Enhancement of TCP over wired/wireless networks with packet loss classifiers inferred by supervised learning

TCP is suboptimal in heterogeneous wired/wireless networks because it reacts in the same way to losses due to congestion and losses due to link errors. In this paper, we propose to improve TCP performance in wired/wireless networks by endowing it with a classifier that can distinguish packet loss causes. In contrast to other proposals we do not change TCP’s congestion control nor TCP’s error recovery. A packet loss whose cause is classified as link error will simply be ignored by TCP’s congestion control and recovered as usual, while a packet loss classified as congestion loss will trigger both mechanisms as usual. To build our classification algorithm, a database of pre-classified losses is gathered by simulating a large set of random network conditions, and classification models are automatically built from this database by using supervised learning methods. Several learning algorithms are compared for this task. Our simulations of different scenarios show that adding such a classifier to TCP can improve the throughput of TCP substantially in wired/wireless networks without compromizing TCP-friendliness in both wired and wireless environments.     

多个TCP连接的拥塞丢包模型

本文提出了有关TCP连接的拥塞丢包分析模型.网络瓶颈一般承载许多TCP连接,瓶颈处不可避免的拥塞和缓存溢出,是导致网上丢包的主要原因.网络瓶颈处的行为很大程度上左右了网络性能.本文的模型估计了存在大量持续TCP连接时,网络瓶颈的丢包概率和网络传输中断概率,给出了对实际网络的良好近似.这对于研究TCP对网络性能的影响,提出改善网络性能的新算法,以及分析(从长远来看)TCP还应做哪些改进,都是非常有用的.     

Modelling of wireless TCP for short‐lived flows

The transmission control protocol (TCP) is one of the most important Internet protocols. It provides reliable transport services between two end‐hosts. Since TCP performance affects overall network performance, many studies have been done to model TCP performance in the steady state. However, recent researches have shown that most TCP flows are short‐lived. Therefore, it is more meaningful to model TCP performance in relation to the initial stage of short‐lived flows. In addition, the next‐generation Internet will be an unified all‐IP network that includes both wireless and wired networks integrated together. In short, modelling short‐lived TCP flows in wireless networks constitutes an important axis of research. In this paper, we propose simple wireless TCP models for short‐lived flows that extend the existing analytical model proposed in [IEEE Commun. Lett. 2002; 6 (2):85–88]. In terms of wireless TCP, we categorized wireless TCP schemes into three types: end‐to‐end scheme, split connection scheme, and local retransmission scheme, which is similar to the classification proposed in [IEEE/ACM Trans. Networking 1997; 756–769]. To validate the proposed models, we performed ns‐2 simulations. The average differences between the session completion time calculated using the proposed model and the simulation result for three schemes are less than 9, 16, and 7 ms, respectively. Consequently, the proposed model provides a satisfactory means of modelling the TCP performance of short‐lived wireless TCP flows. Copyright © 2005 John Wiley & Sons, Ltd.