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。     

Ad Hoc 网络中一种基于端节点的启发式TCP改进方法

针对无线移动自组织网络(Mobile Ad Hoc Network,MANET)中网络拥塞和较高误码所引起的TCP性能下降的问题,本文提出了一种端到端的、启发式TCP改进机制.通过该机制,接收端可以推断出网络丢包的真正原因,以及可能的网络拥塞.根据推断结果,发送端采用ECN和(或)ELN机制向发送端尽早反馈,使得发送端可以针对不同的情况采取合适的措施,从而可以改进TCP在MANET中的性能.NS-2试验结果表明在MANET中该方法的性能优于传统TCP.     

TCP丢包检测技术分析

在有线网络环境中,分组丢失往往是由于网络拥塞造成的,因此传输控制协议(TCP)能够良好运行;然而当TCP运行在无线环境中,由于误码率高,信号衰落以及频繁的移动等特性造成丢包时,TCP拥塞窗口依旧盲目减半,导致其性能大幅度下降。本文首先阐述了无线环境中出现的拥塞丢包、随机丢包、突发丢包以及发生包的重新排序的原因,在此基础上,总结了处理这些问题的方法并对他们进行了评价,最后,分析了进一步的研究方向。     

无线网络中传输控制协议的研究

传统TCP(传输控制协议)拥塞控制协议本是为有线网络设计,它假设包丢失完全是由网络拥塞引起。在无线网络环境下除了拥塞丢包外,还存在较高的比特误码率、路由故障等因素引起的丢包现象。当出现非拥塞丢包时,传统TCP将错误地触发拥塞控制,从而引起TCP性能低下。文章在分析传统TCP在无线网络中存在问题的基础上,对目前无线TCP发展和技术进行归纳和比较,进一步给出无线传输协议的研究和发展方向。     

全IP蜂窝通信系统TCP性能的改进

在无线网络中,造成丢包的主要原因是无线链路的高误码率(BER)及主机在区域间移动。如果TCP的丢包处理简单采用启动拥塞控制机制,势必导致网络传输性能恶化。本文以全IP蜂窝通信系统为例,概述TCP/IP协议组用于无线链路的性能问题,提出解决这些问题的方案,分析该方案的优点和局限性。     

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.     

一种改进的TCP over OBS网络拥塞控制策略

首先深入分析了TCP over OBS网络中造成TCP性能下降的各种原因.在此基础上,考虑将近年来提出的GAIMD算法与OBS网络的特点相结合,提出了一种改进的拥塞控制策略.该策略能在一定程度上改善OBS网络由于随机竞争丢包给TCP性能带来的负面影响.     

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。     

无线网络TCP协议性能分析

改进的TCP协议又称TCP-AP(Transport Control Protocol with AP),主要是针对无线网络下TCP性能下降做出的改动。TCP发送方的丢包问题能此办法解决,该协议区分丢包是由比特错误造成还是网路拥塞造成主要通过层与层之间的通信,此改动对于提高TCP在无线网络中的性能具有重大意义。本文通过仿真工具Network Simulator对各种场景进行模拟后试验,对于TCP-AP协议性能的提高进行验证。     

无线Mesh网中基于网络编码的多路径TCP研究

在无线多跳网络中,本地重传和网络编码已经被成功地应用到多路径技术上以增加吞吐量并减少丢包。然而,在提高UDP传输性能的同时,也产生了数据包重排序和延迟等副作用,严重影响了TCP性能。针对此问题,主要提出一种基于网络编码的多路径传输方案NC-MPTCP,即在无线mesh网络的多条路径中引入网络编码、执行拥塞控制以及使用一个基于信用的方法控制节点的传输速率,提高网络的吞吐量以及增加网络传输的可靠性。该方案使用一个简单的算法,评估丢包率以及发送线性组合数据包的速率,用来降低目的节点的数据包解码延迟和防止TCP的超时重传。仿真结果表明设计的NC-MPTCP有效。     

Packet Recycling and Delayed ACK for Improving the Performance of TCP over MANETs

Most of the schemes that were proposed to improve the performance of transmission control protocol (TCP) over mobile ad hoc networks (MANETs) are based on a feedback from the network, which can be expensive (require extra bandwidth) and unreliable. Moreover, most of these schemes consider only one cause of packet loss. They also resume operation based on the same stand-by parameters that might vary in the new route. Therefore, we propose two techniques for improving the performance of TCP over MANETs. The first one, called TCP with packet recycling (TCP-PR), allows the nodes to recycle the packets instead of dropping them after reaching the retransmission limit at the MAC layer. In the second technique, which is called TCP with adaptive delay window (TCP-ADW), the receiver delays sending TCP ACK for a certain time that is dynamically changed according to the congestion window and the trip time of the received packet. TCP-PR and TCP-ADW are simple, easy to implement, do not require network feedback, compatible with the standard TCP, and do not require distinguishing between the causes of packet loss. Our thorough simulations show that the integration of our two techniques improves the performance of TCP over MANETs.     

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.     

A comparison of mechanisms for improving TCP performance overwireless links

Reliable transport protocols such as TCP are tuned to perform well in traditional networks where packet losses occur mostly because of congestion. However, networks with wireless and other lossy links also suffer from significant losses due to bit errors and handoffs. TCP responds to all losses by invoking congestion control and avoidance algorithms, resulting in degraded end-to end performance in wireless and lossy systems. We compare several schemes designed to improve the performance of TCP in such networks. We classify these schemes into three broad categories: end-to-end protocols, where loss recovery is performed by the sender; link-layer protocols that provide local reliability; and split-connection protocols that break the end-to-end connection into two parts at the base station. We present the results of several experiments performed in both LAN and WAN environments, using throughput and goodput as the metrics for comparison. Our results show that a reliable link-layer protocol that is TCP-aware provides very good performance. Furthermore, it is possible to achieve good performance without splitting the end-to-end connection at the base station. We also demonstrate that selective acknowledgments and explicit loss notifications result in significant performance improvements     

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.     

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.     

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.     

Bandwidth estimation in high mobility scenarios of IEEE 802.11 infrastructure‐less mobile ad hoc networks

Bandwidth estimation in mobile ad hoc networks (MANET), where each node can move randomly and is capable of frequently changing its link with other nodes, is a challenging task. Motivation of this work is in contrast with TCP new‐reno which decreases the congestion window both in the event of link failure and congestion, which in the case of packet loss due to link failure should be close to available channel bandwidth. The proposed novel approach capture the node's mobility behavior in broadcast and unicast scenarios of IEEE 802.11 standard to efficiently estimate the sender's window size. This proposal introduces a data structure and source‐to‐destination path stability metric to imitate the mobility behavior of network and presents the analytic characterization of steady‐state throughput as a function of packet loss, round trip time, and path stability over IEEE 802.11 infrastructure‐less MANET. The performance is evaluated over random‐walk, random‐waypoint, and Gauss‐Markov mobility models in 2D and 3D environments using QualNet 7.4 network simulator. The proposed analytical model is also evaluated through two‐tailed statistical test. Analytical, statistical, and simulation‐based comparisons demonstrate the effectiveness of proposed method in high‐mobility scenarios.     

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