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.     

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.     

TCP Veno: TCP enhancement for transmission over wireless access networks

Wireless access networks in the form of wireless local area networks, home networks, and cellular networks are becoming an integral part of the Internet. Unlike wired networks, random packet loss due to bit errors is not negligible in wireless networks, and this causes significant performance degradation of transmission control protocol (TCP). We propose and study a novel end-to-end congestion control mechanism called TCP Veno that is simple and effective for dealing with random packet loss. A key ingredient of Veno is that it monitors the network congestion level and uses that information to decide whether packet losses are likely to be due to congestion or random bit errors. Specifically: (1) it refines the multiplicative decrease algorithm of TCP Reno-the most widely deployed TCP version in practice-by adjusting the slow-start threshold according to the perceived network congestion level rather than a fixed drop factor and (2) it refines the linear increase algorithm so that the connection can stay longer in an operating region in which the network bandwidth is fully utilized. Based on extensive network testbed experiments and live Internet measurements, we show that Veno can achieve significant throughput improvements without adversely affecting other concurrent TCP connections, including other concurrent Reno connections. In typical wireless access networks with 1% random packet loss rate, throughput improvement of up to 80% can be demonstrated. A salient feature of Veno is that it modifies only the sender-side protocol of Reno without changing the receiver-side protocol stack.     

编码TCP的建模与性能分析

研究了无线网络端到端分组丢失和编码纠错对TCP协议工作过程的影响机制,基于三维Markov链建模描述TCP拥塞窗口和可用窗口的变迁过程,在忽略慢启动阶段的条件下,以最大窗口尺寸、端到端分组丢失率和编码冗余系数为输入参数,通过数值计算求解编码TCP的吞吐率,据此定量分析影响编码TCP性能的因素。基于NS2的模拟实验结果表明基于Markov链的数值计算结果具有较高的准确度。     

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.     

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。     

Using TCP rate control for queue input‐output rate matching

In explicit TCP rate control, the receiver's advertised window size in acknowledgment (ACK) packets can be modified by intermediate network elements to reflect network congestion conditions. The TCP receiver's advertised window (i.e. the receive buffer of a TCP connection) limits the maximum window and consequently the throughput that can be achieved by the sender. Appropriate reduction of the advertised window can control the number of packets allowed to be sent from a TCP source. This paper evaluates the performance of a TCP rate control scheme in which the receiver's advertised window size in ACK packets are modified in a network node in order to match the generated load to the assigned bandwidth in the node. Using simulation and performance metrics such as the packet loss rates and the cumulative number of TCP timeouts, we examine the service improvement provided by the TCP rate control scheme to the users. The modified advertised windows computed in the network elements and the link utilization are also examined. Copyright © 2002 John Wiley & Sons, Ltd.     

TCP SkyLine:数据中心网络高吞吐率传输

针对数据中心网络在"多对一"并发流量模式下,TCP（Transmission Control Protocol）及其现有改进方案在单轮数据传输和多轮数据传输下吞吐率低下问题,提出了一种通过数据包标记实现丢包快速发现和快速重传并动态调整拥塞窗口初始值的策略,称为TSL（TCP SkyLine）.TSL同时解决了传统TCP Incast问题和多轮数据传输下由遗留窗口引发的TCP Incast问题.实验表明,TSL在单轮数据传输和多轮数据传输下均能获得90%以上的带宽利用率.在10Gbps网络中,其支持的并发连接数与传统TCP和DCTCP相比分别提升了5倍和1倍,有效吞吐率分别提升了18倍和8.6倍;在1Gbps网路中,支持的并发连接数较传统TCP和DCTCP分别提升了5.8倍和1倍.     

A stochastic model for the behavior of multiple TCP NewReno sources over optical burst switching network

This work proposes a stochastic model to characterize the transmission control protocol (TCP) over optical burst switching (OBS) networks which helps to understand the interaction between the congestion control mechanism of TCP and the characteristic bursty losses in the OBS network. We derive the steady-state throughput of a TCP NewReno source by modeling it as a Markov chain and the OBS network as an open queueing network with rejection blocking. We model all the phases in the evolution of TCP congestion window and evaluate the number of packets sent and time spent in different states of TCP. We model the mixed assembly process, burst assembler and disassembler modules, and the core network using queueing theory and compute the burst loss probability and end-to-end delay in the network. We derive expression for the throughput of a TCP source by solving the models developed for the source and the network with a set of fixed-point equations. To evaluate the impact of a burst loss on each TCP flow accurately, we define the burst as a composition of per-flow-bursts (which is a burst of packets from a single source). Analytical and simulation results validate the model and highlight the importance of accounting for individual phases in the evolution of TCP congestion window.     

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.     

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…     

Performance evaluation of high speed TCP over optical burst switching networks

Study of TCP performance over OBS networks has been an important problem of research lately and it was found that due to the congestion control mechanism of TCP and the inherent bursty losses in the Optical Burst Switching (OBS) network, the throughput of TCP connections degrade. On the other hand, High Speed TCP (HSTCP) was proposed as an alternative to the use of TCP in high bandwidth-delay product networks. HSTCP aggressively increases the congestion window used in TCP, when the available bandwidth is high and decreases the window cautiously in response to a congestion event. In this work, we make a thorough simulation study of HSTCP over OBS networks. While the earlier works in the literature used a linear chain of nodes as the network topology for the simulation, we use the popular 14-node NSFNET topology that represents an arbitrary mesh network in our study. We also study the performance of HSTCP over OBS for different bandwidths of access networks. We use two different cases for simulations where in the first HSTCP connections are routed on disjoint paths while in the second they contend for resources in the network links. These cases of simulations along with the mesh topology help us clearly distinguish between the congestion and contention losses in the OBS network and their effect on HSTCP throughput. For completeness of study, we also simulate TCP traffic over OBS networks in all these cases and compare its throughput with that of HSTCP. We observe that irrespective of the access network bandwidth and the burst loss rate in the network, HSTCP outperforms TCP in terms of the throughput and robustness against multiple burst losses up to the expected theoretical burst loss probability of 10⁻³.     

Enhancing Fairness and Throughput of TCP in Heterogeneous Wireless Access Networks

Most of the recent research on TCP over heterogeneous wireless networks has concentrated on differentiating between packet drops caused by congestion and link errors, to avoid significant throughput degradations due to the TCP sending window being frequently shut down, in response to packet losses caused not by congestion but by transmission errors over wireless links. However, TCP also exhibits inherent unfairness toward connections with long round-trip times or traversing multiple congested routers. This problem is aggravated by the difference of bit-error rates between wired and wireless links in heterogeneous wireless networks. In this paper, we apply the TCP Bandwidth Allocation (TBA) algorithm, which we have proposed previously, to improve TCP fairness over heterogeneous wireless networks with combined wireless and wireline links. To inform the sender when congestion occurs, we propose to apply Wireless Explicit Congestion Notification (WECN). By controlling the TCP window behavior with TBA and WECN, congestion control and error-loss recovery are effectively separated. Further enhancement is also incorporated to smooth traffic bursts. Simulation results show that not only can the combined TBA and WECN mechanism improve TCP fairness, but it can maintain good throughput performance in the presence of wireless losses as well. A salient feature of TBA is that its main functions are implemented in the access node, thus simplifying the sender-side implementation.     

A Novel Congestion Detection Scheme in TCP Over OBS Networks

This paper introduces a novel congestion detection scheme for high-bandwidth TCP flows over optical burst switching (OBS) networks, called statistical additive increase multiplicative decrease (SAIMD). SAIMD maintains and analyzes a number of previous round-trip time (RTTs) at the TCP senders in order to identify the confidence with which a packet loss event is due to network congestion. The confidence is derived by positioning short-term RTT in the spectrum of long-term historical RTTs. The derived confidence corresponding to the packet loss is then taken in the developed policy for TCP congestion window adjustment. We will show through extensive simulation that the proposed scheme can effectively solve the false congestion detection problem and significantly outperform the conventional TCP counterparts without losing fairness. The advantages gained in our scheme are at the expense of introducing more overhead in the SAIMD TCP senders. Based on the proposed congestion control algorithm, a throughput model is formulated, and is further verified by simulation results.      

The impact of multihop wireless channel on TCP performance

This paper studies TCP performance in a stationary multihop wireless network using IEEE 802.11 for channel access control. We first show that, given a specific network topology and flow patterns, there exists an optimal window size W* at which TCP achieves the highest throughput via maximum spatial reuse of the shared wireless channel. However, TCP grows its window size much larger than W* leading to throughput reduction. We then explain the TCP throughput decrease using our observations and analysis of the packet loss in an overloaded multihop wireless network. We find out that the network overload is typically first signified by packet drops due to wireless link-layer contention, rather than buffer overflow-induced losses observed in the wired Internet. As the offered load increases, the probability of packet drops due to link contention also increases, and eventually saturates. Unfortunately the link-layer drop probability is insufficient to keep the TCP window size around W'*. We model and analyze the link contention behavior, based on which we propose link RED that fine-tunes the link-layer packet dropping probability to stabilize the TCP window size around W*. We further devise adaptive pacing to better coordinate channel access along the packet forwarding path. Our simulations demonstrate 5 to 30 percent improvement of TCP throughput using the proposed two techniques.     

联合无线网络编码与TCP Vegas的多播协议优化研究

为解决移动自组网中网络编码多播路由协议因业务传输负载增大,而产生的网络拥塞现象,本文提出了一种可靠的基于TCP Vegas窗口拥塞控制的网络编码多播路由协议。该协议的核心思想是发送节点采用发送窗口自调整和反馈消息触发发送窗口调整的机制,综合的调节数据包的发送速率,来改善网络拥塞现象,从而可以降低丢包率。仿真结果表明,当传输负载增大时,基于窗口拥塞控制的网络编码多播路由协议可使得系统的总开销大大降低,分组投递率获得了相对的提升。     

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.     

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.     

无线网络中TCP拥塞控制算法的性能分析

根据无线网络中存在随机数据包丢失的特定情况,对TCP拥塞控制算法在无线网络中的性能进行了分析。理论分析和仿真结果表明,随着无线链路中随机数据包丢失概率的增加,TCP拥塞控制算法将导致无线网络性能的严重下降。     

PAQM: an adaptive and proactive queue management for end‐to‐end TCP congestion control

Two functions, the congestion indicator (i.e. how to detect congestion) and the congestion control function (i.e. how to avoid and control congestion), are used at a router to support end‐to‐end congestion control in the Internet. Random early detection (RED) (IEEE/ACM Trans. Networking 1993; 1 (4):397–413) enhanced the two functions by introducing queue length averaging and probabilistic early packet dropping. In particular, RED uses an exponentially weighted moving average (EWMA) queue length not only to detect incipient congestion but also to smooth the bursty incoming traffic and its resulting transient congestion. Following RED, many active queue management (AQM)‐based extensions have been proposed. However, many AQM proposals have shown severe problems with detection and control of the incipient congestion adaptively to the dynamically changing network situations. In this paper, we introduce and analyse a feedback control model of TCP/AQM dynamics. Then, we propose the Pro‐active Queue Management (PAQM) mechanism, which is able to provide proactive congestion avoidance and control using an adaptive congestion indicator and a control function under a wide range of traffic environments. The PAQM stabilizes the queue length around the desired level while giving smooth and low packet loss rates and high network resource utilization. Copyright © 2004 John Wiley & Sons, Ltd.