Analysis of TCP Vegas and TCP Reno

In this paper we use an analytic fluid approach in order to analyze the different features of both Vegas and Reno TCP versions. We then use simulations to confirm our analytic results. When the available bandwidth is high, indeed Vegas can retransmit less than one‐fifth as much data as Reno does, so that the higher the available bandwidth is, the more efficient Vegas is. However, under heavy congestion Vegas behaves like Reno and does not manage to make efficient use of its new mechanism for congestion detection. The analytic results that we obtain are the evolution of the window size, round trip times and their averages, and the average throughput. This revised version was published online in June 2006 with corrections to the Cover Date.     

Potential performance bottleneck in Linux TCP

Transmission control protocol (TCP) is the most widely used transport protocol on the Internet today. Over the years, especially recently, due to requirements of high bandwidth transmission, various approaches have been proposed to improve TCP performance. The Linux 2.6 kernel is now preemptible. It can be interrupted mid‐task, making the system more responsive and interactive. However, we have noticed that Linux kernel preemption can interact badly with the performance of the networking subsystem. In this paper, we investigate the performance bottleneck in Linux TCP. We systematically describe the trip of a TCP packet from its ingress into a Linux network end system to its final delivery to the application; we study the performance bottleneck in Linux TCP through mathematical modelling and practical experiments; finally, we propose and test one possible solution to resolve this performance bottleneck in Linux TCP. Copyright © 2007 John Wiley & Sons, Ltd.     

认知无线电网络中一种改进的传输层协议

在认知无线电网络中,传统的TCP协议无法适应认知用户周期性感知频谱、切换信道等特点,使得网络的传输性能大大降低。在TCP-Reno基础上提出了一种基于传输预判与慢启动门限值(Slow Start Threshold,SSThresh)的自适应调整的传输层协议———TCP-Cog,该协议通过预测下一个数据包的传输时延,调整发送端的发送,并通过预测切换信道的带宽调整SSThresh。通过在NS2工具中进行仿真,验证了TCP-Cog能够提高TCP的传输速率,降低重传率,提升传输性能。     

TCP在移动IP中的性能优化方案

分析了TCP有关拥塞控制的假设在无线和移动IP中的不正确性,由此对TCP协议提出了一些修改方法,以改善TCP在无线和移动点中的性能。     

Average-value analysis of 802.11 WLANs with persistent TCP flows

In this letter we develop an average-value analysis of the TCP performance in 802.11 WLANs. Our model characterizes the equilibrium conditions of the network, and this method yields a simple, yet precise, estimate of the throughput obtained by persistent TCP flows. Via simulations we study the accuracy of the model predictions.     

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.     

TCP/IP在ATM网络中的性能分析和优化

分析了造成ＴＣＰ／ＩＰ在ＡＴＭ上性能低的原因,以提高链路利用率和吞吐量为目的对ＴＣＰ协议进行了改进,提出了两种解决方法：反馈重传策略和分组级的流量成形,来提高ＴＣＰ／ＩＰ在ＡＴＭ网络上的性能。     

基于TCP／IP协议的校园网构筑

１９９４年３月国家教委启动了中国教育计算机网络项目。ＣＥＲＮＥＴ将覆盖全国大专院校等教育科研机构，已于１９９４年１０月与国际ｉｎｔｅｒｎｅｔ联网。这样，全国有条件的大学开始了基于Ｉｎｔｅｒｎｅｔ的校园网络建设工作。     

TCP smart framing: a segmentation algorithm to reduce TCP latency

TCP Smart Framing, or TCP-SF for short, enables the Fast Retransmit/Recovery algorithms even when the congestion window is small. Without modifying the TCP congestion control based on the additive-increase/multiplicative-decrease paradigm, TCP-SF adopts a novel segmentation algorithm: while Classic TCP always tries to send full-sized segments, a TCP-SF source adopts a more flexible segmentation algorithm to try and always have a number of in-flight segments larger than 3 so as to enable Fast Recovery. We motivate this choice by real traffic measurements, which indicate that today's traffic is populated by short-lived flows, whose only means to recover from a packet loss is by triggering a Retransmission Timeout. The key idea of TCP-SF can be implemented on top of any TCP flavor, from Tahoe to SACK, and requires modifications to the server TCP stack only, and can be easily coupled with recent TCP enhancements. The performance of the proposed TCP modification were studied by means of simulations, live measurements and an analytical model. In addition, the analytical model we have devised has a general scope, making it a valid tool for TCP performance evaluation in the small window region. Improvements are remarkable under several buffer management schemes, and maximized by byte-oriented schemes.     

A new TCP for persistent packet reordering

Most standard implementations of TCP perform poorly when packets are reordered. In this paper, we propose a new version of TCP that maintains high throughput when reordering occurs and yet, when packet reordering does not occur, is friendly to other versions of TCP. The proposed TCP variant, or TCP-PR, does not rely on duplicate acknowledgments to detect a packet loss. Instead, timers are maintained to keep track of how long ago a packet was transmitted. In case the corresponding acknowledgment has not yet arrived and the elapsed time since the packet was sent is larger than a given threshold, the packet is assumed lost. Because TCP-PR does not rely on duplicate acknowledgments, packet reordering (including out-or-order acknowledgments) has no effect on TCP-PR's performance. Through extensive simulations, we show that TCP-PR performs consistently better than existing mechanisms that try to make TCP more robust to packet reordering. In the case that packets are not reordered, we verify that TCP-PR maintains the same throughput as typical implementations of TCP (specifically, TCP-SACK) and shares network resources fairly. Furthermore, TCP-PR only requires changes to the TCP sender side making it easier to deploy.     

高速网络环境中的TCP拥塞控制算法

现有的TCP Reno在高速网络环境下存在的带宽利用率等问题,从而引出了诸如HS—TCP,S—TCP和Fast—TCP等改进的TCP拥塞控制算法。本文简单介绍三种算法,对比分析了它们存在的问题和造成的原因,并提出了可能的研究方向和目前最新的改进算法。     

Shielding TCP from last hop wireless losses

The pervasiveness of the transport control protocol (TCP) and the proliferation of wireless local area networks (WLAN) of the 802.11 type make the topic of TCP performance over last hop wireless networks very relevant. The Snoop protocol, a link layer solution introduced several years ago to improve the performance of TCP in this scenario, has been shown to neglect its benefits to the most widely used TCP version, TCP SACK. In this paper, we introduce the TCP SACK‐Aware Snoop protocol to address this problem. Our results indicate that the TCP SACK‐Aware Snoop protocol improves the performance of TCP SACK by around 30% compared with the original Snoop protocol and by about 8% in an environment where no TCP enhancing mechanism is in place. In addition, we introduce further modifications to the proposed protocol to make its advantages available to any TCP sender. We also show that the mechanism does not introduce unfairness among TCP sources and somewhat protects TCP against UDP traffic. Our results show important throughput improvements to all TCP versions and demonstrate that the TCP SACK‐Aware Snoop protocol shields TCP from last hop wireless losses providing throughtput values very close to the maximum possible. Copyright © 2006 John Wiley & Sons, Ltd.     

Experimental Evaluation of TCP Protocols for High-Speed Networks

In this paper, we present experimental results evaluating the performance of the scalable-TCP, HS-TCP, BIC-TCP, FAST-TCP, and H-TCP proposals in a series of benchmark tests. In summary, we find that both scalable-TCP and FAST-TCP consistently exhibit substantial unfairness, even when competing flows share identical network path characteristics. Scalable-TCP, HS-TCP, FAST-TCP, and BIC-TCP all exhibit much greater RTT unfairness than does standard TCP, to the extent that long RTT flows may be completely starved of bandwidth. Scalable-TCP, HS-TCP, and BIC-TCP all exhibit slow convergence and sustained unfairness following changes in network conditions such as the start-up of a new flow. FAST-TCP exhibits complex convergence behavior.     

IP网络中TCP拥塞控制协议的稳定性分析

文章用泛函微分方程建立了TCP拥塞控制协议的动态模型,通过该模型分析了当前Internet上使用的TCP拥塞控制算法,导出了稳定性条件,结果表明该算法在一定网络环境下是不稳定的.     

Parameter Conditions for Global Stability of FAST TCP

In this letter, we study the global asymptotic stability of FAST TCP in the presence of network feedback delays. Based on a continuous-time dynamic model of FAST TCP and a static approximation of queuing delay at the link, we derive stability conditions for FAST TCP. The derived conditions are explicitly appeared as tuning parameter conditions of FAST TCP, and hence can be satisfied in a distributed way. The simulation results illustrate the validity of the conditions for the global asymptotic stability.     

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.     

Reorder notifying TCP (RN‐TCP) with explicit packet drop notification (EPDN)

Numerous studies have shown that packet reordering is common, especially in networks where there is high degree of parallelism and different link speeds. Reordering of packets decrease the TCP performance of a network, mainly because it leads to overestimation of the congestion in the network. In this paper, we analyse the performance of networks when reordering of packets occur. We propose a proactive solution that could significantly improve the performance of the network when reordering of packets occurs. We report results of our simulation experiments, which support this claim. Our solution is based on enabling the senders to distinguish between dropped packets and reordered packets. Copyright © 2005 John Wiley & Sons, Ltd.     

On-board satellite "split TCP" proxy

Several satellite systems currently in operation or under development claim to support broadband Internet applications. In these scenarios, transmission control protocol (TCP) plays a critical role. Unfortunately, when used with satellite links, TCP suffers from a number of well-known performance problems, especially for higher data rates and high altitude satellites with longer delays. In response to these difficulties, the satellite and Internet research communities have developed a large gamut of solutions ranging from architectural modifications to changes in the TCP protocol. Among these, one approach requiring minimal modifications involves splitting the TCP connection in two or more segments with one segment connecting terrestrial nodes across the satellite network. In this paper, we consider an evolution of this idea: placing a TCP proxy on board the satellite that further subdivides the end-to-end connection into separate TCP connections between ground and space. We focus upon the efficient use of buffer resources on board the satellite, while at the same time enhancing TCP performance. We evaluate two TCP protocol versions, TCP NewReno and TCP Westwood. We consider various geosynchronous earth orbit satellite scenarios, with and without the split proxy, and with different channel error conditions (random errors, shadowing, etc.). Using simulation, we show that an on-board proxy provides a number of distinct advantages and can enhance throughput up to threefold for both TCP New Reno and TCP Westwood, in some scenarios, with relatively modest on-board buffering requirements. The main contributions of this paper are: the on-board split proxy concept, the buffer management strategy, the use of a realistic "urban shadowing" model in the evaluation, and the extensive comparison of the recently announced TCP Westwood with the traditional TCP New Reno.     

On detecting TCP path saturation in LTE networks

A significant part of the data in mobile networks is transferred as bulk data with transmission control protocol (TCP) as apps or video downloads. When the video takes too long to start, users are more prone to abandon watching, which eventually leads to decreased revenue for the content provider. While it is widely known that TCP has performance issues in mobile networks, end‐to‐end measurement studies, especially based on real data, should be studied further. In this paper, we measure the efficiency of TCP in long term evolution (LTE) networks and provide an analysis on the reasons of unoptimal performance based on 235 000 measurements from deployed mobile networks. For this purpose, we propose an algorithm for LTE networks that detects the periods during a TCP connection where the path is saturated, ie, the network is the limiting factor. From this data, we find that TCP is a source for unoptimal performance, and assumed reasons from other papers are partly confirmed and partly refuted. Most importantly, we find that the amount of queueing on the path has significant impact on the achieved protocol performance. Lastly, we learn from the LTE experience and put the findings into a 5G context.     

Weighted proportional loss rate differentiation of TCP traffic

Relative service differentiation refers to service models that provide assurances for the relative quality ordering between classes, rather than for the absolute service level in each class. An example is the proportional differentiation model which provides a way to control the quality spacing between classes locally at each hop. In this model, certain forwarding metrics are ratioed proportional to the class differentiation parameters that the network operator chooses. In this paper, we propose a new proportional loss rate differentiation mechanism that integrates relative loss rate differentiation directly into active queue management using random early detection. Copyright © 2004 John Wiley &Sons, Ltd.