Comparative study of various TCP versions over a wireless link with correlated losses

We investigate the behavior of the various transmission control protocol (TCP) algorithms over wireless links with correlated packet losses. For such a scenario, we show that the performance of NewReno is worse than the performance of Tahoe in many situations and even OldTahoe in a few situations because of the inefficient fast recovery method of NewReno. We also show that random loss leads to significant throughput deterioration when either the product of the square of the bandwidth-delay ratio and the loss probability when in the good state exceeds one, or the product of the bandwidth-delay ratio and the packet success probability when in the bad state is less than two. The performance of Sack is always seen to be the best and the most robust, thereby arguing for the implementation of TCP-Sack over the wireless channel. We also show that, under certain conditions, the performance depends not only on the bandwidth-delay product but also on the nature of timeout, coarse or fine. We have also investigated the effects of reducing the fast retransmit threshold.     

Lost retransmission detection for TCP SACK

There have been a lot of works to avoid retransmission timeout (RTO) of transmission control protocol (TCP) that takes place in an unnecessary situation. However, most current TCP implementations, even if selective acknowledgment (SACK) option is used, do not have a mechanism to detect a lost retransmission and avoid subsequent RTO. In this letter, we propose a simple modification that enables a TCP sender using SACK option to detect a lost retransmission, which is called TCP SACK+ in simple. We use a stochastic model to evaluate the performance of TCP SACK+. Numerical results evaluated by simulations show that TCP SACK+ improves the loss recovery of TCP SACK significantly in presence of random losses.     

Retransmission loss recovery by duplicate acknowledgment counting

The performance of transmission control protocol (TCP) is largely dependent upon its loss recovery. Therefore, it is a very important issue whether the packet losses may be recovered without retransmission timeout (RTO) or not. TCP always evokes RTO if a retransmitted packet is lost again. In order to alleviate this problem, we propose an algorithm called duplicate acknowledgment counting (DAC). We use a stochastic model to evaluate the performance of DAC, and compare it with TCP NewReno. Numerical results evaluated by simulations show that DAC can improve TCP loss recovery significantly in presence of random losses.     

Energy Efficiency of TCP in a Local Wireless Environment

The focus of this paper is to analyze the energy consumption performance of various versions of TCP, namely, Tahoe, Reno and NewReno, for bulk data transfer in an environment where channel errors are correlated. We investigate the performance of a single wireless TCP connection by modeling the correlated packet loss/error process (e.g., as induced by a multipath fading channel) as a first-order Markov chain. Based on a unified analytical approach, we compute the throughput and energy performance of various versions of TCP. The main findings of this study are that (1) error correlations significantly affect the energy performance of TCP (consistent with analogous conclusions for throughput), and in particular they result in considerably better performance for Tahoe and NewReno than iid errors, and (2) the congestion control mechanism implemented by TCP does a good job at saving energy as well, by backing off and idling during error bursts. An interesting conclusion is that, unlike throughput, the energy efficiency metric may be very sensitive to the TCP version used and to the choice of the protocol parameters, so that large gains appear possible.     

光突发交换网络中的传输控制协议性能

光突发交换(OBS)是IP over WDM核心网络采用的交换技术。在OBS网络中,送往同一边缘节点的IP分组汇聚成传输和交换的基本单元———数据突发(DB),DB丢弃会导致大量IP分组丢失,显著影响传输层的性能。文中分析OBS网络的参数对传输控制协议(TCP)吞吐量和时延的影响。仿真结果表明,DB丢包率越高,TCP性能越差。在低丢包率情况下,随着DB长度的增加,TCP吞吐量和端到端时延明显增加,高丢包率情况下则不明显;随着汇聚周期的增加,TCP吞吐量逐渐下降,端到端时延逐渐增加。     

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.     

TCP拥塞控制机制浅析

TCP是当今网络中主要的传输协议,它采用了慢启动、拥塞避免、快速重传、快速恢复四种算法,能满足IP网络中数据的可靠传输。但是当出现多个数据包丢失时,由于TCP采用了累计确认机制,造成系统吞吐量下降。文章介绍了一种SACK拥塞控制机制,与传统的Tahoe、Reno对比,并通过仿真实验说明了SACK是一种最好的TCP恢复机制。     

Sliding window protocol with selective-repeat ARQ: performance modeling and analysis

A sliding window mechanism is well known to be a convenient way to integrate cumulative acknowledgement and flow control functions in a simple manner. Hence, widely used reliable data-transfer protocols such as TCP-SACK and RLC utilize this sliding window mechanism in conjunction with a selective-repeat ARQ (automatic repeat request) function, namely, the selective-repeat sliding window protocol (SR-SWP). We demonstrate that SR-SWP yields a throughput restraint effect (TORE) in particular situations when packet losses occur. We propose an analytical performance model to represent the TORE in the case of heavy traffic. Using this model, we derive explicit forms of goodput and mean resequencing delay, which are fundamental qualities of service parameters. From numerical results, we demonstrate that the impact of TORE on the performance is not negligible for high packet-loss probabilities. Furthermore, we show that the SR-SWP model exhibits the lower goodput and the smaller mean resequencing delay than those obtained from the traditional SR-ARQ protocol model.     

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.     

Taxonomy and Evaluation of TCP-Friendly Congestion-Control Schemes on Fairness, Aggressiveness, and Responsiveness

Many TCP-friendly congestion control schemes have been proposed to pursue the TCP-equivalence criterion, which states that a TCP-equivalent flow should have the same throughput with TCP if it experiences identical network conditions as TCP. Additionally, the throughput should converge as fast as TCP when the packet-loss conditions change. This study classifies eight typical TCP-friendly schemes according to their underlying policies on fairness, aggressiveness, and responsiveness. The schemes are evaluated to verify whether they meet TCP-equivalence and TCP-equal share. TCP-equal share is a more realistic but more challenging criterion than TCP-equivalence and states that a flow should have the same throughput with TCP if competing with TCP for the same bottleneck. Simulation results indicate that one of the selected schemes, TCP-friendly rate control (TFRC), meets both criteria under more testing scenarios than the others. Additionally, the results under non-periodic losses, low-multiplexing, two-state losses, and bursty losses reveal the causes that bring fault cases to the schemes. Finally, appropriate policies are recommended for an ideal scheme.     

Improving TCP performance for vertical handover in heterogeneous wireless networks

The incorporation of wireless local area networks (WLANs) into existing cellular networks as supplementary access technologies has become an issue of great interest. However, vertical handover (VHO), which allows users to roam between a WLAN and a cellular network, causes an abrupt change in certain link characteristics such as the round trip time and data rate. Owing to such changes, reordering problem and premature timeout occur and trigger unnecessarily fast retransmission during VHO, causing throughput degradation. Thus, we propose a new transmission control protocol (TCP) mechanism, which resolves the reordering problem by suppressing unnecessary retransmission caused by spurious duplicate acknowledgments (dupacks) incurred because of the reordering problem, and prevents premature timeout by employing an adaptive retransmission timer. We analytically investigate the throughput of our proposed TCP scheme. The numerical and simulation results show that our proposed TCP performs better in terms of throughput than other schemes appearing in the literature. Copyright © 2009 John Wiley & Sons, Ltd.     

Modeling TCP Reno performance: a simple model and its empiricalvalidation

The steady-state performance of a bulk transfer TCP flow (i.e., a flow with a large amount of data to send, such as FTP transfers) may be characterized by the send rate, which is the amount of data sent by the sender in unit time. In this paper we develop a simple analytic characterization of the steady-state send rate as a function of loss rate and round trip time (RTT) for a bulk transfer TCP flow. Unlike the models of Lakshman and Madhow (see IEE/ACM Trans. Networking, vol.5, p.336-50, 1997), Mahdavi and Floyd (1997), Mathis, Semke, Mahdavi and Ott (see Comput. Commun. Rev., vol.27, no.3, 1997) and by by Ott et al., our model captures not only the behavior of the fast retransmit mechanism but also the effect of the time-out mechanism. Our measurements suggest that this latter behavior is important from a modeling perspective, as almost all of our TCP traces contained more time-out events than fast retransmit events. Our measurements demonstrate that our model is able to more accurately predict TCP send rate and is accurate over a wider range of loss rates. We also present a simple extension of our model to compute the throughput of a bulk transfer TCP flow, which is defined as the amount of data received by the receiver in unit time     

How a large ATM MTU causes deadlocks in TCP data transfers

The implementation of protocols, such as TCP/IP, and their integration into the operating system environment is crucial for protocol performance. Putting TCP on high-speed networks, e.g., ATM, with large maximum transmission units causes the TCP maximum segment size to be relatively large. What Nagle's algorithm considers a “small” segment is no longer small, which affects the TCP performance. The authors report on TCP/IP throughput and RPC response time performance measurements for various sizes of send and receive socket buffers, using the Sparc10 architecture machines Axil 311/5.1 running SunOS 4.1.3 connected to a FORE System's ATM network. For some common combinations of socket buffer sizes the authors observe a dramatic performance degradation to less than 1% of expected throughput and to one order-of-magnitude longer response time than expected. The performance degradation is caused by a deadlock situation in the TCP connection which is resolved by the 200 ms spaced timer generated TCP delayed acknowledgment. The authors explain the causes of the deadlock situations, and discuss means to avoid or prevent them     

解决FAST TCP缓存溢出相关问题的改进pacing technique算法和α参数调整算法

分析FAST TCP在缓存溢出发生时的性能,发现在缓存溢出场景中,收敛中的FAST TCP流经历严重的报文段丢失。相反,已经收敛了的FAST TCP流维持着高吞吐量和低报文段丢失概率。这种不公平是由FAST TCP缩减其窗口时的零传输率导致的。通过修改FAST TCP pacing算法,可以解决此问题。文中提出的α-adjusting算法,通过动态调整FAST TCP协议中的α参数来避免频繁的缓存溢出。通过分析ns2仿真结果,证明该算法在公平性和稳定性方面可获得令人满意的性能。     

Adding network-layer intelligence to mobile receivers for solving spurious TCP timeout during vertical handoff

With the help of mobile IP/IPv6 and soft handoff, ongoing TCP sessions can remain active and handoff packet loss can be avoided. However, TCP still faces several performance degradation issues due to the disparities in bandwidth and propagation delay between different access networks. Particularly, during vertical handoffs, some undesirable phenomena may erroneously trigger TCP congestion-control actions and thus degrade TCP performance. In this article we tackle the spurious timeout problem frequently associated with handovers from fast to slow links. We propose three network-layer schemes: fast ACK, slow ACK, and ACK delaying. These schemes require only minor modifications to the network layer of mobile receivers and no change to the TCP protocol and the TCP sender. The simulation results show that these schemes can effectively improve TCP performance during soft vertical handoffs     

TCP performance over mobile ad hoc networks: a quantitative study

In this paper, we study the performance of the transmission control protocol (TCP) over mobile ad‐hoc networks. We present a comprehensive set of simulation results and identify the key factors that impact TCP's performance over ad‐hoc networks. We use a variety of parameters including link failure detection latency, route computation latency, packet level route unavailability index, and flow level route unavailability index to capture the impact of mobility. We relate the impact of mobility on the different parameters to TCP's performance by studying the throughput, loss‐rate and retransmission timeout values at the TCP layer. We conclude from our results that existing approaches to improve TCP performance over mobile ad‐hoc networks have identified and hence focused only on a subset of the affecting factors. In the process, we identify a comprehensive set of factors influencing TCP performance. Finally, using the insights gained through the performance evaluations, we propose a framework called Atra consisting of three simple and easily implementable mechanisms at the MAC and routing layers to improve TCP's performance over ad‐hoc networks. We demonstrate that Atra improves on the throughput performance of a default protocol stack by 50%–100%. Copyright © 2004 John Wiley & Sons, Ltd.     

TCP CERL: congestion control enhancement over wireless networks

In this paper, we propose and verify a modified version of TCP Reno that we call TCP Congestion Control Enhancement for Random Loss (CERL). We compare the performance of TCP CERL, using simulations conducted in ns-2, to the following other TCP variants: TCP Reno, TCP NewReno, TCP Vegas, TCP WestwoodNR and TCP Veno. TCP CERL is a sender-side modification of TCP Reno. It improves the performance of TCP in wireless networks subject to random losses. It utilizes the RTT measurements made throughout the duration of the connection to estimate the queue length of the link, and then estimates the congestion status. By distinguishing random losses from congestion losses based on a dynamically set threshold value, TCP CERL successfully attacks the well-known performance degradation issue of TCP over channels subject to random losses. Unlike other TCP variants, TCP CERL doesn’t reduce the congestion window and slow start threshold when random loss is detected. It is very simple to implement, yet provides a significant throughput gain over the other TCP variants mentioned above. In single connection tests, TCP CERL achieved an 175, 153, 85, 64 and 88% throughput gain over TCP Reno, TCP NewReno, TCP Vegas, TCP WestwoodNR and TCP Veno, respectively. In tests with multiple coexisting connections, TCP CERL achieved an 211, 226, 123, 70 and 199% throughput improvement over TCP Reno, TCP NewReno, TCP Vegas, TCP WestwoodNR and TCP Veno, respectively.     

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.     

Performance Enhancement of On-Board Communication Networks Using Outage Prediction

A research area that has become increasingly important in recent years is that of on-board mobile communication, where users on a vehicle are connected to a local network that attaches to the Internet via a mobile router and a wireless link. In this architecture, link disruptions (e.g., due to signal degradation) may have an immediate impact on a potentially large number of connections. We argue that the advance knowledge of public transport routes, and their repetitive nature, allows a certain degree of prediction of impending link disruptions, which can be used to offset their catastrophic impact. Focusing on the transmission control protocol (TCP) and its extension known as Freeze-TCP, we present a detailed analysis of the performance improvement of TCP connections in the presence of disruption prediction. In particular, we propose a Markov model of Freeze-TCP that captures both the TCP behavior and the prediction+“freezing” feature and, using simulations, show that it accurately predicts the performance of the protocol. Our results demonstrate the significant throughput improvement that can be gained by disruption prediction, even with random packet losses or imperfect timing of the predicted disruptions.     

TCP over optical burst-switched networks with controlled burst retransmission

For optical burst-switched (OBS) networks in which TCP is implemented at a higher layer, the loss of bursts can lead to serious degradation of TCP performance. Due to the bufferless nature of OBS, random burst losses may occur, even at low traffic loads. Consequently, these random burst losses may be mistakenly interpreted by the TCP layer as congestion in the network. The TCP sender will then trigger congestion control mechanisms, thereby reducing TCP throughput unnecessarily. In this paper, we introduce a controlled retransmission scheme in which the bursts lost due to contention in the OBS network are retransmitted at the OBS layer. The OBS retransmission scheme can reduce the burst loss probability in the OBS core network. Also, the OBS retransmission scheme can reduce the probability that the TCP layer falsely detects congestion, thereby improving the TCP throughput. We develop an analytical model for evaluating the burst loss probability in an OBS network that uses a retransmission scheme, and we also analyze TCP throughput when the OBS layer implements burst retransmission. We develop a simulation model to validate the analytical results. Simulation and analytical results show that an OBS layer with controlled burst retransmission provides up to two to three orders of magnitude improvement in TCP throughput over an OBS layer without burst retransmission. This significant improvement is primarily because the TCP layer triggers fewer time-outs when the OBS retransmission scheme is used.