Performance evaluation of TCP connections in ideal and non-ideal network environments

In this paper, we study the performance of TCP in both ideal and non-ideal network environments. For the ideal environments, we develop a simple analytical model for the throughput and transfer time of TCP as a function of the file size and TCP parameters. Our simulation measurements demonstrate that this model can accurately predict the throughput for ideal TCP connections characterized by no packet loss due to congestion or bit errors. If these ideal conditions are not met, the model gives an upper bound for throughput and lower bound for transfer time. For the non-ideal environments, we concentrate on wireless links. While our ideal model provides an easy to use tool to calculate bounds on the performance of all TCP implementations in such environments, we also show through simulation the relative performance of four well-known TCP implementations. We also present simulation results that demonstrate the dominant factors affecting the performance of wireless TCP.     

A simulation-based study of TCP dynamics over HFC networks

New broadband access technologies such as hybrid fiber coaxial (HFC) are likely to provide fast and cost effective support to a variety of applications including video on demand (VoD), inter-active computer games, and Internet-type applications such as Web browsing, ftp, e-mail, and telephony. Since most of these applications use TCP as the transport layer protocol, the key to their efficiency largely depends on TCP protocol performance.We investigate the performance of TCP in terms of effective throughput in an HFC network environment using different load conditions and network buffer sizes. We find that TCP experiences low throughput as a result of the well-known problem of ACK compression. An algorithm that controls ACK spacing is introduced to improve TCP performance.     

On the predictability of large transfer TCP throughput

Predicting the throughput of large TCP transfers is important for a broad class of applications. This paper focuses on the design, empirical evaluation, and analysis of TCP throughput predictors. We first classify TCP throughput prediction techniques into two categories: Formula-Based (FB) and History-Based (HB). Within each class, we develop representative prediction algorithms, which we then evaluate empirically over the Resilient Overlay Network (RON) testbed. FB prediction relies on mathematical models that express the TCP throughput as a function of the characteristics of the underlying network path. It does not rely on previous TCP transfers in the given path, and it can be performed with non-intrusive network measurements. We show, however, that the FB method is accurate only if the TCP transfer is window-limited to the point that it does not saturate the underlying path, and explain the main causes of the prediction errors. HB techniques predict the throughput of TCP flows from a time series of previous TCP throughput measurements on the same path, when such a history is available. We show that even simple HB predictors, such as Moving Average and Holt-Winters, using a history of few and sporadic samples, can be quite accurate. On the negative side, the accuracy of HB predictors is highly path-dependent.     

IDTCP: An effective approach to mitigating the TCP incast problem in data center networks

Recently, TCP incast problem in data center networks has attracted a wide range of industrial and academic attention. Lots of attempts have been made to address this problem through experiments and simulations. This paper analyzes the TCP incast problem in data centers by focusing on the relationships between the TCP throughput and the congestion control window size of TCP. The root cause of the TCP incast problem is explored and the essence of the current methods to mitigate the TCP incast is well explained. The rationality of our analysis is verified by simulations. The analysis as well as the simulation results provides significant implications to the TCP incast problem. Based on these implications, an effective approach named IDTCP (Incast Decrease TCP) is proposed to mitigate the TCP incast problem. Analysis and simulation results verify that our approach effectively mitigates the TCP incast problem and noticeably improves the TCP throughput.     

Packet Spacing: An Enabling Mechanism for Delivering Multimedia Content in Computational Grids

Streaming multimedia with UDP has become increasingly popular over distributed systems like the Internet. Scientific applications that stream multimedia include remote computational steering of visualization data and video-on-demand teleconferencing over the Access Grid. However, UDP does not possess a self-regulating, congestion-control mechanism; and most best-effort traffic is served by congestion-controlled TCP. Consequently, UDP steals bandwidth from TCP such that TCP flows starve for network resources. With the volume of Internet traffic continuing to increase, the perpetuation of UDP-based streaming will cause the Internet to collapse as it did in the mid-1980's due to the use of non-congestion-controlled TCP.To address this problem, we introduce the counter-intuitive notion of inter-packet spacing with control feedback to enable UDP-based applications to perform well in the next-generation Internet and computational grids. When compared with traditional UDP-based streaming, we illustrate that our approach can reduce packet loss over 50% without adversely affecting delivered throughput.     

基于模糊综合评判的网络认知模型

针对传统传输控制协议(TCP)应用于异构网络的局限性,提出了一种基于模糊综合评判的网络认知模型。该模型通过建立隶属度函数和不同网络环境下的动态权重分布,运用模糊综合评判的方法来区分无线误码丢包和网络拥塞丢包。仿真实验证明：与传统TCP协议相比,该模型在不同的网络条件下,能够较准确地区分无线误码丢包和网络拥塞丢包,提高了TCP的吞吐量,改善了网络性能。     

基于TCP NewReno的稳态吞吐量分析模型

最近的研究表明,在当前网络未启用SACK选项的TCP流中,有超过一半的数据流采用TCP New Reno的快速恢复算法.而Padhye提出的基于TCP Reno的TCP吞吐量分析模型,不能准确反映TCP New Reno数据流的吞吐量.Padhye模型在建立过程中采用猝发性丢包模型,同时忽略了快速恢复阶段和超时后的慢启动阶段,影响了预测的准确性.基于此,提出了一种基于TCP New Reno的吞吐量分析模型.在分析过程中,采用了更符合真实网络丢包特征的丢包模型,并且充分考虑了快速恢复阶段和超时后的慢启动阶段对吞吐量的影响.仿真实验表明,该模型可以准确地预测TCP New Reno数据流的吞吐量.     

Marking algorithms for service differentiation of TCP traffic

Class-based service architectures for quality-of-service (QoS) differentiation typically provide loss, throughput, and delay differentiation. However, proposals for class-based service differentiation generally do not account for the needs of TCP traffic, which are characterized by a coupling of packet losses and achievable throughput. Ignoring this coupling may result in poor service differentiation at the microflow level. This paper shows how Explicit Congestion Notification (ECN) can be used to achieve service differentiation for TCP traffic classes at the microflow level. We present a traffic-marking algorithm for routers, which, if used in conjunction with ECN, regulates the transmission rate of TCP sources in such a way that packet drops due to buffer overflows are avoided. We demonstrate how the algorithm can be integrated in a service architecture with absolute and proportional QoS guarantees. Simulation results illustrate the effectiveness of the presented algorithms at avoiding packet losses and regulating traffic for meeting service guarantees, and provide a comparison with other algorithms proposed in the literature.     

基于测量的TCP拥塞控制的公平性研究

通过分析传统TCP算法的局限性，讨论TCP Vegas、TCPW两种基于源端实时带宽测量拥塞控制算法的原理以及带宽分配的公平性，结合主动队列管理技术，提出一种基于加权缓存区容量分配RED算法．理论分析和仿真实验表明该算法提高了带宽分配的公平性．保持了网络的高吞吐量，并实现服务QoS保证.     

无线异构网络环境中基于拥塞状态区分的TCP Vegas 改进算法

在无线网络环境中,TCP Vegas应用时会受到无线信道干扰和噪声的影响,对往返延时（RTT）难以进行准确的估计,导致其性能大幅度降低。结合TCP New Vegas和TCP Vegas A+等的优点提出一种改进算法TCP Vegas-P。该算法针对慢启动过早结束和拥塞避免阶段拥塞出现在反向链路上导致吞吐量下降的问题,以及在与NewReno共存时公平性恶化的问题进行了综合的改进。经仿真实验,改进的算法在无线网络中能够进行比较好的RTT估计,对解决上述Vegas存在的问题达到了良好的效果。     

Packet Spacing: An Enabling Mechanism for Delivering Multimedia Content in Computational Grids

Streaming multimedia with UDP has become increasingly popular over distributed systems like the Internet. Scientific applications that stream multimedia include remote computational steering of visualization data and video-on-demand teleconferencing over the Access Grid. However, UDP does not possess a self-regulating, congestion-control mechanism; and most best-effort traffic is served by congestion-controlled TCP. Consequently, UDP steals bandwidth from TCP such that TCP flows starve for network resources. With the volume of Internet traffic continuing to increase, the perpetuation of UDP-based streaming will cause the Internet to collapse as it did in the mid-1980's due to the use of non-congestion-controlled TCP. To address this problem, we introduce the counter-intuitive notion of inter-packet spacing with control feedback to enable UDP-based applications to perform well in the next-generation Internet and computational grids. When compared with traditional UDP-based streaming, we illustrate that our approach can reduce packet loss over 50% without adversely affecting delivered throughput.     

Receiver-driven bandwidth sharing for TCP and its application to video streaming

Applications using Transmission Control Protocol (TCP), such as web-browsers, ftp, and various peer-to-peer (P2P) programs, dominate most of the Internet traffic today. In many cases, users have bandwidth-limited last mile connections to the Internet which act as network bottlenecks. Users generally run multiple concurrent networking applications that compete for the scarce bandwidth resource. Standard TCP shares bottleneck link capacity according to connection round-trip time (RTT), and consequently may result in a bandwidth partition which does not necessarily coincide with the user's desires. In this work, we present a receiver-based bandwidth sharing system (BWSS) for allocating the capacity of last-hop access links according to user preferences. Our system does not require modifications to the TCP protocol, network infrastructure or sending hosts, making it easy to deploy. By breaking fairness between flows on the access link, the BWSS can limit the throughput fluctuations of high-priority applications. We utilize the BWSS to perform efficient video streaming over TCP to receivers with bandwidth-limited last mile connections. We demonstrate the effectiveness of our proposed system through Internet experiments.     

Architecture of a Web server accelerator

We describe the design, implementation and performance of a high-performance Web server accelerator which runs on an embedded operating system and improves Web server performance by caching data. It can serve Web data at rates an order of magnitude higher than that which would be achieved by a high-performance Web server running on similar hardware under a conventional operating system such as Unix or NT. The superior performance of our system results in part from its highly optimized communications stack. In order to maximize hit rates and maintain updated caches, our accelerator provides an API which allows application programs to explicitly add, delete, and update cached data. The API allows our accelerator to cache dynamic as well as static data. We describe how our accelerator can be scaled to multiple processors to increase performance and availability. The basic design alternatives include a content router or a TCP router (without content routing) in front of a set of Web cache accelerator nodes, with the cache memory distributed across the accelerator nodes. Content-based routing reduces cache node CPU cycles but can make the front-end router a bottleneck. With the TCP router, a request for a cached object may initially be sent to the wrong cache node; this results in larger cache node CPU cycles, but can provide a higher aggregate throughput, because the TCP router becomes a bottleneck at a higher throughput than the content router. We quantify the throughput ranges in which different designs are preferable. We also examine a combination of content-based and TCP routing techniques. In addition, we present statistics from critical deployments of our accelerator for improving performance at highly accessed Sporting and Event Web sites hosted by IBM.     

TCP Vegas拥塞控制机制的改进

针对Vegas拥塞控制技术的不足，提出了一种改进方法。首先对发送分组根据长度进行分类，分别计算各长度类别的“往返时间”（RTT）和RTT最小值（base_rtt）；在更新发送窗口以及设置重传定时器时，根据分组长度选择属于同一长度范围的RTT和base_rtt值作为输入参数。这在一定程度上消除分组长度差异对发送窗口更新和重传定时器设定值所产生的影响，进一步提高协议吞吐率。仿真结果表明，改进的Vegas能够将原协议的吞吐率提高多达24％。     

一种改进的TCP拥塞控制算法

TCP Reno拥塞控制机制是目前互联网中采用的主流拥塞控制算法.根据TCP Reno实现拥塞避免与拥塞控制的AIMD算法中加性因子与减性因子过于武断,对可用带宽的探测缺乏细分,造成信道利用率未达合理水平等缺陷,为此,提出一种根据拥塞窗口的增长情况进行更为平滑的信道容量探测算法,采用基于对数的增长算法和下降算法,通过N...     

An accurate closed-form formula for the throughput of long-lived TCP connections in IEEE 802.11 WLANs

There is a vast literature on the performance modeling of the 802.11 MAC protocol, but the interplay between the TCP dynamics and the self-regulating behavior of the 802.11 CSMA/CA access method has not been sufficiently investigated. In addition, it has been observed that TCP stations in a WLAN are sporadically active due to the TCP flow control mechanisms. This makes traditional saturated models of the network capacity of 802.11 WLANs not very accurate. We presented a rigorous analytical model to calculate the distribution of the number of active TCP stations in a WLAN with multiple long-lived TCP flows [R. Bruno, M. Conti, E. Gregori, Performance modelling and measurements of TCP transfer throughput in 802.11-based WLAN, in: Proceedings of the IEEE MsWiM 2006, Torremolinos, Malaga, Spain, 2006, pp. 4–11, [1]]. Starting from this analysis, in this paper we derive a simple but accurate closed-form expression of the per-connection TCP throughout as a function of the average duration of collisions, the average backoff period and the TCP packet size. We validate this formula through performance tests carried out on a real WLAN. Then, we use our model to mathematically study the optimal minimum contention window size to maximize the TCP throughput. Our analytical results indicate that the initial window size specified in the 802.11b standard is about two times larger than the optimal one. However, we also show that the TCP throughput performance is not very sensitive to changes of the minimum contention window size, and that the TCP flows significantly underutilize the channel bandwidth also in the optimal MAC operating state.     

TCP Performance in Flow-Based Mix Networks: Modeling and Analysis

Anonymity technologies such as mix networks have gained increasing attention as a way to provide communication privacy. Mix networks were developed for message-based applications such as e-mail, but researchers have adapted mix techniques to low-latency flow-based applications such as anonymous Web browsing. Although a significant effort has been directed at discovering attacks against anonymity networks and developing countermeasures to those attacks, there is little systematic analysis of the quality of service (QoS) for such security and privacy systems. In this paper, we systematically address TCP performance issues of flow-based mix networks. A mix's batching and reordering schemes can dramatically reduce TCP throughput due to out-of-order packet delivery. We developed a theoretical model to analyze such impact and present formulas for approximate TCP throughput in mix networks. To improve TCP performance, we examined the approach of increasing TCP's duplicate threshold parameter and derived formulas for the performance gains. Our proposed approaches will not degrade the system anonymity degree since they do not change the underlying anonymity mechanism. Our data matched our theoretical analysis well. Our developed theoretical model can guide the deployment of batching and reordering schemes in flow-based mix networks and can also be used to investigate a broad range of reordering schemes.     

TCP Slow Start阶段流量稳定现象的研究

该文提出了在TCPSlowStart阶段出现的一种流量稳定的现象,并对这种现象产生的原因进行了详尽的分析,然后通过仿真试验,验证了自己的观点,并说明了这种现象对Internet流量研究的重要性。     

Adaptive end-to-end compression for variable-bandwidth communication

In this paper, we present a model for transport level compression with dynamic compression level adaptation, and we present an implementation of this model in Linux TCP, together with measurements showing what performance to expect when using adaptable end-to-end compression over wireless radio links and over an Ethernet. Measurements from our implementation show that for link speeds up to 10 Mbps, employing compression yields a throughput gain for most types of data. For slower speeds (e.g. 1–2 Mbps wireless radio links) this gain is significant (for the Calgary Compression Corpus files, the throughput increase is typically in the range of 50–200%, depending on link speed and data characteristics).     

Evaluating TCP-friendliness in light of Concurrent Multipath Transfer

In prior work, a CMT protocol using SCTP multihoming (termed SCTP-based CMT) was proposed and investigated for improving application throughput. SCTP-based CMT was studied in (bottleneck-independent) wired networking scenarios with ns-2 simulations. This paper studies the TCP-friendliness of CMT in the Internet. In this paper, we surveyed historical developments of the TCP-friendliness concept and argued that the original TCP-friendliness doctrine should be extended to incorporate multihoming and SCTP-based CMT.Since CMT is based on (single-homed) SCTP, we first investigated TCP-friendliness of single-homed SCTP. We discovered that although SCTP’s congestion control mechanisms were intended to be “similar” to TCP’s, being a newer protocol, SCTP specification has some of the proposed TCP enhancements already incorporated which results in SCTP performing better than TCP. Therefore, SCTP obtains larger share of the bandwidth when competing with a TCP flavor that does not have similar enhancements. We concluded that SCTP is TCP-friendly, but achieves higher throughput than TCP, due to SCTP’s better loss recovery mechanisms just as TCP-SACK and TCP-Reno perform better than TCP-Tahoe.We then investigated the TCP-friendliness of CMT. Via QualNet simulations, we found out that one two-homed CMT association has similar or worse performance (for smaller number of competing TCP flows) than the aggregated performance of two independent, single-homed SCTP associations while sharing the link with other TCP connections, for the reason that a CMT flow creates a burstier data traffic than independent SCTP flows. When compared to the aggregated performance of two-independent TCP connections, one two-homed CMT obtains a higher share of the tight link bandwidth because of better loss recovery mechanisms in CMT. In addition, sharing of ACK information makes CMT more resilient to losses. Although CMT obtains higher throughput than two independent TCP flows, CMT’s AIMD-based congestion control mechanism allows other TCP flows to co-exist in the network. Therefore, we concluded that CMT is TCP-friendly, similar to two TCP-Reno flows are TCP-friendly when compared to two TCP-Tahoe flows.