Modeling TCP SACK performance over wireless channels with completely reliable ARQ/FEC

In this paper, we propose an analytical cross‐layer model for a Transmission Control Protocol (TCP) connection running over a covariance‐stationary wireless channel with a completely reliable Automatic Repeat reQuest scheme combined with Forward Error Correction (FEC) coding. Since backbone networks today are highly overprovisioned, we assume that the wireless channel is the only one bottleneck in the system which causes packets to be buffered at the wired/wireless interface and dropped as a result of buffer overflow. We develop the model in two steps. At the first step, we consider the service process of the wireless channel and derive the probability distribution of the time required to successfully transmit an IP packet over the wireless channel. This distribution is used at the next step of the modeling, where we derive expressions for the TCP long‐term steady‐state throughput, the mean round‐trip time, and the spurious timeout probability. The developed model allows to quantify the joint effect of many implementation‐specific parameters on the TCP performance over both correlated and non‐correlated wireless channels. We also demonstrate that TCP spurious timeouts, reported in some empirical studies, do not occur when wireless channel conditions are covariance‐stationary and their presence in those measurements should be attributed to non‐stationary behavior of the wireless channel characteristics. Copyright © 2011 John Wiley & Sons, Ltd.     

TCP throughput and fairness performance in presence of delay spikes in wireless networks

Wireless networks are likely to experience delay spikes exceeding several times the typical round‐trip‐time figures, which can cause spurious timeouts that lead to unnecessary retransmissions and reduction of the TCP sender's transmission rate, and thus, the throughput of the TCP is degraded. This paper presents some research results on the effect of delay spikes caused by handover on TCP performance by using three different mobility models. It is shown that the throughput of TCP connection over a single bottleneck link is decreased in the presence of delay spikes significantly. Furthermore, it is shown that the fairness feature of TCP is also severely affected in the presence of delay spikes. Copyright © 2005 John Wiley & Sons, Ltd.     

Congestion control for sudden bandwidth changes in TCP

In this paper, we propose a novel technique to deal with sudden bandwidth changes in transmission control protocol (TCP). In the current Internet, sudden bandwidth changes may occur because of vertical handovers between heterogeneous access networks, routing path changes, cognitive ratio, and multi‐rate wireless local area network. The current implementation of TCP is designed and optimized for stable networks and does not adapt well upon sudden bandwidth changes. Consequently, it might suffer from packet losses in burst upon sudden bandwidth decrement and under‐utilization upon sudden bandwidth increment. To resolve this problem, we propose to modify the current TCP algorithm to include a new phase, called fast adaptation (FA). The FA phase is triggered upon detecting sudden bandwidth changes, and a TCP sender in the FA phase attempts to recover lost packets quickly to avoid spurious timeouts upon sudden bandwidth decrement. Upon sudden bandwidth increment, it increases its window size drastically to realize full utilization. Through extensive simulations, experiments, and analysis, it is shown that the proposed scheme can effectively deal with sudden bandwidth changes. Copyright © 2011 John Wiley & Sons, Ltd.     

QoS aware transmission control scheme for multihop WiMAX network

The IEEE 802.16j Mobile Multihop Relay (MMR) WiMAX network allows the number of hops between the end user and the base station to be more than two hops. It supports non‐real‐time Polling Service, which considers the minimum reserved rate and the maximum sustained rate as a QoS requirements. The reliability of sending the data over MMR WiMAX is achieved by using Transmission Control Protocol (TCP) in transport layer and automatic repeat request in the link layer. However, the use of automatic repeat request in the link layer makes the round trip time fluctuate rapidly, which increases the possibility of retransmission timeout (RTO) expiration. TCP performance degrades because of frequent timeout, and hence the QoS transmission rates cannot be satisfied. Therefore, this paper presents an RTO smoothing scheme and QoS aware transmission control to enhance the performance of data transmission over MMR WiMAX networks. The RTO smoothing scheme aims to reduce the frequent timeout occurrences. The slow start threshold and maximum congestion window are adjusted to satisfy the required QoS and it provides transmission rate fairness for the users at different hops. The results showed that, the proposed schemes reduce the timeout, and improve the utilization of the allocated resources and TCP throughput. Copyright © 2012 John Wiley & Sons, Ltd.     

802.11网络中Eifel算法性能的探讨

相对于有线网络来说,无线局域网的无线链路具有带宽低、误码率高、易受影响和电磁波能量容易被吸收等特点。因此,用户数据分组在无线链路上的传输具有传输时延高、传输时延变化大、链路突发中断的特点。这可能会引起错误的超时,触发TCP重传。分析了传统TCP触发无用重发的原因,介绍了Eifel算法消除错误重传的原理,并在NS-2环境下对Eifel和其他版本的TCP进行了仿真和比较。     

Cross-Layer Enhancement to Support TCP-Based Traffics in WLANs

Widespread deployment of wireless local area networks and a gradual increase in streaming applications have brought about a demand for improved quality of service (QoS) in wireless networks. However, increasing user datagram protocol based high priority multimedia traffic and the class differentiation introduced in QoS protocols, has resulted into transmission control protocol (TCP) starvation and increased spurious timeouts. While today’s Internet traffic is still dominated by TCP based applications, the negative effects of IEEE 802.11e enhanced distributed coordination function (EDCF) scheme on TCP performance in the presence of high priority traffic have not been extensively explored. In this paper, the performance of TCP in 802.11e WLAN competing with high priority traffic is examined. The prioritised adaptive enhanced scheme (PAD_EDCF) is proposed. The proposed scheme gives priority to TCP control packets in order to improve the low traffic transmission flow and acquires additional capability of adjusting the MAC parameters based on the traffic load condition. Simulation results demonstrate that the proposed scheme significantly improves TCP performances in terms of traffic efficiency, throughput and reduces delay.     

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.     

区分服务网络中的TCP友好公平体系

区分服务(Differentiated Services)是IETF为实现IP服务质量(QoS)而定义的一个体系结构。研究表明,在该体系中存在不公平问题,该文将TCP友好(TCP Friendly)的概念引入到DS网络中,并定义了DS网络中的TCP友好的公平性,仿真验证了目前IETF定义的流量调节(TrafficConditioning)以及丢包策略等机制不能很好地实现TCP友好公平性,因而提出了直接拥塞控制机制来实现这一公平性。     

TCP连接竞争资源的公平性研究

Internet的迅速发展使得计算机网络的资源分配成为数据通信研究领域的热点.在Internet由单一服务质量过渡到支持多种服务质量的过程中,传输控制协议(TCP)将一直是主要的端到端的资源分配方案.本文深入分析了具有不同传输延时的TCP连接竞争资源时的公平性.研究表明,TCP算法对具有较大传输延时的连接具有不公平性.当竞争带宽的各连接的传输延时按比例减小时,小传输延时的连接对大传输延时的连接的抑制性相对减弱,这种不公平性将减弱.结论对于TCP算法的改进具有重要意义,同时这种不公平性对于设计分级服务方案将有很好的应用前景.     

Optimal retrial and timeout strategies for accessing network resources

The notion of timeout (i.e., the maximal time to wait before retrying an action) occurs in many networking contexts. Use of timeouts is encountered especially in large-scale networks, where negative acknowledgments (NACKs) on failures have significantly higher delays than positive acknowledgments (ACKs) and frequently are not employed at all. Selection of a proper timeout involves a tradeoff between waiting too long and loading the network needlessly by waiting too little. The common approach is to set the timeout to a large value, such that, unless the action fails, it is acknowledged within the timeout duration with a high probability. This approach leads to overly long, far from optimal, timeouts. Our quantitative approach has the purpose of computing and studying the optimal timeout strategy. The tradeoff is modeled by introducing a "cost" per unit time (until success) and a "cost" per repeated attempt. The optimal strategy is then defined as one that a selfish user would follow to minimize its expected cost. We discuss various practical interpretations of these costs. We then derive formulas for the optimal timeout values and study some of their fundamental properties. We identify the worthwhile conditions for making parallel attempts from the outset. We also demonstrate a striking property of positive feedback and study the interaction resulting when many users selfishly apply the optimal timeout strategy; we use a noncooperative game model and show that it suffers from an inherent instability problem. Some implications of these results on network design are discussed.     

A Timeout-Based Congestion Control Scheme for Window Flow-Controlled Networks

During overload, most networks drop packets due to buffer unavailability. The resulting timeouts at the source provide an implicit mechanism to convey congestion signals from the network to the source. On a timeout, a source should not only retransmit the lost packet, but it should also reduce its load on the network. Basedon this realization, we have developed a simple congestion control scheme using the acknowledgment timeouts as indications of packet loss and congestion. This scheme does not require any new message formats, therefore, it can be used in any network with window flow control, e.g., ARPAnet or ISO.     

Channel assignment for initial and handoff calls to improve the call-completion probability

The paper focuses on how to assign channels for initial and handoff calls. Previous schemes give priority to handoff calls by queuing handoff calls, reserving some channels for handoff calls, or subrating existing calls for handoff calls. We queue both initial and handoff calls. We take this idea from derivations of the optimal value for an approximation to the call-completion probability. Our goal is to have higher call-completion probability and still keep forced-termination probability low. We propose four schemes: SFTT (single-queue, FIFO, timeout, average timeout), SPTT (single-queue, priority, timeout, average timeout), DFTS (dual-queues, FIFO, timeout, statistical TDM), and DPTS (dual-queues, priority, timeout, statistical TDM). The four schemes, along with the NPS and FIFO schemes, were simulated and compared. For the SFTT scheme, we also simulated different average timeouts for initial calls. All four proposed schemes have better call-completion probabilities than the NPS and FIFO schemes. Call-completion probabilities can be improved by implementing a priority scheme which serves the waiting call with the least remaining time first. The implementation of statistical multiplexing also has the effect of increasing call-completion probability when the average new-call arrival rates are high. However, both the priority scheme and statistical multiplexing may increase forced-termination probability.     

A study on a receiver‐based management scheme of access link resources for QoS‐controllable TCP connections

Although the bandwidth of access networks is rapidly increasing with the latest techniques such as DSL and FTTH, the access link bandwidth remains a bottleneck, especially when users activate multiple network applications simultaneously. Furthermore, since the throughput of a standard TCP connection is dependent on various network parameters, including round‐trip time and packet loss ratio, the access link bandwidth is not shared among the network applications according to the user's demands. In this thesis, we present a new management scheme of access link resources for effective utilization of the access link bandwidth and control of the TCP connection's throughput. Our proposed scheme adjusts the total amount of the receive socket buffer assigned to TCP connections to avoid congestion at the access network, and assigns it to each TCP connection according to characteristics in consideration of QoS. The control objectives of our scheme are (1) to protect short‐lived TCP connections from the bandwidth occupation by long‐lived TCP connections, and (2) to differentiate the throughput of the long‐lived TCP connections according to the upper‐layer application's demands. One of the results obtained from the simulation experiments is that our proposed scheme can reduce the delay of short‐lived document transfer perceived by the receiver host by up to about 90%, while a high utilization of access link bandwidth is maintained. Copyright © 2006 John Wiley & Sons, Ltd.     

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.     

Comparative performance analysis of versions of TCP in a localnetwork with a lossy link

We use a stochastic model to study the throughput performance of various transport control protocol (TCP) versions (Tahoe (including its older version that we call OldTahoe), Reno, and NewReno) in the presence of random losses on a wireless link in a local network. We model the cyclic evolution of TCP, each cycle starting at the epoch at which recovery starts from the losses in the previous cycle. TCP throughput is computed as the reward rate in a certain Markov renewal-reward process. Our model allows us to study the performance implications of various protocol features, such as fast retransmit and fast recovery. We show the impact of coarse timeouts. In the local network environment the key issue is to avoid a coarse timeout after a loss occurs. We show the effect of reducing the number of duplicate acknowledgements (ACKs) for triggering a fast retransmit. A large coarse timeout granularity seriously affects the performance of TCP, and the various protocol versions differ in their ability to avoid a coarse timeout when random loss occurs; we quantify these differences. We show that, for large packet-loss probabilities, TCP-Reno performs no better, or worse, than TCP-Tahoe. TCP-NewReno is a considerable improvement over TCP-Tahoe, and reducing the fast-retransmit threshold from three to one yields a large gain in throughput; this is similar to one of the modifications in the TCP-Vegas proposal. We explain some of these observations in terms of the variation of fast-recovery probabilities with packet-loss probability. The results of our analysis compare well with a simulation that uses actual TCP code     

An SDN-based true end-to-end TCP for wireless LAN

Segment losses due to intermittent connectivity and mobility lead to sub optimal performance of the Transmission Control Protocol (TCP). This is due to the fact that segment loss is considered as a binary signal for triggering congestion control and retransmission mechanisms at the TCP sender. In wired networks, segments are dropped due to congestion at the routers and the strategy of taking missed acknowledgment as an implicit signal for congestion control performs well. However, in wireless networks, segment losses are primarily due to mobility and transmission errors. Unlike many previous efforts, this paper proposes the design and implementation of Software Defined Network (SDN) assisted TCP which does not require the wireless Access Points (APs) to be TCP-aware and preserves end to end semantics. Further, no changes are required to be done in TCP protocol implementation at the end-hosts. The proposed approach utilizes the programmability provided by the SDN paradigm to intelligently trigger the spurious timeout detection and response algorithms, already implemented in standard TCP. The proposed approach is compared with the standard TCP and SDN assisted Zero Window based approach on Linux kernels using virtual data-plane switches and APs provided by the Mininet-WiFi platform. The implementation results establish the applicability of the proposed approach.

     

A New Analytical Model for TCP Reno with Bursts Error Considered

1　IntroductionInthepastfewyears,moststudiesofTCPpro tocolhaveconcentratedonexperimentsandsimula tions[1～2 ,9～ 1 4 ] .Theprevioustheoreticalstudieson lyconsideredTCP sowncapacity ,butthecharac teristicsofwirelesslinkswerenotcaredaboutenough .Evensomeofthemconsideredwirelesslinks,butmostofthemfailedtoconsidertheeffectsofbursterror[3,1 1 ] whichisanimportantfeatureofwirelesslinks.InRef.[4],ZORZIM proposesaanalyticalmodelforTCP ,buthismodelisverycomplicated ,andalsoatsomeplacesheusesuppe…     

Design of coordinator to eliminate redundant local retransmission in wireless networks

Owing to limited bandwidth, high bit error rate, and bursty error in the wireless environment, the performance of the transmission control protocol (TCP) degrades greatly in wireless networks.Up to now, many researchers have contributed greatly to the wireless TCP field.However, in most of their works, the wireless TCP module usually works in the TCP layer and has no idea of the actual time of the packet transmission, which is determined by the Scheduler in the media access control (MAC) layer, and this will bring the inaccuracy to the local retransmission timeout and induce the redundant local retransmission.In this article, a coordinator is introduced into the base-station (BS), which can provide efficient cooperation between the TCP module and the scheduler module.On the bais of the performance analysis and simulation results, the proposed method is shown to eliminate redundant local retransmission, increase throughput, and improve TCP-level fairness in wireless networks.Moreover, this scheme is orthogonal to those existing wireless TCP schemes, thus it can give great compatibility to the current networks, and further enhance the performance of TCP under the condition that the performance improvement benefiting from the existing approaches will not be affected.     

A Novel M‐ary Code‐Selected Direct Sequence BPAM UWB Communication System

In this letter, a novel M‐ary code‐selected direct sequence (DS) ultra‐wideband (UWB) communication system is presented. Our purpose is to achieve a high data rate by an M‐ary code‐selected direct sequence bipolar pulse amplitude modulation (MCSDS‐BPAM) scheme. In this system, a particular DS code sequence is selected by the log₂M/2 bits from the DS gold code set. This scheme can accomplish both a high data rate without increasing the system bandwidth or changing the pulse shape and improve the BER with an increase of modulation level M even at a lower signal‐to‐noise ratio (SNR). The receiver signal processing algorithm is given for an MCSDS‐BPAM UWB system over an ideal AWGN channel and correlation receivers.     

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