适合TCP的网络编码重传机制

通过理论分析,看出基于反馈的重传方法比定量重传的方法有更低的解码延迟。提出了一种新型的基于反馈的网络编码(FNC)重传机制,利用seen机制中的隐含信息来获取接收方解码所需的重传分组个数,并改变了编码规则使部分分组可以提前解码。该机制不仅可以处理有固定误码率的随机分组丢失,还可以有效地应对大量突发性分组丢失。仿真结果显示,该机制在高误码率下也能保持较高的吞吐量,且极大地减少了解码延迟,传输过程基本不受分组丢失的影响,有效地对拥塞控制协议隐藏了链路错误。算法简单有效,更适于在实际系统中应用。     

A Segment-Based SACK Scheme for TCP Over the Error-Prone Links

The TCP Selective ACK (SACK) is used to notify information on the sequence space of the receiving buffer to the sender. By definition of RFC 2018, however, each SACK block has to be described by the two 32-bit unsigned integers, which may introduce significant extra load in the reverse ACK path when TCP connection is error-prone. Moreover, due to the 40-byte limitation of TCP option field, the available option space may not be sufficient to report all of the sequence blocks when the errors occur in a sudden burst. This may result in the unnecessary spurious retransmissions while the reverse SACK path is severely congested. To overcome such shortcomings, we propose a segment-based TCP SACK scheme in this paper. Both the theoretical analysis and simulation results show that the proposed scheme not only can significantly reduce the SACK overhead but also can avoid the unwanted spurious retransmissions in some specific error-prone scenarios.     

无线Ad Hoc网中的TCP SACK与TCP Vegas

本文用仿真方法分析了TCP SACK和Vegas在无线ab hoc网络中的性能，提出了一种改进的SACK选项格式（ASACK）和一种称为环回时间通知（RN）的新技术以分别用于提高TCP SACK和TCP Vegas在无线ab hoc网中的性能。为了研究路由稳定性TCP Vegas性能的影响，我们实现了一种基于相关性的选路协议（ABR）。     

TCP SACK突发分组丢失吞吐量模型

利用Gilbert分组丢失模型描述端对端突发分组丢失特性,提出了基于RFC6675的快重传和快恢复模型,推导并基于该模型建立TCP SACK吞吐量模型。数值实验和仿真实验表明,快重传和快恢复模型能准确描述基于RFC6675的快重传和快恢复过程;TCP SACK流吞吐量模型估计的准确性得到提升。     

TCP over OBS边缘节点重传机制的研究

使用传输控制协议(TCP)的拥塞控制机制在传统互联网技术中是解决问题的很好方法.然而在面对新型的光网络技术OBS网络,简单地采用TCP技术因受到OBS自身特性的影响,性能表现不佳,吞吐量低下.为此,通过对OBS技术和TCP拥塞控制机制深入的研究,提出一种基于OBS边缘节点的ACK重传机制来提升TCP over OBS性能的方法,提高OBS网络吞吐量.     

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.     

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

Providing reliable data communications over wireless channels is a challenging task because time-varying wireless channel characteristics often lead to bit errors. These errors result in loss of IP packets and, consequently, TCP segments encapsulated into these packets. Since TCP cannot distinguish packet losses due to bit corruption from those due to network congestion, any packet loss caused by wireless channel impairments leads to unnecessary execution of the TCP congestion control algorithms and, hence, sub-optimal performance. Automatic Repeat reQuest (ARQ) and Forward Error Correction (FEC) try to improve communication reliability and reduce packet losses by detecting and recovering corrupted bits. Most analytical models that studied the effect of ARQ and FEC on TCP performance assumed that the ARQ scheme is perfectly persistent (i.e., completely reliable), thus a frame is always successfully transmitted irrespective of the number of transmission attempts it takes. In this paper, we develop an analytical cross-layer model for a TCP connection running over a wireless channel with a semi-reliable ARQ scheme, where the amount of transmission attempts is limited by some number. The model allows to evaluate the joint effect of stochastic properties of the wireless channel characteristics and various implementation-specific parameters on TCP performance, which makes it suitable for performance optimization studies. The input parameters include the bit error rate, the value of the normalized autocorrelation function of bit error observations at lag 1, the strength of the FEC code, the persistency of ARQ, the size of protocol data units at different layers, the raw data rate of the wireless channel, and the bottleneck link buffer size.     

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.     

Analytic models for the latency and steady-state throughput of TCP Tahoe, Reno, and SACK

Continuing the process of improvements made to TCP through the addition of new algorithms in Tahoe and Reno, TCP SACK aims to provide robustness to TCP in the presence of multiple losses from the same window. In this paper we present analytic models to estimate the latency and steady-state throughput of TCP Tahoe, Reno, and SACK and validate our models using both simulations and TCP traces collected from the Internet. In addition to being the first models for the latency of finite Tahoe and SACK flows, our model for the latency of TCP Reno gives a more accurate estimation of the transfer times than existing models. The improved accuracy is partly due to a more accurate modeling of the timeouts, evolution of cwnd during slow start and the delayed ACK timer. Our models also show that, under the losses introduced by the droptail queues which dominate most routers in the Internet, current implementations of SACK can fail to provide adequate protection against timeouts and a loss of roughly more than half the packets in a round will lead to timeouts. We also show that with independent losses SACK performs better than Tahoe and Reno and, as losses become correlated, Tahoe can outperform both Reno and SACK.     

Multiuser detection for packet-switched CDMA networks with retransmission diversity

In this paper, multiuser detection is developed with exploitation of retransmission diversity. A maximum likelihood detector (MLD) that has the same order of complexity as the MLD using only the data in one transmission is developed. It is shown that the retransmission diversity increases the minimum signal distance and, therefore, significantly improves the performance of MLD. The linear MMSE, decorrelating, and MF detectors are developed under two design approaches. In the first approach,we have considered the weighted sum of outputs of a number of linear detectors, where each is implemented for one transmission. The optimum set of weights that maximizes the signal-to-interference-and-noise ratio (SINR) is obtained. In the second approach, a linear detector jointly utilizes the statistics collected from all transmissions. When the SINR of a linear detector is the same in each transmission, the retransmission diversity is shown to increase the SINR by 10log/sub 10/J dB for a packet of J transmissions. The limiting bit error rate for these three linear detectors in large networks with random spreading sequences is obtained in closed form, which enables further network throughput evaluation.     

Nonbinary error detection codes for data retransmission and biterror rate monitoring

The author discusses the use of (n, n-1) polynomial codes for data retransmission and bit error rate (BER) monitoring in nonbinary data transmission systems. For a particular type of polynomial code, called a simple polynomial code, a simple error detection scheme which exploits the Gray coding commonly employed in nonbinary data transmission systems can be devised. Even though its algebraic structure is the same as that of general (n, n -1) polynomial code, the simple polynomial code's performance, when using this detection algorithm, is either equal to or better than that of the corresponding general polynomial code, for data retransmission and BER monitoring. The improvement in the BER monitoring performance of the simple code relative to that of the corresponding general code increases as the data alphabet size becomes larger     

On the impact of link layer retransmission schemes on TCP over 4G satellite links

We study the impact of reliability mechanisms introduced at the link layer on the performance of transport protocols in the context of 4G satellite links. Specifically, we design a software module that performs realistic analysis of the network performance, by utilizing real physical layer traces of a 4G satellite service. Based on these traces, our software module produces equivalent link layer traces, as a function of the chosen link layer reliability mechanism. We further utilize the link layer traces within the ns‐2 network simulator to evaluate the impact of link layer schemes on the performance of selected Transmission Control Protocol (TCP) variants. We consider erasure coding, selective‐repeat automatic request (ARQ) and hybrid‐ARQ link layer mechanisms, and TCP Cubic, Compound, Hybla, New Reno and Westwood. We show that, for all target TCP variants, when the throughput of the transport protocol is close to the channel capacity, using the ARQ mechanism is most beneficial for TCP performance improvement. In conditions where the physical channel error rate is high, hybrid‐ARQ results in the best performance for all TCP variants considered, with up to 22% improvements compared to other schemes. Copyright © 2014 John Wiley & Sons, Ltd.     

FRED-fair random early detection algorithm for TCP over ATMnetworks

The authors present the FRED (fair random early detection) algorithm as a congestion control mechanism for TCP over ATM networks. The FRED algorithm enhances the RED gateway algorithm, by using the fact that TCP connections should be allocated buffer space in proportion to their bandwidth-delay products. Through simulation, the effectiveness of the proposed FRED algorithm is shown as compared with the drop-tail and the original RED algorithms     

Cross-layered retransmission schemes for TCP-based services

Various retransmission schemes for wireless communication systems have been used to improve performance such as reliability and throughput. Each retransmission scheme is designed to improve the performance according to characteristics of each layer of protocol stacks, such as delay components and error control. Especially, a cross-layered retransmission scheme has been proposed to maximize the spectral efficiency by combining a retransmission scheme and adaptive modulation and coding (AMC). However, the cross-layered retransmission scheme is designed for performance improvement at the wireless access networks. The end-to-end performance is not taken into account for modeling of the cross-layered retransmission schemes. It is difficult to design retransmission schemes for the end-to-end performance improvement. In this paper, we analyze the delay and the throughput at the transport layer for the end-to-end performance when a system uses a cross-layered retransmission scheme and the transmission control protocol as the reliable transmission protocols. We also propose a cross-layered retransmission strategy, AMC combined with automatic repeat request (ARQ) and hybrid ARQ (HARQ), to improve end-to-end throughput. From the evaluation results, it is shown that the proposed cross-layered retransmission strategy is suitable for delay insensitive services that require high throughput.     

Reliability-based retransmission criteria for hybrid ARQ

Bit error probability (BEP) and word error probability (WEP) are used as reliability-based retransmission criteria in conjunction with hybrid ARQ (HARQ) protocols. Instead of exploiting an error-detecting code, the decision for a retransmission is based on the error probability of the decoded word, which can be calculated in or after the decoding process.     

基于UVM验证方法学的存储转发系统验证

针对存储转发系统数据随机性、不同接口之间时序异步的特点,提出了不同于典型平台的事物级数据结构和参考模型设计,构建基于System Verilog语言的通用验证方法学(UVM)的验证平台。验证结果表明,此验证平台能够灵活控制随机约束和验证进程,优化验证事务。该平台提高了验证的效率和验证平台的可重用性,较好地满足了超大规模可编程逻辑器件验证需要。     

Hybrid-ARQ scheme using different TCM for retransmission

A new hybrid automatic repeat-request (ARQ) scheme that uses supplementary trellis codes for retransmission is described. In this scheme, supplementary codes are designed to compensate for the squared free distance (d/sub free//sup 2/) of the trellis code used for the previous transmission. Differently encoded codewords are combined and decoded by the same Viterbi decoder at the receiver. This scheme provides improved (d/sub free//sup 2/) upon retransmission compared with that provided by the conventional trellis coded hybrid-ARQ scheme where the same coding scheme is used for all transmissions.