混合网络下的TCP拥塞控制跨层优化算法

针对混合网络中并行链路间TCP流的不公平性,提出一种新的算法.此算法利用跨层设计的思想,以传输层的数据重传率为参数来调整TCP流不公平性,也就是说MAC层上的竞争窗口将根据重传率的动态变化而改变,其目的在于抑制并行链路TCP流接入信道能力的不公平性.并且用仿真工具NS2进行仿真的结果表明,采用改进算法后的网络公平性指数比未改进前提高了17.9％.该算法能明显改善并行链路间TCP流的不公平性.     

Performance limitation of the leaky bucket algorithm for ATMnetworks

The performance limitation of the “leaky bucket algorithm” is analyzed for usage parameter control and traffic management in asynchronous transfer mode (ATM) networks. Simulation results show that the conventional statistical bandwidth allocation method, which uses “the worst pattern derived from the cell interarrival time moments” permitted by the leaky bucket algorithm, can not guarantee the quality of service (QOS). As a result, this paper proves that the VC/VP bandwidth allocation method based on the leaky bucket algorithm is unsatisfactory     

Efficient consolidation algorithm for multicast ABR service in ATMnetworks

A new efficient consolidation algorithm at a branch-point switch for multicast available bit rate (ABR) service is proposed. The scheme can achieve a faster response to congestion status and has a lower consolidation noise compared with existing algorithms. Particularly network environments with a large number of branch points     

Fair queueing algorithm with rate independent delay for ATMnetworks

A new scheduling algorithm based on self-clocked fair queueing is proposed. The algorithm employs an array of sorting bins for managing the virtual finishing time of connections and uses token buckets as a behaviour-indicator. It remedies the drawbacks of existing fair queueing algorithms and is able to reserve bandwidth to connections while guaranteeing a class-dependent and rate-independent delay for behaving connections     

自适应RED算法在OBS网络中的应用

队列管理是光突发交换(OBS)网络边缘节点的重要功能之一.对随机早期检测(RED)队列管理算法及两种自适应RED算法在OBS网络中的应用进行了研究和仿真分析.仿真结果表示:动态调整队列权重的自适应RED算法和动态调整最大丢包率的自适应RED算法均能改进RED算法的性能,但是后者的性能优于前者.     

Optimal bit allocation for coding of video signals over ATMnetworks

We consider optimal encoding of video sequences for ATM networks. Two cases are investigated. In one, the video units are coded independently (e.g., motion JPEG), while in the other, the coding quality of a later picture may depend on that of an earlier picture (e.g., H.26x and MPEGx). The aggregate distortion-rate relationship for the latter case exhibits a tree structure, and its solution commands a higher degree of complexity than the former. For independent coding, we develop an algorithm which employs multiple Lagrange multipliers to find the constrained bit allocation. This algorithm is optimal up to a convex-hull approximation of the distortion-rate relations in the case of CBR (constant bit-rate) transmission. It is suboptimal in the case of VBR (variable bit-rate) transmission by the use of a suboptimal transmission rate control mechanism for simplicity. For dependent coding, the Lagrange-multiplier approach becomes rather unwieldy, and a constrained tree search method is used. The solution is optimal for both CBR and VBR transmission if the full constrained tree is searched. Simulation results are presented which confirm the superiority in coding quality of the encoding algorithms. We also compare the coded video quality and other characteristics of VBR and CBR transmission     

The ERICA switch algorithm for ABR traffic management in ATMnetworks

This paper describes the “explicit rate indication for congestion avoidance” (ERICA) scheme for rate-based feedback from asynchronous transfer mode (ATM) switches. In ERICA, the switches monitor their load on each link and determine a load factor, the available capacity, and the number of currently active virtual channels. This information is used to advise the sources about the rates at which they should transmit. The algorithm is designed to achieve high link utilization with low delays and fast transient response. It is also fair and robust to measurement errors caused by the variations in ABR demand and capacity. We present performance analysis of the scheme using both analytical arguments and simulation results. The scheme is being considered for implementation by several ATM switch manufacturers     

Virtual path long-term bandwidth allocation algorithm for ATMnetworks using simulated annealing

The long-term virtual path (VP) bandwidth allocation process requires the use of heuristics or artificial intelligence to find the number of virtual circuits that can multiplex by each VP to maximise the quality of service, which is defined here in terms of the call blocking probabilities. The authors propose a new technique, based on simulated annealing, for performing the allocation process. It substantially improves upon previous proposals     

Scalable QoS-based IP multicast over label-switching wireless ATMnetworks

A Mobile IP multicast prototype that integrates a label-switching wireless asynchronous transfer mode, the mobile core-based multicast architecture, and an Internet multicast infrastructure is presented. MCOM creates multiple core-based layer 2 multicast trees that are independently established in member networks. They are interconnected via the Internet using layer 3 multicast routing. Gateways on the border of the Internet and wireless ATM networks convert ATM multicast traffic to suitable IP packets as well as converting from IP packets to ATM cells for MCOM. To solve the cell interleaving problem that results, ATM block transfer/immediate transmission capability is reasonably modified. Additionally, class-based block buffer management for ATM multicast connections is built into wireless ATM switches for soft quality of service control. Dynamic group management, multicast channel rerouting, and reliable multicasting are also studied in relation to existing Internet protocols like Mobile IP, Internet group management protocols, and multicast routing protocols     

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.     

拥塞控制中的随机检测算法研究

作为主动队列管理（AQM）的典型代表，RED（Random Early Detection）算法得到了较为广泛的应用。出于改进其稳定性和公平性的考虑，研究者开发出了ARED，SRED，FRED，RIO和WRED等几种算法。重点研究了RED及其多种变种方法的设计思想、算法，并比较了它们的优缺点。在此基础上，分析了IP复接器中适用的算法。     

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.     

Differentiated services TCP algorithm for the Internet

A new concept of the slow-start threshold hopping coefficient in the TCP Reno algorithm is proposed. A modified TCP algorithm is devised for implementing differentiated services over the Internet. A theoretical analysis and simulations are presented to validate the effectiveness of the algorithm     

Measurement-based simulation model for TCP over ATM

Cell level analysis of ATM networks by means of simulation requires an accurate model for traffic sources. We present a simple model for TCP over ATM traffic sources in an ATM LAN, which captures the fundamental characteristics of the behaviour of tcp in this environment. The model was developed by extensive statistical analysis of numerous traffic traces recorded in an atm testbed. Simulated traffic generated by our model has roughly the same properties as observed in real traffic, even on the time scale of milliseconds. This makes it suitable for use by the analysis of various scheduling and congestion management algorithms.     

A Markovian model for TCP over ATM

This paper focuses on the development of an approximate Markov chain model and a detailed simulation model for the performance analysis of TCP connections supporting long file transfers within highspeed ATM networks. The Markov model construction is first illustrated, emphasizing the simplifying assumptions introduced to maintain an acceptable complexity in the performance study. Then the model validation is described, comparing the analytical results with those produced by a very detailed simulation setup in which the officially distributed BSD 4.3 TCPreno code was adapted to run on top of a software tool for the simulation of ATM networks. Numerical results indicate that the analytical model is able to capture the overall behavior of the system with respect to the performance measures we consider, with a cost which is orders of magnitude less than the one necessary to obtain reliable estimates from the detailed simulation model.     

Bandwidth estimation schemes for TCP over wireless networks

The use of enhanced bandwidth estimation procedures within the congestion control scheme of TCP was proposed recently as a way of improving TCP performance over links affected by random loss. This paper first analyzes the problems faced by every bandwidth estimation algorithm implemented at the sender side of a TCP connection. Some proposed estimation algorithms are then reviewed, analyzing and comparing their estimation accuracy and performance. As existing algorithms are poor in bandwidth estimation, and in sharing network resources fairly, we propose TIBET (time intervals based bandwidth estimation technique). This is a new bandwidth estimation scheme that can be implemented within the TCP congestion control procedure, modifying only the sender-side of a connection. The use of TIBET enhances TCP source performance over wireless links. The performance of TIBET is analyzed and compared with other schemes. Moreover, by studying TCP behavior with an ideal bandwidth estimation, we provide an upper bound to the performance of all possible schemes based on different bandwidth estimates.     

TCP Performance over UMTS-HSDPA System

The performance of TCP based applications in a UMTS-HSDPA system is evaluated, using an analytical model. HSDPA (High Speed Downlink Packet Access) is an evolution of the UMTS standard over the air interface to achieve higher bit rates through the introduction of adaptive modulation and coding, Hybrid ARQ, fast scheduling, fast cell selection and MIMO. The proposed model is used to evaluate the effect of the TCP protocol on the bit rate of various data services (at 32, 64 and 128 kbps). The results show that the bit rate per flow decreases strongly if congestion in the wired network increases. The interaction between cell capacity and the TCP layer is also studied. A semi-analytical model to evaluate the cell capacity is also presented in this paper, and the correspondence between cell capacity and guaranteed TCP bit rate is reported in this paper for a Round Robin scheduler. The findings are supported by simulation results.     

基于随机检测算法的网络拥塞控制系统设计

为解决由传输数据堆积而引发的网络拥塞问题,达到缩减网络滞后延迟的目的,设计一种基于随机检测算法的新型网络拥塞控制系统。利用LAIDS/LIDS主机随机检测架构,对拥塞控制器、网络数据过滤器进行调节,完成新型系统的硬件运行环境搭建。通过传输数据拥塞定义的方式对网络中可能出现的拥塞情况进行分类处理,再根据具体判断结果选择合适的网络拥塞控制方式,实现新型系统的软件运行环境搭建。结合软、硬件结构,完成基于随机检测算法的网络拥塞控制系统设计。对比实验结果表明,与现有控制系统相比,应用新型控制系统后单位时间内传输数据堆积量的最大值不超过5.5×10~(13)TB,网络滞后延迟平均值始终低于60 ms,网络拥塞现象得到有效缓解。     

An image change detection algorithm based on Markov random field models

This paper addresses the problem of image change detection (ICD) based on Markov random field (MRF) models. MRF has long been recognized as an accurate model to describe a variety of image characteristics. Here, we use the MRF to model both noiseless images obtained from the actual scene and change images (CIs), the sites of which indicate changes between a pair of observed images. The optimum ICD algorithm under the maximum a posteriori (MAP) criterion is developed under this model. Examples are presented for illustration and performance evaluation.