移动IP的移动性管理机制

本文简单介绍了移动性管理原理，并讨论了移动IP的移动性管理机制，详述了其实现过程。     

移动IP的移动性管理机制

本简单介绍了移动性管理原理，并讨论了移动IP的移动性管理机制，详述了其实现过程。     

移动IP的移动性协议及管理机制

本主要从移动网络协议结构开始，讨论移动IP有关现行的和将来的Inernet移动性协议极其管理机制。     

IP over ATM简介

IP业务急剧发展,全世界大约每6～9个月其业务量就翻一番,因此要求IP网的带宽急剧增长,而且要求IP能够提供新的业务功能(如实时的音频、视频业务)。这就迫使IP业务提供商(ISP)重新规划和设计现有网络;另一方面,ATM技术日趋成熟,并被越来越多的运营商采用。于是解决好IP与ATM的综合发展,是良好的选择之一。     

IP城域网传输技术简介

业务的开展和用户的覆盖，使各个运营商城域网的建设成为一个近期的重点。而业务的泛IP化趋势，导致了城域网上IP类业务的比重在快速地增大，出现了IP城域网的概念。     

VSIA IP标准简介

VSIA（Virtual Socket Interface Alliance）成立于1996年9月．是最早出现的国际性IP标准组织．成员包括系统设计公司、半导体供应商、EDA公司、IP提供商等。其目的是为系统芯片工业建立统一的技术规范和标准,这些规范和标准可以作到使不同来源的IP进行集成并相匹配。2004年以前VSIA是以工作组的形式开展工作，陆续成立了模拟／混合信号、功能验证、依靠硬件的软件、实现／验证、IP保护、与制造相关的测试、片上总线、系统级设计、虚拟组件质量、基于平台的设计和虚拟组件转让，等共11个开发工作组，     

移动IP技术简介

文中主要介绍了当今移动IP技术的基本原理、相关概念和简要工作过程。     

面向宽带IP网络的拥塞控制研究进展

讨论了IP网络中拥塞控制的基本原理,分析和比较了目前一些有代表性的可升级的拥塞控制算法,指出了其中存在的问题,而后总结了控制理论在分析和设计网络拥塞控制算法方面所取得的一些进展,并提出了未来的工作方向。     

对TCP/IP计算机网络拥塞控制的研究

为提升计算机的网络性能,更好地避免拥塞现象的发生,需要对其进行必要的技术控制。鉴于此,对基于TCP/IP协议的网络拥塞控制方法进行分析。在TCP拥塞控制中主要采用TCP Tahoe,TCP Reno,TCP New Reno以及TCP Sack四种方法,其中TCP New Reno对快速恢复算法进行了改进,通过对TCP协议中的Reno进行可视化处理,实行对网络拥塞的有效管理。而IP拥塞控制方法则分为FIFO,FQ和WFQ,RED以及ECN四种类型,通过队列调度管理方式实现了对网络拥塞的有效管理。     

TCP/IP performance with random loss and bidirectional congestion

With the growth in Internet access services over networks with asymmetric links such as asymmetric digital subscriber line (ADSL) and cable-based access networks, it becomes crucial to evaluate the performance of TCP/IP over systems in which the bottleneck link speed on the reverse path is considerably slower than that on the forward path. In this paper, we provide guidelines for designing network control mechanisms for supporting TCP/IP. We determine the throughput as a function of buffering, round-trip times, and normalized asymmetry (defined as the ratio of the transmission time of acknowledgment (ACK) in the reverse path to that of data packets in the forward path). We identify three modes of operation which are dependent on the forward buffer size and the normalized asymmetry, and determine the conditions under which the forward link is fully utilized. We also show that drop-from-front discarding of ACKs on the reverse link provides performance advantages over other drop mechanisms in use. Asymmetry increases the TCP already high sensitivity to random packet losses that occur on a time scale faster than the connection round-trip time. We generalize the by-now well-known relation relating the square root of the random loss probability to obtained TCP throughput, originally derived considering only data path congestion. Specifically, random loss leads to significant throughput deterioration when the product of the loss probability, the normalized asymmetry and the square of the bandwidth delay product is large. Congestion in the reverse path adds considerably to TCP unfairness when multiple connections share the reverse bottleneck link. We show how such problems can be alleviated by per-connection buffer and bandwidth allocation on the reverse path     

A framing strategy for congestion management

A congestion management strategy for integrated services packet networks that is robust with regard to transmission speed and network size is proposed. The strategy supports several classes of services with zero loss and different delay bounds as well as services without stringent loss and delay guarantees. Loss-free and bounded-delay transmission is accomplished by means of an admission policy which ensure smoothness of the traffic at the network edge, and by a service discipline called stop-and-go queuing, which maintains the traffic smoothness throughout the network. Both the admission policy and the stop-and-go queuing are based on a time framing concept described elsewhere by the author (IEEE Trans. Commun., vol.39, Dec.1991). This concept is further developed to incorporate several frame sizes into the strategy, thereby providing flexibility in meeting throughput and delay requirements of different applications. Stop-and-go queueing is realizable with minor modification to a first-in first-out (FIFO) queueing structure     

Quality of service for voice over IP in networks with congestion avoidance

This paper assesses the impact of active queue management schemes on the quality of service of voice over Internet protocol applications. A new analytical method based on a fixed point approach to estimate the end-user satisfaction is proposed. The results obtained were validated using discrete event simulation techniques. In all the studied cases, it was observed a great deal of agreement between the analytical results and the results obtained through simulation. The theoretical predictions, as well as the presented empirical evidences confirm, as demonstrated in previous works, that the use of active queue management offers better quality of service than the traditional queue control mechanisms used in Internet. From these results, we may reasonably conclude that the presented method can be used for network design in the presence of voice traffic.     

Fairness and stability of congestion control mechanisms of TCP

In this paper, we focus on fairness and stability of the congestion control mechanisms adopted in several versions of TCP by investigating their time–transient behaviors through an analytic approach. In addition to TCP Tahoe and TCP Reno, we also consider TCP Vegas which has been recently proposed for higher throughput, and enhanced TCP Vegas, which is proposed in this paper for fairness enhancements. We consider the homogeneous case, where two connections have the equivalent propagation delays, and the heterogeneous case, where each connection has different propagation delay. We show that TCP Tahoe and TCP Reno can achieve fairness among connections in the homogeneous case, but cannot in the heterogeneous case. We also show that TCP Vegas can provide almost fair service among connection, but there is some unfairness caused by the essential nature of TCP Vegas. Finally, we explain the effectiveness of our enhanced TCP Vegas in terms of fairness and throughput. This revised version was published online in June 2006 with corrections to the Cover Date.     

Analysis of congestion control mechanisms in an intelligent network

An SCP overload control method is proposed and its characteristics are evaluated and compared with other methods (window and call gapping) under various overload traffic patterns by using the network simulator that models SCP, SSPs, and their relationships in the IN. © 1998 John Wiley & Sons, Ltd.     

Bandwidth management and congestion control in plaNET

The protocols and mechanisms necessary for network bandwidth management and congestion control are addressed. The discussion draws heavily on the lessons learned from the design and implementation of plaNET, a high-speed packet-switching system for integrated voice, video, and data communications. A general overview of the mechanisms involved is given. The individual components of the system are discussed. Most of the conclusions are general and can be applied to other high-speed networks, including asynchronous transfer mode (ATM) systems     

QoS-driven adaptive congestion control for voice over IP in multiservice wireless cellular networks

This work establishes a QoS-driven adaptive congestion control framework that provides QoS guarantees to VoIP service flows in mixed traffic scenarios for wireless cellular networks. The framework is composed of three radio resource management algorithms: admission control, packet scheduling, and load control. The proposed framework is scalable to several services and can be applied in any current or future packet-switched wireless system. By means of dynamic system-level simulations carried out in a specific case study where VoIP and Web service flows compete for shared access in an HSDPA wireless network, the proposed framework is able to increase the overall system capacity twofold depending on the traffic mix, while keeping the system operating optimally in its target QoS profile.     

An investigation of multilevel service provision for voice over IP under catastrophic congestion

QoS has been the dominating issue faced in VoIP development today. In this article we investigate and empirically evaluate three DiffServ (Kilkki 1999, Blake et al. 1998) design strategies, WRED Dropper (Cisco Systems), PQ meter, and multiple differentiated meter (MDM), and study how multilevel communication services can be guaranteed for multiple VoIP classes. The novel aspect of our work is that we are trying to provide solutions with the use of features that are not currently available in real routers (Westhead, 2002). By taking advantage of network simulation techniques, a series of experiments have been designed and carried out under different network conditions by using the intersim simulation tool (Westhead, 2002). Our simulation results effectively reflect that: 1) all three designs can provide service differentiation for multiple VoIP classes; 2) the highest-priority VoIP class can always be protected under catastrophic congestion; and 3) each design has some pros and cons in terms of performance stability under varied traffic distribution and changed packet size mix. Based on this investigation, we conclude that MDM is comparatively better at providing even service distribution and staying robust against traffic distribution variation and packet size changes.