一种新的IPv6任播地址解析协议及应用

任播技术是一种新的网络应用。任播通信是支特导向服务地址的，但是目前IPv6中它的很多定义是模糊的，并且在控制路由方面没有选成一致的协议标准，因此节点间任播通信仍然无法实现。首先说明了基于IPv6的任播通信及目前在该领域中存在许多可能的应用，并指出一些问题和与之对应的解决办法，随后介绍了一种使用特殊任播地址来建立TCP连接的任播地址解析协议(AARP)，使得任播地址在适应已经存在的应用和高层协议时，不需要修改或者扩展的情况下更加有效。     

NAT-PT技术在IPv4和IPv6互联中的实现

介绍NAT-PT及与其相关的网间协议转换技术,采用NAT-PT技术设计IPv4/IPv6转换网关。通过该网关在实际网络环境中运行,测试NAT-PT对UDP协议和TCP协议的翻译,最终实现IPv4和IPv6之间的双向通信功能,满足网络高速发展的需求。     

NAT-PT技术在IPv4和IPv6互联中的实现

介绍NAT-Prr及与其相关的网间协议转换技术,采用NAT-PT技术设计IPv4/IPv6转换网关.通过该网关在实际网络环境中运行,测试NAT-PT对UDP协议和TCP协议的翻译,最终实现IPv4和IPv6之间的双向通信功能,满足网络高速发展的需求.     

部署IPv6面临的问题及解决方法

IPv6针对IPv4的缺陷在网络层做了很大的改变,除了采用128位地址解决IPv4存在的地址资源不足之外,还对报头进行了较大的改进,使之与IPv4的编址方式相比具有更大的灵活性。IPv6只对传输层的TCP、UDP和下层的数据链路层做了很少的修改,这点在工程化过程中非常重要,因为部署单纯的IPv6可以沿用IPv4的经验,甚至比IPv4还要简单。然而,目前部署IPv6还将面临下面两个主要问题。     

嵌入式IPv6协议栈的设计与实现

提出了一种基于平台无关性的嵌入式IPv6协议栈的实现方法.根据嵌入式的应用,运用了Mbuf技术实现对缓冲区的管理,设计了系统功能模块,裁减了IPv6中ICMPv6和邻居发现模块的部分功能,并扩展了TCP、UDP模块.在开发平台FS2410上对此协议栈进行测试,实测效果良好.     

三种支持共存的技术

1IPv6/IPv4双协议栈技术简单地说,双栈机制就是使IPv6网络节点具有一个IPv4栈和一个IPv6栈,同时支持IPv4和IPv6协议。IPv6和IPv4是功能相近的网络层协议,两者都应用于相同的物理平台,以及承载相同的传输层协议TCP或UDP,如果一台主机同时支持IPv6和IPv4协议,那么该主机就可以和仅支持IPv4或IPv6协议的主机通信。2隧道技术隧道机制就是必要时将IPv6数据包作为数据封装在IPv4数据包里,使IPv6数据包能在已有的IPv4基础设施(主要是指IPv4路由器)上传输。随着IPv6的发展,出现了一些被运行IPv4协议的骨干网络隔离开的局部IPv6…     

IPv4-IPv6组播过渡技术

要使IPv4主机与IPv6主机进行组播通信，必须做协议转换工作。采用多播转换网关（MTG）技术方案能较好地实现IPv4网和IPv6网之间的组播互通。方法是将MTG部署在lPv4和IPv6网络的边界，将IPv4网络和IPv6网络视为地位对等的两个异构网络。IPv6主机可以加入组播源位于IPv4网络的组播组，IPv4主机也可以加入组播源位于IPv6网络的组播组。在IPv4中，MTG作为IPv6的代理，参与IPv4的组播；同样，MTG在IPv6中则作为IPv4的代理。在MTG系统内部，两个代理之间进行协议转换。     

基于Mtrace的Ipv6组播监听的实现

针对日益增多的高带宽需求的多媒体应用，下一代互联网协议——IPv6提供了增强的组播技术。在IPv4向IPv6推进的过程中，一些IPv6环境下组播相关的问题亟待解决，其中之一就是提供用于组播管理的工具。介绍了IPv6组播相关技术，并基于Mtrace在开源的IPv6协议栈——KAME上实现了一个IPv6组播路由查找工具。     

基于2-D稳定条件的混合流网络参数设置

由于网络容量的限制,任何一个网络都不可能避免拥塞问题。传统的RED算法只考虑了少量TCP用户,没有涉及UDP用户的情况,同时随着网络应用的多样化,如越来越多的UDP用户接入网络,传统的RED机制无法控制它们,保证不了TCP用户的服务质量(QoS)。针对TCP/UDP混合流多用户的情况,本文提出TCP/UDP混合流的区分控制,这里TCP和UDP流使用不同的带宽,及TCP/UDP混合流单瓶颈网络的2-D稳定条件。基于该稳定条件可以选择一个合适的RED控制参数Pmax,获得满意的网络拥塞控制性能。本文建立了一个TCP和UDP流单瓶颈网络的线性时滞系统模型,利用2-D拉普拉斯-Z变换,推导出基于稳定条件的混合流网络参数配置。利用NS2仿真验证所提出的混合流网络参数配置能够有效获得关于路由器队列长度和TCP窗口的稳定性。     

嵌入式精简IPv6协议栈的设计

总结了设计嵌入式TCP/IP协议栈使用的技术，重点论述了协议栈中IPv6与IPv4不同部分的设计思想。     

Towards end-host-based identification of competing protocols against TCP in a bottleneck link

Classical Transmission Control Protocol (TCP) designs have never considered the identity of the competing transport protocol as useful information to TCP sources in congestion control mechanisms. When competing against a TCP flow on a bottleneck link, a User Datagram Protocol (UDP) flow can unfairly occupy the entire link bandwidth and suffocate all TCP flows on the link. If it were possible for a TCP source to know the type of transport protocol that deprives it of link access, perhaps it would be better for the TCP source to react in a way which prevents total starvation. In this paper, we use coefficient of variation and power spectral density of throughput traces to identify the presence of UDP transport protocols that compete against TCP flows on bottleneck links. Our results show clear traits that differentiate the presence of competing UDP flows from TCP flows independent of round-trip times variations. Signatures that we identified include an increase in coefficient of variation whenever a competing UDP flow joins the bottleneck link for the first time, noisy spectral density representation of a TCP flow when competing against a UDP flow in the bottleneck link, and a dominant frequency with outstanding power in the presence of TCP competition only. In addition, the results show that signatures for congestion caused by competing UDP flows are different from signatures due to congestion caused by competing TCP flows regardless of their round-trip times. The results in this paper present the first steps towards development of more ’intelligent’ congestion control algorithms with added capability of knowing the identity of aggressor protocols against TCP, and subsequently using this additional information for rate control.     

Fixed-point analysis of a network of routers with persistent TCP/UDP flows and class-based weighted fair queuing

Fixed-point models have already been successfully used to analytically study networks consisting of persistent TCP flows only, or mixed TCP/UDP flows with a single queue per link and differentiated buffer management for these two types of flows. In the current study, we propose a nested fixed-point analytical method to obtain the throughput of persistent TCP and UDP flows in a network of routers supporting class-based weighted fair queuing allowing the use of separate queues for each class. In particular, we study the case of two classes where one of the classes uses drop-tail queue management and is intended for only UDP traffic. The other class targeting TCP, but also allowing UDP traffic for the purpose of generality, is assumed to employ active queue management. The effectiveness of the proposed analytical method is validated in terms of accuracy using ns-3 simulations and the required computational effort.     

传输实时多媒体流的新协议TFRC

当前实际网络中传输实时多媒体流使用UDP协议；但UDP与TCP间不公平。基于此近几年提出了TCP友好的概念，TFRC是其中发展较好的一个协议，是当前的一个研究热点。由于国内对其研究较少，对TFRC很有介绍的必要。     

以太网通信在网络化测试中的实现

文章详细介绍了网络化测试的一个基本通信技术----以太网通信技术的研究和实现。使用INTEL386EX设计一块以太网通信测试单板NCEX,使用C语言实现一个在该单板上运行的嵌入式TCP/IP协议栈,单板成为UDP服务器;并使用Visual C＋＋ 6.0的Socket功能在计算机上设计UDP客户端,实现多台计算机通过以太网和UDP/IP协议与NCEX测试板同时通信。     

Wireless video streaming with TCP and simultaneous MAC packet transmission (SMPT)

Video streaming is expected to account for a large portion of the traffic in future networks, including wireless networks. It is widely accepted that the user datagram protocol (UDP) is the preferred transport protocol for video streaming and that the transmission control protocol (TCP) is unsuitable for streaming. The widespread use of UDP, however, has a number of drawbacks, such as unfairness and possible congestion collapse, which are avoided by TCP. In this paper we investigate the use of TCP as the transport layer protocol for streaming video in a multi‐code CDMA cellular wireless system. Our approach is to stabilize the TCP throughput over the wireless links by employing a recently developed simultaneous MAC packet transmission (SMPT) approach at the link layer. We study the capacity, i.e. the number of customers per cell, and the quality of service for streaming video in the uplink direction. Our extensive simulations indicate that streaming over TCP in conjunction with SMPT gives good performance for video encoded in a closed loop, i.e. with rate control. We have also found that TCP is unsuitable (even in conjunction with SMPT) for streaming the more variable open‐loop encoded video. Copyright © 2004 John Wiley & Sons, Ltd.     

Visual C++6.0环境下Winsock的研究与应用

首先对Visual C＋＋6．0环境下Winsock API函数进行研究分析和比较，并且结合TCP／IP和UDP协议进行对比分析。然后研究Winsock网络通信的基本方法和步骤，同时使用扩展的UDP协议实现在Intemet环境下大问题文件的传送和接收过程。最后给出了部分程序的代码。     

用Indy9.0组件实现局域网C/S模式的通信

单纯用TCP或UDP实现局域网通信存在很多不足,为了建立高效、稳定、可靠的通信。并以多媒体教学系统的软件开发为例,在Delphi平台上,运用Indy9.0组件,将UDP和TCP结合起来,实现了局域网中C/S模式的通信。首先简述TCP,UDP协议的特点,对网络中三种基本的数据传输模式进行了比较,然后介绍Indy技术,最后写出实现这种方式的关键步骤和程序。     

Sharing small optical buffers between real-time and TCP traffic

Buffers in emerging optical packet routers are expensive resources, and it is expected that they would be able to store at most a few tens of KiloBytes of data in the optical domain. When TCP and real-time (UDP) traffic multiplex at an optical router with such small buffers (less than 50 KB), we recently showed that UDP packet loss can increase with increasing buffer size. This anomalous loss behaviour can negatively impact the investment made in larger buffers and degrade quality of service. In this paper we explore if this anomalous behaviour can be alleviated by dedicating (i.e., pre-allocating) buffers to UDP traffic. Our contributions within this context are two fold. First, we show using extensive simulations that there would seem to be a critical buffer size above which UDP benefits with dedicated buffers that protect it from the aggressive nature of TCP. However, for smaller buffers that are below this critical value, UDP can benefit by time-sharing the buffers with TCP. Second, we develop a simple linear model that quantitatively captures the combined utility of TCP and real-time traffic for shared and dedicated buffers, and propose a method to optimise the buffer split ratio with the objective of maximising the overall network utility. Our study equips designers of optical packet switched networks with quantitative tools to tune their buffer allocation strategies subject to various system parameters such as the ratio of traffic mix and relative weights associated with TCP and UDP traffic.