一种IP网络多媒体传输方案

随着网络技术的迅速发展,图像和视频等多媒体业务已逐渐成为信息处理领域主要的信息媒体形式。IP多媒体业务就是利用IP网络(Internet、Intranet和LAN等)传送多媒体通信的业务,包括语音、数据和视频等,既有非实时业务,又有实时业务,它们对通信网络的要求并不相同。多媒体数据传输具有实时性要求,对网络传输时延和时延抖动等较为敏感。因此,网络必须对多媒体应用的分组进行特殊处理,以保证以最小的时延和时延抖动提供可预见的服务。文中从多媒体通信的特征入手,从QoS的角度介绍了综合业务和区分业务,针对它们各自的优缺点,提出了一种IP多媒体传输的解决方案。     

IP网络发展新趋势

近几年来，随着技术的进步和市场的推动，人们正不断开发出许多具有实时和多媒体特性的网络应用，譬如远程教育、远程医疗、视频点播、会议电视等。这些业务的共同特点是：对网络提出的QOS（服务质量，即时延、叶延抖动、吞吐量等）要求，尤其对实时性要求很高。另外它们一股建立在点到多点或多点到多点的多播（multicast）通信之上，点到点的通信只是一种特例。传统的Internet中，由于网络对业务流的控制能力很弱，故只能提供点到点的“besteffort”服务。不能保证端到端的时延上限。因此它无法有效支持上述业务。为了在Internet上提供包…     

保障流媒体时间结构的网络时延分析

流媒体同步对端到端时延和时延抖动提出了确定的要求,而终端抖动缓存一方面能消除时延抖动的影响,一方面却增加了端到端时延,流媒体同步保障对网络时延的要求不明确。论文从概率保障流媒体同步的角度,确定了保障流媒体同步的抖动缓存容量范围,提出了流媒体同步网络保障的充分条件,针对基于Internet VoIP(Voice over IP)业务的实际网络测试结果,给出了应用流媒体同步网络保障充分条件进行同步保障评价的应用实例并验证了其正确性。     

IP网的QoS保证机制

Qos即服务质量，指发送和接收信皂的用户之间以及用户与传输信息的服务网络之间关于信息传输的质量约定。QoS包括用户要求和网络服务提供的行为两个方面。用户要求指用户在Internet网络上进行多媒体通信时所要求的服务类型以及相应的传输性能和质量。用户最关心的是端到端的QoS，由于多媒体应用的自身特点，所以多媒体应用对IP网QoS的要求同时体现：丢包率，传送时延、时延抖动，网络带宽等方面。     

端到端IP QoS业务平台网络架构

随着互联网的快速发展，基于IP的业务越来越丰富，未来各种业务统一到IP平台是网络发展的大趋势。以多媒体、语音、视频等为代表的应用对IP网的性能提出了更高的要求，要求网络提供服务质量(QoS)保证。当前IP网所提供的是“尽力而为”(best-effort)的服务，由于不具备服务质量保障特性，不能预留带宽，不能限定网络时延，因此无法支持许多新的应用。     

IP QoS服务模型及其相关问题的探讨

1 采用QoS机制的必要性 传统的IP网络只提供“尽力而为”的数据传输能力。随着网络上主机数量的不断增加,网络服务的需求将超过网络提供的能力,从而造成传输时延变化（抖动）、传输时延过大甚至引起分组丢失,也就是说出现了网络拥塞。网络拥塞对一些Internet应用（如电子邮件,文件传输和Web应用）一般不会造成太大影响,但对传输时延要求比较苛刻的实时应用（如多媒体业务）及大多数双向通信业务（如电话业务）却是不能容忍的。     

PC-Phone中的语音质量要素及其测量（VQM）

PC-Phone，也称为宽带电话业务，是指贯通传统电话网和IP网络的一种新的电信业务，这类业务的典型网络配置如图1所示。它是一个融合的业务网络，融合点就是图中的网关设备。通过网关设备进行PSTN和VoIP的信令流和媒体流相互转换，PSTN用户和VoIP用户可以实现业务互通。PSTN网络之所以长盛不衰，是因为它针对语音质量和人与人之间话音交谈的特性进行了大量优化。经过优化的PSTN特别适合承载要求低时延、小抖动和连续低带宽占用的语音业务。     

以太网电路仿真中抖动缓冲器性能对业务时延影响的研究

电路仿真中影响业务时延的因素很多，本文主要研究抖动缓冲器的性能对业务时延的影响。通过对抖动缓冲器建模、仿真，分析了仿真分组大小以及抖动缓冲器的存储容量和业务时延的关系并得出一些结论。在此基础上提出了划分缓冲子区的存储策略并证明了该策略的有效性。     

基于SDH／PDH的网络业务节点抖动指标测试

抖动特性是基于光同步数字体系(SDH)的网络业务节点的重要技术指标。—般来说，网络要能正常工作，业务节点的抖动就必须符合一定的抖动指标。因此，测试抖动等性能指标是验证网络业务节点可靠性的重要工作之一。本着重探讨了网络业务节点抖动指标在工程上的测试方法。     

IP网络的QoS研究

Internet最初的设计目的是高效数据传输,采用的TCP/IP协议族是无连接、基于数据报的传输模式。传统的IP网络(IPv4)提供的是“尽力而为(besteffort)”数据传输,无法保证吞吐量和传送时延等网络服务质量(QoS)。随着网络上越来越多的主机连接在一起,网络服务需求超过网络的服务能力,造成传输时延变化(抖动),甚至引起分组丢失,造成网络拥塞。虽然网络拥塞对某些Internet应用(如E-mail、ftp或http)影响不大,但对传输时延要求较苛刻的实时应用(如多媒体业务)及大多数双向通信业务(如电话业务)影响很大,因此如何在IP网络传输多媒体业务成为…     

Towards a better-than-best-effort forwarding service for multimediaflows

A salient requirement of interactive multimedia applications is that they transmit data continuously at uniform rates with minimum possible end-to-end delay. The majority of these applications do not require hard and fast guarantees of network performance, but the current best-effort forwarding model of the Internet is frequently insufficient for realizing these requirements. Worse still, the requirement of uniform-rate transmission puts many multimedia applications at odds with current and proposed Internet network management practices that assume or require TCP-like reactions to packet loss. We are investigating router-based active queue management, specifically the use of queue occupancy thresholds to isolate TCP flows and to provide a better-than-best-effort forwarding service for flows in need of uniform-rate transmissions. Our current scheme, class-based thresholds (CBT), relies on a packet marking mechanism such as those proposed for realizing differentiated services on the Internet. CBT, when combined with existing active router queue management schemes such as random early detection (RED), provides a performance for TCP that approximates that achievable under a packet scheduling scheme and acceptable performance for multimedia flows. CBT is a simple and efficient mechanism with implementation complexity and run-time overhead comparable to that of RED     

认知 mesh 网络服务区分的动态频谱接入策略

动态频谱接入策略是实现认知无线电网络高效利用频谱的关键。与传统认知无线电网络不同,认知mesh网络中不同QoS需求的多类型业务共同接入,为适应这一特点,提出服务区分的动态频谱接入策略。策略依据业务的QoS需求确立优先级,针对不同优先级业务采取不同的信道接入方案,实时业务依据最优传输延迟期望选择接入信道集合,在减小传输延迟的同时降低数据传输过程授权用户出现的概率,普通业务选择最优理想传输成功概率的信道,降低信道切换概率。理论与实验结果表明,与传统的认知网络频谱接入策略相比,提出的策略能提供不同业务的服务区分,满足实时业务的低延迟需求,降低数据传输的中断率,同时在授权信道空闲率与网络负载较大时吞吐量性能较优。     

Multimedia transmission with adaptive QoS based on real‐time protocols

In this paper, we describe a mechanism for adaptive transmission of multimedia data, which is based on real‐time protocols. The proposed mechanism can be used for unicast or multicast transmission of multimedia data over heterogeneous networks, like the Internet, and has the capability to adapt the transmission of the multimedia data to network changes. In addition, the implemented mechanism uses an inter‐receiver fairness function in order to treat the group of clients with fairness during the multicast transmission in a heterogeneous environment. The proposed mechanism uses a ‘friendly’ to the network users congestion control policy to control the transmission of the multimedia data. We implement a prototype application based on the proposed mechanism and we evaluate the proposed mechanism both in unicast and multicast transmission through a number of experiment and a number of simulations in order to examine its fairness to a group of clients and its behaviour against transport protocols (TCP) and UDP data streams. Copyright © 2003 John Wiley & Sons, Ltd.     

Adaptive and reliable video transmission over UMTS for enhanced performance

This paper proposes a mechanism for the congestion control for video transmission over universal mobile telecommunications system (UMTS). Our scheme is applied when the mobile user experiences real‐time multimedia content and adopts the theory of a widely accepted rate control method in wired networks, namely equation‐based rate control. In this approach, the transmission rate of the multimedia data is determined as a function of the packet loss rate, the round trip time and the packet size and the server explicitly adjusts its sending rate as a function of these parameters. Furthermore, we examine the performance of the UMTS for real‐time video transmission using real‐time protocols. Through a number of experiments, we measure performance parameters such as end‐to‐end delay, delay in radio access network, delay jitter and throughput in the wireless link. Copyright © 2006 John Wiley & Sons, Ltd.     

基于RTP协议的IP电话QoS监测及提高策略

介绍了多媒体通信的发展趋势和当前存在的问题,对基于RTP协议的网络电话中音频数据传输技术进行了研究,对影响实时传输质量QoS的典型因素进行了分析。在局域网的环境下进行了语音包分析实验,探讨了基于RTP协议的QoS动态监测方法,并提出可行的改进方案。     

The integrated services in the Internet: state of the art

This paper is about the evolution of the Internet from a simple data network into a true multiservice network that can support the emerging multimedia applications and their protocols with appropriate performance and costs. The real-time delivery and specific bandwidth requirements of these multimedia applications have created a need for an integrated services Internet in which traditional best effort datagram delivery can coexist with additional enhanced quality of service delivery classes. The integrated services Internet will be able to commit to meet bandwidth, packet loss, and delay specifications for individual data flows by using the resource reservation protocol together with appropriate packet forward scheduling policies     

网络化多媒体实时监控系统的应用研究

采用MPEG－1编码解码国际标准和TCP／IP网络协议,在网络环境下,实现视音频信息和控制数据的网上传输。介绍了网络化多媒体实时监控系统的的构成、工作原理和功能特点,并对视音频实时编码技术和网络编程技术进行了详细探讨。     

综合业务Internet及其在ATM上的实现

Internet已成为全球范围的网络平台，随着实时和多媒体业务的广泛应用，它正朝提供综合服务的综合业务Internet过渡。同时ATM由于其突出的优点，也开始投入广泛的使用。如何将它们的优势互补，实现有效集成，已成为广泛关注的焦点。文章对综合业务Internet的基本内容进行简述，并对在IP-oer-ATM上实现综合服务的主要问题进行讨论。     

基于Tikhonov正则化的高分辨率群时延测量与计算方法

在衡量传输网络对信号传输时间延迟和信号失真影响时,群时延是非常重要的一项指标.基于差分法计算群时延是目前测量仪器普遍使用的方法,该方法存在着分辨率和精度之间的矛盾,在提高频率分辨率的同时势必引起测量精度的下降.本文在分析差分法误差来源的基础上,基于Tikhonov正则化给出了一种新的群时延计算方法.比较分析得出该方法能够在存在测量误差的情况下,精确得到具有较高频率分辨率的群时延.在实际给出的测量验证中,通过与矢量网络分析仪得到的群时延数据对比,验证了该方法的有效性.     

CORS: A cooperative overlay routing service to enhance interactive multimedia communications

As interactive multimedia communications are developing rapidly on the Internet, they present stringent challenges on end-to-end (E2E) performance. On the other hand, however, the Internet’s architecture (IPv4) remains almost the same as it was originally designed for only data transmission purpose, and has experienced big hurdle to actualize QoS universally. This paper designs a cooperatively overlay routing service (CORS) aiming to overcome the performance limit inherent in the Internet’s IP-layer routing service. The key idea of CORS is to efficiently compose a number of eligible application-layer paths with suitable relays in the overlay network. Besides the direct IP path, CORS can transfer data simultaneously through one or more application-layer paths to adaptively satisfy the data’s application-specific requirements on E2E performance. Simulation results indicate the proposed schemes are scalable and effective. Practical experiments based on a prototype implemented on PlanetLab show that CORS is feasible to enhance the transmission reliability and the quality of multimedia communications.