自动化090425王晓康17孙挺24杨凌云16张瑞
These days a maive On the deployment of VoIP in Ethernet
networks:
methodology and case study
Khaled Salah, Department of Information and Computer Science, King Fahd University of Petroleum and Minerals, P.O.Box 5066, Dhahran 31261, Saudi ArabiaReceived 25 May 2004;revised 3 June 2005;accepted 3 June 2005.Available online 1 July 2005.
1Introduction
deployment of VoIP is taking place over data networks.Most of these networks are Ethernet based and running IP protocol.Many network managers are finding it very attractive and cost effective to merge and unify voice and data networks into one.It is easier to run, manage, and maintain.However, one has to keep in mind that IP networks are best-effort networks that were designed for non-real time applications.On the other hand, VoIP requires timely packet delivery with low latency, jitter, packet lo, and sufficient bandwidth.To achieve this goal, an efficient deployment of VoIP must ensure these real-time traffic requirements can be guaranteed over new or existing IP networks.When deploying a new network service such as VoIP over existing network, many network architects, managers, planners, designers, and engineers are faced with common strategic, and sometimes challenging, questions.What are the QoS requirements for VoIP? How will the new VoIP load impact the QoS for currently running network services and applications? Will my existing network support VoIP and satisfy the standardized QoS requirements? If so, how many VoIP calls can the network support before upgrading prematurely any part of the existing network hardware? These challenging questions have led to the development of some commercial tools for testing the performance of multimedia applications in data networks.A list of the available commercial tools that support
VoIP is listed in [1,2].For the most part, these tools use two common approaches in aeing the deployment of VoIP into the existing network.One approach is based on first performing network measurements and then predicting the network readine for supporting VoIP.The prediction of the network readine is based on aeing the health of network elements.The second approach is based on injecting real VoIP traffic into existing network and measuring the resulting delay, jitter, and lo.Other than the cost aociated with the commercial tools, none of the commercial tools offer a comprehensive approach for succeful VoIP deployment.In particular, none gives any prediction for the total number of calls that can be supported by the network taking into account important design and engineering factors.These factors include VoIP flow and call distribution, future growth capacity, performance thresholds, impact of VoIP on existing network services and applications, and impact background traffic on VoIP.This paper attempts to addre those important factors and layout a comprehensive methodology for a succeful deployment of any multimedia application such as VoIP and video conferencing.However, the paper focuses on VoIP as the new service of interest to be deployed.The paper also contains many useful engineering and design guidelines, and discues many practical iues pertaining to the deployment of VoIP.These iues include characteristics of VoIP traffic and QoS requirements, VoIP flow and call distribution, defining future growth capacity, and measurement and impact of background traffic.As a case study, we illustrate how our approach and guidelines can be applied to a typical network of a small enterprise.The rest of the paper is organized as follows.Section 2 presents a typical network topology of a small enterprise to be used as a case study for deploying VoIP.Section 3 outlines practical eight-step methodology to deploy succefully VoIP in data networks.Each step is described in considerable detail.Section 4 describes important design and engineering decisions to be made based on the analytic and simulation studies.Section 5 concludes the study and identifies future work.
外文文献译文
以太网网络电话传送调度：方法和案例分析
Khaled Salah , 信息与计算机科学, Fahd University 皇家石油矿物大学, 信箱5066, Dhahran 31261, 沙特阿拉伯半岛 收到日期2004 年5月25 日; 校正日期2005 年6月3 日; 接受日期2005 年6月3 日。网上可用日期2005 年7月1 日。
1绪论
最近大量的网络电话调度在数据网中占有相当的比例。其中大部分基于以太网和运行IP协议。不少网络管理员发现把语音和数据网合二为一非常有吸引力和具成本效应。这将更易于运行、管理和维护。然而，人们应该认识到IP网络目的是为了非实时应用服务；另一方面，网络电话要求带有低延迟、低抖动、低丢包率和充足的带宽。为了达到这一目标，必须保证在现有或新的IP网络中完成实时通信要求的高效网络电话调度。当在现有的网络部署譬如VoIP这样的新网络服务，许多网络架构师、经理、计划师、设计师和工程师面临着共同的目标和挑战。什么是网络电话的服务质量要求？新的网络电话负荷怎样冲击了当今运行中的网络服务和应用的质量？我们现有的网络将支持网络电话和将满足标准的服务质量要求吗？如果那样，在过早升级任何现有的网络硬件的零件前，能支持多少个的VoIP电话网络？这些富挑战性的问题导致了用于测试网络多媒体数据应用性能的商业工具的发展。支持网络电话的商用工具名单如表[1,2]。很大程度上，这些工具使用两种共同的方法把VoIP 部署入现有的网络。一种方法根据第一次执行的网络测量和然后预测支持VoIP的网络应该就绪情况。预测的网络情况是根据网络要素来估计的。第二种是根据加入到现有网络中VoIP的实时通信信息，然后测出延时时间、时基误差和丢包率。除相关的商业工具费用外，没有其它工具能够提供全面兼容的成功网络电话调度方法。特别是，无任何一个预测能给出网络可以支持的呼叫次数已提到设计和工程的议事日程上来。这些因素包括网络电话流程和呼叫分配、今后增长容量、性能极限、VoIP对现有网络服务和应用以及后台通信能力的冲击。本文尝试研究这些重要因素，规划出一个支
持像网络电话和视频会议系统的全面可行方法。
无论如何，本文集中论述了网络电话服务调度新方法。文章同样包含了许多工程和设计指南，同时也讨论了许多与网络电话相关的实际问题。这些问题包括网络电话的通信和服务质量要求，网络电话数据流向和呼叫分配，定义今后的增长容量以及我们的方法和纲领怎样应用在像小型的企业网这样的典型网络中。其余部分作如下安排：第二部分论述了小型企业网的典型网络拓扑如何在网络电话中实现调度。第三部分大致描述了网络电话数据网调度实用的八个步骤。每一步骤都有详细的说明。第四部分介绍了基于分析和仿真研究得出的重要设计结论和工程结论。第五部分谈到了今后研究还需做的具体工作。
