外文文献原文
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 Arabia
Received 25 May 2004; revised 3 June 2005; accepted 3 June 2005. Available online 1 July 2005.
Abstract
Deploying IP telephony or voice over IP (VoIP) is a major and challenging task for data network researchers and designers. This paper outlines guidelines and a step-by-step methodology on how VoIP can be deployed successfully. The methodology can be used to assess the support and readiness of an existing network. Prior to the purchase and deployment of VoIP equipment, the methodology predicts the number of VoIP calls that can be sustained by an existing network while satisfying QoS requirements of all network services and leaving adequate capacity for future growth. As a case study, we apply the methodology steps on a typical network of a small enterprise. We utilize both analysis and simulation to investigate throughput and delay bounds. Our analysis is based on queuing theory, and OPNET is used for simulation. Results obtained from analysis and simulation are in line and give a close match. In addition, the paper discusses many design and engineering issues. These issues include characteristics of VoIP traffic and QoS requirements, VoIP flow and call distribution, defining future growth capacity, and measurement and impact of background traffic.
Keywords: Network Design，Network Management，VoIP，Performance Evaluation，
Analysis，Simulation，OPNET
1 Introduction
These days a massive 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 loss, 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 assessing the deployment of VoIP into the existing network. One approach is based on first performing network measurements and then predicting the network readiness for supporting VoIP. The prediction of the network readiness is based on assessing 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 loss. Other than the cost associated with the commercial tools, none of the commercial tools offer a comprehensive approach for successful 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
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network services and applications, and impact background traffic on VoIP. This paper attempts to address those important factors and layout a comprehensive methodology for a successful 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 discusses many practical issues pertaining to the deployment of VoIP. These issues 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 successfully 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.
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2 Existing network
Fig. 1 illustrates a typical network topology for a small enterprise residing in a high-rise building. The network shown is realistic and used as a case study only; however, our work presented in this paper can be adopted easily for larger and general networks by following the same principles, guidelines, and concepts laid out in this paper. The network is Ethernet-based and has two Layer-2 Ethernet switches connected by a router. The router is Cisco 2621, and the switches are 3Com Superstack 3300. Switch 1 connects Floors 1 and 2 and two servers; while Switch 2 connects Floor 3 and four servers. Each floor LAN is basically a shared Ethernet connecting employee PCs with workgroup and printer servers. The network makes use of VLANs in order to isolate broadcast and multicast traffic. A total of five LANs exist. All VLANs are port based. Switch 1 is configured such that it has three VLANs. VLAN1 includes the database and file servers. VLAN2 includes Floor 1. VLAN3 includes Floor2. On the other hand, Switch 2 is configured to have two VLANs. VLAN4 includes the servers for E-mail, HTTP, Web and cache proxy, and firewall. VLAN5 includes Floor 3. All the links are switched Ethernet 100 Mbps full duplex except for the links for Floors 1–3 which are shared Ethernet 100 Mbps half duplex.
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3 Step-by-step methodology
Fig. 2 shows a flowchart of a methodology of eight steps for a successful VoIP deployment. The first four steps are independent and can be performed in parallel. Before embarking on the analysis and simulation study, in Steps 6 and 7, Step 5 must be carried out which requires any early and necessary redimensioning or modifications to the existing network. As shown, both Steps 6 and 7 can be done in parallel. The final step is pilot deployment.
3.1. VoIP traffic characteristics, requirements, and assumptions
For introducing a new network service such as VoIP, one has to characterize first the nature of its traffic, QoS requirements, and any additional components or devices. For simplicity, we assume a point-to-point conversation for all VoIP calls with no call conferencing. For deploying VoIP, a gatekeeper or Call Manager node has to be added to the network [3,4,5]. The gatekeeper node handles signaling for establishing, terminating, and authorizing connections of all VoIP calls. Also a VoIP gateway is required to handle external calls. A VoIP gateway is responsible for converting VoIP calls to/from the Public Switched
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英文文献 科技类 原文及翻译（电子 电气 自动化 通信）1
