英文文献_科技类_原文及翻译_(电子_电气_自动化_通信)1

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1、外文文献原文On the deployment of VoIP in Ethernet networks: methodology and case studyAbstractDeploying 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

2、 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 netwo

3、rk 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

4、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 capaci

5、ty, and measurement and impact of background traffic. Keywords: Network Design，Network Management，VoIP，Performance Evaluation，Analysis，Simulation，OPNET 1 IntroductionThese days a massive deployment of VoIP is taking place over data networks. Most of these networks are Ethernet based and running IP p

6、rotocol. 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 application

7、s. 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

8、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 net

9、work 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 s

10、ome 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 appro

11、ach 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 mea

12、suring 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 netw

13、ork 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 address

14、 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

15、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 ill

16、ustrate 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

17、 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.2 Existing networ

18、kFig. 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, guid

19、elines, 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 serv

20、ers. 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

21、. 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

22、Mbps full duplex except for the links for Floors 13 which are shared Ethernet 100 Mbps half duplex.3 Step-by-step methodologyFig. 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 embark

23、ing 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 characteri

24、stics, 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 confer

25、encing. 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 responsi

26、ble for converting VoIP calls to/from the Public Switched Telephone Network (PSTN). As an engineering and design issue, the placement of these nodes in the network becomes crucial. We will tackle this issue in design step 5. Other hardware requirements include a VoIP client terminal, which can be a

27、separate VoIP device, i.e. IP phones, or a typical PC or workstation that is VoIP-enabled. A VoIP-enabled workstation runs VoIP software such as IP Soft Phones .Fig. 3 identifies the end-to-end VoIP components from sender to receiver 9. The first component is the encoder which periodically samples t

28、he original voice signal and assigns a fixed number of bits to each sample, creating a constant bit rate stream. The traditional sample-based encoder G.711 uses Pulse Code Modulation (PCM) to generate 8-bit samples every 0.125 ms, leading to a data rate of 64 kbps . The packetizer follows the encode

29、r and encapsulates a certain number of speech samples into packets and adds the RTP, UDP, IP, and Ethernet headers. The voice packets travel through the data network. An important component at the receiving end, is the playback buffer whose purpose is to absorb variations or jitter in delay and prov

30、ide a smooth playout. Then packets are delivered to the depacketizer and eventually to the decoder which reconstructs the original voice signal. We will follow the widely adopted recommendations of H.323, G.711, and G.714 standards for VoIP QoS requirements. Table 1 compares some commonly used ITU-T

31、 standard codecs and the amount of one-way delay that they impose. To account for upper limits and to meet desirable quality requirement according to ITU recommendation P.800, we will adopt G.711u codec standards for the required delay and bandwidth. G.711u yields around 4.4 MOS rating. MOS, Mean Op

32、inion Score, is a commonly used VoIP performance metric given in a scale of 15, with 5 is the best. However, with little compromise to quality, it is possible to implement different ITU-T codecs that yield much less required bandwidth per call and relatively a bit higher, but acceptable, end-to-end

33、delay. This can be accomplished by applying compression, silence suppression, packet loss concealment, queue management techniques, and encapsulating more than one voice packet into a single Ethernet frame.3.1.1. End-to-end delay for a single voice packet Fig. 3 illustrates the sources of delay for

34、a typical voice packet. The end-to-end delay is sometimes referred to by M2E or Mouth-to-Ear delay. G.714 imposes a maximum total one-way packet delay of 150 ms end-to-end for VoIP applications . In 22, a delay of up to 200 ms was considered to be acceptable. We can break this delay down into at lea

35、st three different contributing components, which are as follows (i) encoding, compression, and packetization delay at the sender (ii) propagation, transmission and queuing delay in the network and (iii) buffering, decompression, depacketization, decoding, and playback delay at the receiver. 3.1.2.

36、Bandwidth for a single callThe required bandwidth for a single call, one direction, is 64 kbps. G.711 codec samples 20 ms of voice per packet. Therefore, 50 such packets need to be transmitted per second. Each packet contains 160 voice samples in order to give 8000 samples per second. Each packet is

37、 sent in one Ethernet frame. With every packet of size 160 bytes, headers of additional protocol layers are added. These headers include RTP+UDP+IP+Ethernet with preamble of sizes 12+8+20+26, respectively. Therefore, a total of 226 bytes, or 1808 bits, needs to be transmitted 50 times per second, or

38、 90.4 kbps, in one direction. For both directions, the required bandwidth for a single call is 100 pps or 180.8 kbps assuming a symmetric flow.3.1.3. Other assumptionsThroughout our analysis and work, we assume voice calls are symmetric and no voice conferencing is implemented. We also ignore the si

39、gnaling traffic generated by the gatekeeper. We base our analysis and design on the worst-case scenario for VoIP call traffic. The signaling traffic involving the gatekeeper is mostly generated prior to the establishment of the voice call and when the call is finished. This traffic is relatively sma

40、ll compared to the actual voice call traffic. In general, the gatekeeper generates no or very limited signaling traffic throughout the duration of the VoIP call for an already established on-going call. In this paper, we will implement no QoS mechanisms that can enhance the quality of packet deliver

41、y in IP networks. A myriad of QoS standards are available and can be enabled for network elements. QoS standards may include IEEE 802.1p/Q, the IETFs RSVP, and DiffServ. Analysis of implementation cost, complexity, management, and benefit must be weighed carefully before adopting such QoS standards.

42、 These standards can be recommended when the cost for upgrading some network elements is high and the network resources are scarce and heavily loaded.3.2. VoIP traffic flow and call distributionKnowing the current telephone call usage or volume of the enterprise is an important step for a successful

43、 VoIP deployment. Before embarking on further analysis or planning phases for a VoIP deployment, collecting statistics about of the present call volume and profiles is essential. Sources of such information are organizations PBX, telephone records and bills. Key characteristics of existing calls can

44、 include the number of calls, number of concurrent calls, time, duration, etc. It is important to determine the locations of the call endpoints, i.e. the sources and destinations, as well as their corresponding path or flow. This will aid in identifying the call distribution and the calls made inter

45、nally or externally. Call distribution must include percentage of calls within and outside of a floor, building, department, or organization. As a good capacity planning measure, it is recommended to base the VoIP call distribution on the busy hour traffic of phone calls for the busiest day of a wee

46、k or a month. This will ensure support of the calls at all times with high QoS for all VoIP calls. When such current statistics are combined with the projected extra calls, we can predict the worst-case VoIP traffic load to be introduced to the existing network. Fig. 4 describes the call distributio

47、n for the enterprise under study based on the worst busy hour and the projected future growth of VoIP calls. In the figure, the call distribution is described as a probability tree. It is also possible to describe it as a probability matrix. Some important observations can be made about the voice tr

48、affic flow for inter-floor and external calls. For all these type of calls, the voice traffic has to be always routed through the router. This is so because Switchs 1 and 2 are layer 2 switches with VLANs configuration. One can observe that the traffic flow for inter-floor calls between Floors 1 and

49、 2 imposes twice the load on Switch 1, as the traffic has to pass through the switch to the router and back to the switch again. Similarly, Switch 2 experiences twice the load for external calls from/to Floor 3.3.3. Define performance thresholds and growth capacity In this step, we define the networ

50、k performance thresholds or operational points for a number of important key network elements. These thresholds are to be considered when deploying the new service. The benefit is twofold. First, the requirements of the new service to be deployed are satisfied. Second, adding the new service leaves

51、the network healthy and susceptible to future growth. Two important performance criteria are to be taken into account. First is the maximum tolerable end-to-end delay; and second is the utilization bounds or thresholds of network resources. The maximum tolerable end-to-end delay is determined by the

52、 most sensitive application to run on the network. In our case, it is 150 ms end-to-end for VoIP. It is imperative to note that if the network has certain delay sensitive applications, the delay for these applications should be monitored, when introducing VoIP traffic, such that they do not exceed t

53、heir required maximum values. As for the utilization bounds for network resources, such bounds or thresholds are determined by factors such as current utilization, future plans, and foreseen growth of the network. Proper resource and capacity planning is crucial. Savvy network engineers must deploy

54、new services with scalability in mind, and ascertain that the network will yield acceptable performance under heavy and peak loads, with no packet loss. VoIP requires almost no packet loss. In literature, 0.15% packet loss was generally asserted. However, in 24 the required VoIP packet loss was cons

55、ervatively suggested to be less than 10. A more practical packet loss, based on experimentation, of below 1% was required in 22. Hence, it is extremely important not to utilize fully the network resources. As rule-of-thumb guideline for switched fast full-duplex Ethernet, the average utilization lim

56、it of links should be 190%, and for switched shared fast Ethernet, the average limit of links should be 85% 25. The projected growth in users, network services, business, etc. must be all taken into consideration to extrapolate the required growth capacity or the future growth factor. In our study,

57、we will ascertain that 25% of the available network capacity is reserved for future growth and expansion. For simplicity, we will apply this evenly to all network resources of the router, switches, and switched-Ethernet links. However, keep in mind this percentage in practice can be variable for eac

58、h network resource and may depend on the current utilization and the required growth capacity. In our methodology, the reservation of this utilization of network resources is done upfront, before deploying the new service, and only the left-over capacity is used for investigating the network support

59、 of the new service to be deployed.3.4. Perform network measurementsIn order to characterize the existing network traffic load, utilization, and flow, network measurements have to be performed. This is a crucial step as it can potentially affect results to be used in analytical study and simulation.

60、 There are a number of tools available commercially and noncommercially to perform network measurements. Popular open-source measurement tools include MRTG, STG, SNMPUtil, and GetIF 26. A few examples of popular commercially measurement tools include HP OpenView, Cisco Netflow, Lucent VitalSuite, Pa

61、trol DashBoard, Omegon NetAlly, Avaya ExamiNet, NetIQ Vivinet Assessor, etc. Network measurements must be performed for network elements such as routers, switches, and links. Numerous types of measurements and statistics can be obtained using measurement tools. As a minimum, traffic rates in bits pe

62、r second (bps) and packets per second (pps) must be measured for links directly connected to routers and switches. To get adequate assessment, network measurements have to be taken over a long period of time, at least 24-h period. Sometimes it is desirable to take measurements over several days or a

63、 week. One has to consider the worst-case scenario for network load or utilization in order to ensure good QoS at all times including peak hours. The peak hour is different from one network to another and it depends totally on the nature of business and the services provided by the network. Table 2

64、shows a summary of peak-hour utilization for traffic of links in both directions connected to the router and the two switches of the network topology of Fig. 1. These measured results will be used in our analysis and simulation study.外文文献译文以太网网络电话传送调度：方法论和案例分析摘 要对网络数据研究者和设计师来说，IP电话或语音IP电话调度是一项重大而艰巨的

65、任务。本文概述的准则和循序渐进的方法，解释了怎样在IP上成功调度传送语音。该方法可用于评估的支持，并准备用在现有的网络。此前购买并部署的网络电话设备，这种方法预算出了在保证现有网络服务质量要求和日后足够扩充能力基础上的网络电话调用次数。作为一个研究的课题，我们把这种方法在一个典型的小型企业网上得到逐步应用。我们运用分析和模拟吞吐量和延迟区域。我们的分析基于排队理论，并且OPNET用于模拟。理论分析和模拟结构比较一致。此外，本文谈论了许多设计和工程问题。这些问题包括网络电话通信的特征和服务质量要求，网络电话流程和呼叫分配，定义未来增长容量，测定后台通信的影响。关键词：网络设计，网络管理，VoIP

66、，性能评估，分析，模拟，OPNET8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