VoIP Handbook

Applications, Technologies, Reliability, and Security

Syed A. Ahson, Mohammad Ilyas

• Publisher: CRC Press
• Year: 2018
• ISBN:
  9781351834681
• Edition: 1
• Topic: Technology & Engineering
• Subtopic: Electrical Engineering & Telecommunications
• Index: Technology & Engineering

The number of worldwide VoIP customers is well over 38 million. Thanks to the popularity of inexpensive, high-quality services, it's projected to increase to nearly 250 million within the next three years.

The VoIP Handbook: Applications, Technologies, Reliability, and Security captures the state of the art in VoIP technology and serves as the comprehensive reference on this soon-to-be ubiquitous technology. It provides:

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• A step-by-step methodology to evaluate VoIP performance prior to network implementation
• An invaluable overview of implementation challenges and several VoIP multipoint conference systems
• Unparalleled coverage of design and engineering issues such VoIP traffic, QoS requirements, and VoIP flow

As this promising technology's popularity increases, new demands for improved quality, reduced cost, and seamless operation will continue to increase. Edited by preeminent wireless communications experts Ahson and Illyas, the VoIP Handbook guides you to successful deployment.

Part I

Introduction

1

Deploying VoIP in Existing IP Networks

Khaled Salah

CONTENTS

1.1 Introduction

1.2 Step-by-Step Methodology

1.2.1 VoIP Traffic Characteristics, Requirements, and Assumptions

1.2.1.1 End-to-End Delay for a Single Voice Packet

1.2.1.2 Bandwidth for a Single Call

1.2.1.3 Other Assumptions

1.2.2 VoIP Traffic Flow and Call Distribution

1.2.3 Define Performance Thresholds and Growth Capacity

1.2.4 Network Measurements

1.2.5 Upfront Network Assessment and Modifications

1.2.6 Analysis

1.2.6.1 Bandwidth Bottleneck Analysis

1.2.6.2 Delay Analysis

1.2.7 Simulation

1.2.8 Pilot Deployment

1.3 Case Study

1.4 Summary and Conclusion

References

1.1 Introduction

Many network managers find it attractive and cost effective to merge and unify voice and data networks. A unified network is easier to run, manage, and maintain. However, the majority of today’s existing data networks is Ethernet-based and use Internet Protocols (IP). Such networks are best-effort networks in that they were not designed to support real-time applications such as Voice over Internet Protocol (VoIP). VoIP requires timely packet delivery with low latency, jitter, packet loss, and sufficient bandwidth. To achieve this, efficient deployment of VoIP must ensure that 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 data networks, many network architects, managers, planners, designers, and engineers are faced with common strategic, and sometimes challenging, questions. What are the quality of service (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 standardized QoS requirements? If so, how many VoIP calls can the network support before it becomes necessary to upgrade any part of the existing network hardware?

Commercial tools can answer some of these challenging questions, and a list of the commercial tools available for VoIP can be found in [1,2]. For the most part, these tools use two common approaches to assess the deployment of VoIP into the existing network. One approach is based on first performing network measurements and then predicting the readiness of the network to support VoIP by assessing the health of network elements. The second approach injects real VoIP traffic into the existing network and measures the resulting delay, jitter, and loss.

There is a definite financial cost associated with the use of commercial tools. Moreover, no commercial tool offers a comprehensive approach to successful VoIP deployment. Specifically, none is able to predict the total number of calls that can be supported by the network, taking into account important design and engineering factors, including VoIP flow and call distribution, future growth capacity, performance thresholds, the impact of VoIP on existing network services and applications, and the impact of background traffic on VoIP. This chapter attempts to address these important factors and lays out a comprehensive methodology to successfully deploy any multimedia application such as VoIP and videoconferencing. Although the chapter focuses essentially on VoIP, it also contains many useful engineering and design guidelines, and discusses many practical issues pertaining to the deployment of VoIP. These issues include the characteristics of VoIP traffic and QoS requirements, VoIP flow and call distribution, defining future growth capacity, and the 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 chapter is organized as follows. Section 1.2 outlines an eight-step methodology to successfully deploy VoIP in data networks. Each step is described in considerable detail. Section 1.3 presents a case study of a VoIP introduced to a typical data network of a small enterprise, using the methods described in the previous section. Section 1.4 summarizes and concludes the study.

1.2 Step-by-Step Methodology

In this section, an eight-step methodology is described for the successful deployment of a VoIP (Figure 1.1). The first four steps are independent and can be performed in parallel. Steps 6 and 7, an analysis and simulation study, respectively, can also be done in parallel. Step 5, however, involves the early and necessary re-dimensioning or modification to the existing network. The final step is pilot deployment.

FIGURE 1.1 Flowchart of an eight-step methodology. (Source: K. Salah, “On the deployment of VoIP in ethernet networks: Methodology and case study,” International Journal of Computer Communications, Elsevier Science, vol. 29, no. 8, 2006, pp. 1039–1054. With permission.)

This methodology can be used to deploy a variety of network services other than VoIP, including videoconferencing, peer to peer (p2p), online gaming, internet protocol television (IPTV), enterprise resource planning (ERP), or SAP services. The work in [3,4] show how these steps can be applied to assess the readiness of IP networks for desktop videoconferencing.

1.2.1 VoIP Traffic Characteristics, Requirements, and Assumptions

In order to introduce a new network service such as VoIP, one must first characterize the nature of the traffic, QoS requirements, and the need for additional components or devices. For simplicity, we assume a point-to-point conversation for all VoIP calls with no call conferencing. First, a gatekeeper or CallManager node, which can handle signaling to establish, terminate, and authorize all VoIP call connections, has to be added to the network [5–7]. Also, a VoIP gateway responsible for converting VoIP calls to/from the Public Switched Telephone Network (PSTN) is required to handle external calls. From an engineering and design standpoint, the placement of these nodes in the network is critical (see Step 5). Other hardware requirements include a VoIP client terminal, which can be a separate VoIP device (i.e., IP phone) or a typical PC or workstation that is VoIP-enabled and which runs VoIP software such as IP SoftPhones [8–10].

Figure 1.2 identifies the end-to-end VoIP components from sender to receiver [11]. The first component is the encoder, which periodically samples the 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 [12]. Following the encoder is the packetizer, which 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 to the receiver where an important component called the playback buffer is placed for the purpose of absorbing variations or jitt...

Table of contents

• Cover
• Half Title
• Title Page
• Copyright Page
• Table of Contents
• Preface
• Editors
• Contributors
• Part I Introduction
• Part II Technologies
• Part III Applications
• Part IV Reliability and Security
• Index