An analytical simulator for deploying IP telephony

Deploying IP telephony or voice over IP (VoIP) is a major and challenging task. This paper describes an analytical design and planning simulator to assess the readiness of existing IP networks for the deployment of VoIP. The analytical simulator utilizes techniques used for network flows and queuing network analysis to compute two key performance bounds for VoIP: delay and bandwidth. The simulator is GUI‐based and has an interface with drag‐and‐drop features to easily construct any generic network topology. The simulator has an engine that automates and implements the analytical techniques. The engine determines the number of VoIP calls that can be sustained by the constructed network while satisfying VoIP QoS requirements and leaving adequate capacity for future growth. As a case study, the paper illustrates how the simulator can be utilized to assess the readiness to deploy VoIP for a typical network of a small enterprise. We have made the analytical simulator publicly available in order to improve and ease the process of VoIP deployment. Copyright © 2008 John Wiley & Sons, Ltd.     

IP Telephony问题研究综述

文中从对集成服务网络的主要应用之一分组网络语音传输应用的体系结构分析入手,从集成服务网络研究的观点出发,综述了以ＩＰ Ｔｅｌｅｐｈｏｎｙ为代表的分组网络语音传输技术体系中上前尚有待研究解决的问题。通过文中的分析,我们讨论了电信网与因特网通过诸多技术方面的融合,走向合一网络的发展趋势。     

局域网IP电话系统PSTN网关实现

本提出在企业内部构建基于局域网的IP电话系统的迫切性和技术可行性，并结合目前最新的CTI技术，给出局域网IP电话系统重要组成模块——PSTN网关的一种具体实现方案。     

STEM:安全IP电话的支撑部件

目前,包括IP电话在内的动态应用还没有被企业广泛接受,其主要原因在于现有的网络基础设施,像防火墙和网络地址转换设备等静态网元不能有效地支持动态应用。安全IP电话的支撑部件(Secure Telephony EnabledMiddlebox,STEM)体系结构能够增强现有的网络设计并消除由于静态设备而带来的问题。目前,在数据网络上提供的业务已经从短的文本消息发展到了实时的语音和视频。这种转变要求在数据通路上采用更复杂和更智能化的网元。现有企业网通过大量的包括路由器、防火墙和网络地址转换设备(中间件)在内的静态设备连接在了一起。这些设备通过一…     

Supplementary services in the H.323 IP telephony network

Traditionally, different networks were developed to handle voice, data, and video. The circuit-switched telephone network carried voice and the packet network carried data. Due to different deployment of these networks, different services were developed, such as voice mail in the telephone network and electronic mail on the Internet. With the revolution of multimedia in the computer industry, voice, video, and data are now being carried on both networks. Supplementary services, such as transfer and forwarding (which were originally developed for private telephone networks and later migrated to public telephone networks) are now being developed for packet networks. The standards for packet networks are being defined in the H.323-based series of ITU-T recommendations. This article provides the H.323 architecture for supplementary services, the differences in deployment of these services between the circuit-switched and packet-switched networks, and interworking of these services across hybrid networks     

Vulnerability analysis and best practices for adopting IP telephony in critical infrastructure sectors

IP telephony has been rapidly introduced to replace the traditional circuit switched infrastructure for telephony services. This change has had an enormous impact on critical-infrastructure (CI) sectors, which are expected to become increasingly dependent on IP telephony services. Reliable and secure telephony service is a key concern confronting most organizations in the critical-infrastructure sector today. With the proliferation of voice over IP (VoIP) services in these organizations, it is important for them to understand the security vulnerabilities and come up with a set of best practices during the evolution of the IP telephony services. This article outlines the potential security issues faced by CI sectors as they transform their traditional phone systems into VoIP systems. Vulnerability analyses are conducted to understand the impact of VoIP security challenges in the new convergent network paradigm. The most common security measures are analyzed to identify their strengths and limitations in combating these new security challenges. A set of recommendations and best practices are offered to address the key issues of VoIP security as IP telephony is being introduced into critical infrastructure.     

构建基于全IP架构的RAN接入网络

对基于全IP架构的RAN接入网络构建方案作了相关探讨,主要分析了MSTP、RPR、TMPLS、PBT等各种技术在全IPRAN接入网络中的选择与应用。     

基于H.323协议的IP电话网络合法侦听体系

针对近年来IP网络电话业务的大量涌现，简要介绍了IP网络电话合法侦听的概念，分析了与传统PSTN相比，对IP电话实施侦听的难点所在；接着设计给出了适用于H．323网络的、符合ETSI规范的合法侦听体系，并对比其他侦听体系分析了其优点所在。     

建立可扩展验证核的方法研究

提出一种可扩展验证核的结构,根据该结构建立了一个面向光通信应用领域SDH系列芯片验证的可扩展验证核VIP,验证人员通过文本编辑,可以产生验证所需要的XML配置文件,VIP根据配置XML文件,产生仿真激励并在线检查仿真结果.     

Integrated circuits for telephony

Many integrated circuits for large-scale application to telephone networks have been designed, including subscriber line interfaces, antialiasing filters, analog-to-digital-to-analog converters, and tone receivers. This paper summarizes in a unified fashion both interfacing and functional requirements for these devices, as well as the related circuit and technology approaches which have been utilized.     

Adaptive delta modulator for telephony

An adaptive version of the basic delta modulator employing full-width pulses and RC integration has been designed. A digital technique is used to sense the level of the input signal and to control the amplitude of the pulses supplied to the RC integrator in the feedback circuit. For an 800 Hz sinewave input a signal/quantisation-noise ratio of 32 dB has been obtained over a dynamic input range of 30 dB.     

Speech enhancement for mobile telephony

The properties of noise fields in automobile interiors are discussed with a view toward speech enhancement for voice-activated mobile telephony. The limitations on performance of adaptive noise cancellation are explained in the context of the spatial correlation properties of the noise field. A simple delay-equalized near-field array of directional microphones is analyzed and found to be effective for increasing the signal-to-noise density ratio (SNR) and reducing the reverberant distortion of the speech, without introducing any further distortion. An array of N microphones, each with a delay and weight chosen according to its distance from the speech source, is a viable solution. Such an array gives gains on the order of N over the speech band, reduces reverberation, and does not introduce waveform distortion. Experimental results confirming the predicted performance are presented     

Internet protocol support for telephony

In this paper, we consider the evolution of telephone networks from time-division multiplexing circuit switching to packet switching and, in particular, to packet switching-based on Internet Protocol (IP-supported telephony). We analyze IP-supported telephony design solutions by proposing a layered reference model in which each layer is associated to a subset of the functions that support telephony. We use the reference model to establish a terminology and a framework for the comparison of the design solutions. We group the design solutions in scenarios and compare them in terms of the reference model proposed. We then focus on IP telephony, in which IP is used in telephone company networks, and on Internet telephony, in which the Internet is used to support telephony. We show that they both can be seen as implementations of the same architecture, which consists of a set of components, associated to functions, and of the interactions among these components. We then consider the issue of voice-data integration and analyze the variety of design solutions that can be adopted to integrate voice and data.     

Screen-based telephony

Experimentation with screen-based telephony is discussed in the framework of Apple Computer's user-centered design philosophy. The approach described enables programs to be developed for the most ubiquitous technologies today-plain old telephone service (POTS) and proprietary private branch exchanges (PBXs)-but also allows for an easy, logical upgrade to future technologies like ISDN. The model has been built in three layers, which reflect the three constituencies who must cooperate to bring ISDN applications (and their forerunners) to market, in the volume required to spark the growth of this area: telecommunications vendors, personal computer vendors, and independent software developers. The Call Manager Application Programmers Interface is the code/specification from Apple that enables voice and data applications to be written independently of the vendor or technology. Telecommunications vendors can write drivers to the Application Program Interface (API) and build the appropriate hardware interfaces, without having to support thousands of software developers. Some of the features of Apple's prototype human interface to the voice network on this API are described     

宽带IP城域网建设中的典型问题

当前运营商的竞争重点从骨干向用户侧转移，城域网得到了蓬勃发展，本通过对城域网承载技术选择、LAN接入、MPLS在城域中的应用，建设可盈利的宽带网络等进行讨论，探讨了宽带IP城域网建设中的典型问题。     

为快速开发通信系统构建模块

概述Internet的爆炸性增长要求通信企业迅速地开发出新产品，它们具有高度灵活的结构体现，支持多种协议。随着在线用户数量快速地增长，访问图形丰富的的网址，传送大文件和在线收看视频，大量的Internet数据业务持续攀升。从用户中心到局“最后一公里”的带宽的需要增加，DSL和Cable Modem的逐渐普及满足了客户的需求。另外，随着高速光纤的铺设，骨干网的容量继续增加，每个通道的数据速率从OC-12到OC-48和OC-192，采用密集波分复用(DWDM)技术，每条光纤承载的数据通道数量大大地增加了。虽然用户对带宽的需求加快了网络骨干网容…     

RF CMOS transceivers for cellular telephony

This article describes system and circuit issues related to cellular transceivers, presenting design techniques that have provided high performance in CMOS technology. Following an overview of relevant GSM and WCDMA specifications, the article identifies four trends in RF design that have continued to improve the performance. Examples of CMOS transceivers, and circuit and device concepts are then described that meet the stringent requirements of cellular telephony.