Trunking of TDM and narrowband services over IP Networks

The recent interest in IP as the vehicle for transporting TDM and narrowband services stems from the possibility of using a common transport network for voice, video, and data, and the flexibility with which new services can be introduced. A key step in the evolution of networks towards a ‘broadband’ IP‐based environment is the ‘graceful’ interworking of the IP networks with the existing networks and services, particularly with the circuit switched telephone network. A ‘graceful’ interworking approach is one whose complexity is minimal and preserves the user's perceived quality of service (QoS). To interwork with a circuit switched network whose services are pre‐dominantly time‐sensitive, the IP network must essentially behave as a transparent ‘link’ in the end‐to‐end connection. This paper presents an overview of the main technical problems to be addressed when trunking TDM and narrowband services over IP networks. Copyright © 2002 John Wiley & Sons, Ltd.     

分组传送网标准进展

全球运营商正致力于构建自己的下一代IP包交换网络,网络逐步迈向全IP化。不断增长的数据业务对网络的带宽、服务质量（QoS）、操作维护管理、可靠性等提出了新的要求。IP网络业务融合（如视频、语音、数据融合）需求以及多业务统一的承载需求,进一步驱动传送网技术与数据网络通信技术融合。分组传送网技术应变化而生,分组传送网标准发展迅速。     

Perceptual QoS assessment technologies for VoIP

Since quality is not generally guaranteed in an IP network, the proper design and management of networks and/or terminals for high-quality voice over IP services and maintenance of service levels is important. In terms of quality design and management, methodologies for appropriately and effectively evaluating the perceptual QoS of VoIP are indispensable. This article gives an overview of the state of the art of quality assessment technologies for VoIP, including recent work on improving their accuracy.     

Real-time voice over packet-switched networks

We discuss the architecture and technical viability of transporting real-time voice over packet-switched networks such as the Internet. The value of integrating voice and data networks onto a common platform is well known. The telephony industry has proposed the ATM standard as a means of upgrading the Internet to provide both real-time and data services. In contrast, voice services may be added to traditional IP networks that were originally designed for data transmission alone. We consider the feasibility and expected quality of service of audio applications over IP networks such as the Internet. In particular, we examine possible architectures for voice over IP and discuss measured Internet delay and loss characteristics     

Differentiated QoS for survivable WDM optical networks

Optical networks based on WDM technology have become a promising solution to realize transport networks that can meet the ever-increasing demand for bandwidth. As WDM networks carry a huge volume of traffic, maintaining a high level of survivability is an important and critical issue. The. development of GMPLS switching technology led to the direct integration of IP and WDM. In these IP-over-WDM networks different applications/end users need different levels of fault tolerance and differ in how much they are willing to pay for the service they get. The current trend in network development is moving toward a unified solution providing support for voice, data, and various multimedia services. Therefore, it imperative that WDM networks incorporate fault tolerance to single or multiple component failures, protection bandwidth, recovery time, and recovery granularity besides resource utilization and call acceptance ratio. This article presents a survey of various methods that have been proposed for providing service differentiation in survivable WDM networks and discuss their performance. Such methods are broadly classified under various paradigms such as differentiated reliability, R-connections, quality of protection, and quality of recovery.     

Joint Routing and Playout Buffering of IP Telephony Flows in MANETs

The low-cost of packet-based networking technologies with respect to traditional circuit-switched ones and the reliability of the current (wired) IP networks have brought to a considerable employment of the VoIP (Voice over IP) technologies in the voice services market. This success is expected to happen also in mobile ad hoc networks (MANETs), which may offer a good platform for the fast deployment of VoIP mobile networks. However, efforts must be made to improve performance before MANETs can be used for this purpose. One of the main limitations is related to the highly variability of the network topology and channel behavior, which heavily influences the service quality due to route losses and significant delay variations. In this paper, we propose a strategy where these impairments are jointly addressed. The source is responsible for jointly selecting the transmission paths and adjusting the playout delay, with an adaptive inter-talkspurt approach. These tasks are accomplished on the basis of historical data on network connectivity and transmission delays, and are driven by a quality-based approach. The collection of statistics of the network status relies on the QOLSR routing algorithm, whereas the voice quality is measured by means of the ITU-T E-Model.     

城域传送网骨干层未来发展趋势

基于对现有移动传送网现状的分析,对于专线、话音等小颗粒业务,目前的SDH/MSTP网络已经提供了完善的解决方案。但面对爆炸式增长的大颗粒带宽业务,现有的网络和技术则无法很好的解决。城域网必须具备大容量、高带宽、多业务、高可靠性等能力,从而实现骨干层层面的宽带IP化业务支持能力。     

Speech transmission performance planning in hybrid IP/SCN networks

The introduction of Internet protocol technology into traditional telecommunications networks is changing the nature of speech communication in those networks. As the current switched network infrastructure is augmented by packet networks, the needs of voice users of these hybrid networks must be given due consideration. These users are accustomed to high-quality connections with low distortion in the speech signal and low transmission delay in the speech path. In order for packetized networks to gain widespread acceptance for speech transmission services, it is necessary to maintain this high-quality performance in the evolving networks. We discuss relevant speech transmission performance requirements and the associated activities taking place in regional and international standards bodies     

Mobile and Multicast IP Services in PACS: System Architecture, Prototype, and Performance

Traditionally, wireless cellular communication systems have been engineered for voice. With the explosive growth of Internet applications and users, there is an increasing demand on providing Internet services to mobile users based on the voice-oriented cellular networks. However, Internet services add a set of radically different requirements on to the cellular wireless networks, because the nature of communication is very different from voice. It is a challenge to develop an adequate network architecture and necessary systems components to meet those requirements.This paper describes our experience on developing Internet services, in particular, mobile and multicast IP services, in PACS (Personal Access Communication Systems). Our major contributions are five-fold: (i) PACS system architecture that provides wireless Internet and Intranet access by augmenting the voice network with IP routers and backbone links to connect to the Internet; (ii) simplified design of RPCU (Radio Port Controller Unit) for easy service maintenance and migration to future IP standards such as IPv6; (iii) native PACS multicast to efficiently support dynamic IP multicast and MBone connectivity; (iv) optimization and incorporation of Mobile IP into PACS handoff mechanism to efficiently support roaming within a PACS network as well as global mobility between PACS networks and the Internet; (v) successful prototype design of the new architecture and services verified by extensive performance measurements of IP applications. Our design experience and measurement results demonstrate that it is highly feasible to seamlessly integrate the PACS networks into the Internet with global IP mobility and IP multicast services.     

Standardization Progress of Packet Transport Networks

Global operators are in the process of constructing next generation IP packet-switched networks, with the view of moving towards full IP-based networks. As data services increase, improvements in bandwidth, Quality of Service (QoS), Operation Administration and Maintenance (OAM), and network reliability are required. Convergence of network data transmission and network communications technology has been driven by converged IP services (including video, voice, and data), and unified multi-service load demand. Packet Transport Networking (PTN) has come into being to meet these requirements, and PTN Standards are being developing rapidly.     

Operation, administration, and maintenance in MPLS networks

The boundaries between packet and circuit networks have long disappeared, with many traditional circuit-switched applications such as voice and video now being carried over packet-switched IP/MPLS or Ethernet networks. However, this transition has happened so fast that many OAM functions supported by circuit-switched networks such as SONET/SDH are still unmatched in packet networks. In order to match the quality sustained by circuit-switched networks, OAM functions must also be developed for such packet networks. A number of recent efforts have started to address OAM functions for IP/MPLS and Ethernet packet technologies. Service providers and carriers alike are the driving force behind the work, as there is general recognition that to generate sustained revenues, services must be efficiently managed. In this article we discuss issues in providing OAM features and capabilities for MPLS-based packet networks.     

Voice communications over zigbee networks

This article provides an overview of ZigBee-enabled wireless networks and discusses the feasibility of supporting voice communications over ZigBee networks. We begin by providing an overview of the ZigBee technology followed by an evaluation of voice quality and performance over such an impoverished wireless channel. Two types of voice communications, namely full-duplex voice over IP (VoIP) and half-duplex push-to-talk (PTT) are considered. Voice quality of VoIP is measured using the R-factor [1] (a well known objective speech quality metric). The quality of PTT, however, is evaluated based on packet-loss rate, delay, and jitter. The simulation results demonstrate that a low-power, low-rate wireless sensor network can support a limited range of voice services.     

Quality of experience (QoE) driven adaptation scheme for?voice/video over IP

Network quality of service (NQoS) of IP networks is unpredictable and impacts the quality of networked multimedia services. Adaptive voice and video schemes are therefore vital for the provision of voice over IP (VoIP) services for optimised quality of experience (QoE). Traditional adaptation schemes based on NQoS do not take perceived quality into consideration even though the user is the best judge of quality. Additionally, uncertainties inherent in NQoS parameter measurements make the design of adaptation schemes difficult and their performance suboptimal. This paper presents a QoE-driven adaptation scheme for voice and video over IP to solve the optimisation problem to provide optimal QoE for networked voice and video applications. The adaptive VoIP architecture was implemented and tested both in NS2 and in an Open IMS Core network to allow extensive simulation and test-bed evaluation. Results show that the scheme was optimally responsive to available network bandwidth and congestion for both voice and video and optimised delivered QoE for different network conditions, and is friendly to TCP traffic.     

Integrated management architecture for IP-based networks

IP telephony will bring about a dramatic change in the way IP services are planned, provisioned, managed, and billed. In order to build and retain a strong customer base for these new services, service providers need to meet, if not exceed, the customer expectations set by today's traditional voice services. Acceptance of IP telephony will depend on the quality and efficiency with which service providers offer, deliver, and manage IP services. Installation, configuration, and activation must be rapid and error-free. Furthermore, customers will want direct control over the reconfiguration of services and real-time visibility into the impact change has on their operating costs. Once the service is activated, customers will want the provider to guarantee service quality as defined by industry standards. Corporate customers in particular will need to be assured that the provider is proactively monitoring performance to avoid problems and providing them visibility into the performance data collected. This article discusses an integrated management support system for IP-based networks illustrating the functions needed to support the unique challenges of managing VoIP services. An example of a service management system is also described     

VoIP over DVB-RCS with QoS and bandwidth on demand

Motivated by the need for compliance/interoperability above the satellite-specific layers, this article proposes a consolidated approach for voice over IP over satellite networks based on the ETSI DVB-RCS standard. Voice communication is a real-time service that needs priority over other services in IP environments with limited bandwidth, such as IP satellite networks. Bandwidth utilization in such networks needs to be optimized in order to reduce service costs, and this requires the use of dynamic bandwidth allocation schemes. This article therefore addresses the role of bandwidth on demand in the optimization of bandwidth allocation for VoIP and assesses the impact of BoD mechanisms on voice quality. The trade-off between voice quality and bandwidth efficiency is investigated under different DVB-RCS-specific capacity request/allocation strategies, and it is demonstrated that DVB-RCS provides an efficient platform for integrated support for a variety of VoIP applications over satellite. The main contribution of this article consists of the identification of the mechanisms capable of responding to the key challenges raised by the VoIP application in the satellite environment.     

下一代互联网研究方向与趋势

"业务综合"和"网络融合"是通信界多年来努力的目标。IP网络具有承载包括语音、数据和视频等多种业务应用能力,这使业界将全IP网络看作是未来融合网络的雏形。但目前的IP网络无法保证实时业务的服务质量以及网络和业务的安全性。为了在下一代互联网发展方面处于领先地位,全球各国都在不断投入力量进行下一代互联网的研究工作。本文主要介绍IP网络和业务的发展情况、目前IP网络存在的问题以及全球在发展下一代互联网方面所做的努力。     

IP resilience within an autonomous system: current approaches, challenges, and future directions

Network survivability is gaining increasing attention from the Internet research community. The ubiquity of IP services has fueled increasing interest in ensuring their dependability, by making IP networks more disruption-tolerant. After providing a brief overview of how routing is accomplished in the Internet, this article reviews how the protocols react to failures or changes in network state within an autonomous system. The problems associated with current restoration schemes, with respect to newer and more stringent requirements posed by emerging services such as voice over IP, are identified. We present an overview of the schemes proposed to ameliorate fault recovery and critique their contributions. We also outline future research directions for improving IP resilience.     

Service-oriented broadband wireless network architecture - [Guest editorial]

In the past few years, broadband wireless networks have been very attractive as a key technology for providing Internet access because of quick and cost-effective deployments. In addition to the features of high data rate and large coverage, broadband wireless networks also promise to rapidly provide access to locations in the world?s rural and developing areas where broadband is currently unavailable, as well as compete for urban market shares. Till now, most existing efforts have been focused on basic access capability. However, to successfully operate broadband wireless networks, a crucial issue must be addressed: how to support a variety of services and applications within the broadband wireless networks? As an example, there is an increasing need to efficiently support applications such as voice over IP (VoIP), video streaming, music downloading, IP TV, and many others.     

Implementing VoIP: a voice transmission performance progress report

Aiming to introduce voice over IP networks and services in ways that satisfy the voice quality expectations of our customers, we have been conducting laboratory studies of how VoIP transmission affects voice quality while also carefully monitoring and managing several field implementations of VoIP. This article summarizes much of what we have learned in this work, and we hope it provides a useful progress report on the industry's evolution to VoIP. We review our data on the voice quality effects of packet loss, delay, speech coders, packet loss concealment algorithms, and the compression option of suppressing transmission during silence. Because the familiar problem of echo has emerged repeatedly in the VoIP environment, we review this issue in some detail. Packet loss and delay variation measurements made on private VoIP networks are reviewed, and the data here are encouraging. We finish by making our case that the network planning tool known as the E-model is currently an inexact predictor of VoIP network performance.     

VoWLAN通信平台的设计与实现

VoWLAN也叫VoWiFi或者WiFi VoIP。它是基于无线网络技术和VoIP网络,是两者的有机结合。即是通过WLAN提供VoIP业务,使得终端用户通过WLAN拨打IP电话成为现实。本文提出了在基于Linux操作系统的SIP应用服务器及VoIP网关中,如何通过SIP信令和传统的PSTN数据通信线路与无线网络无缝连接方案,从而实现IP网络与传统电话间的实时语音通信、电话会议、语音信箱、视频通信、短消息、数据传输等业务。本设计已成功应用于某企业的实时语音通信平台,获得良好的效果。