统一消息业务在IP电话系统中的实现

文章分析了将UM业务融入IP电话系统中实现的优越性 ,确定了通过IP电话系统网守实现对UM服务器的管理机制 ,研究了该UM服务器的主要技术、系统结构、媒体信息的收发方式和安全策略等关键问题 ,完成了基于IP电话系统的UM业务的实现方案并成功开发了这一方案的UM原型系统。     

基于开放业务的网络融合

文章以IP电话为应用背景论述了网络融合的两种典型模型,讨论了采用Parlay/OSA 4.0开放业务规范实现网络融合的可能性,提出了一种能提供跨越因特网与传统电话网之间IP电话业务的网络融合方案:一种基于开放业务的网络融合方案,并讲座了该方案在实现网络融合相关的电信业务方面的优点和不足.     

国际经济电话业务与IP电话业务

今年中国电信开通了国际经济电话（宜通卡）业务和中国电信IP电话实验业务。它们都是利用IP同来传送语音,实现电话到电话的业务。表1列出了两者的异同。国际经济电话和中国电信IP电话业务采用相同的语音压缩和IP打包技术,在技术本质上没有什么差别。但国际经济电话通过IP专线开通国际方向的业务,而中国电信IP电话采用国际互联网为传输网络,这是它们最主要的差别。正是由于这一差别,通常情况下,国际经济电话比中国电信IP电话的通话质量要好,特别在忙时,这种优势更加明显。而利用因特网作传输网络的中国电信IP电话组网灵活,投…     

IP电话与IP传真业务融合的关键问题

1概述目前在绝大多数情况下,IP电话是不能用于发传真的。也就是说,IP电话与IP传真通常是分离的,用户不能够像使用传统的固定电话那样,在一条电话线上随意地使用语音业务和传真业务。这主要是因为IP电话采用了压缩技术,如G郾729A或G郾723等编码方式,它们主要是针对语音的,而传真与语音具有完全不同的数据特性,这些语音压缩算法并不适用于传真,并且IP电话和IP传真的数据封装格式也不一样。因此,要实现这两种业务的融合,满足用户的需要,必须对IP电话的相关设备尤其是网关采取一些技术措施。本文针对IP电话与IP传真业务融合时面临的主…     

中国IP电话业务的发展

中国于1999年4月开始了IP电话业务的商用试验，尽管现阶段中国IP电话业务还处于市场成长期，但已显露出了相当大的市场潜力。在业务总量大幅度增加的同时，国内IP电话业务量所占比重在不断上升；IP电话作为长途电话业务的一种实现方式，对PSTN长途电话的替代作用越来越明显，在2001年，IP电话国际业务通话时长的绝对值已超过了PSTN国际长途电话业务。     

下一代通信网络中的SIP语音业务

介绍了最新的基于因特网的语音技术。在该方案设计中采用了融合传统电路交换网和IP网的结构思想,不仅将语音的传输放到了IP网上,更为集成将来的业务,诸如网页、电子邮件、多媒体技术等提供了灵活的结构。其初期目的在于替代传统的企业级电话业务。     

基于H.323标准的IP电话附加业务

随着IP电话的快速发展，用户对IP电话新业务的需求也日益增长，传统电话业务正在被移植到IP多媒体电话系统中。由于IP电话系统在信令结构方面与传统电话系统的集中模型有本质的区别，因此基于IP电话系统附加业务的开发和控制协议具有不同的特点。描述了基于H.323标准的附加业务的信令结构和业务设计，并将其与传统电话交换业务模型进行了比较。     

中国电信推出IP电话业务

经信息产业部批准,中国电信将于1999年4月底至12月底进行IP电话业务试验,北京成为首批试验城市之一。中国电信此次IP电话业务试验以电话记帐卡方式实现从普通电话机到普通电话机的通信。中国电信IP电话业务试验的接入码为17900,主要为试验用户提供国际、国内长途IP电话及IP传真业务。     

IP智能语音终端业务介绍

IP集合通信打破传统单一模式的界限，实现各种通信方式的融合，将电话、传真、电子邮件、软件客户端等多种通信手段整合在统一的IP通信平台上。本文对IP智能语音终端提供的三类特色业务进行了详细的介绍。通过基本语音类业务、统一消息业务、协同工作业务，能实现音频、视频及数据等多种通信模式的融合；实时通信业务与非实时消息业务的融合；通信技术功能与办公及业务流程的融合．     

IP智能语音终端业务介绍

IP集合通信打破传统单一模式的界限，实现各种通信方式的融合，将电话、传真、电子邮件、软件客户端等多种通信手段整合在统一的IP通信平台上。本文对IP智能语音终端提供的三类特色业务进行了详细的介绍。通过基本语音类业务、统一消息业务、协同工作业务，能实现音频、视频及数据等多种通信模式的融合；实时通信业务与非实时消息业务的融合：通信技术功能与办公及业务流程的融合。     

A Study of the Next Generation Intelligent Networks Interworking with IP Networks

A Study of the Next Generation Intelligent Networks Interworking with IP Networks     

True number portability and advanced call screening in a SIP-basedIP telephony system

Custom local area signaling service features offered in the PSTN have certain limitations due to the closed nature of PSTN network signaling. The adoption of telephony over IP (IP telephony) will enable a new paradigm of services and features that are not possible to implement in today's PSTN. This is especially the case for services that make use of personal, trusted information, which can be provided by a user's personal digital assistant. We demonstrate how personal information can be coupled with an IP telephony service to provide user-customized call handling by the network. In particular, we describe a demonstration architecture that includes Ethernet-attached phones running SIP, with an interface to synchronize with PDAs that supply personal information. The proposed architecture is quite flexible; it can support enhanced versions of the current PSTN and private branch exchange services, in addition to many new features and services. We describe true number portability and advanced call screening as examples of new services in a hybrid PSTN/IP telephony environment     

SIP security issues: the SIP authentication procedure and its processing load

Session Initiation Protocol (SIP) is currently receiving much attention and seems to be the most promising candidate as a signaling protocol for the current and future IP telephony services, also becoming a real competitor to the plain old telephone service. For the realization of such a scenario, there is an obvious need to provide a certain level of quality and security, comparable to that provided by the traditional telephone systems. While the problem of QoS mostly refers to the network layer, the problem of security is strictly related to the signaling mechanisms and the service provisioning model. For this reason, at present, a very hot topic in the SIP and IP telephony standardization track is security support. In this work, the security model used by SIP is described, and the different open issues are highlighted. We focus, in particular, on the problem of authentication providing a short tutorial on the solution under standardization. The architecture of a possible commercial IP telephony service including user authentication is also described. Finally, we focus on performance issues. By means of a real testbed implementation, we provide an experimental performance analysis of the SIP security mechanisms, based on our open source Java implementation of a SIP proxy server. The performance of the server has been compared with and without security support, under various scenarios.     

Integration of call signaling and resource management for IPtelephony

IP telephony presents a tremendous opportunity to service providers to offer both traditional services as well os a range of creative new services. However, there are substantial challenges to be faced in supporting a resource management framework that is adequate for telephony, and in providing a signaling architecture that enables these services while preserving user privacy and preventing theft of service. This article describes the distributed open signaling architecture, a framework for call signaling and resource management that meets these needs. A key contribution of our work is a recognition of the need for coordination between signaling, which controls access to telephony-specific services, and resource management, which controls access to network-layer resources. We evaluate one approach to resource management in the backbone, consistent with our architecture, using signaling for aggregates of flows. Using traces from cells on the AT&T long distance network, we show that the multiplexing gains achieved by such aggregation con achieve most of the benefits of per-flow signaling, while avoiding its overheads. We also evaluate scheduling algorithms in order to understand their effect on the end-to-end delay experienced by voice packets     

IP网实时业务服务质量问题的研究

分析了ＩＰ电话／传真业务对于服务质量的要求,对解决 其服务质量问题做了较深入的分析和探讨。列出了应用 层可采取的提高服务质量的方法,并详细分析了国际上 解决网络服务质量的两大研究模型： ＲＳＶＰ和 ＤＩＦＦ－Ｓｅｒｖ 的技术内容和优缺点。提出Ｄｉｆｆ－Ｓｅｒｖ才是解决网络服务 质量问题的研究方向。     

IP Telephony Solutions for the Customer Premises

Recent developments in IP telephony equipment have made it possible to provide a fully converged communications solution. This paper specifically concentrates on the customer-premises-based IP telephony solutions for the business market and in particular it looks at the next generation in IP-based telephones and private branch exchanges (PBXs). This also leads to the importance both of the network infrastructure, including local and wide-area networks, and of providing end-to-end quality of service to support voice as a real-time service. Suppliers are striving to develop the systems that will exploit the new converged world while at the same time deliver the reliability and functionality that customers demand. It is currently very much an early adopter market for these IP telephony solutions and experience drawn from delivering trials to external customers as well as the internal pilot work is used to provide a practical snapshots of today's solutions.     

QoS enabled voice support in the next generation Internet: issues,existing approaches and challenges

The Internet is under rapid growth and continuous evolution in order to accommodate an increasingly large number of applications with diverse service requirements. In particular, Internet telephony, or voice over IP is one of the most promising services currently being deployed. Besides the potentially significant cost reduction, Internet telephony can offer many new features and easier integration with widely adopted Web-based services. Despite these advantages, there still exist a number of barriers to the widespread deployment of Internet telephony. The most prominent one, however, is how to ensure the QoS needed for voice conversation. The purpose of this article is to survey the state-of-the-art technologies in enabling the QoS support for voice communications in the next-generation Internet. In this article, we first review the existing technologies in supporting voice over IP networks, including the basic mechanisms in the IETF Internet telephony architecture and ITU-T H.323-related Recommendations. We then discuss the IETF QoS framework, specifically the Intserv and Diffserv framework. Finally, we present two leading companies' (Cisco and Lucent) solutions to offering IP telephony services as examples to illustrate how real systems are implemented     

VoIP mobility in IP/cellular network internetworking

This article explores VoIP mobility in the context of IP and cellular networks interworking. ITU-T Rec. H.323 gateways provide the interconnection between IP networks and switched circuit networks. They allow a call originating from an SCN phone to be transmitted over an IP network to an H.323 terminal, or bridged to another SCN phone. While H.323 provides interoperability with other SCN terminals, the major efforts have been focused on IP/wired SCN (PSTN, ISDN, etc.) interworking. In this article we discuss the challenges associated with the interworking between IP networks and cellular networks through H.323 gateways, and propose an innovative approach using the existing call transfer supplementary service to provide VoIP mobility in the H.323 IP telephony networks. The proposed approach uses existing components in the H.323 standard, thereby allowing VoIP mobility service in hybrid IP/cellular networks to be a value-added feature in the existing H.323-compliant Internet telephony systems     

Mobile agent and CORBA technologies in the broadband intelligentnetwork

This article presents an IN architecture that is based on distributed object and mobile agent technologies. It deviates from traditional IN in two prominent ways: first, by substituting message-based protocols with a CORBA distributed processing environment. The new communications infrastructure not only serves to convey IN flows, but also functions as a potent unifying location-transparent layer. Second, the new architecture makes heavy use of mobile agent technology. Service logic programs are implemented as mobile agents, and are thus not constrained to control the switch's operations remotely from a central location. Instead, they can migrate and control its operations locally. The infrastructure elements of the architecture exploit this ability of service logic programs by cloning them and dragging their clones to the appropriate locations in response to excess processing or signaling load. The architecture is in this way self-balancing, in contrast to the centralized nature of traditional IN. A prototype implementation is presented along with a service creation framework that enables the IN service designer to disregard environment-related issues and concentrate on the IN logic of the services per se     

Multimedia services interoperability in next-generation networks for the residential environment

This article presents an open and functional architecture based on a next-generation network solution in order to seize its converged nature to guarantee interoperability among different platforms and terminals. The defined architecture is an integrated solution for end-to-end communication of various IP multimedia subsystem platforms marketed by different vendors, belonging to several operators located in different countries across Europe and supporting multiple terminals and technologies in order to experiment with a new broadband telephony service for the residential environment. This work addresses, in particular, the interoperability problems of name resolution and implementation of the Session Initiation Protocol Interface at the terminal as a key use in such a scenario. Also, several advanced services are developed and tested within the proposed architecture as a proof of concept of the IMS capability for quick service creation and deployment. The architecture proposed contributes to the reduction of obstacles and barriers among the European countries in order to adopt NGN architectures, allowing a new type of communication that is more complete and flexible.