传统长途电话网向分组化网络的演进之路──BICC

传统ＰＳＴＮ网向分组化网络演进是历史的必然,而长途电话网的演进则是首当其冲。本文首 先分析了这种演进的必然性及演进中遇到的问题,指出承载无关的呼叫控制（ＢＩＣＣ）为此铺 就了既面向未来又能充分地利用已有承载网投资的演进之路。接着本文介绍了ＢＩＣＣ的功能和 协议模型,以及ＢＩＣＣ在信令消息及信令处理过程方面与ＩＳＵＰ的差别,并用一个例子说明 了ＢＩＣＣ的动态流程。最后本文还介绍了ＢＩＣＣ协议标准化的现状和未来。     

新一代通信网的信令技术

文章深入讨论了以IP分组化为基本特征的新一代通信网的信令技术,描述了多媒体会话控制信令和通用信令传送协议,介绍了与承载无关的呼叫控制信令体系和IP网络QOS支持信令的研究动态,并展望了未来开放式信令的发展方向。     

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     

The ITU-T BICC protocol: the vital step toward an integratedvoice-data multiservice platform

At the end of 1999, the ITU-T completed the first edition of the Bearer-Independent Call Control (BICC) protocol, just nine months after the start of this activity. The development of BICC can be considered historic. For the first time a full-fledged operator-grade PSTN/ISDN service can be offered over a variety of packet networks, using only standardized protocols. This tutorial explains the BICC architecture and the operation of the BICC protocol     

IMS与传统PSTN网络的互通问题研究

目前,下一代网络（NGN）的核心技术正从传统软交换向IP多媒体子系统（IMS）衍化。在IMS这个全新的网络构架下,如何实现与传统公用交换电话网（PSTN）的互通成为目前网络转化过程中急需解决的问题之一。从阐述IMS网络的系统架构入手,通过典型的呼叫实例分析,探讨从IMS终端到PSTN终端的呼叫流程。着重分析互通过程中各IMS网络功能实体的详细作用,以及会话发起协议（SIP）与综合业务数字网用户部分（ISUP）之间的转换映射关系。     

Voice service interworking for PSTN and IP networks

This article presents an overview of the main technical problems to be addressed for the provision of interoperable services between IP telephony and the PSTN. The pivotal element of the solution resides in an interworking function. This function is typically implemented in a gateway whose requirements and behavior are here analyzed in terms of signaling and control protocols (control plane) as well as user data transfer (user plane). The presentation is structured around these two planes. The control plane defines the set of signaling protocols to be used in each networking context and the translation between them. Detailed scenarios illustrate the signal translation in the gateway allowing for the establishment of a hybrid phone call. The user plane is responsible for adapting the user data to the properties of each network channel and determines the quality of service of the voice call in terms of delay and speech quality     

基于现有VOIP实现多方通话的方案

随着网络技术的不断进步和网络间融合趋势的不断增强,IP网络开始承载越来越多的传统电信网和电视网上的业务,尤其是VoIP业务的应用在Internet网络上的发展速度尤其惊人。在VoIP的业务中,近年来的热点和难点在于,如何利用极少的内存消耗以实现更多用户可同时进行交流的会议电话,给出了基于现有voip调度实现多方通话的解决方案,并简介了多方通话的混叠技术,该方案具有可拓展性,不仅可以实现4方用户同时通话,还可以实现参与用户数大于4的多方通话。     

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     

一种基于FPGA的媒体网关设计与实现

基于IP的软交换网络实现话音业务并能与PSTN交换网实现互联互通,关键要解决网络之间的信令互通和话音互通。现给出了一种与PSTN交换网络能够实现互通的一种媒体网关的硬件、软件和FPGA设计的实现方案介绍,该设计方案采用模块化设计,单板接口数量多,可通过控制软件灵活配置,还兼具有可扩展性。对媒体网关中FPGA的整体设计方案,以及上行数据处理和下行数据处理部分中各个模块的设计和功能进行了比较详尽的介绍,其中下行数据处理模块设计中还兼具有保证Qos的一种方法。     

新一代电信网信令协议BICC

根据ITU－T SG11的最新研究动态,介绍支撑电信网演进的重要信令与承载无关的呼叫控制（BICC）信令的应用背景,网络结构,协议模型,能力集1和2（CS－1和CS－2）的信令标准及其进一步研究方向。     

RIDE Replacement — A Hybrid PSTN/VoIP Application Solution

This paper, co-authored by BT and Telspec, describes the architecture, development and deployment of the recorded information distribution equipment (RIDE) replacement platform. The platform delivers large volumes of announcements into the PSTN yet has an internal architecture built on a multicast QoS-enabled VoIP network and provides a ready-made evolution path to terminate calls originated from IP networks. It is therefore seen as one of the first platforms to be deployed by BT to deliver consistent functionality into both the legacy PSTN and the evolving IP world.     

Distinctive characteristics of call-handling procedures andsignaling logic in a Russian public telephone network

The article describes the main peculiarities of the former Soviet Union public switched telephone network (PSTN), signalling systems, call handling procedures, charging principles, etc. General information about the PSTN is also included. The primary goal of the paper is to show some specific PSTN features and give the classification of the existing interfaces rather than to describe existing exchanges in detail     

Java call control, coordination, and transactions

Future telecommunications networks will consist of integrated packet-switched (IP and/or ATM), circuit-switched (PSTN), and wireless networks. Service providers will offer a wide portfolio of innovative applications over these integrated networks. Doing so rapidly and efficiently requires open network APIs, with a key API being that for call control, as well as for coordination and transactions. The JAIN community is defining an API for Java call control and Java coordination and transactions. The JCC API defines the interface for applications to initiate and manipulate calls, while JCAT defines the facilities for applications to be invoked and return results before or during calls. Note that in this context a call refers to a multimedia, multiparty, multiprotocol communications session. The JCC/JCAT Edit Group of JAIN is in the process of defining the first version of the JCC/JCAT API specification, which is expected to be released in early 2000. This article describes the background of and motivation for the design of the JCC/JCAT API. We describe the AIN and JTAPI call models on which the JCC/JCAT API is based. We then describe the scope of JCC/JCAT and its relationship to other JAIN Edit Groups defining facilities for enabling service creation. Finally, we describe the requirements and example service drivers for JCC/JCAT, as well as the initial proposed design and structure for JCC and JCAT     

NGN的演进

文章介绍了下一代网络(NGN)的起因、定义和特点,分析和比较了狭义NGN及可支持多种接入的IP多媒体子系统(IMS)结构以及将移动网络、固定网络和Internet融合的广义NGN。重点介绍了广义NGN的通用模型和网元的基本功能,说明NGN的演进是电信发展的必然趋势。     

流控制传输协议概述

本文旨在扩充计算机网络课程的授课内容。流控制传输协议(SCTP),最初设计用于在IP网络上运输PSTN信令,现已作为互联网的一种新型运输层协议。SCTP可运行在IP之上,为用户提供可靠数据传送服务,并支持使用cookie机制的关联建立、多穴端点、多流、选择确认、流内有序和无序交付等新特性。文中主要阐述SCTP的体系结构、功能、应用,及其与两个主要的运输协议TCP和UDP的异同,可供教学参考。     

话音无阻塞的可视电话系统设计与实现

为实现话音无阻塞、视频图像清晰流畅和低延迟的高品质可视通话,提出了一种使用PSTN单独传送话音,使用IP网络单独传送通话者视频并在终端同步还原的可视电话系统及其实现方法,系统由终端和支撑服务器群构成。设计用PSTN的话音拨号信号或来电振铃信号作为视频呼叫的触发信号,终端通过捕获电话号码封装为SIP的Invite消息并发起视频呼叫,与对端建立P2P连接并使用UDP报文传送RTP视频流来实现视频通话。系统测试表明通话话音始终无阻塞;在带宽为256kbit／s、IP网络无拥塞的条件下,实现了视频CIF分辨率,帧频30fps,视频延迟小于60ms;用户使用习惯如电话号码和拨号方式无需任何改变。其综合性能优于现有的PSTN可视电话和IP可视电话。     

软交换技术的应用与发展

IMS技术的逐渐兴起，对软交换技术的发展产生了新的挑战。与IMS相比软交换更适合于实现传统的窄带话音业务，因此软交换后续的发展应该定位于在分组交换网络上继承和发展传统语音业务。在这一定位的基础上，软交换可以应用于公共交换电话网（PSTN）长途网以及网络智能化改造当中，同时软交换的一般网络架构也会随着业务的发展和提高服务质量的需求而有所变化。     

软交换中基于VoIP语音识别的业务部署方案

本文提出了一套基于VoIP语音捕获及其识别的业务部署方案,该方案可以在不改变现有通信网络结构和业务部署的情况下,利用语音识别技术,继承或扩展非IP网的通信业务,也可以在IP网中部署新的增值业务。实验环境下的验证结果表明,该方案具备实际部署的可行性。     

基于VoIP技术视频会议系统研究与设计

为了实现视频会议的IP网络与传统的PSTN网络互通,由于现有的SmartMeeting系统是一个纯粹的IP网络视频会议,这样就不能够利用传统的PSTN网络下的电话进行视频会议,故该系统采用增加VoIP网关进行对H．232协议和SIP协议做了相应的协议装换的分析,在现有SmartMeeting系统的基础上对IP网络与PSTN网络进行互通功能实现进行详细的研究与设计。系统是在VC＋＋6．0平台下开发,采用C／S模式实现的。