A Distributed Experimental Communications System

The packet experimental communications system (packet XCS) is a new experimental voice and data switch. It uses a local-area network (LAN) for digital voice transmission, with local intelligence for switching. The packet XCS also has highly distributed control. The individual sites cooperate to provide user services as well as internal data management. We have learned that several local networks, including CSMA/CD networks, can be made to work well for voice transmission and that highly distributed control is practical in such a system. A system has been constructed which is used as a testbed for distributed voice and data communications experiments. This system is purely for experimentation and does not indicate a direction for future Bell System product offerings.     

Microprocessor Application to Telephone Terminal Switching System Architecture

Microprocessor application to a switching system, especially to telephone terminals, is expected to change switching system architecture. From this viewpoint, Basic Experimental Switching System (BESS) is designed as a functionally distributed control switching system accommodating intelligent terminals. This paper describes BESS hardware and software structures. BESS has the following characteristics. 1) Microprocessor controlled terminals can store and forward dialed digits, and can exchange data between them. Almost all services can be realized merely by a program stored in each terminal itself. 2) Functionally distributed controller of only three untailored microprocessors can treat 100 basic calls. 3) Program flexibility is especially realized by distinct exchange program separation into algorithm, data management and network control parts.     

Distributed Microprocessors Control Architecture for Versatile Business Communications

This paper describes the NEAX®2400 Information Management System (IMS), which is no longer a conventional telephone switching system, but is instead a switching hub for various office automation equipment. The NEAX2400 IMS not only provides various circuit-switching functions such as conventional voice communication (telephone) switching, low-speed and high-speed data switching, but also provides stored and forward capabilities for voice (voice mail) and data (text mail, facsimile mail, etc.). This paper especially emphasizes the basic EPBX features and functions of the NEAX2400 IMS. The standard 64 kbit/s PCM technique is used for digitalization of voice signals, and switched through a nonblocking architecture time-division digital network. All digital data signals are transmitted through the same time-division digital network at the 64 kbit/s rate intermixed with digital voice signals. The controls of the switching functions, station service features, and maintenance service are performed by functionally divided distributed microprocessers. The most outstanding attribute of the NEAX2400 IMS is the unique building-block architecture of the equipment configuration. Modules are stacked above the basic module as the number of line and trunk ports, or additional stored and forward features are required. Up to four additional modules can be stacked up as a single module group. This unique arrangement permits the NEAX2400 IMS to be very flexible in its system applications and expandability. Practically, the NEAX2400 IMS will economically service as few as 184 ports (mixture of voice/data, line/trunk) and can be continuously expanded to as many as 23 184 ports.     

A Digital Cross-Connect System for Digital Data Circuits

The AT&T subrate data cross-connect system (SRDC) brings the capabilities of the digital access and cross-connect system (DACS) used primarily for analog circuits to the digital data world, including the capability of mechanized remote provisioning and maintenance. The SRDC performs DACS-equivalent functions on 64 kbit/s channels of the Digital Data System (DDS) and other similar emerging data networks at the 2.4, 4.8, 9.6, and 56 kbit/s rates. An SRDC-equipped DACS frame can process both voice and digital data in a shared arrangement. It can also provide all the functions necessary for a DDS hub office environment, including subrate multiplexing, multipoint functions, dataport error correction and secondary channel processing. The operational functions supported by SRDC include circuit testing, facility performance monitoring, circuit and equipment provisioning, memory administration, and equipment maintenance.     

High-Speed Packet Switching Systems for Multimedia Communications

High-speed packet switching (HPS) systems can Provide flexible, economical, high-quaiity services for integrated voice, video, and data communications. To realize such HPS systems, methods have been developed to bring about high-speed protocol processing as well as a system architecture for facilitating high-throughput switching. Adopting the parallel processing algorithm into protocol processing allows us to achieve high-speed packet protocol processing of about 100 times faster than conventional processing. Furthermore, a fully distributed system architecture in addition to hierarchical interconnection networks can achieve high-capacity packet switching systems. The proposed HPS system is thus capable of accommodating lines of up to 10-50 Mbits/s, of providing high-throughput switching capability of 1 000 000 packets/s, and of having an average delay of less than 2 ms. Furthermore, an evaluation of network delay performances of video conferencing and voice communications indicate that HPS systems are quite suitable for handling such multimedia communications.     

Paris: An approach to integrated high-speed private networks

This paper describes a design of a high-speed packet switching system for integrated voice, video and data communications. The system makes use of a simplified network architecture in order to achieve the low packet delay and high nodal throughput necessary for the transport of voice and video. A prototype of this system has been implemented and is now being tested under a variety of packet traffic loads. We have demonstrated that this system provides a cost-effective solution for private integrated networks.     

基于公众移动通信网的集群通信系统的设计和实现

本文提出了一种基于公众移动通信网的集群通信系统PTOS的体系结构,其基本原理是:在控制和管理层面,利用智能网的思想和体系结构对群组通信进行管理和控制;在资源层面,使用语音板卡提供的多方会议和混音功能实现群组的实时多方通话.首先介绍了PTOS的软硬件体系结构,然后对其实现中的关键技术如多点控制模块设计、软硬件可靠性、应用层消息路由、跨板卡的会议等进行了研究,最后对本系统的商用前景和局限性进行了讨论.     

基于AMBE-2000的数字语音传输系统

介绍了AMBE-2000的工作原理及其应用,并结合实例给出了数字语音传输系统中AMBE-2000与TLV320AIC10的A/D、D/A接口电路。AMBE-2000是DVSI(Digital Voice System·Inc)公司的语音压缩解压缩芯片,它采用AMBE语音压缩技术,具有语音音质好和编码波特率低、功耗低等优点;TLV320AIC10是TI公司的低功耗∑-Δ型16位A/D、D/A音频接口芯片。应用AMBE-2000的数字语音传输系统可以在低速率下提供优良的语音质量,并能实现全双工、实时语音压缩解压缩功能,因此在卫星通信、移动通信等领域有着广泛的应用。     

自愈通信系统的结构与功能

文章介绍了自愈通信系统的网络结构,讨论了自愈系统的基本单元、链接和开放业务平台.对于自愈通信系统的功能,文章重点讨论了链接多种技术和网络的业务网关如何以多种方法实现即插即用、寻址和移动、对等通信链接和应用等功能.文章认为自愈通信系统实现的难点在于用户适应特定环境的发现和寻址机理.在动态网络结构基础上,提供含有文本意识、个性化和自适应能力的会话;在中间件结构上实现自适应运行能力等是自愈通信系统的重要研究课题.     

Very high speed and high capacity packet switching for broadbandISDN

The authors discuss and propose a very-high-speed and high-capacity packet-switching (HPS) architecture for a future broadband ISDN (integrated-services digital network). The HPS network accommodates various communication services, such as voice, high-speed data, high-speed still picture, and video services. The proposed architecture has three significant principles: a high-speed oriented simple network protocol, separation of signaling and network control from data transfer, and hardware switching. These principles provide fast- and high-throughput transmission for data packets and reliable transmission and processing for call-control packets. The HPS protocol structure is addressed, which provides high flexibility for various communications services as well as high-speed capability. A 3-Gb/s capacity and building-block-structured packet-switching system architecture, using bus- and loop-type switch fabric, is also presented     

A software architecture for workstations supporting multimediaconferencing in packet switching networks

The software architecture of a multimedia workstation connected to an experimental packet switching network which is capable of switching voice, data, and video simultaneously is described. The user of a multimedia workstation may engage in several multimedia calls, each using multiple information streams consisting of voice, data, and images. The network interface software in the workstation maintains the temporal relationship among a given call's information streams. For each call, the network interface provides a multiple information channel interface to upper layer software supporting the notion of a multimedia virtual circuit. A connector mechanism is provided to control the information exchange among the network interface, the various voice, data, and image peripherals, and the computer programs used in a multimedia application. These information sources and sinks share a common abstraction called active devices     

A modeling approach to software cost estimation

The author describes a novel software estimation modeling process as well as the important productivity factors and the productivity measurement metrics used in the 5ESS project, one of the largest telecommunication projects at AT&T Bell Laboratories. The 5ESS switch is a modern digital electronic switching system with a distributed hardware and software architecture. The model estimation approach has greatly improved the quality of estimates. The study of productivity factors has resulted in some significant productivity and quality improvement processes in the 5ESS development community     

Personal Access Communications System: fixed wireless local loopand mobile configurations and services

The Personal Access Communications System (PACS) supports wireline quality voice, voiceband data, digital data, and messaging services and is based on a low-power, low-complexity design. Operating environments for PACS include indoor and outdoor; fixed, low mobility, and vehicular mobility; and private and public access. PACS is a microcellular system providing high frequency reuse efficiency and thus is capable of supporting high traffic density. These attributes make PACS ideal for providing fixed wireless local loop and mobile services in moderate to densely populated areas, including developing countries. PACS mobile and fixed wireless local loop service can be offered by interfacing PACS with advanced intelligent network/ISDN switches, any traditional wireline switches, or with cellular/personal communications system mobile switching centers. The PACS air interface has been standardized by the American National Standards Institute     

Software Protocol for Handling ISDN in the Packet-Switched Network

The Integrated Services Digital Network (ISDN) provides basic architecture for existing, as well as future residential plus business communications. ISDN overlayed with CCS#7 of a digital PSTN (Public Switched Telephone Network) can be the ultimate, ubiquitous network for circuit switch (voice, data), packet switch (voice, data), and private line (voice, data) applications. Assuming that the present ISDN has to interwork in the present physically separate overlayed networks (voice and data), significant problems are expected to emerge for designing hardware and linking softwares for handling packet traffic. In this paper, the software-related problems, when ISDN packet distribution nodes have to handle an ISDN interface, will be outlined with an ISDN software protocol solution. An approximation of the delay involved in the telephone switching system which is part of ISDN processing as well as the delay for the interface gateways, the HOST computer nodes, and the LAN and WAN computer nodes will be identified and formulated to reflect the total performance measure defined. Major emphasis is given to flow and congestion control performance measures in the ISDN Gateways, which are analyzed and simulated with the assistance of the basic delay table transfer software model developed for the IMPS and gateways in the ARPANET, MILNET, and MINET. The performance evaluation of this basic ISDN interfacing software, which only involved one ISDN level, i.e., the HOST or gateway and its related subnetworks, is simulated on sections of these networks to illustrate its congestion control effectiveness. There are six mathematical software techniques to account for end-to-end delay, which form the basis for the solution to these ISDN software-hardware problems in the Interface Gateways connecting the electronic switch to the computer network components.     

中国电信移动通信网络管理系统

中国电信移动通信网络管理系统的由一个部级网管中心和３１个省市级网管中心构成，将对中国邮电部模拟及数字移动通信网内的ＴＡＣＳ系统及ＧＳＭ系统进行综合性的网络管理。本文综述了邮电部移动通信的现状，发展指标，网元的现状，开发网管系统的指导思想，以及逐步实施ＩＴＵ－ＴＴＭＮ建议的必要性。     

Economic analysis of integrated voice and data networks: A case study

This paper describes the results of a study to evaluate alternative switching strategies for future integrated voice and data networks. Three fundamental problems are addressed: 1) the economics of integrating voice and data applicatiom in a common communications system; 2) the comparison of alternative switching technologies for integrated voice and data networks; 3) the cost-effectiveness of alternative voice digitization rates and strategies. Three broad switching technologies and variations thereof are investigated and compared. The switching technologies are: circuit switching, packet switching, and hybrid (circuit-packet) switching. Each switching technology can accommodate either voice or data applications separately or combined voice and data requirements in an integrated fashion. Results of studies regarding communications systems option are provided and the sensitivity of the results are tested with respect to traffic variations, cost trends of switching and transmission, and network performance variables. The significant variables which affect the results are identified and quantified. The intent of this study is to identify and quantify network technologies which demonstrate long-term low operating costs, This is a necessary effort to provide the basis for determining the most cost-effective evolutionary path for future communication systems. It is recognized that transition problems and associated costs may be other important factors determining the ultimate evolutionary path. However, determining these costs was not an objective of this study. Neverthless, this study provides a framework and a target technology for detailed evolution, planning, and cost analysis.     

An integrated voice and data multiple-access scheme for a land-mobile satellite system

A Land-Mobile Satellite System (LMSS) is a satellite-based communications network which provides voice and data communications to mobile users in a vast geographical area. By placing a "relay tower" at a height of 22300 mi, an LMSS can provide ubiquitous radio communication to vehicles roaming in remote or thinly populated area. LMSS is capable of supporting a variety of services, such as two-way alphanumeric service, paging service, full-duplex voice service, and half-duplex dispatch service. A Network Management Center (NMC) will handle the channel requests, channel assignments, and in general the network control functions. A pool of channels is managed at the NMC to be shared by all mobile users. An integrated demand-assigned multiple-access protocol has been developed for the experimental LMSS. The pool of channels is divided into reservation channels and information channels. The information channels can be assigned by the NMC to be either voice channels or data channels. Each mobile user must send a request through one of the reservation channels to the NMC via the ALOHA random-access scheme. Once the request is received and processed, the NMC will examine the current traffic condition and assign an information channel to the user. NMC will periodically update the partitions between the reservation channels, voice channels, and data channels to optimize system performance. Data channel requests are queued at the NMC while voice channel requests are blocked calls cleared. Various operational scenarios have been investigated. Tradeoffs between the data and voice users for a given delay requirement and a given voice call blocking probability have been studied. In addition, performance impacts of such technological advancements as satellite on-board switching and variable bandwidth assignment are discussed.     

PRANA at the Age of Four: Multiservice Loops Reach Out

Can the French loop plant support voice and multiservice data traffic simultaneously? If so, what techniques are available for practical ISDN operations before 1990, at prices attractive to the domestic subscriber? Four years ago CNET, the French National Center for Telecommunications Research, agreed to back our plan for finding practical answers to those questions. This paper presents our methods and findings. Our program groups capabilities for: voice communications, in either analog or digital form; telematics services, up to 64 kbits/s; and conventional datacomm, at selectable rates up to 9600 bits/s. We call the system PRANA, a French-language acronym for Digital Subscriber Loop System.     

基于SMS4的加密通信可编程片上系统设计实现

针对面向工业用、商用以及民用电子设备加密串行通信需求,通过基于SMS4的加密通信SOPC软硬件架构、SMS4加密通信控制器IP技术、SMS4加密通信软件技术等内容的研究,实现了基于SMS4加密算法的串行通信技术,并在非128位明文的加密传输数据控制和基于8位异步模式的128位数据串行收发上效率明显提升。基于SMS4的加密通信SOPC提高了系统的加密通信功能可定制能力,且使加密通信核心算法自主可控。