Dual digital transmission system for multipair cables based on ?crank-shaft coding?

A transmission system which provides two virtually independent digital channels for digital voice or data on a single two-wire connection is described. The basis of this system is the proposed ?crank-shaft code?. Full-duplex operation can be obtained in a straightforward way by including echo cancellation.     

Echo cancellation and applications

Practical echo cancellation techniques, in particular, those used in telecommunications, are reviewed. The various situations in which echoes are generated are examined. Echo path modeling techniques and adaptive algorithms for coefficient control are reviewed. Current international standardization activities are discussed, and echo canceler implementation considerations are set forth. These include echo cancelers for telephone circuits, echo cancelers for full-duplex data transmission over voice channels, acoustic echo cancelers, and echo cancelers for ISDN digital loop transmission     

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.     

密话在INMARSAT数据信道上的传输

介绍了INMARSAT系统的发展与应用,提出了应用海事卫星数据信道进行加密话音双向传输的实现方案,给出了系统组成及功能实现,并对话音传输中应用的一些关键技术,包括:话音的流控实现、话音的实时转发及话音的存储与回放等,进行了详细介绍。经过方案论证及试验验证,实现了密话在数据信道上的双向传输,证明了该实现机制是切实可行的。     

话音引接倍增设备的设计与实现

话音引接倍增设备是一种采用话音压缩编码和复分接技术、将一定数量的64 kbps PCM话音集中在窄带信道中传输的设备。介绍了话音引接倍增设备的应用背景和工作原理。针对具体的应用环境,设计了话音引接倍增设备技术方案,描述了设备组成和功能划分,并对语音压缩编码技术和回波抵消与透传双音多频联合处理进行了分析和设计。通过模拟测试环境和卫星链路功能测试,验证了方案设计合理可行,提高了传输链路的带宽利用率,设备各项功能指标满足应用需求。     

VoIP通信中的回声消除方法研究

介绍了语音通信声学回声产生模型和自适应AEC回声消除算法原理,分析了AEC应用于VoIP语音通信中存在的问题,设计了一种基于短时能量的非线性回声消除方法,在NGN网络的VoIP通信中,使用该方法实现了极高的回声抑制比。测试结果表明该方法的消回声效果、算法稳定性和实现复杂度等指标明显优于自适应AEC算法,适合于嵌入式VoIP通信终端设备的开发。     

The Effects of Large Interference on the Tracking Capability of Digitally Implemented Echo Cancellers

Adaptive mean-square-tapped-delay-line echo cancellers for voice applications are conventionally designed to stop adjustment during periods of "double-talking", i.e., when a large informationbearing signal is present along with the echo signal to be cancelled. Continuous adaption is, however, desirable in full-duplex, two-wire data transmission where the periods of double-talking are so long that the echo channel may vary. We presume that the tap weights of an echo canceller have converged during a training period free of double talking, and address the problem of subsequent echo-canceller tap adjustment via the estimated-gradient algorithm in the presence of double talking. In the estimated-gradient algorithm the tap increment should be proportional to the product of the residual echo and the tap voltage. However, when double talking occurs the residual echo can only be estimated. For an idealized double-talking model, it is demonstrated, from infinite-precision considerations, that use of the memoryless maximum-likelihood estimate of the residual echo is nearly equivalent to abrupt reduction of the step size of the adjustment algorithm when double-talking begins, and could provide an automatic mechanism for recognizing double-talking. Unfortunately, the response of a digitally implemented canceller to a sharply reduced step size can be a deterioration in performance. In fact, the use of an exceedingly small step size during periods of doubletalking may lead to a cancellation error considerably larger than that predicted by coefficient precision. It is demonstrated how averaging the estimated gradient can significantly decrease the mean-squared tap error during periods of double talking. To a first approximation, the tap-weight error can be reduced by a factor proportional to the averaging interval, with an equivalent decrease in tracking capability.     

基于CVSD编码的无线语音系统方案的设计

简要介绍了连续可变斜率增量(CVSD)调制的原理。与目前应用较为广泛的其它语音编码方式相比较,CVSD拥有更优的数字特性。着重介绍了由CML公司研制开发的基于CVSD的语音编码芯片———CMX649的特点及相关的应用方式。CMX649能够成功地应用在广泛的语音编码系统中,尤其适合无线语音系统应用。在此基础上,给出了一种实用性很强的低成本、低功耗无线语音系统的设计与应用方案。该方案可提供清晰可靠的语音传输,可广泛应用于农村地区,具有广阔的市场空间。     

Design of a broadcast packet switching network

An overview is given of a system designed to handle a heterogeneous and dynamically changing mix of applications. It is based on fiber-optic transmission systems and high-performance packet switching and can handle applications ranging from low-speed data to voice to full-rate video. A novel feature is a flexible multipoint connection capability suitable for broadcast and conferencing applications. The architecture of a switching systems that can be used to support this network is described     

一种基于SDH技术的雷达数据宽带传输方案

雷达数据包括回波、工作状态监控、雷达控制和话音四类,如何对此进行高效传输是雷达组网过程中必须解决的问题。本文提出了基于SDH(同步数字系列)传输技术实现雷达数据宽带传输的方案,给出了实现雷达数据传输的SDH网络拓扑结构和虚级联的数据通道分配方案。     

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     

基于窄带电台的无线AAL2传输及组网技术

提出了一种基于窄带电台的无线AAL2传输及组网方案,在双向带宽19200bps的信道下,实现了压缩语音和数据的同时组网传输。针对窄带信道的特性,提出采用AAL2适配来满足语音的实时性,提出增长信元技术和带宽自适应技术来提高数话同传时的带宽利用率,较好地解决了语音数字业务跨电台传输组网的QoS问题。给出了窄带电台无线AT...     

Design approaches and performance criteria for integrated voice/data switching

This paper examines some of the fundamental problems associated with the design and performance of integrated systems and networks that switch both voice and data. Specifically, the need for an integrated approach to the switching and transmission of voice and data is explored and alternative design considerations are discussed. One approach, described in detail, utilizes a distributed architecture to implement variable width channel allocations for the dynamic union of voice and data. Key performance criteria which aid the systems designer in evaluating the merits of a proposed unified design are identified. Examples are illustrated and supportive material is provided by a comprehensive bibliography.     

Architecture and Design of a Reliable Token-Ring Network

Architecture, performance, transmission system, and wiring strategy of a token-ring local area network implemented at the IBM Zurich Research Laboratory are described. In the design of the system, particular emphasis was placed on high reliability, availability, and serviceability. To ensure robustness of the token-access protocol, we employ the concept of a monitor function which is responsible for fast recovery from access-related errors. Our protocol supports asynchronous transmission of data frames concurrently with full-duplex synchronous channels, e.g., for voice services or other applications requiring guaranteed delay. The delay-throughput performance of the token ring is shown to depend very little on data rate and distance. The transmission system of the ring is fully bit synchronous and allows insertion/removal of stations in/from the ring at any time. A mixed ring/star wiring strategy is used which provides the means for both fault detection and isolation, and system reconfiguration, and allows wiring of a building systematically.     

Expressnet: A High-Performance Integrated-Services Local Area Network

Expressnet is a local area communication network comprising an inbound channel and an outbound channel to which the stations are connected. Stations transmit on the outbound channel and receive on the inbound channel. The inbound channel is connected to the outbound channel so that all signals transmitted on the outbound channel are duplicated on the inbound channel, thus achieving broadcast communication among the stations. In order to transmit on the bus, the stations utilize a distributed access protocol which achieves a conflict-free round-robin scheduling. This protocol is more efficient than existing round-robin Schemes as the time required to switch control from one active user to the next in a round is minimized (on the order of a carrier detection time), and is independent of the end-to-end network propagation delay. This improvement is particularly significant when the channel data rate is so high, or the end-to-end propagation delay is so large, Or the packet size is so small as to render the end-to-end propagation delay a significant fraction of, or larger than, the transmission time of a packet. Moreover, some features of Expressnet make it particularly suitable for voice applications. In view of integrating voice and data, a simple access protocol is described which meets the bandwidth requirement and maximum packet delay constraint for voice communication at all times, while guaranteeing a minimum bandwidth requirement for data traffic. Finally, it is noted that the voice/data access protocol constitutes a highly adaptive allocation scheme of channel bandwidth, which allows data users to recover the bandwidth unused by the voice application. It can be easily extended to accommodate any number of applications, each with its specific requirements.     

Technologies for global communications

Some of the technological breakthroughs underlying today's global communication are reviewed, and the relationship between business needs and the evolution and revolution of telecommunications technology is discussed. The introduction of trans-Atlantic transmission cables and the advent of satellite transmission are described. The role of photonics in reducing transport cost and improving quality and the development of information storage and processing techniques that use various voice and video applications to take advantage of this processing power are outlined. The development of global intelligent networks consisting of a distributed switching fabric with both the local switches and the long distance/international gateway switches interconnected and communicating among themselves with standardized common channel signaling protocols and the applications of such networks are also reviewed     

android系统通话中回声消除的实现

为了消除android系统电话免提通话时产生的声学回声,利用静音检测（VAD）机制,在android系统开源代码软件asterisk模块中,加入声学回声消除算法。通过不断比较来话音和去话音数据,判断是否为声学回声并进行白噪声替换,测试结果表明在一般的通话环境中,可以消除正常语音通话时90%以上的回声,实现半双工通信,适合于嵌入式android终端设备的开发。     

基于小波变换的声学回波消除系统

声学回波消除技术一直是语音通信领域的研究热点。在声学回波消除系统中,通过估计回波路径中的固定时延区域来提高自适应滤波算法的收敛速度。提出了一种基于小波变换的固定时延估计算法以及基于小波变换的声学回波消除系统,解决传统时延估计算法在声学回波消除系统中估计误差大、抗干扰能力弱的问题。测试结果表明,算法稳健性、有效性等指标明显优于传统时延估计算法,基于小波变换的声学回波消除系统具有良好的消回波性能。     

A multiplexing scheme for H.323 voice-over-IP applications

Voice communications such as telephony are delay sensitive. Existing voice-over-IP (VoIP) applications transmit voice data in packets of very small size to minimize packetization delay, causing very inefficient use of network bandwidth. This paper proposes a multiplexing scheme for improving the bandwidth efficiency of existing VoIP applications. By installing a multiplexer in an H.323 proxy, voice packets from multiple sources are combined into one IP packet for transmission. A demultiplexer at the receiver-end proxy restores the original voice packets before delivering them to the end-user applications. Results show that the multiplexing scheme can increase bandwidth efficiency by as much as 300%. The multiplexing scheme is fully compatible with existing H.323-compliant VoIP applications and can be readily deployed.