The telephone network of the 1960s

In 1960 the transmission technology of the public switched telephone network was ordinary cable pairs, carrier systems on both paired and coaxial cable, and point-to-point microwave radio systems. Voice signals were frequency multiplexed for efficient transmission, but remained in analog form from end to end of a connection. Electromechanical switching systems provided circuit-switched connections, and about half of the customers could directly dial long distance calls. The following decade saw the beginnings of electronic switching and direct distance dialing on a global basis. Meanwhile, applying digital technology to achieve lower costs and better quality for voice transport began in earnest, and the interconnection of data terminals and computer centers became an increasingly important communications need. The initial data communications approach, still used with ever increasing sophistication and speed, was to convert data to speech-like form with data modems. Later, the direct connection of digital data to digital network facilities became the method of choice for building computer communications networks and the backbone of the Internet. It now appears likely that one day all signals, including voice, will be transported by unified high-speed digital networks with a common set of protocols     

Microwave Testing with Millimicrosecond Pulses

Pulse testing techniques have been used for many purposes for a long time. In studies of waveguides as radio and radar components and for possible communications systems use, we have employed various types of pulse testing. The need for greater resolution through the use of very short pulses has always been apparent. For this reason, equipment has recently been built to generate and display 9,000 megacycle pulses having a length of about 6 millimicroseconds. In a pulse of this length there are less than 100 cycles of radio frequency energy, and the pulse occupies less than 10 feet of path length in the transmission medium. The r-f bandwidth required is about 500 megacycles.     

基于蓝牙的电台自动控制装置设计与实现

蓝牙技术是一种可靠且成熟的无线传输技术,介绍一种以蓝牙发送接收模块为基础,以RS232串行通信接口为接口的电台自动控制装置的设计与实现。该系统实现了无线电台发送和接收的语音自动控制,通过该装置用户可以用语音来控制电台的收发而不需要人工按下发射键,为无线电台的使用提供了一种方便的方法,提高了无线电台的使用性能。     

Office of telecommunications policy proposed biomedical telemetry standard for emergency medical communications march 1975

This standard defines the minimum requirements for a multiplex or nonmultiplex voice/telemetry channel to assure compatibility and interoperability of portable physiologic monitoring equipment with both radio and telephone equipments and hospital emergency and display devices. Specifically, it defines a National telemetry subcarrier frequency within the voice band which is compatible with both radio and telephone transmission characteristics. This standard applies only to voice and telemetry modulation-demodulation electronic instrumentation used in emergency medical care.     

Orthogonal Cochannel Operation for High Spectrum Efficiency Based on the Example of 16 QAM-140 Mbit/s Systems Using Rolloff 0.19

In view of the rapidly increasing demands for transmission capacity of public networks and the limited number of radio frequencies available, the band efficiency of existing systems in general will become insufficient for future high-capacity requirements. This fact is becoming increasingly evident for 16 QAM-140 Mbit/s radio systerns at 3.9, 6.7, and 11.2 GHz which are in widespread use in todays long-haul transmission networks. Conceptual and technological measures for making cochannel operation feasible and practical are presented using the example of a system family. By these means, even the capacity of existing 16 QAM-140 Mbit/s radio routes with 40 MHz interleaved channel arrangements can be doubled with the particular advantage of reutilizing the existing antennas and radio equipment with only minor modifications. A progressive filter concept makes orthogonal cochannel transmission possible with negligible adjacent channel spectrum interference. Surface acoustic wave IF filters are used to convert existing systems from 0.5 rolloff to 0.19 rolloff. Special RF channel/antenna arrangements were developed for the transmission of all RF channels in both directions, including space diversity and using only two antennas. A field experiment verified the feasibility of the concept and indicated that a crosspolar interference canceller is generally not necessary for 16 QAM140 Mbit/s cochannel operation.     

Broadband access using wireless-over-fibre technologies

This paper describes a number of wireless-over-fibre technologies for broadband access which are being investigated in the Photonics Group at University College London (UCL). In particular, the paper will report the asymmetric Fabry-Perot modulator/detector (AFPM), functioning not only as an optical modulator in the uplink direction, but also as a conventional photodetector in the downlink direction in a single device. Compared to the existing waveguide type electro-absorption modulator, the AFPM is polarisation-insensitive to light, simpler to fabricate and uses only a single optical fibre for both uplink and downlink transmissions, and therefore has the potential to lower the costs of future picocell deployments. Direct modulation of a semiconductor laser is the simplest way to generate modulated optical signals. In the presence of multiple large modulation signals, however, intermodulation or inter-channel distortion can easily be generated in the semiconductor laser and affect other lower power neighbouring channels. A feed-forward linearisation technique for directly modulated semiconductor lasers, capable of operating at frequencies above 5 GHz, has been developed and will be described in detail in this paper. Most modern commercial buildings already have a multimode optical fibre (MMF) infrastructure for carrying the Ethernet data. To provide cost-effective and reliable indoor cellular and wireless local area network (WLAN) coverage without dependence on the radio penetration from outside base-stations, it is highly desirable that the same MMF infrastructure be used to carry these additional services between the equipment room and the remote antenna units around the building. Although not previously regarded as having sufficient bandwidth, it has now been shown that MMF can successfully carry microwave modulated optical signals, including GSM1800 and UMTS cellular radio. Use of MMF for multiservice, multioperator in-building radio coverage has been demonstrated in a collaboration between UCL and the University of Cambridge and will be described in this paper. As the number of cellular and broadband WLAN devices increases, the lower parts of the radio spectrum are becoming more and more congested. To meet the demand for ever higher data transmission rates, other parts of the radio spectrum are being considered for these applications, particularly the millimetre-wave region (30 GHz and above). One of the challenges facing such systems is the generation and delivery of a low phase noise precisely synchronised local oscillator signal. In this paper, a simple optical heterodyne source with two injection locked slave lasers and a more environmentally robust source using an optical injection phase lock loop will be described, together with signal distribution using a bidirectional semiconductor optical amplifier in a coarse wavelength division multiplex fibre ring architecture. Finally it will be argued in the conclusions that future broadband wireless-over-fibre access networks will be required to provide multiservice and multi-operator coverage for buildings, and having the required technologies at sufficiently low cost will be the key factor to success.     

Trading CDPD availability and voice blocking probability incellular networks

Cellular digital packet data (CDPD) offers mobile users access to a low-cost, ubiquitous, wireless data network. CDPD can be overlaid on existing AMPS analog cellular systems and share their infrastructure equipment on a noninterfering basis. Since CDPD utilizes the idle radio resources of AMPS, it is important to ensure that the normal AMPS activities are not affected by CDPD. Several timers and system parameters are defined in CDPD for this purpose. This article investigates how these timers and parameters affect the performance of AMPS/CDPD systems. Our study provides guidelines for the selection of input parameters to aid CDPD network planning. Specifically, we study the trade-off between CDPD availability and voice blocking probability. CDPD availability is defined as the proportion of time that the RF channels can be used to transmit CDPD data. We observe that selecting too large values for CDPD dwell time and the maximum number of CDPD channel streams will degrade the voice blocking probability without improving CDPD availability. On the other hand, selecting a too large value for the CDPD layoff time will degrade CDPD availability without improving voice blocking probability     

智能电路调度系统

智能电路调度系统在固定台站通信中具有非常重要的意义。本文就此展开讨论,并给出了一套完整可行的电路调度系统方案。经过PCM基群设备,从配线架来的30个模拟话音信号被组合成一个标准的2 Mb/s基群信号。经过智能电路调度系统,4个2 Mb/s基群信号被分解成120个64 kb/s数字话音信号。在计算机的控制下,120个64 kb/s数字话音信号进行交叉连接后,再重新组合成4个2 Mb/s基群信号,从而实现了智能电路调度功能。     

Wideband CDMA system for personal radio communications

This article describes a wideband spread code-division multiple access (W-CDMA) system for high-capacity and high-quality personal radio communication. This system has been authorized as an EIA/TIA Interim Standard IS-665, T1P1 Trial Use Standard J-STD-015, and ITU-R Recommendation M 1073. The system uses wideband spreading to accomplish good interference immunity, high-quality speech, and high-speed data transmission. The system uses coherent detection (CD) and an interference canceller system (ICS) to enhance the capacity. The CD and ICS use continuous pilot signals in the forward/reverse links to estimate the propagation path parameters. PN and Hadamard sequences are used as the spreading code for minimal mutual interference between traffic and pilot/sync/paging channels. A robust 32 kb/s waveform speech coder, ITU-T COM101+, has been developed to achieve toll-quality speech in the radio environment. This system provides up to 128 voice channels per cell and data transmission up to 64 kb/s by 5 MHz spreading. Unification of low- and high-mobility applications and reduced complexity of system and hardware configurations are accomplished     

Design issues relevant to developing an integrated voice/datamobile radio system

The various design issues related to developing an integrated voice/data mobile radio system, including high speed digital radio frequency modulation in a mobile environment, statistics for the talkspurt/silence gap composition of speech, switching schemes for voice/data integration, encoding techniques, and voice and data traffic statistics are discussed. A performance analysis is conducted for a typical design, showing that a voice-only mobile radio system can be upgraded to an integrated voice/data system capable of carrying the full voice and data loads without requiring additional radio channels and without compromising voice performance. Data traffic is only minimally delayed (46.2 ms mean delay) for a fully loaded system     

The DTX-240D digital circuit multiplication system

This paper describes the DTX-240D digital circuit multiplication system (DCMS) offered by ECI Telecom. It will accept up to 240 × 64 kb/s trunks carrying either 64 kb/s voice, voice band analogue non-speech signals, or digital data for transmission over a 2·048 Mb/s digital link. Over 1000 are currently ‘on-line’ and carrying traffic. The system comprises a pair of terminals, one on each side of the interterminal digital link (bearer). It will normally operate in the network at a concentration ratio of 5:1, in which case 150 × 64 kb/s trunks, carrying voice, voice band data or digital data can be concentrated into one 2·048 Mb/s bearer. The users are able to increase the number of trunks up to 240 per 2·048 Mb/s bearer, when time zone differences cause a spread of busy-hour traffic carried on a single system. Each terminal will normally be located at an international switching centre (ISC) but may also be located at an earth-station. The system uses a DSI (digital speech interpolation) stage providing a 2·5:1 multiplication, followed by an additional 2:1 multiplication by means of ADPCM (adaptive differential pulse code modulation). In addition, the VBR (variable bit rate) technique is used to prevent clipping, due to overload congestion. The system can also be used with 1·544 Mb/s digital bit streams (trunk side or bearer).     

基于AMBE2000的新型无线同播系统

在无线多通道语音通信中,距离长短使各接收机接收信号的时间不一,同步输出需要精确的延时。介绍一种新型的基于AMBE2000和AD73311的高质量语音同步通信系统的设计与实现。该系统能够在2.0～2.4kb/s的低比特速率产生高话音质量,实现最小步长为1us的输出延时,达到多路同步传输效果。因此,该系统可以广泛应用于软件无线电,多模式电台及多路广播等场合。     

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.     

A narrowband, high capacity integrated voice/data mobile radiosystem for dispatch-related broadcast applications

This paper presents the design and performance analysis of a narrowband, mobile radio system optimized for dispatch-oriented broadcast applications in the 450 and 800 MHz frequency bands. The proposed system is significant since present-generation systems have proven incapable of providing the increased capacity demanded by new data services such as mobile data terminals, automatic vehicle location, and mobile facsimile. The centerpieces of the proposed design are a dynamic multiplexing technique known as packetized data, voice dedicated (PDVD) burst switching, which allows transmission of data within the silence gaps inherent in speech, and a noncausal signal processing technique called lookahead, which provides the data sources with advance knowledge of where the silence gaps will occur. Simulation results are given for two dispatch-oriented broadcast applications with widely differing traffic statistics, showing the flexibility and efficiency of the proposed system. The results show that for either application, the proposed integrated voice/data system can be retrofitted into existing 450 and 800 MHz public and private land mobile radio channels and, compared to conventional voice-only systems, can provide the user with full data services plus a tripling of voice capacity     

A Data Modem for GSM Voice Channel

This paper introduces a novel approach to data communication over the global system for mobile communications (GSM) voice channel. It is based on the concept of ldquosymbolsrdquo-a set of predefined signals with finite bandwidths. Data are encoded into the symbols, and the symbols are voice coded as they were speech, modulated into the GSM signal, sent over the air, GSM demodulated, voice decoded, and converted back to data. The symbols are synthesized by a genetic algorithm with the aim of maintaining separability after passing them through the voice codec. This method enables data transfer over communication networks that do not have dedicated data channels and could also be used in conjunction with other data services to balance the system load between data and voice channels, allowing optimization of system resources. We present the full algorithmic structure of the system, which performs data communications over the GSM voice channel, and we also give the results of the performance tests.     

Operating aspects of new LRE/DSI DCMEs

Digital circuit multiplication equipment (DCME) that uses a combination of talk-spurt interpolation and low-rate encoding (32 kb/s ADPCM rather than the traditional 64 kb/s PCM) is about to be introduced in operating satellite systems. These DCMEs possess a number of important features that have been introduced to achieve a suitable balance between high channel multiplication ratios and high quality voice and data transmission. The methods used to accomplish this are generation of overload channels to virtually eliminate talk-spurt clipping, establishing a limit on the fraction of channels that undergo bit reduction by use of dynamic load control, automatic routeing of trunks carrying in-band data signals to non-interpolated bearer channels, use of a bit-bank approach to provide five-bits/sample ADPCM coding for in-band data channels and rotation of the server channels among the talk-spurts to spread the degradation caused by bit reduction uniformly across all talk-spurts. This paper analyses the performance of the DCME embodying the above features, and presents results in terms of the input-to-output channel multiplication ratio (CMR) as a function of the number of bearer channels (over a range from 4 to 61 bearer channels) and the number of input channels carrying in-band data. Extending the operation range down to as few as four bearer channels permits evaluation of the effectiveness of multiclique operation. In addition, the influence on CMR of (a) non-bit-bank operation, that would be possible if an ADPCM codec capable of supporting in-band data carriers at four-bits/sample were used, and (b) a short-hangover-time speech detector are examined. Use of hangover times less than 30 ms are shown to achieve a CMR of as great as 3.3: 1.     

Digital audio broadcasting: High-grade service quality through on-board processing techniques

A digital audio broadcasting (DAB) satellite system is presented here which provides a high-grade quality of service for some hundreds of channels. This, with minimum cost of the radio set and of the broadcasting station, greatly improves the radio broadcasting service throughout the world, especially in areas where deployment of conventional relay infrastructure is difficult and not yet cost-effective. Potential broadcasters located in the visible hemisphere can directly up-link their programming to the DAB satellites. Broadcasters can also copy each up-link channel and steer flexibility to any beam or combination of satellite beams. The paper illustrates the advantages and the feasibility of a DAB system based on geostationary satellites with on-board processing generating signals of MCPC (multiple channel per carrier) type. This multiplexing technique allows large system margins. This solution allows the satellite L-band TWTA amplifiers to be operated in saturation and eliminates the intermodulation noise associated with the transmission of FDMA channels. A powerful coding scheme has been selected to create a ‘robust’ down-link. In this way several high-power DAB channels of radio programming, with selectable data rate, are directly delivered to the users.     

Dispersion Signatures, a Statistically Based, Dynamic, Digital Microwave Radio System Measurement Technique

Outage in digital microwave radio systems during multipath fading is highly correlated with the channel amplitude dispersion. This characteristic is exploited in a new digital microwave radio system measurement technique which yields dispersion signatures. These dispersion signatures can be readily applied to assess the effects of time varying channels on digital radio equipment. Thus, dispersion signatures can be used for in-service monitoring of equipment operation during actual fading. Routine maintenance assessment and laboratory evaluation of digital microwave radio equipment can also be readily carried out. The laboratory evaluation results can be used for system design both with and without diversity.     

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.