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.     

Wideband packet technology in the integrated access and cross-connect system (IACS)

The integrated access and cross-connect system (IACS) is an AT&T product based on the wideband packet technology. The IACS has two components: the integrated access terminal (IAT) and the integrated access controller (IAC). The IAC provides centralized control, operations and maintenance for a network of IATs. An IAT provides direct digital connectivity at 1·536 Mb/s or 2·048 Mb/s to customer terminals and to circuit- and packet-switched networks. It integrates voice, voiceband data, facsimile traffic, digital data, signalling, image and network control into packets of common format. It allows networking in both the circuit and packet domains, offers bandwidth on demand, and achieves graceful degradation of voice quality during congestion.     

Efficiency of packet voice with deterministic delay

Packet switching is appealing for carrying real-time traffic because it can benefit from (possibly variable bit rate) compression schemes and statistical multiplexing to more efficiently exploit network resources. This work explores the efficiency of IP telephony in terms of the volume of voice traffic carried with deterministically guaranteed quality related to the amount of network resources used. An IP network carrying compressed voice is compared to circuit switching carrying PCM (64 kb/s) encoded voice, and some design choices affecting IP telephony efficiency are discussed     

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。     

Advanced techniques for circuit multiplication: Celtic 3G

The Celtic 3G is Alcatel CIT's new digital circuit multiplication system designed to enhance the cost-effectiveness of long-haul links by increasing their capacity by a factor of between 5 and 8. The Celtic 3G offers an effective, versatile solution to congestion problems on 2048 and 1544 kb/s digital transmission links. The paper describes the compression modes used by Celtic 3G family to transmit the various types of signals (speech, voice band data, 64 kb/s digital data), the applications and the operating principles of the equipment. A number of telecommunication administrations and agencies have adopted and will this year deploy the Celtic 3G because of its benefits, technical qualities and the savings which it will achieve on digital links.     

Ku-band satellite digital transmission systems

A nation-wide data communications service is being placed in service by a major telecommunications carrier. This service will rely on a satellite network being provided and installed by Harris Satellite Communications Corporation. The network supports information data rates from 56 kb/s to 2·048 Mb/s. This data communications network has been architectured to include a number of innovative monitor and control functions, up-link power control, modular equipment shelter design and a modular antenna/feed subsystem. This paper will detail this application.     

An efficient traffic control scheme for store‐and‐forward video‐on‐demand services

According to the MPEG‐1/2 standard, full motion video can be compressed and stored in an information warehouse. Multiplexing with normal voice calls, it is retrieved and delivered to the customer's local BISDN central office via 155 Mb/s trunks. These voice calls have higher priority than video‐on‐demand (VOD) so that normal voice services will not be influenced by VOD transmission. The number of voice calls always fluctuates in real‐time. Thus, an efficient traffic control scheme is highly desired to guarantee a given level of performance and achieve as great as possible use of available bandwidth. Based on the fact that the number of real‐time voice calls is variable, and the use of available capacity is less than 100 per cent, this paper will present an efficient traffic control scheme for store‐and‐forward VOD services, and demonstrate the scheme by various simulation results. Copyright © 2001 John Wiley & Sons, Ltd.     

Traffic studies of a multiplexer in an ATM network and applications to the future broadband ISDN

In the first part of this paper the influence of the arrival process on these three performance parameters is discussed. The values of these three performance parameters depend strongly on the utilization distribution over the input trunks of the multiplexer while the total utilization on the input trunks is fixed. If there is the same utilization on all input trunks, the values of these three performance paramters reach a maximum. This result is also valid for an ATM switching matrix with a separate FIFO buffer at each output trunk. In the second part of this paper some examples are provided of how to apply the results of these studies to the future broadband ISDN which is intended to transmit services with very high bit rates (e.g. 135 Mb/s) as well as services with very low bit rates (e.g. 64 kb/s).     

Application of 32 and 16 kb/s speech encoding techniques to digital satellite communications

This paper describes the performance of various voice encoding techniques at 32 and 16 kb/s for applying to digital satellite communication systems. The subjective performances of adaptive differential PCM (ADPCM), adaptive predictive coding (APC), subband coding (SBC) and adaptive delta modulation (ADM) are compared under various satellite channel environments, that is, random and burst channel errors in satellite link and an ambient noise in the ship-to-shore direction in a maritime satellite channel. The performance of the voiceband data at 4·8 and 2·4 kb/s is also evaluated for these coders. ADPCM encoding at 32 kb/s is very attractive for conventional fixed satellite systems, keeping the equivalent quality to 64 kb/s PCM. On the other hand, APC encoding at 16 kb/s is also most suitable for maritime satellite communication systems at the sacrifice of a small degradation of speech quality.     

IsoEthernet: an integrated services LAN

IsoEthernet is an integrated services LAN directed at multimedia applications. It enables the carrying of 6.144 Mb/s of isochronous data (96 B-channels at 64 kb/s each) in addition to 19 Mb/s of 10Base-T traffic using the existing 10Base-T wiring infrastructure. A 10Base-T mode of operation accommodates existing 10Base-T equipment. Interoperability with 10Base-T networks is provided for packet traffic. IsoEthernet is being standardized by the IEEE 802.9 standards committee     

16 Kb/s high quality voice encoding for satellite communication networks

This paper describes applications of adaptive predictive coding (APC) with maximum likelihood quantization (MLQ) which can cover a wide range of coding rates from 4.8 to 16 kb/s for low C/N satellite communication systems, such as maritime, aeronautical mobile and thin-route satellite communication systems, and also for speech and data integration, including digital circuit multiplication equipment (DCME) in business communication systems, such as INTELSAT business services (IBS). A 16 kb/s APC–MLQ hardware codec has been implemented by NEC–7720 DSP chips and the performance has been confirmed in subjective quality of speech through conversational tests. The objective performance has also been evaluated for non-voice signals, such as single and multi-frequency tones, and 1200 and 2400 b/s voiceband data signals. The APC-MLQ codec can transmit the voice-band data at 1200 b/s over two asynchronous tandem links and at 2400 b/s over one link. It was noted that the APC-MLQ codec is superior in speech performance at 16 kb/s to a narrow-band companded FM and meets requirements for low C/N satellite communication systems. For voice and data integration into 16 kb/s for 64 kb/s links, we propose a multi-media multiplexing for low C/N digital satellite communication systems and also a small-scale circuit multiplication system for business use. In these systems, a variable rate coding of APC-MLQ from 4.8 to 16 kb/s can be effectively introduced for voice and data integration.     

Telematic services via satellite in Europe

The word ‘telematics’ was used for the first time in France in 1978 by Simon Nora and Alain Minc; it is a contraction of the French words ‘tétécommunications’ and ‘informatique’, and defines a marriage between computers and data transmission networks. This paper describes the five systems which will be implemented to provide telematic services via satellite in Europe during the next few years: EUTELSAT, TELECOM 1, UNISAT, DFS and ITALSAT. The main characteristics of the ‘special service’ transponders are presented, showing that, in Europe, (a) 2 to 5 transponders per satellite will be devoted to the special (telematic) services; (b) 4 satellites out of the 5 use the 12.5–12.75 GHz down-link band for this purpose (only ITALSAT uses the 20 GHz down-link for the same purpose), and (c) e.i.r.p.s at 12 GHz are in the range 40–49 dBW. The ground and space segments corresponding to this new service are presented. The main peculiarity of the ground segment is the exclusive use of ‘small’ earth-stations equipped with fixed antennas and SCPC or 25–60 Mb/s TDMA multiple access equipment. Telematic services include videoconference at 2048 kb/s, voice services, computer-to-computer transfer of data, data distribution, remote printing, electronic mail, high-speed facsimile, offshore voice and data services, digital network overflow, transatlantic digital business etc… Customer bit rates start from 2.4, 4.8, 9.6 and 32 kb/s (for TELECOM 1) up to 2048 kb/s by 64 kb/s steps. Bit error rates are maintained below 10^?6 (standard) or 10^?10 (higher grade) during 99 per cent of the year using FEC. The five networks for telematics via satellite in Europe are independent. A large efficiency improvement of the different services should be obtained through a future interconnection of the various networks via intersatellite links.     

Beyond 3G: wideband wireless data access based on OFDM and dynamicpacket assignment

The rapid growth of wireless voice subscribers, the growth of the Internet, and the increasing use of portable computing devices suggest that wireless Internet access will rise rapidly over the next few years. Rapid progress in digital and RF technology is making possible highly compact and integrated terminal devices, and the introduction of sophisticated wireless data software is making wireless Internet access more user-friendly and providing more value. Transmission rates are currently only about 10 kb/s for large cell systems. Third-generation wireless access such as WCDMA and the evolution of second-generation systems such as TDMA IS-136+, EDGE, and CDMA IS-95 will provide nominal bit rates of 50-384 kb/s in macrocellular systems. This article discusses packet data transmission rates of 2-5 Mb/s in macrocellular environments and up to 10 Mb/s in microcellular and indoor environments as a complementary service to evolving second- and third-generation wireless systems. Dynamic packet assignment for high-efficiency resource management and packet admission; OFDM at the physical layer with interference suppression, space-time coding, and frequency diversity; as well as smart antennas to obtain good power and spectral efficiency are discussed in this proposal. Flexible allocation of both large and small resources also permits provisioning of services for different delay and throughput requirements     

MPEG-4标准起草工作近况

本文简略报道国际活动图像专家组近年对多媒体通信草拟新标准MPEG 4的进展情况。其中音频编码包括语音、音乐 (自然的和合成的 ) ,比特率从 2至 6 4kb /s。视频编码包括甚低比特率 5～ 6 4kb /s和较高比特率 6 4kb /s至 2Mb /s。视频编码可将图像中每一对象分开编码为不同的比特流层 ,又可操纵对象的尺度、位置等 ,具有以内容为基础的交互式功能。除了核心编码器外 ,对输入视频序列的每一帧分成若干个任意形状的“视频对象平面” ,编码成各个分开的“视频对象层”。另外 ,利用“子图形”编码技术 ,将图像的背景以及前景中的每一对象分开编码成视频序列后传输 ,可以改善视像质量。     

语音数字信号压缩技术在TES系统中的应用

介绍ADPCM标准、RLPC编码原理,编、解码器方框图及工作过程。民航卫星通信网TES系统为节省卫星转发器频率资源对传输的语音信号进行压缩处理,其信道单元基带信号处理器对语音信号进行CCITT推荐的G.721-ADPCM编码和修斯公司专利技术开发的RLPC编码处理,将64 kb/s语音数字信号压缩至32 kb/s,16 kb/s,9.6 kb/s传输,实现语音质量满足一般通信要求的低速率语音信号传输。     

France telecom's introduction of switched broadband business services

In order to cope with the growth of video transmissions in France, France Telecom will commercially introduce switched broadband business services by June 1990. A broadband network based on a meshed network of optical fibres will provide one-way point-to-multipoint (including point-to-point) connections, available on reservation. This network will link non-blocking 32 × 32 or 64 × 64 space switches that can cross-connect different signals, including analog SPFM (square pulse frequency modulation) video, HDB3 34 Mb/s and CMI 140 Mb/s digital signals.     

An algorithm for sizing trunks carrying mixtures of constant and variable bit rate traffic

As new broadband networks based on the asynchronous transfer mode (ATM) protocol emerge to support a variety of services, the need for analytical models and tools for sizing network components carrying mixtures of different kinds of traffic will be increasingly felt. This paper presents a simple, yet efficient algorithm for sizing trunks carrying mixtures of constant and variable bit rate traffic. It combines theoretical results for the multi-dimensional Erlang model and the concept of equivalent bandwidth with some heuristic arguments to produce an analytic method for sizing trunks subject to blocking and quality of service constraints for traffic classes.     

无比特丢弃的AAL2分组话音系统最优分组长度的研究

该文根据分组话音业务的特点,结合分组话音业务服务质量要求,特别是分组丢弃概率和端到端分组传送时延的要求,研究AAL2分组话音系统中AAL2分组最优长度的确定方法,得出结论:对于无比特丢弃的AAL2分组话音系统,当话音采用32kb/s的编码时,AAL2分组的最优长度大约为31个字节;当话音采用16kb/s的编码时,AAL2分组的最优长度大约为27个字节。此时AAL2分组的分组头开销小,话音分组的丢弃概率和端到端分组传送时延低,所得的分组话音质量高。     

Expérimentation sur site ďun système de multiplication de circuits par mise en paquets

This article describes the main results of the field trial conducted by France Telecom andatt of theiacs. This packetized circuit multiplication equipment has been submitted to the following tests: subjective evaluation of speech (subscribers interview), performances of voice band data transmission, performances with a bit error rate on the bearer, test of the fax demodulation process.     

Integrated video and voice transmission using packet reservation multiple access

This paper proposes a new protocol for the integration of voice and video transmission over the packet reservation multiple access (PRMA) system that is a modification of reservation‐ALOHA protocol. We focus on low bit‐rate video applications like video conferencing and visual telephony for wireless communications. The ITU–T H.263 standard provides a solution to the need for low bit‐rate video compression under 64 kbytes/s. The proposed protocol assumes that each voice terminal follows a traffic pattern of talk spurts and silent gaps with fixed permission probability (p=0.3), and each video terminal has the higher permission probability (p=1) to access the available slot based on ITU–T H.263 standard. Again, we present a ‘pseudo‐reservation’ scheme to release slots reserved by video terminals according to the contents of each video transmission buffer, and the active voice terminals can temporarily access the additional slots to improve the performance without sacrificing the video capacity of the system. The packet dropping probability of the active voice terminals and bandwidth utilization of the system are superior to the original PRMA, as indicated in simulation results. Copyright © 2001 John Wiley & Sons, Ltd.