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).     

光纤环形数模混合局域网

本文介绍一种价格低廉的数模混合光纤环形综合业务局域网。我们将统一的光纤网分解为数字数据传输子网和模拟电视传输子网两部分。数字子网速率为20Mb/s,采用报文分组交换和令牌环介质访问方式。模拟子网采用副载波多路复用传输多路电视信号和频道分配交换方式。模拟网的交换功能受数字网控制,使网络的构成简化并降低了成本。     

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     

Distributed source control: a network access control for integratedbroadband packet networks

The authors propose a congestion control strategy called distributed source control (DSC), designed to address the integration of diverse traffic types in broadband packet networks. DSC is a rate-based network access control implemented at the asynchronous transfer mode (ATM) layer as a feedforward control, in conjunction with an adaptive end-to-end control between network edges. The authors examine the performance of high-speed data traffic such as large file transfers, still images, and document retrievals. The authors study DSC's integrability with real-time traffic such as voice and video. With the help of a simulation model of a broadband packet network (150 Mb/s), the authors quantify the improvement in network performance due to DSC. An analytic model for an access node under DSC is developed, and guidelines for sizing of buffers in the switch and for choosing the end-to-end window size are provided     

Packet CDMA versus dynamic TDMA for multiple access in anintegrated voice/data PCN

A comparative evaluation of dynamic time-division multiple access (TDMA) and spread-spectrum packet code-division multiple access (CDMA) approaches to multiple access in an integrated voice/data personal communications network (PCN) environment are presented. After briefly outlining a cellular packet-switching architecture for voice/data PCN systems, dynamic TDMA and packet CDMA protocols appropriate for such traffic scenarios are described. Simulation-based network models which have been developed for performance evaluation of these competing access techniques are then outlined. These models are exercised with example integrated voice/data traffic models to obtain comparative system performance measures such as channel utilization, voice blocking probability, and data delay. Operating points based on typical performance constraints such as voice blocking probability 0.01 (for TDMA), voice packet loss rate 10^-3 (for CDMA), and data delay 250 ms are obtained, and results are presented     

Fuzzy/neural congestion control for integrated voice and dataDS-CDMA/FRMA cellular networks

The paper proposes congestion control using fuzzy/neural techniques for integrated voice and data direct-sequence code division multiple access/frame reservation multiple access (DS-CDMA/FRMA) cellular networks. The fuzzy/neural congestion controller is constituted by a pipeline recurrent neural network (PRNN) interference predictor, a fuzzy performance indicator, and a fuzzy/neural access probability controller. It regulates the traffic input to the integrated voice and data DS-CDMA/FRMA cellular system by determining proper access probabilities for users so that congestion can be avoided and the throughput can be maximized. Simulation results show that the DS-CDMA/FRMA fuzzy/neural congestion controllers perform better than conventional DS-CDMA/PRMA with channel access function in terms of voice packet dropping ratio, corruption ratio, and utilization. In addition, the neural congestion controller outperforms the fuzzy congestion controller     

无线侧提升VoLTE用户语音感知的优化策略研究

VoLTE是在4G网络全IP条件下的端到端语音解决方案,能提供更短的接入时延和更好的语音质量。在建立语音质差模型和分析空口丢包原理的基础上,梳理无线侧VoLTE语音感知的优化流程,对无线侧影响VoLTE语音感知的6个维度进行专项优化整治,实施效果较好,为今后VoLTE用户语音感知优化提供参考和指导。     

关于发展融合型接入网的探讨

接入网是电信网的重要组成部分。由于新技术的大量应用,使传统纯双绞线铜缆接入网逐渐被光纤接入网取代。目前的接入网主要是V5综合业务接入网,提供窄带综合业务。随着IP、多媒体业务的飞速增长,接入网进一步向宽带化和综合化方向发展,于是出现了宽窄带融合型接入网,这种网络基于统一平台提供传统/分组语音接入和宽窄带数据接入。文章针对接入网宽带化、综合化发展趋势,探讨了一种能提供宽窄带综合业务和向下一代网络(NGN)演进的融合型接入网。     

Broadband ISDN technology and architecture

Broadband integrated services digital network (BISDN) network architecture for local public serving areas is described. It is designed to offer end-users access to bandwidths up to 155 Mb/s. The BISDN protocols described support the simultaneous delivery of voice, data, image, and video services to business and residential customers. The objective of the broadband architecture described is to suggest designs that can evolve from today's base of narrowband equipment toward BISDN. It supports the use of fiber for narrowband services, in order to ensure that a broadband infrastructure is in place to serve as a platform from which broadband services can be offered. It also uses optional broadband modules on existing narrowband systems to facilitate evolutionary growth of broadband services and recognizes the metropolitan area network (MAN) protocol defined by IEEE 802.6 as a possible early implementation of BISDN     

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.     

Output and delay process analysis for slotted CDMA wirelesscommunication networks with integrated voice/data transmission

This paper presents the output and delay process analysis of integrated voice/data slotted code division multiple access (CDMA) network systems with random access protocol for packet radio communications. The system model consists of a finite number of users, and each user can be a source of both voice traffic and data traffic. The allocation of codes to voice calls is given priority over that to data packets, while an admission control, which restricts the maximum number of codes available to voice sources, is considered for voice traffic so as not to monopolize the resource. Such codes allocated exclusively to voice calls are called voice codes. In addition, the system monitoring can distinguish between silent and talkspurt periods of voice sources, so that users with data packets can use the voice codes for transmission if the voice sources are silent. A discrete-time Markov process is used to model the system operation, and an exact analysis is presented to derive the moment generating functions of the probability distributions for packet departures of both voice and data traffic and for the data packet delay. For some cases with different numbers of voice codes, numerical results display the correlation coefficient of the voice and data packet departures and the coefficient of variation of the data packet delay as well as average performance measures, such as the throughput, the average delay of data packets, and the average blocking probability of voice calls     

TRANSMISSION CAPABILITY OF DISCRETE MULTITONE MODULATION FOR TAIWAN’S SUBSCRIBER LOOPS

The transmission of 1.5 Mb/s and beyond data rate in the conventional twisted-pair local loops, called asymmetrical digital subscriber line (ADSL) technology, will be possible to create a new transport access capability to provide economic wideband voice/data/video integrated services directly to residential customers. The term ‘asymmetric’ in ADSL refers to the much higher data rate in the direction from central office (CO) to the customer and the lower rate of return (control) data from the customer to the CO. Discrete multitone (DMT) modulation has been selected as the modulation scheme in ADSL. Because the crosstalk of local loops in Taiwan is 10–15 dB worse than that in the US, the service capability of DMT ADSL for Taiwan’s local loops may be different from that in other countries, and thus needs to be evaluated. On the basis of the characteristics of Taiwan’s local loops, transmission capability is estimated to be 1.544 Mb/s and 6 Mb/s in Taiwan. Simulation results also show what percentage of users in Taiwan may have 1.544 Mb/s or 6 Mb/s of ADSL services. Far-end crosstalk (FEXT) and additive white Gaussian noise (AWGN) are considered to be the dominant noise sources in the work.     

FLC-Based MBWIMA/UMTS Protocol for Multimedia Services in UTRA TDD Mode

An intelligent medium access control (MAC) protocol based on fuzzy logic control (FLC) is proposed and compared with a general packet radio system in UMTS (GPRS/UMTS), priority scheme and the movable boundary wireless integrated multiple access in UMTS (MBWIMA/UMTS) protocols. The integrated video/voice/data services of UMTS in UTRA TDD mode have different transmission properties. By fuzzy logic control, the resources of the wireless communication can be intelligent assigned for different types of mediums. The voice-video dropping probability and data packet delay are input to FLC to optimally select the maximum number of voice/video slots. Voice activity detector (VAD) and multiple access interference in single cell are also considered in the simulations.     

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.     

Implementation of a broadband integrated services hybrid network

The issues involved in the design of a broadband integrated services digital network (B-ISDN), that utilizes optical fibers in the subscriber loop, are considered. A hybrid scheme that provides integrated access to ISDN and broadband services by delivering voice/data signals in a digital format while using an analog format for full-motion video services to subscribers is proposed. The discussion covers planning for a B-ISDN, the system concept, implementation, the loop fiber network, subscriber equipment, and cost comparison and analysis     

Dynamic Code Assignment Medium Access (DCAMA) Protocol for Wireless Integrated Services Networks

A medium access control (MAC) protocol for wireless mobile networks that supports integrated services and provides quality of service (QoS) support is presented and evaluated via simulation. A controlled random access protocol which allows all terminals to dynamically share a group of spread spectrum spreading codes is used. The protocol provides mobile terminals the access control required for efficient transfer of integrated traffic with QoS guarantees. Two service classes are provided; "best-effort" service, with priority queueing, and reserved bandwidth circuit service. The performance of the protocol is evaluated via simulation for traffic consisting of integrated voice, data and compressed video. The performance assessment measure is packet delay.     

Digital cellular-its roots and its future

Cellular radio started in the early 1980s by using analog technologies. Research in voice coding, modulation, and channel coding resulted in second-generation cellular radio based on digital technologies, which were introduced in the early 1990s. These were all based on advanced time division multiple access technology, resulting in better capacity and lower cost. Today, these digital technologies-based on the Global System for Mobility, digital Advanced Mobile Phone System, and personal digital cellular-have more than 100 million subscribers worldwide. The next step is to introduce wide-band packet services for wireless Internet up to 2 Mb/s. These so-called third-generation systems (Universal Mobile Telecommunications Services, IMT2000) are planned to be introduced in the early 2000s     

A novel self‐adaptive transmission scheme over an IEEE 802.11 WLAN for supporting multi‐service

The IEEE 802.11 wireless local area network (WLAN) media access control (MAC) specification is a hybrid protocol of random access and polling when both distributed coordination function (DCF) and point coordination function (PCF) are used. Data traffic is transmitted with the DCF, while voice transmission is carried out with the PCF. Based on the performance analysis of the MAC protocol for integrated data and voice transmission by simulation, this paper puts forward a self‐adaptive transmission scheme to support multi‐service over the IEEE 802.11 WLAN. The simulation results show that, on the premise of satisfying the maximum allowable delay of packet voice, the self‐adaptive transmission scheme can improve the data traffic performance and increase the WLAN capacity through dynamic and appropriate adjustment of the protocol parameters. Especially, voice traffic is sensitive to delay jitter, and the self‐adaptive scheme can effectively decrease it. Finally, it is worth noting that the adaptive scheme is easy to be realized, whereas no change in the MAC protocol is needed. Copyright © 2006 John Wiley & Sons, Ltd.     

Admission policies for integrated voice and data traffic in CDMApacket radio networks

The authors derive optimal admission policies for integrated voice and data traffic in packet radio networks employing code division multiple access (CDMA) with direct-sequence spread spectrum (DS/SS) signaling. The network performance is measured in terms of the average blocking probability of voice calls and the average delay and packet loss probability of data messages. The admission scheme determines the number of newly arrived voice users that are accepted in the network so that the long-term blocking probability of voice calls is minimized. In addition, new data arrivals are rejected if the mean delay or the packet loss probability of data exceeds a desirable prespecified level. A semi-Markov decision process (SMDP) is used to model the system operation. Then, a value iteration algorithm is used to derive the optimal admission control. Two models for the other-user interference of the CDMA system are considered: one based on thresholds and another based on the graceful degradation of the CDMA system performance, and their performance is compared. These admission policies find application in emerging commercial CDMA packet radio networks including cellular networks, personal communication networks, and networks of LEO satellites for global communications     

Terminal‐Assisted Hybrid MAC Protocol for Differentiated QoS Guarantee in TDMA‐Based Broadband Access Networks

This paper presents a terminal‐assisted frame‐based packet reservation multiple access (TAF‐PRMA) protocol, which optimizes random access control between heterogeneous traffic aiming at more efficient voice/data integrated services in dynamic reservation TDMA‐based broadband access networks. In order to achieve a differentiated quality‐of‐service (QoS) guarantee for individual service plus maximal system resource utilization, TAF‐PRMA independently controls the random access parameters such as the lengths of the access regions dedicated to respective service traffic and the corresponding permission probabilities, on a frame‐by‐frame basis. In addition, we have adopted a terminal‐assisted random access mechanism where the voice terminal readjusts a global permission probability from the central controller in order to handle the ‘fair access’ issue resulting from distributed queuing problems inherent in the access network. Our extensive simulation results indicate that TAF‐PRMA achieves significant improvements in terms of voice capacity, delay, and fairness over most of the existing medium access control (MAC) schemes for integrated services.