一种动态利用码资源的CDMA系统性能研究

本文提出了一种基于随机预约ＡＬＯＨＡ访问方式,能支持话音和数据业务的动态使用码资源的码分多址访问协议。该协议中,话音终端采用预约请求排队访问方式。数据终端采用时隙ＡＬＯＨＡ方式传输数据分组。理论分析和计算机仿真结果表明,该协议能有效地提高系统码资源的利用率。在系统处于重负载情况下该协议能优先保证话音业务服务质量,而处于轻负载下系统码资源能力数据业务充分使用。     

具有多重码的预约Slotted ALOHA协议性能分析

本文对具有多重码的预约Slotted ALOHA协议进行了性能分析。它是基于帧的协议且在一个时隙内有多重码可用来传输分组信息。文中利用离散时间,离散状态和Markov链来分析一个小区上行链路的话音分组吞吐量和数据分组延时性能,并提出了两种改进碰撞解决的方法,得到了较好的系统特性。     

MC-CDMA ALOHA系统的性能分析

文章提出了一种多码CDMA ALOHA(MC-CDMA ALOHA)系统用以支持多速率数据通信，分析研究了系统在高斯噪声信道的吞吐量性能，得出了在结合ALOHA和多码CDMA两种技术的优势的情况下，多码CDMA ALOHA系统能支持多速率数据业务并具有较好的性能表现的结论。     

CSMA的仿真与分析

本文章描述对1-坚持型CSM(A载波侦听多址访问协议)网络和纯ALOHA网络分别建立仿真模型并对吞吐量和信道负载进行性能分析。在OPNET仿真工具下进行仿真建模,改变节点数目后再不断进行仿真,完成对1-坚持型CSMA协议较纯ALOHA协议的优越性的研究。主要分析了在不同协议下,不同节点数网络中CSMA较ALOHA协议在吞吐量以及稳定性方面的优势。最后仿真证明了CSMA协议在不同的业务量下,吞吐量均比ALOHA协议高。     

移动通信系统中一种改进的综合业务接入控制方案

提出了一种适用于多码CDMA S-ALOHA系统,以各级数据传输概率为控制因子的综合业务接入控制方案。仿真结果表明,方案可使话音掉线率明显改善,并且即使在信道负载很重时也能将数据业务吞吐量保持在峰值附近。     

基于HFC网络上信道CDAM—预约ALOHA多接入方式吞吐量分析

为了在HFC网络上实现交互式业务,人们对基于HFC网络的介质访问协议进行了广泛的研究,已出现的几种基于HFC的MAC协议基本上都采用了竞争与预约相结合的访问机制,基于CDMA的HFC是新一代的HFC系统。本文提出了一种基于CDMA的双向HFC网络竞争与预约相结合的上行信道多址接入方式,并且为该协议系统建立了四状态Markov链分析模型,对该协议进行了性能分析,并给出了计算机实验结果与分析。     

时隙ALOHA DS/CDMA系统与多载波时隙ALOHA系统吞吐性能的比较研究

该文对同样带宽下时隙 ALOHA DS/CDMA系统和多载波时隙 ALOHA系统的吞吐量进行了理论计算、比较和仿真。结果表明,在总负载较大时,采用高纠错能力的时隙 ALOHA DS/CDMA系统可以在吞吐量上有更好的性能。但若网络负载过重,时隙ALOHA DS/CDMA系统的吞吐性能较多载波时隙ALOHA的系统下降快;码字总数受限会带来码字选择的冲突,从而降低系统的吞吐性能。     

基于HFC网络上行信道CDMA-预约ALOHA多接入方式吞吐量分析

为了在HFC网络上实现交互式业务,人们对基于HFC网络的介质访问协议进行了广泛的研究.已出现的几种基于HFC的MAC协议基本上都采用了竞争与预约相结合的访问机制.基于CDMA的HFC是新一代的HFC系统.本文提出了一种基于CDMA的双向HFC网络竞争与预约相结合的上行信道多址接入方式.并且为该协议系统建立了四状态Markov链分析模型,对该协议进行了性能分析,并给出了计算机实验结果与分析.     

分组无线网络信道接人协议ALOHA性能仿真研究

文章分析了分组无线网络信道接入协议ALOHA（夏威夷加性在线链路系统）的工作原理及其性能。在性能分析中,主要就ALOHA系统的吞吐量和网络负载两个性能指标进行了分析研究。在此基础上给出了ALOHA系统的仿真流程．并就其中的帧到达时间和随机时延的选定进行了阐述．仿真结果表明采用文章提出的仿真模型得到的仿真结果与理论结果基...     

话音/数据综合的多时隙预约多址协议及其性能分析

本文提出了一种综合话音和数据的多时隙预约多址协议.该协议在保证话音终端的优先权的情况下,允许数据终端在报文的传输期间在连续多个帧中预约多个信息时隙.文中对协议进行了理论分析,并推导出了协议的重要性能指标(如话音分组丢失率、数据报文平均接入时延、系统平均吞吐率等)的解析表达式.研究表明,该协议可以支持比IPRMA、NC-IPRMA更高的等效数据终端速率,而且系统平均吞吐率在很大的负载范围内接近最大值.     

A reservation random-access protocol for voice/data integratedspread-spectrum multiple-access systems

Most code-division multiple-access (CDMA) systems described in the literature provide only one single service (voice or data) and employ the strategy of “one-code-for-one-terminal” for code-assignment. This assignment, though simple, fails to efficiently exploit the limited code resource encountered in practical situations. We present a new protocol called reservation-code multiple-access (RCMA), which allows all terminals to share a group of spreading codes on a contention basis and facilitates introducing voice/data integrated services into spread-spectrum systems. The RCMA protocol can be applied to short-range radio networks, and microcell mobile communications, and can be easily extended to wide area networks if the code-reuse technique is employed. In RCMA, a voice terminal can reserve a spreading code to transmit a multipacket talkspurt while a data terminal has to contend for a code for each packet transmission. The voice terminal will drop a long delayed packet while the data terminal just keeps it in the buffer. Therefore, two performance measures used to assess the proposed protocol are the voice packet dropping probability and the data packet average delay. Theoretical performance is derived by means of equilibrium point analysis (EPA) and is examined by extensive computer simulation     

A Medium Access Control Protocol for Voice/Data Integrated Wireless CDMA Systems

In this paper, a medium access control protocol is proposed for integrated voice and data services in wireless local networks. Uplink channels for the proposed protocol are composed of time slots with multiple spreading codes per slot based on slotted code division multiple access (CDMA) systems. The proposed protocol uses spreading code sensing and reservation schemes. This protocol gives higher access priority to delay‐sensitive voice traffic than to data traffic. The voice terminal reserves an available spreading code to transmit multiple voice packets during a talkspurt. On the other hand, the data terminal transmits a packet without making a reservation over one of the available spreading codes that are not used by voice terminals. In this protocol, voice packets do not come into collision with data packets. The numerical results show that this protocol can increase the system capacity for voice service by applying the reservation scheme. The performance for data traffic will decrease in the case of high voice traffic load because of its low access priority. But it shows that the data traffic performance can be increased in proportion to the number of spreading codes.     

Design and performance evaluation of an RRA scheme for voice‐data channel access in outdoor microcellular environments

In PCS networks, the multiple access problem is characterized by spatially dispersed mobile source terminals sharing a radio channel connected to a fixed base station. In this paper, we design and evaluate a reservation random access (RRA) scheme that multiplexes voice traffic at the talkspurt level to efficiently integrate voice and data traffic in outdoor microcellular environments. The scheme involves partitioning the time frame into two request intervals (voice and data) and an information interval. Thus, any potential performance degradation caused by voice and data terminals competing for channel access is eliminated. We consider three random access algorithms for the transmission of voice request packets and one for the transmission of data request packets. We formulate an approximate Markov model and present analytical results for the steady state voice packet dropping probability, mean voice access delay and voice throughput. Simulations are used to investigate the steady state voice packet dropping distribution per talkspurt, and to illustrate preliminary voice-data integration considerations. This revised version was published online in June 2006 with corrections to the Cover Date.     

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     

Spreading code protocols for distributed spread-spectrum packetradio networks

Spreading code protocols for a distributed spread-spectrum packet radio network are presented. A distributed single-hop system (i.e. each terminal can hear all other terminals) with the users approximately synchronized and a set of prespecified spreading codes are presented. The spreading code protocol is a policy for choosing a spreading code to be used, given that a terminal has a packet to send, and a policy for monitoring spreading codes, given that a terminal is idle. A slotted system where a packet occupies a number of slots is considered, and two protocols that involve changing the spreading code of a transmission after an initial header is transmitted are presented. In one protocol, the header is transmitted on a common code, and in the other it is transmitted on a receiver-based code, the rest of the packet being transmitted on a transmitter-based code. In the receiving mode, a terminal monitors either a common code, in the first case, or a receiver-based code in the latter. Upon recognizing its own address and the source address, the receiver dynamically switches to a despreading code corresponding to the source. Throughput results are obtained for the case of geometrically distributed packet lengths     

A New Performance Measurement for Voice Transmission in Burst and Packet Switching

Voice transmission in burst switching is characterized by the process of talkspurt clipping, while in packet switching, it is characterized by the process of packet delay. In most analyses, the talkspurt clipping has been measured by the clipping probability averaged over all bits, and the packet delay has been measured by the delay performance averaged over all packets. The resulting measures overlook the duration of clipping in a talkspurt and the significant difference of delay in packets arriving at different times. Because of the nature of voice, different effects of these may result in substantially different degrees of voice distortion. This paper studies the worst case performance of both processes. The voice traffic is modeled as a process alternating between overload and underload periods. Statistically, more clipping and delay will be incurred while in the overload period. By worst case we mean that, in burst switching, we measure the worst case of talkspurt clipping duration in an overload period, while in packet switching, we measure the worst case of packet delay in an overload period. Furthermore, a simple closed form equation is derived which gives a very good approximation of the worst case mean packet delay performance. This equation can be more generally applied when the packet service time is to be geometrically distributed or when voice and data are to be integrated. The voice performances in burst switching and packet switching are also compared.     

Performance analysis of an enhanced DQRUMA/MC-CDMA protocol for voice traffic

The original distributed-queueing request update multiple-access (DQRUMA)/multicode code-division multiple-access (MC-CDMA) protocol was developed as a channel access protocol for wireless packet CDMA networks. This protocol has recently attracted considerable attention. We modify the original protocol, which was designed for data traffic only, to additionally accommodate voice traffic and call it the A-Protocol. We propose a new packet CDMA protocol that enhances the A-Protocol by improving the utilization of receivers in a base station and call it the E-Protocol. In the E-Protocol, an access request is attempted with a randomly chosen code at a request minislot. We analytically evaluate the performance of both protocols and compare analytical results with computer simulation. Analytical results agree well with simulation results.     

Voice and data integration in the air-interface of a microcellularmobile communication system

A multiple-access protocol and a call acceptance algorithm for voice and data integration in a microcellular mobile communication system are presented. The protocol supports circuit-mode voice, burst-mode voice, and data. A hybrid multiplexing scheme with no boundaries performs statistical multiplexing, the call-level (for circuit-mode voice) and the talkspurt/message-level (for burst-mode voice and data). This scheme achieves high utilization of the available bandwidth compared to pure circuit switching, but with a lower quality in the latter two classes, due to delay during channel access on each talkspurt/message. A two-party transaction model for each class is implemented, giving a realistic load on uplink and downlink. A unified access procedure is presented, and the structure of the required control bursts is described. Performance is analyzed using simulation, and the optimum data-segment size is obtained. The maximum acceptable load is determined for various traffic mixes. A call acceptance algorithm is implemented, and typical simulation results for delay and call blocking are given     

An approximate analysis of TDMA out-of-slot random access protocols for microcellular mobile communications

In this paper, out-of-slot random access protocols for voice services that operate in microcellular environment are studied and simulated. The bearer service is assumed to be structured as time division multiple access/frequency division multiple access/frequency division duplex (TDMA/ FDMA/FDD). According to a stratification of information flow ascall, talkspurt, andpacket, the protocols are implemented at the talkspurt level. During a call, talkspurts generate a stream of packets. Each talkspurt has to reserve a voice time slot with a special control packet sent in a dedicate control slot (out of slot signaling). After a successful access, a voice slot is assigned for the duration of the talkspurt. This work concentrates on the out of slot random access method. When a transition from the idle state to the active state occurs, a voice terminal starts generating a talkspurt. Access for a voice slotV is then initiated via a dedicated control slotC. The time spent in gaining aV slot depends on the kind of random access protocol used in theC slots. Once the access reservation phase is successful, the talkspurt starts the second phase of information transmission in a freeV slot. If allV slots are occupied by other talkspurts, the new talkspurt is queued until aV slot becomes free. If the sum of the access and queueing times exceeds a thresh-old, a portion of the talkspurt is clipped. In our work we define an analytical model to evaluate the percentage of clipped voice packets. Simulations validate the analytical model.The second version of this work was rewritten while the author was a visiting scholar at WINLABThe IS-54 standard itself has the TDMA/FDMA structure. The ETDMA enhancement appears to be very much like what is described in this paper.     

Approximate performance analysis of heavily loaded slotted downlink channel in a wireless CDMA system supporting integrated voice/data services

This paper is concerned with the performance analysis of a slotted downlink channel in a wireless code division multiple access (CDMA) communication system with integrated packet voice/data transmission. The system model consists of a base station (BS) and mobile terminals (MT), each of which is able to receive voice and/or data packets. Packets of accepted voice calls are transmitted immediately while accepted multipacket data messages are initially buffered in first in, first out (FIFO) queues created separately for each destination. The BS distinguishes between silence and talkspurt periods of voice sources, so that packets of accepted data messages can use their own codes for transmission during silent time slots. To fulfill QoS requirements for both traffic types, the number of simultaneous packet transmissions over the downlink channel must be limited. To perform this task, a fair, single-priority multiqueueing scheduling scheme is employed. Discrete-time Markov processes are used to model the system operation. Statistical dependence between queues is the main difficulty which arises during the analysis. This dependence leads to serious computational complexity. The aim of this paper is to present an approximate analytical method which enables one to evaluate system performance despite the dependence. Therefore, it is assumed that the system is heavily loaded with data traffic, and a heuristic assumption is made that makes the queueing analysis computationally tractable. Typical system performance measures (i.e., the data message blocking probability, the average data throughput and delay) are evaluated, however, due to the accepted heuristic assumption, the analysis is approximate and that is why computer simulation is used to validate it.