Performance analysis of voice/data cellular CDMA with SIR-based admission control

In this paper, we analyze the performance of a signal-to-interference ratio (SIR)-based admission control strategy on the uplink in cellular code-division multiple-access (CDMA) systems with voice and data traffic. Most studies in the current literature to estimate CDMA system capacity with both voice and data traffic do not take into account admission control based on SIR constraints. Here, we present an analytical approach to evaluate the outage probability for voice traffic, the average system throughput, and the mean delay for data traffic in a voice/data CDMA system, which employs an SIR-based admission control. We make two main approximations in the voice call outage analysis-one based on the central limit theorem (CLT) and the other based on the Fenton's method. We apply the Fenton's method approximation to compute the retransmission probability and the mean delay for data traffic, and the average system throughput. We show that for a voice-only system, a capacity improvement of about 30% is achieved with the SIR-based admission control as compared with the code availability-based admission control. For a mixed voice/data system with 10 Erlangs of voice traffic, an improvement of about 40% in the mean delay for data is shown to be achieved. Also, for a mean delay of 50 ms with 10 Erlangs of voice traffic, the data Erlang capacity improves by about 50%.     

An uplink CDMA system architecture with diverse QoS guarantees forheterogeneous traffic

The uplink access control problems for cellular code-division multiple-access (CDMA) systems that service heterogeneous traffic with various types of quality-of-service (QoS) and use multicode CDMA to support variable bit rates are addressed. Considering its distinct QoS requirements, class-I real-time traffic (e.g., voice and video) is differentiated from class-II non-real-time traffic (e.g., data). Connection-oriented transmission is achieved by assigning mobile-oriented code channels for class-I traffic, where each corresponding mobile needs to pass an admission test. Class-II traffic is transmitted in a best-effort manner through a transmission-rate request access scheme which utilizes the bandwidth left unused by class-I traffic. Whenever a mobile has class-II messages to transmit, the mobile requests code channels via a base station-oriented transmission-request code channel, then, according to the base station scheduling, the transmission is scheduled and permitted. Addressed are the admission test for class-I connections, transmission power allocation, and how to maximize the aggregate throughput for class-II traffic. The admission region of voice and video connections and the optimum target signal-to-interference ratio of class-II traffic are derived numerically. The performance of class-II traffic transmissions in terms of average delay is also evaluated and discussed     

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     

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     

Integrating data traffic into a CDMA cellular voice system

Part one of this paper analyzes the effects of data traffic integration into a CDMA cellular voice system. The figure of merit used for the quality of service seen by the voice users is measured by the probability of blocking. The CDMA system under consideration is a power controlled, cellular architecture in which blocking occurs when the total interference level exceeds the background noise level by 10 dB [1]. It is shown that the introduction of data can be done at little or no increase in the probability of blocking on the voice users. In part two we propose and analyze a protocol which achieves the efficient integration of data by maximizing the utilization of the resources and minimizing the delay experienced by the voice users. The proposed protocol admits data traffic into the CDMA cellular system based on the current aggregate voice interference level, and allows for the efficient integration of voice and data without degrading the quality of service for the delay-critical voice traffic. A Markovian model for this protocol is developed, evaluated and compared to computer simulation results.     

非完备功控CDMA无线网络上行准入控制算法及性能分析

本文研究了非完备功控宽带CDMA无线网络上行准入控制问题。首先为非完备功控CDMA网络上行链路建立了系统模型，然后设计相应的准入控制算法，其中无线资源被划分为供各类用户专用的部分和在各类用户之间共享的部分。为了利用非实时用户可以容忍延时的特点，又引入了队列结构以降低阻塞率。通过对算法进行Markov分析，从理论上计算了用户阻塞率和平均吞吐率性能。仿真结果验证了理论模型的正确性。     

Link layer retransmission schemes for circuit-mode data over the CDMA physical channel

In the last few years, wide-area data services over North American digital (TDMA and CDMA) cellular networks have been standardized. The standards were developed under three primary constraints: (i) compatibility with existing land-line standards and systems, (ii) compatibility with existing cellular physical layer standards that are optimized for voice, and (iii) market demands for quick solutions. In particular, the IS-95 CDMA air interface standard permits multiplexing of primary traffic (e.g., voice or circuit data) and secondary traffic (e.g., packet data) or in-band signaling within the same physical layer burst. In this paper, we describe two radio link protocols for circuit-mode data over IS-95. The first protocol, Protocol S, relies on a single level of recovery and uses a flexible segmentation and recovery (FSAR) sublayer to efficiently pack data frames into multiplexed physical layer bursts. We next describe Protocol T, that consists of two levels of recovery. Protocol T has been standardized for CDMA circuit-mode data as IS-99 (Telecommunications Industry Association, 1994). We provide performance comparisons of the two protocols in terms of throughput, delay and recovery from fades. We find that the complexity of the two level recovery mechanism can buy higher throughput through the reduced retransmission data unit size. However, the choice of TCP (and its associated congestion control mechanism) as the upper layer of recovery on the link layer, leads to long fade recovery times for Protocol T. The two approaches also have significant differences with respect to procedures and performance at handoff and connection establishment. This revised version was published online in June 2006 with corrections to the Cover Date.     

Adaptive S-ALOHA CDMA as an alternative way of integrating servicesin mobile environments

Code-division multiple-access (CDMA) schemes appear to be very promising access techniques for coping with the requirements of third-generation mobile systems, mainly because of their flexibility. This paper proposes an adaptive S-ALOHA DS-CDMA access scheme as a method for integrating nonreal-time (i.e., Internet applications) and real-time (i.e., voice) services in a multicell scenario by exploiting the potentials of CDMA under time-varying channel load conditions. The adaptive component makes data terminals autonomously change their transmission rate according to the total (voice+data) channel occupancy, so that the minimum possible data delay, which can be analytically obtained by defining a birth-death process, is almost always achieved. Moreover, by means of a simplified cellular model, the proposed algorithm revealed the same behavior, i.e., it tries to select the most suitable transmission rate at any time slot, when it is affected by intercell interference and even by power control imperfections. Finally, in order to gain more insight into the potentials of such an access strategy, the adaptive S-ALOHA CDMA scheme is then compared to a reservation time-division multiple-access (TDMA)-based protocol (PRMA++), showing the benefits of the CDMA-based solution in terms of capacity, flexibility, and data delay performance     

Capacity design and performance of call admission control incellular CDMA systems

Since code-division multiple-access (CDMA) capacity is interference limited, call admission control (CAC) must guarantee both a grade of service (GoS), i.e., the blocking rate, and a quality of service (QoS), i.e., the loss probability of communication quality. This paper describes the development of a new capacity design method based on these two concepts. Theoretical expressions for GoS and QoS as functions of traffic intensity and CAC thresholds are first derived from the traffic theory viewpoint, and then a design method using these expressions is presented. At that time, two strategies for CAC are assumed. One is based on the number of users, and the other is based on the interference level. Computer simulation results are presented that strongly support the proposed design method. Furthermore, numerical examples and a performance comparison of the two strategies considering various propagation parameters, nonuniform traffic distributions, and various transmission rates are shown     

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     

Packet Voice on a Local Area Network with Round Robin Service

In this paper, we propose a combined voice/data protocol suitable for multiple access broadcast networks that provide round robin service to the stations. Such networks are well suited to the integration of voice and data since they guarantee bounded delay and provide high utilization even for high bandwidth channels. Using one such network proposal-namely Expressnet-as a representative scheme, we examine the characteristics of the service that voice traffic experiences under the voice/data protocol. We show that the access protocol is able to utilize the channel efficiently to support a large population of voice sources while maintaining low packet delay and guaranteeing some prespecified minimum bandwidth for data traffic. In addition, we show the advantages of silence suppression, i.e., discarding speech that constitutes silent periods, and we examine the cost of overloading the network in terms of the amount of speech discarded.     

A call admission control scheme for packet data in CDMA cellular communications

In wireless cellular communication systems, call admission control (CAC) is to ensure satisfactory services for mobile users and maximize the utilization of the limited radio spectrum. In this paper, we propose a new CAC scheme for a code division multiple access (CDMA) wireless cellular network supporting heterogeneous self-similar data traffic. In addition to ensuring transmission accuracy at the bit level, the CAC scheme guarantees service requirements at both the call level and the packet level. The grade of service (GoS) at the call level and the quality of service (QoS) at the packet level are evaluated using the handoff call dropping probability and the packet transmission delay, respectively. The effective bandwidth approach for data traffic is applied to guarantee QoS requirements. Handoff probability and cell overload probability are derived via the traffic aggregation method. The two probabilities are used to determine the handoff call dropping probability, and the GoS requirement can be guaranteed on a per call basis. Numerical analysis and computer simulation results demonstrate that the proposed CAC scheme can meet both QoS and GoS requirements and achieve efficient resource utilization.     

Congestion control policies for IP-based CDMA radio access networks

As CDMA-based cellular networks mature, the current point-to-point links used in connecting base stations to network controllers evolve to an IP-based radio access network (RAN) for reasons of lower cost due to statistical multiplexing gains, better scalability and reliability, and the projected growth in data applications. In this paper, we study the impact of congestion in a best-effort IP RAN on CDMA cellular voice networks. We propose and evaluate three congestion control mechanisms, admission control, diversity control, and router control, to maximize network capacity while maintaining good voice quality. We first propose two new enhancements to CDMA call admission control that consider a unified view of both IP RAN and air interface resources. Next, we introduce a novel technique called diversity control that exploits the soft-handoff feature of CDMA networks and drops selected frames belonging to multiple soft-handoff legs to gracefully degrade-voice quality during congestion. Finally, we study the impact of router control where an active queue management technique is used to reduce delay and minimize correlated losses. Using simulations of a large mobile network, we show that the three different control mechanisms can help gracefully manage 10-40 percent congestion overload in the IP RAN.     

Multidimensional PRMA with prioritized Bayesian broadcast-a MACstrategy for multiservice traffic over UMTS

Multidimensional packet reservation multiple access is proposed as a medium-access control (MAC) strategy for the uplink channel of the UTRA (UMTS terrestrial radio access) time-division/code-division multiple-access (TD/CDMA) mode to benefit from efficient statistical multiplexing on the large common pool of available resources (i.e., slots defined both in time and code domain). A prioritized Bayesian broadcast algorithm is derived to stabilize multidimensional packet reservation multiple access (MD PRMA) and to allow for access delay discrimination of four different access classes. Access delay spread can be selected, and trading voice-dropping ratio against data-access delay is possible. To control multiple-access interference, Bayesian broadcast can be combined with load-based access control. The performance of both frequency-division duplex (FDD) and time-division duplex (TDD) mode is evaluated, the latter particularly relevant for TD/CDMA. For mixed voice, Worldwide Web (WWW) browsing, and e-mail traffic, the UMTS WWW model is used, while the e-mail traffic model is derived here     

Congestion control for multimedia services

The problem of congestion control in high-speed networks for multimedia traffic, such as voice and video, is considered. It is shown that the performance requirements of high-speed networks involve delay, delay-jitter, and packet loss. A framing congestion control strategy based on a packet admission policy at the edges of the network and on a service discipline called stop-and-go queuing at the switching nodes is described. This strategy provides bounded end-to-end delay and a small and controllable delay-jitter. The strategy is applicable to packet switching networks in general, including fixed cell length asynchronous transfer mode (ATM), as well as networks with variable-size packets     

Call admission control for voice/data integration, in.broadband, wireless networks

This paper addresses bandwidth allocation for an integrated voice/data broadband mobile wireless network. Specifically, we propose a new admission control scheme called EFGC, which is an extension of the well-known fractional guard channel scheme proposed for cellular networks supporting voice traffic. The main idea is to use two acceptance ratios, one for voice calls and the other for data calls in order to maintain the proportional service quality for voice and data traffic while guaranteeing a target handoff failure probability for voice calls. We describe two variations of the proposed scheme: EFGC-REST, a conservative approach which aims at preserving the proportional service quality by sacrificing the bandwidth utilization, and EFGC-UTIL, a greedy approach which achieves higher bandwidth utilization at the expense of increasing the handoff failure probability for voice calls. Extensive simulation results show that our schemes satisfy the hard constraints on handoff failure probability and service differentiation while maintaining a high bandwidth utilization.     

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.     

An analysis of the performance of ARQ schemes for integrated traffic satellite-switched SFH and DS CDMA networks

The delay and throughput performance of satellite-switched Slow Frequency Hopping CDMA network for simultaneous voice and data transmission is analyzed and compared to that of a DS-CDMA system. Two ARQ schemes are suggested for data while Forward Error Correction using the same encoder is used for voice packets. The queueing analysis assumes priority for voice and two models for voice traffic are used (Markovian and IPP). The probability of successful packet transmission is derived for all systems as a function of traffic load allowing us to evaluate the systems using delay, throughput, and voice packet loss as figures of merit. Numerical results show that while voice delay is minimal DS CDMA is much more effective then SFH CDMA in all cases. One interesting result is that SFH systems perform better with S/W schemes and achieve a higher maximum throughput. It is also observed that the IPP and Markovian models gave similar results.This work was supported by an NSERC CRD (Collaborative Industrial Research and Development grant,) with Spar Aerospace, Quebec, Canada     

Adaptive admission/congestion control policies for CDMA‐based wireless internet

Radio resource management (RRM) is vital for the next generation wireless networks. RRM comprises many functionalities and this paper focuses on the investigation of the performance of several adaptive call admission/congestion control policies based on a window‐measurement estimation of the status of the buffer at the base station under the hybrid TDMA/CDMA access scheme. In our study, we interrelate the physical limitations of the base stations (i.e. the number of transmission and reception modems), call and burst level traffic, instantaneous buffer conditions and end‐to‐end bit error performance in one queuing problem. Subsequently, a window‐measurement estimator is implemented to estimate the likelihood of buffer congestion at the base station. Accordingly, the traffic loads shall be controlled. We use event‐driven simulation to simulate the multimedia integrated CDMA networks where heterogeneous traffic users are multiplexed into a simple TDMA frames. The simulation results show outstanding performance of the proposed call admission/congestion control policies in guaranteeing QoS requirements. Copyright © 2005 John Wiley & Sons, Ltd.     

Performance evaluation of a CDMA protocol for voice and dataintegration in personal communication networks

Future wireless personal communication networks (PCN's) will require voice and data service integration on the radio link. The multiaccess capability of the code-division multiple-access (CDMA) technique has been widely investigated in the recent literature. The aim of this paper is to propose a CDMA-based protocol for joint voice and data transmissions in PCN's. The performance of such a protocol has been derived by means of an analytical approach both in terms of voice packet dropping probability and mean data packet delay. Voice traffic has been modeled as having alternated talkspurts and silences, with generation of voice packets at constant rate during talkspurts and no packet generation during silence gaps. A general arrival process is assumed for the data traffic. However, numerical results are derived in the case of a Poisson process. Simulation results are given to validate our analytical predictions. The main result derived here is that the proposed CDMA-based protocol efficiently handles both voice and data traffic. In particular, it is shown that the performance of the voice subsystem is independent of the data traffic