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 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     

Joint admission/congestion control for wireless CDMA systemssupporting integrated services

A new data traffic control scheme is developed for maintaining the packet error rate (PER) of real-time voice traffic while allowing nonreal-time data traffic to utilize the residual channel capacity of the multi-access link in an integrated service wireless CDMA network. Due to the delay constraint of the voice service, voice users transmit their packets without incurring further delay once they are admitted to the system according to the admission control policy. Data traffic, however, is regulated at both the call level (i.e., admission control) and at the burst level (i.e., congestion control). The admission control rejects the data calls that will otherwise experience unduly long delay, whereas the congestion control ensures the PER of voice traffic being lower than a specified quality of service (QoS) requirement (e.g., 10 ^-2). System performance such as voice PER, voice-blocking probability, data throughput, delay, and blocking probability is evaluated by a Markovian model. Numerical results for a system with a Rician fading channel and DPSK modulation are presented to show the interplay between admission and congestion control, as well as how one can engineer the control parameters. The tradeoff of using multiple CDMA codes to reduce the transmission time of data messages is also investigated     

Modeling and efficient handling of handoffs in integrated wireless mobile networks

We propose and analyze two handoff schemes without and with preemptive priority procedures for integrated wireless mobile networks. We categorize the service calls into four different types, namely, originating voice calls, originating data calls, voice handoff request calls, and data handoff request calls and we assume two separate queues for two handoff services. A number of channels in each cell are reserved exclusively for handoff request calls. Out of these channels, few are reserved exclusively for voice handoff request calls. The remaining channels are shared by both originating and handoff request calls. In the preemptive priority scheme, higher priority is given to voice handoff request calls over data handoff request calls and can preempt data service to the queue if, upon arrival, a voice handoff request finds no free channels. We model the system by a three-dimensional Markov chain and compute the system performance in terms of blocking probability of originating calls, forced termination probability of voice handoff request calls, and average transmission delay of data calls. It is observed that forced termination probability of voice handoff request calls can be decreased by increasing the number of reserved channels. On the other hand, as a data handoff request can be transferred from a queue of one base station to another, there is no packet loss of data handoff except for a negligibly small blocking probability.     

Study of information loss in packet voice systems

Once a voice buffer is full, it remains full for a certain period, during which many packets are possibly blocked, resulting in consecutive clippings in voice. The packet loss rate during this period changes slowly and has large fluctuations. It is shown that the temporal behavior of packet loss, especially at high rate, is inherently determined by voice correlation and system capacity and is independent of buffer size. Buffering may reduce the occurrence of short blocking periods associated with low rates packet loss but does not affect long ones associated with high packet loss rates. In fact, increasing the buffer size merely extends nonblocking periods, and thereby reduces the overall average packet loss rate, but packet-loss performance within existing blocking periods is not significantly improved. A simple tool is developed for calculating the boundary performance. It is found that it is possible to design a packet-switched voice system without buffering only at the expense of supporting a fewer number of calls. The issue of voice delay allocation between source and network is discussed, and it is shown that it is more effective to keep the network delay short while extending the source delay     

An Integrated Voice Data Protocol for Slotted CDMA Personal Communication Networks

In this paper we propose a new protocol forintegrated voice/data traffic in personal communicationnetworks (PCNs) employing slotted packet code divisionmultiple access (CDMA). The concept of dynamic boundary is adopted in both code and timedomains to meet the different requirements for data andvoice traffic. This paper outlines and compares theperformance of three channel assignment policies. The network performance is measured in terms of theaverage voice blocking probability and average datadelay. A discrete-event simulation and a two-dimensionalMarkov model are developed for both the fixed boundary scheme and dynamic boundary scheme.The results indicate that an improved performance can beachieved by introducing the dynamic boundary scheme, andthe dynamic assignment policy results in a performance close to the optimal value. It isalso easier to be implemented.     

Channel-aware priority-based groupwise packet scheduling for non-real time data service with burst-switching DS/CDMA system

The algorithm of scheduling scheme of channel-aware priority-based groupwise transmission is investigated for non-real time data service for the uplink direct sequence code division multiple access (DS/CDMA) system using the burst-switching scheme to support the integrated voice/data service. The proposed scheme optimally determines the transmission-time groups and assigns optimal data rates to the users with packets in the transmission-time group depending on priority metric, which involves several parameters such as delay threshold, waiting time, length of packet, and state of the channel, in a way of minimizing the average transmission delay. Simulation results show that the proposed algorithm gives better performance of average transmission delay and packet loss probability than any other conventional algorithms.     

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     

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%.     

Optimal Joint Session Admission Control in Integrated WLAN and CDMA Cellular Networks with Vertical Handoff

This paper considers optimizing the utilization of radio resources in a heterogeneous integrated system consisting of two different networks: a wireless local area network (WLAN) and a wideband code division multiple access (CDMA) network. We propose a joint session admission control scheme for multimedia traffic that maximizes overall network revenue with quality of service (QoS) constraints over both the WLAN and the CDMA cellular networks. The WLAN operates under the IEEE 802.11e medium access control (MAC) protocol, which supports QoS for multimedia traffic. A novel concept of effective bandwidth is used in the CDMA network to derive the unified radio resource usage, taking into account both physical layer linear minimum mean square error (LMMSE) receivers and characteristics of the packet traffic. Numerical examples illustrate that the network revenue earned in the proposed joint admission control scheme is significantly larger than that when the individual networks are optimized independently with no vertical handoff between them. The revenue gain is also significant over the scheme in which vertical handoff is supported, but admission control is not done jointly. Furthermore, we show that the optimal joint admission control policy is a randomized policy, i.e., sessions are admitted to the system with probabilities in some states     

Performance evaluation of a modified PRMA protocol for joint voiceand data packet wireless networks

In this paper, a modified version of the packet reservation multiple-access (PRMA) protocol is proposed to provide spatially dispersed voice and data user terminals wireless access to a base station over a common short-range radio channel. An analytical approach is presented in order to derive system performance in terms of mean data message delay and voice packet dropping probability. A suitable permission probability design is also proposed to enhance system performance. Performance comparisons with an extension of the PRMA protocol to voice data systems previously reported in literature are shown to highlight the better behavior of this approach     

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     

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     

Optimal strategies for admitting voice and data traffic in networks of LEO satellites using CDMA

Efficient policies are derived for admitting voice and data traffic into networks of low-earth-orbit (LEO) satellites using code-division multiple-access (CDMA) with direct-sequence spread-spectrum (DS/SS) signaling. The satellites act as bent-pipes; no on-board processing or intersatellite links are present. Dual satellite diversity is used to mitigate the effects of shadowing. The policies assume a movable boundary form, allocate optimally the CDMA capacity (PN codes) to voice and data users, and can increase significantly the number of users served while satisfying their bit error rate (BER) requirements. In contrast to direct admission policies that do not take into consideration the statistical features of the traffic, the new policies do account for the different nature of voice and data traffic and the history of prior transmissions/admissions. Two priority schemes are considered: voice users have higher priority than data users, or voice and data users have the same priority. A modified version of our policies can handle two classes of data users: one with high priority which requires real-time delivery and another with low priority that can be queued; the BER requirements of the two data types may differ. Optimal policies have lower voice blocking rates and data packet error rates than direct admission policies.     

Design and Analysis of a Channel Assignment Scheme for CDMA/TDMA Mobile Networks

In this paper, a channel assignment scheme is proposed for use in CDMA/TDMA mobile networks carrying voice and data traffic. In each cell, three types of calls are assumed to compete for access to the limited number of available channels by the cell: new voice calls, handoff voice calls, and data calls. The scheme uses the movable boundary concept in both the code and time domains in order to guarantee the quality of service (QoS) requirements of each type. A traditional Markov analysis method is employed to evaluate the performance of the proposed scheme. Measures, namely, the new call blocking probability, the handoff call forced termination probability, the data call loss probability, the expected number of handoff and the handoff link maintenance probability are obtained from the analysis. The numerical results, which are validated by simulation, indicate that the scheme helps meet the QoS requirements of the different call types.     

Performance analysis of Finite-User slotted aloha in wireless networks with multiple packet reception and random traffic

In this paper, we propose an analytical approach for evaluating the performance of finite-user slotted Aloha in wireless networks with multiple packet reception and random traffic. We derive the exact values of the throughput, the average system size, the packet blocking probability, and the average system delay. Our analysis is based on probability theory. We show that our numerical results are identical to simulation results.     

A dynamic rate control mechanism for source coded traffic in a fastpacket network

To achieve better statistical gain for voice and video traffic and to relieve congestion in fast packet networks, a dynamic rate control mechanism is proposed. An analytical model is developed to evaluate the performance of this control mechanism for voice traffic. The feedback delay for the source node to obtain the network congestion information is represented in the model. The study indicates that significant improvement in statistical gain can be realized for smaller capacity links (e.g., links that can accommodate less than 24 voice calls) with a reasonable feedback time (about 100 ms). The tradeoff for increasing the statistical gain is temporary degradation of voice quality to a lower rate. It is shown that whether the feedback delay is exponentially distributed or constant does not significantly affect performance in terms of fractional packet loss and average received coding rate. It is also shown that using the number of calls in talkspurt or the packet queue length as measures of congestion provides comparable performance     

A CDMA wireless packet network for voice-data transmissions

This letter investigates the possibility of integrating voice and data communications in a CDMA wireless packet network to provide access to a base station over a common short-range radio uplink channel for many spatially dispersed voice and data user terminals. Speech activity detection is assumed for voice communications to temporarily devote codes unused by voice user terminals during silence periods to data transmissions. The network proposed exhibits a good performance both in terms of quality of voice communications which is independent of data transmissions and maximum data traffic load supported with bounded delay     

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.     

Channel Allocation for GPRS with Buffering Mechanisms

General Packet Radio Service (GPRS) provides mobile users end-to-end packet-switched services by sharing the radio channels with voice and circuit-switched services. In such a system, radio resource allocation for circuit-switched and packet-switched services is an important issue, which may affect the QoS for both services significantly. In this paper, we propose two algorithms: Dynamic Resource Allocation with Voice and Packet queues (DRAVP) and Dynamic Resource Allocation with Packet and Voice queues (DRAPV) for channel allocation of the voice calls and packets. We propose analytic and simulation models to investigate the performance of DRAVP and DRAPV in terms of voice call incompletion probability, packet dropping probability, average voice call waiting time, and average packet waiting time. Our study indicates that the buffering mechanism for GPRS packets significantly increase the acceptance rate of GPRS packets at the cost of slightly degrading the performance of voice calls.