An integrated CDMA code assignment model for voice and data traffic in wireless communications

Wireless personal communication requires a provision of integrated services of multimedia traffic, such as voice and data, over the radio link. The multiple access protocols of code-division multiple-access (CDMA) techniques have been widely investigated in the recent literature. This paper presents an innovative multiple access protocol for CDMA-based wireless communication systems by fully utilizing the characteristics of voice and data traffic. In other words, a voice terminal can reserve a spreading code to transmit packets in multiple talk spurts, while a data terminal can transmit packets by either using the unassigned codes or borrowing the codes from the voice terminals during their silent periods. We build mathematical models for voice and data subsystems, respectively. Two performance parameters, the average dropping probability for voice packets and the average transmission delay for data packets, are derived based on the equilibrium point analysis. The effects of the two performance parameters on the system performance are discussed by varying the code reservation intervals of the voice terminals.     

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.     

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.     

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     

Output and delay of multi-channel slotted ALOHA systems for integrated voice and data transmission

Packet-switched technology has been developed to offer personal communication services not only for data but also for different types of user-end equipment such as phone-type audio. To satisfy the huge service demand and multi-traffic requirements with limited bandwidth, this paper proposes an efficient procedure of multi-channel slotted ALOHA for integrated voice and data transmission in wireless information networks and presents an exact analysis with which to numerically evaluate the performance of the systems. A channel reservation policy is applied, where a number of channels (called reserved channels) are used exclusively by voice packets, while the remaining channels are used by both voice and data packets, and voice packets select the reserved channels with a given probability (called selection probability). Probability distributions for the numbers of voice and data departures and for the data packet delay are derived. Numerical results compare some cases with different numbers of channels, different numbers of reserved channels and different selection probabilities to discuss what effects they may have on channel utilization, loss probability, average packet delay, coefficient of variation of data packet delay, and correlation coefficient of packet departures. This revised version was published online in June 2006 with corrections to the Cover Date.     

Wireless access to the World Wide Web in an integrated CDMA system

We propose a wireless access mechanism for web traffic packets in an integrated wireless code-division multiple-access system that has both voice and Web traffic. The proposed scheme is a medium-access control layer/link layer (MAC/LL) scheduling algorithm that consists of a two level control: admission control and packet scheduling. The admission control restricts the number of users in the system such that quality-of-service requirements [target signal-to-interference ratio (SIR) and delay] for both voice and Web traffic can be met. The packet scheduling balances the system interference on a slot-by-slot basis such that the target SIRs can be achieved for all users (voice and Web browsing sessions) with a higher scheduling priority for voice. Designing admission control for Web users based on the average offered rate per session is difficult due to the high variations in the offered load generated by heavy tailed distributions for Web traffic. To overcome this problem, we propose an admission control algorithm that adaptively estimates the aggregate average load based on load measurements using a sliding observation window.     

A demand assignment system for mobile users in a community ofinterest

The paper presents a high performance wireless access and switching system for interconnecting mobile users in a community of interest. Radio channel and time slot assignments are made on user demand, while the switch operations are controlled by a scheduling algorithm designed to maximize utilization of system resources and optimize performance. User requests and assignments are carried over a low-capacity control channel, while user information is transmitted over the traffic channels. The proposed system resolves both the multiple access and the switching problems and allows a direct connection between the mobile end users. The system also provides integration of voice and data traffic in both the access link and the switching equipment. The “movable boundary” approach is used to achieve dynamic sharing of the channel capacity between the voice calls and the data packets. Performance analysis based on a discrete time Markov model, carried out for the case of optimum scheduling yields call blocking probabilities and data packet delays. Performance results indicate that data packets may be routed via the exchange node with limited delays, even with heavy load of voice calls. Also the authors have proposed scheduling algorithms that may be used in implementing this system     

Resource allocation for multiple classes of DS-CDMA traffic

We consider a packet data direct-sequence code-division multiple-access (DS-CDMA) system which supports integrated services. The services are partitioned into different traffic classes according to information rate (bandwidth) and quality of service (QoS) requirements. Given sufficient bandwidth, QoS requirements can be satisfied by an appropriate assignment of transmitted power and processing gain to users in each class. The effect of this assignment is analyzed for both a single class of data users and two classes of voice and data users. For a single class of data users, we examine the relationship between average delay and processing gain, assuming that ARQ with forward error correction is used to guarantee reliability. The only channel impairment considered is interference, which is modeled as Gaussian noise. A fixed user population is assumed and two models for generation of data packets are considered: (1) each user generates a new packet as soon as the preceding packet is successfully delivered and (2) each user generates packets according to a Poisson process. In each case, the packets enter a buffer which is emptied at the symbol rate. For the second traffic model, lowering the processing gain below a threshold can produce multiple operating points, one of which corresponds to infinite delay. The choice of processing gain which minimizes average delay in that case is the smallest processing gain at which multiple operating points are avoided. Two classes of users (voice/data and two data classes) are then considered. Numerical examples are presented which illustrate, the increase in the two-dimensional (2-D) capacity region achievable by optimizing the assignment of powers and processing gains to each class     

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     

Hybrid channel assignment scheme for accommodating voice/datatraffic in DS-CDMA cellular systems

A hybrid channel assignment (HCA) scheme in direct sequence-code division multiple access (DS-CDMA) systems for accommodating integrated voice/data traffic is proposed and the required power levels of voice and data traffic are derived. These levels can be used to maintain the minimum required link qualities of all calls. In the proposed scheme, delay-sensitive voice traffic is accommodated in circuit mode and delay-nonsensitive data traffic is accommodated in packet mode. The capacity region is derived and it can be used for controlling voice call admission and scheduling data packets. The proposed scheme can achieve a high link efficiency with reduced control overhead by statistically multiplexing voice and data traffic     

Coded Speech in Packet-Switched Networks: Models and Experiments

This paper considers the possibility of introducing packetized voice traffic into a packet-switched network. It is well known that the network must assure voice packets sufficient delay characteristics for conversational speech, i.e., low delay between speaker and listener and low delay jitter or variance. To reach these goals, simplified protocols and priority rules for voice handling are proposed and evaluated. A model of a packet switching node structure capable of handling both data and voice is derived for both analytical and simulation approaches. The use of low bit rate voice encoders is considered. The necessity of avoiding the transmission of silent intervals is discussed in relation to the behavior of packet voice receivers. Proposed strategies are compared by means of analytical tools and simulation experiments considering the presence of voice, interactive, and batch data packets.     

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.     

Performance of a Dynamic TDMA System for Mobile Wireless Access

A dynamic TDMA system can utilize voice activityand allow the integration of voice and data traffic.This can be achieved by allocating frequency channelsand time slots on demand. In this approach, upon the arrival of a talkspurt or a data packet,the base station is requested to assign a time slot foreach transmission. Message requests and assignments ofmobile users are carried over a Control channel, while the voice and traffic are transmittedover a Traffic channel. Time slot assignments are madefrom a pool of Traffic channels. A numberof slots in the pool will be shared by voice and data, with voice having priority over data, andthe remaining will be used by data only. Voice slots arereserved for the duration of the talkspurt whereas datapackets are assigned on a per-slot basis. Data packets can be buffered whereas voicetraffic can only tolerate limited delay beyond whichtalkspurts will be clipped off. The Control channeluplink access is based on Slotted Aloha so that mobile users have autonomous access to base stations.This paper presents the performance of the dynamic TDMAsystem outlined here. The analysis aims at assessing thecapacity gained by using voice activity and voice/data integration, in terms of theimpairments introduced to voice quality (e.g., speechclipping and/or delay) and the delays to data packets.The analysis has been based on a discrete time Markov model operating on a frame-by-frame basis thatprovides the joint distribution of the number of activevoice and data users in the system. The analysis alsoevaluates the delays of message requests via the uplink control channel. In evaluating theclipping probability, we combine the impact of both theaccess delays at the control channel as well as theunavailability of time slots in the pool. Performance results indicate that the capacity gain mayexceed 80% and the speech clipping can be kept below 1%.Also, data packets may be transmitted with limiteddelays even when all capacity is allocated for voice users. The proposed approach may be used toenhance the capacity of the existing TDMA cellularsystems and to provide integration of voice and dataservices.     

Performance of Integrated Packet Voice/Data Token-Passing Rings

This paper presents performance results that indicate that packetized voice service can be provided on a token-passing ring without adversely affecting the performance of data traffic. This is accomplished by introducing a relatively mild priority structure: stations are limited to a single packet transmission per medium access, and voice packets are given access priority over data packets at the same station. In addition, voice traffic is allowed longer packet lengths than data traffic. Several versions of this basic scheme are considered: 1) the number of active stations is constrained so that voice packets are guaranteed access within one packetization period, 2) no guarantee on access time is provided and voice packets are discarded when the waiting time exceeds one packetization period, and 3) no guarantee on access time is provided and voice packets are buffered until they can be transmitted.     

An Integrated Voice/Data System for VHF/UHF Mobile Radio

This paper presents the basic architecture and performance of a mobile radio multiaccess voice/data system. Natural pauses in conversational speech allow bandwidth saving through interleaving of data packets and talkspurts from different voice sources. A speech detector designed specifically for the mobile environment is presented. Blocking and delay performance of the multiaccess uplink is analyzed for voice traffic, assuming no traffic effects from the low priority data packets. Performance results from simulation are then presented for two downlink strategies in a two-hop virtual circuit in which a base station acts as a relay. The results verify also that the uplink analysis is valid for low voice traffic. For the data traffic, simulation results are presented in terms of data packet transmission delay and probability of collision with talkspurts. The results indicate that data flow may be limited by the collision factor. This work concludes that relative to conventional radio telephoning in which two channels are dedicated to each transmitter/receiver pair, a bandwidth reduction of 30-35 percent can be achieved.     

Contention resolution in random-access wireless networks based on orthogonal complementary codes

This paper proposes a new method for contention resolution in random-access wireless networks. Using orthogonal complementary codes to design access-request packets, users can reserve channel access successfully, even in severe contentions. Collisions among access-request packets can be resolved and exploited, whereas collisions among data packets are avoided. System throughput and delay performance can be enhanced, because random-access contention becomes transparent. Specifically, system throughput approaches the offered load up to the maximum value one with improved average packet delay performance. A joint layer design approach is proposed with both the physical layer signal-detection algorithm and the medium access-control layer random-access protocol. The performance is analyzed with the consideration of signal detection errors. Simulations are performed to demonstrate its superior performance.     

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.     

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.     

A Theoretical Traffic Performance Analysis of an Integrated Voice-Data Virtual Circuit Packet Switch

A time division packet switch capable of concurrently handling both voice and data traffic is proposed, and some of its performance limitations are analyzed. The voice packet traffic is handled at a higher priority level than data traffic, in order to meet stringent timing criteria, and can be shown to be handled just as if it were circuit switched. The data traffic utilizes whatever time slots are not occupied with voice traffic. The principal performance limitations described in this exploratory study are the fraction of time the voice traffic is blocked due to all the available time slots already being used for voice traffic, and an upper bound on the mean delay encountered by the data traffic as it waits to find an available time slot. An illustrative numerical result is the following. If we assume that each voice telephone conversation lasts for a mean of five minutes, and that twenty voice calls are generated over a six hour time span, and each data session lasts for a mean of forty minutes, and that five data calls are generated over a six hour time span, then if separate line switched networks are used for voice and for data with long term blocking probability of one percent, a total of 703 64 kbits links would be required to support 461 voice stations and 882 data terminals. On the other hand, using the integrated voice/data switch described here, and if we assume that the total delay due to the switch alone for data packets cannot exceed a long term mean value of one second, then only 298 64 kbit/s links are required to support 461 voice stations and 882 data terminals, reducing the number of required links by a factor of about two. Moreover, the assumptions leading to this comparison suggest that the packet switch could in fact support significantly more than this number of voice stations and data terminals. This is achieved at the expense of additional buffering for the data in the packet switch approach.     

On Multiple Traffic Type Integration over Wireless TDMA Channels with Adjustable Request Bandwidth

A new medium access control (MAC) protocol for mobile wireless communications is presented and investigated. We explore, via an extensive simulation study, the performance of the protocol when integrating voice, video and data packet traffic over a wireless channel of high capacity (referring to an indoor microcellular environment). Depending on the number of video users admitted into the system, our protocol varies: a) the request bandwidth dedicated to resolving the voice users contention, and b) the probability with which the base station grants information slots to voice users, in order to preserve full priority for video traffic. We evaluate the maximum voice capacity and mean access delay, as well as the aggregate channel throughput, for various voice and video load conditions, and the maximum voice capacity, aggregate channel throughput and average data message delays, for various video, voice and data load conditions. As proven by the comparison with a recently introduced efficient MAC scheme (DPRMA), when integrating voice and video traffic our scheme obtains higher voice capacity and aggregate channel throughput. When integrating all three traffic types, our scheme achieves high aggregate channel throughput in all cases of traffic load.