Congestion relief on power-controlled CDMA networks

A digital cellular radio code-division multiple-access (CDMA) system can only support a finite number of users before the interference plus noise power density, I₀, received at the cellular base station causes an unacceptable frame-error rate. Once the maximum interference level is reached, new arrivals should be blocked. In a power-controlled CDMA system, the base station can direct mobiles to reduce their power and data rate to reduce interference and allow more users on the system. This approach is employed in TIA IS-95 with respect to the time-varying voice activity on cellular voice channels. We investigate an alternative technique where we adjust the power and data rate of mobile data users to the time-varying interference level to allow more users on a congested system. This scheme was simulated for various proportions of voice and data users and offered traffic levels. Blocking probabilities are reduced in some cases by two orders of magnitude. Message wait time, now a random variable, may exceed the wait time for a constant rate system at high traffic levels. If the cellular carrier has a maximum blocking requirement, an adaptive rate/power system can increase the capacity. For example, a base station that normally supports 26.4 Erlangs offered traffic with 2% blocking can support 33.5 Erlangs with the same blocking probability if adaptive rates and power control are used. Thus, the adaptive rate system can increase the capacity by 27%     

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%.     

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.     

Optimal resource allocation and adaptive call admission control for voice/data integrated cellular networks

Resource allocation and call admission control (CAC) are key management functions in future cellular networks, in order to provide multimedia applications to mobiles users with quality of service (QoS) guarantees and efficient resource utilization. In this paper, we propose and analyze a priority based resource sharing scheme for voice/data integrated cellular networks. The unique features of the proposed scheme are that 1) the maximum resource utilization can be achieved, since all the leftover capacity after serving the high priority voice traffic can be utilized by the data traffic; 2) a Markovian model for the proposed scheme is established, which takes account of the complex interaction of voice and data traffic sharing the total resources; 3) optimal CAC parameters for both voice and data calls are determined, from the perspective of minimizing resource requirement and maximizing new call admission rate, respectively; 4) load adaption and bandwidth allocation adjustment policies are proposed for adaptive CAC to cope with traffic load variations in a wireless mobile environment. Numerical results demonstrate that the proposed CAC scheme is able to simultaneously provide satisfactory QoS to both voice and data users and maintain a relatively high resource utilization in a dynamic traffic load environment. The recent measurement-based modeling shows that the Internet data file size follows a lognormal distribution, instead of the exponential distribution used in our analysis. We use computer simulations to demonstrate that the impact of the lognormal distribution can be compensated for by conservatively applying the Markovian analysis results.     

Access control of data in integrated voice/data CDMA systems:benefits and tradeoffs

In previous work, access control for data has been proposed as a method to ensure adequate quality of service (QoS) in an integrated voice/data CDMA system. The motivation behind access control is to schedule data packet transmissions in slots when voice activity is low and to curtail data transmissions when the voice load is heavy. In this work, the class of probabilistic access control schemes, wherein data transmissions are controlled by dynamically changing the permission probability, are considered. The trigger for changing the permission probability is a measure of the current uplink load. Perfect power control is assumed first, and the trigger for access control is the power control feasibility condition, Schemes based on prediction are analyzed. While prediction schemes are complex to implement, they do provide an upper bound for performance of access control schemes. A simple and practical access control scheme, proposed earlier in the literature, is then extended. It controls the permission probability for data based on uplink load and a 1-bit broadcast feedback to all of the mobiles. The performance of this scheme depends on the choice of three parameters. It is demonstrated that, through a combined choice of these parameters, access control can be tuned to perform as desired and to yield significant capacity gains over not using access control. Results are then extended to the case of imperfect power control, where the outage criterion is based on limiting the total received power at the base station. In this case, too, the simple control scheme is shown to work well     

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     

Uplink user capacity in a CDMA macrocell with a hotspot microcell: exact and approximate analyses

This paper studies the number of voice users (user capacity) supported on the uplink of a single-macrocell/single-microcell code-division multiple-access system. A "hotspot" microcell is embedded within a larger macrocell and operates over the same bandwidth as the larger cell. Analytic methods are presented for computing user capacity which account for propagation loss, multiple-access interference, power control, and random locations of user terminals, as well as two distinct methods by which users select base stations (tiers). Along with the exact user capacity, a technique for making accurate approximations is also presented. Simulation results verify both the exact and approximate analytical methods. This simulation is also employed to study the capacity gains of a third, more optimal, tier-selection scheme. These results point to differences in capacity performance based on the tier-selection method, as well as on the traffic density within the hotspot region.     

Performance analysis of a packet-switched synchronous voice/datatransmission system

The queuing behavior of voice and data traffic for a proposed voice/data multiplexing system is investigated. For synchronization of packetized voice in the packet-switched network, the master frame format of time-division multiplexing (TDM) is utilized. This scheme has discrete delay characteristics for packetized voice and provides a simple play-out method for voice signals. The performance of voice and data traffic is investigated by obtaining the cumulative distribution of delay of voice packets and the mean delay time of voice and data packets. The performances of the proposed system and the circuit-switched system are compared to time-assignment speech interpolation (TASI) in terms of the loss probability of voice traffic and the maximum allowable number of input voice channels for a given trunk capacity. The proposed system has been found to be more efficient than the circuit-switched system with TASI for integrating multiple voice channels in a single link. For a given trunk capacity, the system can accommodate about twice the number of input voice channels as the circuit-switched system     

Performance evaluation of a reservation random access scheme for packetized wireless systems with call control and hand-off loading

We consider a packet switched wireless network where each cell's communication channel is shared among packet voice sources. In this paper, we present a method for the design and analysis of wireless cells using a reservation random access (RRA) scheme for packet access control. This scheme is integrated with a call admission control procedure. We model the state process of a single cell as a vector Markov chain. We compute the steady state distribution of the Markov chain. This result is used to calculate the packet dropping probability and the call blocking probability. By setting limits on maximum permissible levels for the call blocking probability and the packet dropping probability, we obtain the Erlang capacity of a single cell, with and without hand-off traffic. For an illustrative RRA scheme, the Erlang capacity of a single cell is shown to be about twice that attained by a comparable fixed assigned TDMA scheme. We show that a cellular network using this RRA scheme and which applies can be no blocking of hand-off calls, exhibits similar call capacity levels.This work is supported by a University of California MICRO and Pacific-Bell Grant No. 94-107.     

Optimum Power Assignment for Maximum Throughput in CDMA Personal Communication Systems with Integrated Voice/Data Traffic

A CDMA personal communication system with integrated voice/data traffic is considered, in which the link error performance is controlled according to the voice error rate requirement, and the acceptable data traffic error rate is ensured by ARQ. Optimum power assignment (or allocation) between voice and data users is investigated to maximize the total system throughput. A graphical method (the tangent method) is described to obtain this optimum power allocation. The maximum throughput is expressed as a function of other system parameters. The tangent method is further used to measure the impact of transmission quality on the maximum data throughput. Numerical results and a design example are given for a power controlled wideband IS-95 type wireless personal communication system.     

Mobility-Aware Admission Control Schemes in the Downlink of Third-Generation Wireless Systems

In this paper, novel connection admission control (CAC) algorithms that take into account the effect of mobility of users both inside and outside the cell in the downlink of third-generation mobile systems are developed. First, the system capacity, including the other-cell interference, subject to feedback between cells is studied. Then, effective bandwidth expressions for calls are obtained as a function of both their location in the cell as well as their class of traffic (i.e., voice versus data). Next, this formulation is used to derive two mobility-aware admission control algorithms, i.e., a priority CAC, where calls are accepted not only upon resource availability, but also through acceptance ratios that reflect their levels of priority, and a squeezing CAC, where elastic calls may be squeezed to a minimum agreed value, giving way to admit more calls in the system and to secure further ongoing mobile users. Using Markovian analysis, several performance measures are obtained, namely the blocking probability, the dropping probability, both intracell and intercell, as well as the overall cell throughput. The authors eventually investigate the performance of our CAC and show how to extend the Erlang capacity bounds, i.e., the set of arrival rates such that the corresponding blocking/dropping probabilities are kept below predetermined thresholds     

Voice over ATM using AAL2 and bit dropping: performance and calladmission control

Asynchronous transfer mode (ATM) adaptation layer 2 (AAL2) has been designed for efficient transport of voice, fax, and voiceband data (VBD) traffic over an ATM virtual circuit. The protocol helps achieve low latency and high bandwidth efficiency while applying suitable compression methods on voice/VBD/fax calls and silence elimination on voice calls. We analyze the performance and capacity of an ATM multiplexer based on AAL2 adaptation. We assume that embedded adaptive differential pulse code modulation (ADPCM) is used to compress voice, and silence elimination is used to achieve statistical multiplexing gain. The embedded ADPCM coding scheme allows selective dropping of less significant bits of voice during congestion in the ATM/AAL2 multiplexer. We compare the call capacities of voice multiplexers with and without bit dropping (BD). The performance models and results presented are based on fairly general assumptions and can be used for traffic engineering and call admission control in land-line or wireless ATM systems for a variety of voice/voiceband compression algorithms. A generalized algorithm for call admission control is also described     

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.     

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     

A dynamic packet reservation multiple access scheme for wireless ATM

The dynamic packet reservation multiple access (DPRMA) scheme, a medium access control protocol for wireless multimedia applications, is proposed and investigated. DPRMA allows the integration of multiple traffic types through a single access control mechanism that permits users to specify their immediate bandwidth requirements. The primary feature of DPRMA is the dynamic matching of the traffic source generation rates with the assigned portion of the channel capacity. This is accomplished by a control algorithm that regulates the actual amount of channel capacity assigned to users. To support multimedia communication, channel capacity assignments are prioritized by traffic type. The performance of the scheme is evaluated and the scheme is shown to perform well in a system with voice, video conferencing, and data users present. It is also shown to provide improved performance over a system with a modified version of the packet reservation multiple access (PRMA) scheme. Furthermore, several system parameters are studied and optimized.     

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

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

Analysis and engineering of a voice/data packet multiplexer

This paper investigates performance and engineering issues concerning a multiplexer scheme that has been implemented in AT&T's Integrated Access Terminal (IAT) to transport packetized voice and data traffic on shared facilities. The multiplexer serves voice and data traffic according to a dynamic bandwidth allocation scheme in order to simultaneously meet their performance requirements. A bit-dropping procedure is employed for voice packets to provide a graceful degradation of voice quality under overload conditions. An analytical model is developed for the multiplexer service scheme that estimates performance parameters given the voice and data offered loads. The model is used to demonstrate the capacity advantages of dynamic bandwidth allocation, and to generate load-service curves that illustrate the tradeoffs of carrying different combinations of voice and data traffic on the multiplexer. Sensitivity of voice and data performance to the multiplexer time-slice parameters is also investigated. The model is readily embedded in a design approach that determines the bandwidth required to carry the voice and data traffic demands while satisfying all desired performance objectives     

Power Control for Packet Voice Service with Application to EDGE Wireless System

Using the enhanced data rates for GSM evolution (EDGE) system with cyclic frequency hopping as an example, we apply a Kalman-filter power control method based on interference tracking to packet voice service in wireless networks. Our results show that the power-control method significantly improves the spectral efficiency by enabling the 1/3 frequency reuse while maintaining a stringent requirement of 2% packet loss probability for voice service. Specifically, for allocated spectrum of 1.8, 3.6 and 5.4 MHz, the 1/3 reuse with the Kalman power control can yield 102.5%, 49.5% and 32.5% improvement in spectral efficiency, respectively, over the 3/9 reuse (regardless of whether or not power control is used). We also compare the performance of the Kalman method with a traditional Signal-to-interference-ratio method and a control method that is based on the last interference measurement. We find that appropriate selection of power for the first packet of each talk spurt and the filtering function for noisy measurements are crucial in providing high system capacity for packet voice service. For the EDGE system, we also identify a need for shortening the power update period, which is 480 ms in the specifications.     

Handoff algorithms in dynamic spreading WCDMA system supporting multimedia traffic

The undergoing third-generation wireless network needs to support the integration of voice and multimedia data services with varying quality-of-service (QoS) requirements. It is critical that the least bit-error rate (BER) for voice traffic, World Wide Web (WWW) traffic, and streaming video traffic be guaranteed at all time. In this paper, we discussed the integration of soft handoff and dynamic spreading factor in wideband code-division multiple-access system in supporting multimedia traffic. The contribution of our work is twofold. First, the processing time of the handoff request is analyzed. We found that intensive mobile handoff might consume significant amount of access channel time and increase the delay of handoff. We, therefore, proposed a batch mechanism such that multiple handoff requests could be processed simultaneously. The average delay is reduced from 1.12 s to 800 ms at heavy handoff rate. Our second contribution is a new resource allocation algorithm, where the spreading factor and transmission power for the handoff mobiles are jointly considered to maximize the throughput. The BER requirements for the handoff mobiles and the target cell are maintained during the handoff process. The original problem is formulated into a nonlinear programming format. We proposed a procedure to simplify it into a linear constraint problem, which is solved by a revised simplex method. Numerical results show a 25% increase in throughput for WWW traffic and a 26% improvement for the video traffic.     

Simulation analysis of a communication link with statisticallymultiplexed bursty noise sources

Variable bit rate (VBR) coding techniques have received great research interest as very promising tools for transmitting bursty multimedia traffic with low bandwidth requirements over a communication link. Statistically multiplexing the multimedia bursty traffic is a very efficient method of maximizing the utilization of the link capacity. The application of computer simulation techniques in analyzing a rate-based access control scheme for multimedia traffic such as voice traffic is discussed. The control scheme regulates the packetized bursty traffic at the user network interface of the link. Using a suitable congestion measure, namely, the multiplexer buffer length, the scheme dynamically controls the arrival rate by switching the coder to a different compression ratio (i.e., changing the coding rate). VBR coding methods can be adaptively adjusted to transmit at a lower rate with very little degradation in the voice quality. Reported results prove that the scheme greatly improves the link performance, in terms of reducing the probability of call blocking and enhancing the statistical multiplexing gain