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.     

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     

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.     

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     

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.     

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.     

AAL2分组话音复接器缓冲器队列门限值的确定方法

该文研究了带比特丢弃的AAL2分组话音复接器缓冲器队列门限值的确定方法,提出用话音分组作为缓冲器队列门限值的单位,给出了确定门限值的计算公式,并对输出链路容量为384kb/s的情况进行了计算机仿真。仿真结果表明,作者提出的门限值的确定方法可获得较小的平均分组时延和较低的平均分组丢失率,计算简便,易于实现,是一种很好的确定缓冲器队列门限值的方法。     

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     

Mean Packet Queueing Delay in a Buffered Two-User CSMA/CD System

We consider a system of two users of slotted CSMA-CD (carrier-sense multiple-access with collision detection). The two users are assumed to have independent identical packet arrival streams, the identical randomizing policy for retransmission, and an infinite capacity for storing queued packets. The mean packet delay (including the queueing and retransmission delays) is derived explicitly.     

Packetized voice/data integrated transmission on a token passingring local area network

The performance of a token-passing ring network with packetized voice/data mixed traffic is investigated through extensive simulations. Both data and voice users are modeled in the simulations. Data users produce bursty traffic. Voice traffic is modeled as having alternating talkspurts and silences, with generation of voice packets at a constant rate during talkspurts and no packet generation during silence periods. Token passing ring local area networks are shown to effectively handle both voice and data traffic. The effects of system parameters (e.g. voice packet length, talkspurt/silence lengths, data traffic intensity, and limited exhaustive service disciplines) on network performance are discussed     

Performance Analysis of Statistical Voice/Data Multiplexing Systems with Voice Storage

In this paper the performances of four statistical voice/data multiplexers are analyzed and compared. The structures of these multiplexers are based on the newly proposed queueing, frame management, and integrated flow control methods. For the queueing purpose, two separate finite voice and data buffers are used to queue voice and data packets independently. For frame management, four versions of master frame management are considered. As for the integrated flow control algorithm, speech coding techniques such as variable rate coding and embedded coding are considered. To evaluate the performances of the multiplexers, analytical formulations are made from the equivalent queueing models of the proposed systems. Then, numerical results on the performance of the multiplexers are obtained by using the Gauss-Seidel iteration method. These results show that there exist tradeoffs between data and voice traffic performances for a given input traffic load and the output channel rate.     

Performance evaluation of packet aggregation scheme for VoIP service in wireless multi-hop network

In a wireless multi-hop network environment, energy consumption of mobile nodes is an important factor for the performance evaluation of network life-time. In Voice over IP (VoIP) service, the redundant data size of a VoIP packet such as TCP/IP headers is much larger than the voice data size of a VoIP packet. Such an inefficient structure of VoIP packet causes heavy energy waste in mobile nodes. In order to alleviate the effect of VoIP packet transmission on energy consumption, a packet aggregation algorithm that transmits one large VoIP packet by combining multiple small VoIP packets has been studied. However, when excessively many VoIP packets are combined, it may cause deterioration of the QoS of VoIP service, especially for end-to-end delay. In this paper, we analyze the effect of the packet aggregation algorithm on both VoIP service quality and the energy consumption of mobile nodes in a wireless multi-hop environment. We build the cost function that describes the degree of trade-off between the QoS of VoIP services and the energy consumption of a mobile node. By using this cost function, we get the optimum number of VoIP packets to be combined in the packet aggregation scheme under various wireless channel conditions. We expect this study to contribute to providing guidance on balancing the QoS of VoIP service and energy consumption of a mobile node when the packet aggregation algorithm is applied to VoIP service in a wireless multi-hop networks.     

Traffic smoothing effects of bit dropping in a packet voicemultiplexer

The performance of a packet voice multiplexer queue in which the less significant bits of voiced packets are dropped during states of congestion in the multiplexer is examined. Using the results of simulation and analytical modeling, it is illustrated that bit dropping of voice packets significantly smooth the burstiness of superposition packet voice traffic by speeding up the packet service rate during critical periods of congestion in the queue. The smoothing effect renders it possible to approximate the superposition by a Poisson process for modeling a packet voice multiplexer with bit dropping. By comparison with a simulation, an analytical model based on the Poisson assumption is shown to produce quite accurate performance predictions. The results indicate that significant capacity and performance advantages are gained in the multiplexer as a result of the bit-dropping scheme     

Adaptive Spatial Filtering for an FH/SSMA Packet Radio Network with Packet Combining

In this paper, packet throughput is analyzed and simulated for a show FH/SSMA packet radio network with adaptive antenna array and packet combining in a Rayleigh fading channel with shadowing. The packet throughput is defined as the average number of captured packets per slot. To enhance the throughput performance, an adaptive spatial filtering through adaptive antenna array and a packet combining scheme are employed. As a random access protocol, slotted ALOHA is considered, and synchronous memoryless hopping patterns are assumed. A packet consists of codewords from an (n, k) RS (Reed-Solomon) code. The tap weights of an adaptive processor is updated by RLS (recursive-least-square) algorithm. From the simulation results, it is shown that a pre-processing by adaptive antenna array and a post-processing by packet combining are very effective to improve reception performance of an FH/SSMA network.     

Optimum arrival-time distribution for delay capture inspread-spectrum packet radio networks

The optimum arrival-time distribution that maximizes the delay-capture probability in spread-spectrum packet radio networks is derived. It is shown that when the optimum arrival-time distribution is employed, the capture probability converges to a finite value as the number of contending packets increases. Normalized throughput and average number of packet retransmissions are computed for slotted ALOHA multipoint-to-point packet radio networks employing direct-sequence spread-spectrum modulation. It is shown that large performance improvement is obtained by optimizing the arrival-time distribution compared to the uniform arrival-time distribution assumed by Davis and Gronemeyer (1980), especially when error-correction coding is employed. Two practical modifications are derived which are shown to provide performance close to the optimum along with an adaptive network load control scheme     

C-PRMA: a centralized packet reservation multiple access for localwireless communications

Packet-switched technology has been demonstrated as effective in cellular radio systems with short propagation delay, not only for data, but also for voice transmission. In fact, packet voice can efficiently exploit speech on-off activity to improve bandwidth utilization over time division multiple access (TDMA). Such an approach has been first suggested in the packet reservation multiple-access (PRMA) technique, an adaptation of the reservation ALOHA protocol to the cellular environment. However, being PRMA-based on a fixed frame scheme, it cannot thoroughly take advantage of the very short propagation delays encountered in microcellular systems that allow, for instance, the immediate retransmission of packets lost because of the interference noise from adjacent cells. We present the centralized PRMA, a natural enhancement of PRMA, in which the base station (BS) plays a central role in scheduling the transmissions of mobile stations (MSs). As a consequence, the transmission scheduling is very flexible and can account for the different traffic rate and delay constraints that emerge from voice and data integration. A packet retransmission policy to recover corrupted packets can be implemented and operated efficiently to provide an acceptable grade of service, even in a very noisy environment. The simulation results presented show the quantitative improvements of the centralized packet reservation multiple-access (C-PRMA) performance with respect to PRMA     

Routing for multimedia traffic in wireless frequency-hopcommunication networks

A new method is described for routing multimedia traffic in a frequency-hop (FH) store-and-forward packet radio network. The method is illustrated for traffic of two types, each type having its own throughput, delay, and error-rate requirements. A typical application is the routing of voice and data packets in a distributed multiple-hop network. In such an application, voice packets cannot tolerate much delay, but they are allowed to contain a small number of frame erasures while data packets must be delivered error-free even if a moderate delay is required to do so. The fully distributed routing protocol presented in the paper takes into account the type of service required for each type of traffic, and it adapts to the interference as seen by the FH radio receivers in the network. Our approach to multimedia routing is based on least-resistance routing with different link and path resistance metrics for different message types. Each of the resistance metrics for a link reflects the ability of the link to provide the service required by the one of the message types. This includes, but is not limited to, a measure of the likelihood of successful reception by the FH radio receiver for that link. The route selection for a particular type of packet depends on the resistances of the links along the routes from that packet's source to its destination. In general, different routes may be selected for different types of packets. The primary conclusion of this paper is that the quality of service increases for each of the two types of multimedia traffic if the routing protocol accounts for the type of message that is being relayed     

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

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

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     

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.