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     

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.     

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     

Delay Related Issues in Integrated Voice and Data Networks

This paper focuses on network delays as they apply to voice traffic. First the nature of the delay problem is discussed and this is followed by a review of enhanced circuit, packet, and hybrid switching techniques: these include fast circuit switching (FCS), virtual circuit switching (VCS), buffered speech interpolation (SI), packetized virtual circuit (PVC), cut-through switching (CTS), composite packets, and various frame management strategies for hybrid switching. In particular, the concept of introducing delay to resolve contention in SI is emphasized, and when applied to both voice talkspurts and data messages, forms a basis for a relatively new approach to network design called transparent message switching (TMS). This approach and its potential performance advantages are reviewed in terms of packet structure, multiplexing scheme, network topology, and network protocols. The paper then deals more specifically with the impact of variable delays on voice traffic. In this regard the importance of generating and preserving appropriate length speech talkspurts in order to mitigate the effects of variable network delay is emphasized. The results indicate that a desirable length of talkspurt "hangover" of about 200 ms will accomplish this without unduly affecting speech activity, and that, under these circumstances, the perceptable threshold of variable talkspurt delay can be as high as about 200 ms average. As such, the results provide a useful guideline for integrated services system designers. Finally, suggestions are made for further studies on performance analysis and subjective evaluation of advanced integrated services systems.     

Output and delay process analysis for slotted CDMA wirelesscommunication networks with integrated voice/data transmission

This paper presents the output and delay process analysis of integrated voice/data slotted code division multiple access (CDMA) network systems with random access protocol for packet radio communications. The system model consists of a finite number of users, and each user can be a source of both voice traffic and data traffic. The allocation of codes to voice calls is given priority over that to data packets, while an admission control, which restricts the maximum number of codes available to voice sources, is considered for voice traffic so as not to monopolize the resource. Such codes allocated exclusively to voice calls are called voice codes. In addition, the system monitoring can distinguish between silent and talkspurt periods of voice sources, so that users with data packets can use the voice codes for transmission if the voice sources are silent. A discrete-time Markov process is used to model the system operation, and an exact analysis is presented to derive the moment generating functions of the probability distributions for packet departures of both voice and data traffic and for the data packet delay. For some cases with different numbers of voice codes, numerical results display the correlation coefficient of the voice and data packet departures and the coefficient of variation of the data packet delay as well as average performance measures, such as the throughput, the average delay of data packets, and the average blocking probability of voice calls     

Packet voice communications over PC-based local area networks

Experimental implementations of packet voice communication systems over two types of PC-based local area networks are described. The first is a Proteon proNET token-passing ring network, and the second is an Ethernet network. System configuration, operation, and performance are described for both networks. Models of network performance for estimating the maximum allowable number of active voice stations without incurring intolerable packet loss are presented for each system. The models are defined for systems with and without silence detection. PC-related implementation issues are also discussed     

SDH以太共享环在城域网的应用

随着因特网和数据业务的爆炸性增长，城域网的主要业务正在从话音向数据转移。然而过去的城域传送网是为话音业务设计的，所以采用的是TDMM网络(SDH、SONET)。本首先介绍了基于SDH网络传送数据业务的几种新技术，如：POS、GFP、LCAS、RPR、共享以太环等，以及可以实现业务隔离、保证安全的VLAN、VLAN嵌套技术，然后对各种城域网数据传输技术进行了比较，最后分新了薪一代的MSTP设各府该具有的几个特点。     

A Comparison of Measured and Calculated Speech Temporal Parameters Relevant to Speech Activity Detection

This paper deals with the measurement and calculation of various speech temporal parameters of interest in an environment where speech activity detection is employed. In particular it is shown that, based on either a measurement or model of the probability density function (pdf) for silence durations for the case of zero talkspurt "hangover" or "fill-in," that the following temporal parameters can be computed for any value of hangover or fill-in: the mean (and pdf) for silence durations, the mean talkspurt duration, the mean talkspurt rate, and the speech activity. Directly measured values of these parameters and those computed from both measured and fitted versions of the pdf for silence durations are compared and are shown to be in reasonable agreement. The illustrated results are based on measurements of about two minutes of taped male monolog source speech. However, the approach to calculating the above parameters is general in the sense that it can be applied to any measured or modeled pdf for silence durations. The significance of this work lies in the important role that talkspurt hangover plays, for example, in minimizing speech detector induced back-end clipping of talkspurts, reducing exposure to the variable talkspurt delay impairment, and in determining signaling overhead and resource occupancy in various speech interpolation, packet voice, and integrated voice/data systems.     

Data Performance in Burst Switching When the Voice Silence Periods Have a Hyperexponential Distribution

The performance of data in burst switching has been analyzed in previous work with a fluid approximation of the data traffic. This study extends the previous model to the case where the silence interval between talkspurts has a hyperexponential, rather than an exponential, distribution. It is shown that data performance is extremely sensitive to the variance of the silence interval, and that, for empirical talkspurt and silence distributions, this model provides a vast improvement on models which assume that both types of intervals are exponentially distributed.     

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     

A Bandwidth Reservation Multiple Access Protocol for Wireless ATM Local Networks

A bandwidth reservation multiple access scheme(BRMA) is proposed to resolve contention and assignbandwidth among multiple users trying to gain access toa common channel such as in mobile users contending for resources in an ATM-based cellular networkor a wireless local area network (LAN) with shortpropagation delays. The protocol is best suited tosupport variable-bit-rate (VBR) traffic that exhibits high temporal fluctuations. Each mobile user isconnected end-to-end to another user over virtualchannels via the base station that is connected to thewired ATM B-ISDN network. The channel capacity is modeled as a time frame with a fixed duration.Each frame starts with minislots, to resolve contentionand reserve bandwidth, followed by data-transmissionslots. Every contending user places a request for data slots in one of the minislots. If therequest is granted by the base station through adownlink broadcast channel, the user then startstransmission in the assigned slot(s). The number ofassigned slots varies according to the required qualityof service (QoS), such as delay and packet lossprobability. A speech activity detector is utilized inorder to indicate the talkspurts to avoid wastingbandwidth. Due to its asynchronous nature, BRMA is ratherinsensitive to the burstiness of the traffic. Since theassignment of the minislots is deterministic, therequest channels are contention-free and the data channels are collision-free. Hence, in spite ofthe overhead (minislots) in each frame, BRMA provideshigher throughput than Packet Reservation MultipleAccess (PRMA) for the same QoS, especially for high-speed systems. A better delay performance is alsoachieved for data traffic compared to Slotted Alohareservation-type protocol PRMA. In addition, BRMAperforms better in terms of bandwidth efficiency thanthe conventional TDMA or the Dynamic TDMA, wherespeech activity detectors are very difficult toimplement.     

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     

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.     

Performance Analysis of Data in Burst Switching

The performance of interactive data traffic in burst switching is analyzed with a model that allows a number of the channels to be reserved for data messages, assumes that voice talkspurts have preemptive priority over data messages in the shared channels, and uses a fluid-flow approximation for the data traffic. Extensive validations are provided for the model.     

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 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 Channel Access Protocol for Integrated Voice/Data Applications

A new multiaccess protocol is proposed for an integrated voice/data application. The protocol, which is a variation of virtual time CSMA (VT-CSMA), takes advantage of the periodicity of voice packets and possesses a number of important features. With this protocol, voice stations appear to have a dedicated time-division multiplexed (TDM) slot, and the delay of a voice packet is bounded by the length of a frame (defined to be the period between two consecutive voice packets from a voice station). Also, the amount of data added to the channel has little effect on the voice traffic. When silence detection is used, many more voice conversations can be supported without losing the dedicated-slot characteristic. This is in contrast to a movingboundary TDM system where the excessive bandwidth saved by silence detection can only be used for data. The protocol requires no global synchronization and is easy to implement. Simulation results are presented to evaluate its performance.     

Analytical Performance Estimation of a Wireless TDMA System with Traffic Classification

This paper describes an analytical model for performance estimation of a mobile communications system based on a TDMA/TDD access scheme. In accordance with the ATM classification of services and its terminology, three types of traffic have been taken into account: CBR for voice connections, VBR for real-time data applications and ABR for non real-time data services. Both voice and data traffic is modeled following Poisson processes. Moreover, three different policies for data resource assignment are evaluated: FCFS, NPPS and silence exploitation (SAD). Theoretical versus simulated results show the suitability of the model when the real-time traffic load is relatively low compared to non real-time traffic.     

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.