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 Distributed Experimental Communications System

The packet experimental communications system (packet XCS) is a new experimental voice and data switch. It uses a local-area network (LAN) for digital voice transmission, with local intelligence for switching. The packet XCS also has highly distributed control. The individual sites cooperate to provide user services as well as internal data management. We have learned that several local networks, including CSMA/CD networks, can be made to work well for voice transmission and that highly distributed control is practical in such a system. A system has been constructed which is used as a testbed for distributed voice and data communications experiments. This system is purely for experimentation and does not indicate a direction for future Bell System product offerings.     

Performance analysis of channel de-allocation schemes for dynamic resource allocation in GSM/GPRS networks

Channel de-allocation for GSM voice call (DASV) has been considered for dynamic resource allocation in GSM/GPRS networks. Two new de-allocation schemes are proposed: de-allocation for GPRS packet (DASP) and de-allocation for both GSM voice call and GPRS packet (DASVP). An analytic model with general GPRS data channel requirement is derived to evaluate the performance of the schemes in terms of GSM voice call incompletion probability, GPRS packet dropping probability, average GPRS packet transmission time and channel utilisation.     

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     

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     

分组卫星话音／数据综合的混合多址协议性能研究

本文讨论了一种用于分组话音／数据综合的混合多址协议，该协议是固定分配与随机访问的混合，并且赋予话音分组优先传输权，从而保证了无重传话音分组有较小的丢失概率。本文进行了详细的理论分析，并得出了性能比较特性，所得结果认为这是一个兼顾话音／数据综合的较好协议，且具有一定的灵活性。     

A reservation random-access protocol for voice/data integratedspread-spectrum multiple-access systems

Most code-division multiple-access (CDMA) systems described in the literature provide only one single service (voice or data) and employ the strategy of “one-code-for-one-terminal” for code-assignment. This assignment, though simple, fails to efficiently exploit the limited code resource encountered in practical situations. We present a new protocol called reservation-code multiple-access (RCMA), which allows all terminals to share a group of spreading codes on a contention basis and facilitates introducing voice/data integrated services into spread-spectrum systems. The RCMA protocol can be applied to short-range radio networks, and microcell mobile communications, and can be easily extended to wide area networks if the code-reuse technique is employed. In RCMA, a voice terminal can reserve a spreading code to transmit a multipacket talkspurt while a data terminal has to contend for a code for each packet transmission. The voice terminal will drop a long delayed packet while the data terminal just keeps it in the buffer. Therefore, two performance measures used to assess the proposed protocol are the voice packet dropping probability and the data packet average delay. Theoretical performance is derived by means of equilibrium point analysis (EPA) and is examined by extensive computer simulation     

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.     

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     

Stability evaluation and design of the PRMA joint voice data system

Packet reservation multiple access (PRMA) provides spatially dispersed voice and data terminals wireless access to a base station over a common short-range radio channel. The author presents an equilibrium point analysis of the joint voice data system. The analysis leads to two coupled nonlinear equations. By a judicious change of variables, the author calculates the equilibrium of the joint system by finding the roots of a univariate equation. The author provides expressions for voice packet dropping probability, the distribution of data packet delays and the system throughput, at equilibrium. For the joint voice data system the design problem is two dimensional. Elementary catastrophe theory leads to a bifurcation of the design space into useful and congested regions. The useful design space is further restricted by target values of the performance criteria. The study of a nominal system verifies the design methodology. Moreover, it shows that PRMA gracefully accepts low bit rate data terminals with moderate data packet delays (few hundred milliseconds), while simultaneously satisfying a 1% speech packet dropping criterion     

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     

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.     

Integrated voice/data multiplexers with slotted and nongated packetservices

Integrated voice/data multiplexers that provide packet services for both voice and data traffic are discussed. A slotted service is assumed, so that packet transmissions are synchronized to slot boundaries. Nongated service, in which packets are transmitted as soon as the transmission capacity becomes available, is also assumed. The performance of nongated and slotted multiplexers is obtained by analytic and simulation approaches. In particular, a PRIO (head-of-the-line priority to voice packets) and a BVFD (busy-voice, fixed-data) multiplexer are shown to be suitable for such a nongated environment     

Proposal and performance evaluation of an efficient multiple-access protocol for LEO satellite packet networks

This paper deals with a modified version of the packet reservation multiple-access (PRMA) protocol suitable for integration of real-time (voice) and best effort (data) traffic in low Earth orbit (LEO) satellite communication systems. The proposed scheme differs from previous alternatives on the method adopted to handle access requests for voice and data terminals, and to transmit data messages. An analytical approach is proposed and validated in the case of voice and classical (i.e., geometric distributed) data traffic in order to derive system performance in terms of mean data message delay and voice packet dropping probability. However, in order to better highlight the advantages of the proposed approach typical interactive and background traffics types have been also considered. Performance comparisons with previous proposed PRMA protocols for voice and data transmission in LEO satellite communication systems are also shown in order to highlight the better behavior of the proposed scheme. Finally, a brief discussion concerning the extension of the proposed S-PRMA protocol to the case of different satellite communication systems is also provided.     

Optimal Packet Length for Packet Voice Communication

It is shown that for typical operating parameters, the optimal packet length for a single link packet voice system is of the order of 300-700 bits. This is contrary to both the optimal length of a data packet (approximately 1000 bits), and documented experimental implementations of such network architecture for voice (approximately 1000 bits).     

A Markov Modulated Characterization of Packetized Voice and Data Traffic and Related Statistical Multiplexer Performance

We study the performance of a statistical multiplexer whose inputs consist of a superposition of packetized voice sources and data. The performance analysis predicts voice packet delay distributions, which usually have a stringent requirement, as well as data packet delay distributions. The superposition is approximated by a correlated Markov modulated Poisson process (MMPP), which is chosen such that several of its statistical characteristics identically match those of the superposition. Matrix analytic methods are then used to evaluate system performance measures. In particular, we obtain moments of voice and data delay distributions and queue length distributions. We also obtain Laplace-Stieitjes transforms of the voice and data packet delay distributions, which are numerically inverted to evaluate tails of delay distributions. It is shown how the matrix analytic methodology can incorporate practical system considerations such as finite buffers and a class of overload control mechanisms discussed in the literature. Comparisons with simulation show the methods to be accurate. The numerical results for the tails of the voice packet delay distribution show the dramatic effect of traffic variability and correlations on performance.     

New voice over Internet protocol technique with hierarchical datasecurity protection

The authors propose a voice over Internet protocol (VoIP) technique with a new hierarchical data security protection (HDSP) scheme. The proposed HDSP scheme can maintain the voice quality degraded from packet loss and preserve high data security. It performs both the data inter-leaving on the inter-frame of voice for achieving better error recovery of voices suffering from continuous packet loss, and the data encryption on the intra-frame of voice for achieving high data security, which are controlled by a random bit-string sequence generated from a chaotic system. To demonstrate the performance of the proposed HDSP scheme, we have successfully verified and analysed the proposed approach through software simulation and statistical measures on several test voices     

Access scheme for integrated voice/data transmissions over commonpacket channel in 3GPP

In this letter, we propose a flexible channel assignment scheme using preemption as an access method for integrated voice/data transmissions over common packet channel (CPCH) in 3GPP. We analyze the proposed scheme and compare the performance of the proposed scheme with the performance of the basic, channel monitoring, and channel assignment schemes in view of the voice packet dropping probability and the average delay of data packet     

A new protocol for the integration of voice and data over PRMA

In packet reservation multiple access (PRMA) the receiver in the mobile terminal is required to listen continuously to monitor the acknowledgment messages broadcasted at the end of every time slot. A new scheme for the integration of voice and data based on PRMA is proposed. The voice and the data subsystems are logically separated. The total available bandwidth is divided into three regions-voice information, voice contention, and data regions. The available bandwidth is dynamically partitioned between the above three regions subject to the fulfillment of the quality of service (QoS) requirements of the voice users. The voice subsystem has been modeled as a Markov chain and an exact analytical method used to compute the voice packet dropping probability is described. A nonlinear programming problem is formulated to optimize the bandwidth allocated for the data users. Solutions to this nonlinear programming problem that are very close to optimum have been obtained heuristically. Numerical results indicate that a significant amount of data traffic can be supported without sacrificing the voice capacity of the system