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     

Coded Speech in Packet-Switched Networks: Models and Experiments

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

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     

An integrated services token-controlled ring network

A ring protocol is proposed that allows voice and data traffic to coexist within the ring. This ring has the following distinct features: (1) it allows synchronous traffic such as voice and video to have a definite access to the channel within each packetization period (or frame); (2) it allows data messages to have a higher channel access priority provided that the synchronous traffic is not delayed by more than one frame; (3) it supports variable rate data circuits. Simulation results show that the data-message delay is much smaller than for other integrated services schemes. Urgent messages can be transmitted with a higher priority over voice. Since the voice packet delay is bounded within one packetization period, no time-stamping is needed and the voice loss can be completely avoided by reserving a sufficient number of slots. Continual speech reception is possible by synchronizing the speech regeneration process to the end of each frame. Since the ring is synchronized, gateway switching to external circuit-switched and packet-switched networks is very simple     

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     

Soft QoS provisioning using the token bank fair queuing scheduling algorithm

Future-generation wireless packet networks will support multimedia applications with diverse QoS requirements. Much of the research on scheduling algorithms has been focused on hard QoS provisioning of integrated services. Although these algorithms give hard delay bounds, their stringent requirements sacrifice the potential statistical multiplexing performance and flexibility of the packet-switched network. Furthermore, the complexities of the algorithms often make them impractical for wireless networks. There is a need to develop a packet scheduling scheme for wireless packet-switched networks that provides soft QoS guarantees for heterogeneous traffic, and is also simple to implement and manage. This article proposes token bank fair queuing (TBFQ), a soft scheduling algorithm that possesses these qualities. This algorithm is work-conserving and has a complexity of O(1). We focus on packet scheduling on a reservation-based TDMA/TDD wireless channel to service integrated real-time traffic. The TBFQ scheduling mechanism integrates the policing and servicing functions, and keeps track of the usage of each connection. We address the impact of TBFQ on mean packet delay, violation probability, and bandwidth utilization. We also demonstrate that due to its soft provisioning capabilities, the TBFQ performs rather well even when traffic conditions deviate from the established contracts.     

Supplementary services in the H.323 IP telephony network

Traditionally, different networks were developed to handle voice, data, and video. The circuit-switched telephone network carried voice and the packet network carried data. Due to different deployment of these networks, different services were developed, such as voice mail in the telephone network and electronic mail on the Internet. With the revolution of multimedia in the computer industry, voice, video, and data are now being carried on both networks. Supplementary services, such as transfer and forwarding (which were originally developed for private telephone networks and later migrated to public telephone networks) are now being developed for packet networks. The standards for packet networks are being defined in the H.323-based series of ITU-T recommendations. This article provides the H.323 architecture for supplementary services, the differences in deployment of these services between the circuit-switched and packet-switched networks, and interworking of these services across hybrid networks     

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.     

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     

Techniques for Packet Voice Synchronization

Packet switching has been proposed as an effective technology for integrating voice and data in a single network. An important aspect of packet-switched voice is the reconstruction of a continuous stream of speech from the set of packets that arrive at the destination terminal, each of which may encounter a different amount of buffering delay in the packet network. The magnitude of the variation in delay may range from a few milliseconds in a local area network to hundreds of milliseconds in a long-haul packet voice and data network. This paper discusses several aspects of the packet voice synchronization problem, and techniques that can be used to address it. These techniques estimate in some way the delay encountered by each packet and use the delay estimate to determine how speech is reconstructed. The delay estimates produced by these techniques can be used in managing the flow of information in the packet network to improve overall performance. Interactions of packet voice synchronization techniques with other network design issues are also discussed.     

Channel Allocation for GPRS with Buffering Mechanisms

General Packet Radio Service (GPRS) provides mobile users end-to-end packet-switched services by sharing the radio channels with voice and circuit-switched services. In such a system, radio resource allocation for circuit-switched and packet-switched services is an important issue, which may affect the QoS for both services significantly. In this paper, we propose two algorithms: Dynamic Resource Allocation with Voice and Packet queues (DRAVP) and Dynamic Resource Allocation with Packet and Voice queues (DRAPV) for channel allocation of the voice calls and packets. We propose analytic and simulation models to investigate the performance of DRAVP and DRAPV in terms of voice call incompletion probability, packet dropping probability, average voice call waiting time, and average packet waiting time. Our study indicates that the buffering mechanism for GPRS packets significantly increase the acceptance rate of GPRS packets at the cost of slightly degrading the performance of voice calls.     

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     

TRACE: time reservation using adaptive control for energy efficiency

Time reservation using adaptive control for energy efficiency (TRACE) is a time frame based media access control (MAC) protocol designed primarily for energy-efficient reliable real-time voice packet broadcasting in a peer-to-peer, single-hop infrastructureless radio network. Such networks have many application areas for various scenarios that obey a strongly connected group mobility model, such as interactive group trips, small military or security units, and mobile groups of hearing impaired people. TRACE is a centralized MAC protocol that separates contention and data transmission, providing high throughput, bounded delay, and stability under a wide range of data traffic. Furthermore, TRACE uses dynamic scheduling of data transmissions and data summarization prior to data transmission to achieve energy efficiency, which is crucial for battery operated lightweight radios. In addition, energy dissipation is evenly distributed among the nodes by switching network controllers when the energy from the current controller is lower than other nodes in the network, and reliability is achieved through automatic controller backup features. TRACE can support multiple levels of quality-of-service, and minimum bandwidth and maximum delay for voice packets are guaranteed to be within certain bounds. In this paper, we describe TRACE in detail and evaluate its performance through computer simulations and theoretical analysis.     

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.     

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

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

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.     

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     

A distributed MAC scheme supporting voice services in mobile ad hoc networks

Future mobile ad hoc networks are expected to support voice traffic. The requirement for small delay and jitter of voice traffic poses a significant challenge for medium access control (MAC) in such networks. User mobility presents unique difficulties in this context due to the associated dynamic path attenuation. In this paper, a MAC scheme for mobile ad hoc networks supporting voice traffic is proposed. With the aid of a low‐power probe prior to DATA transmissions, resource reservation is achieved in a distributed manner, thus leading to small packet transmission delay and jitter. The proposed scheme can automatically adapt to dynamic path attenuation in a mobile environment. Statistical multiplexing of on/off voice traffic can also be achieved by partial resource reservation for off voice flows. Simulation results demonstrate the effectiveness of the proposed scheme. Copyright © 2009 John Wiley & Sons, Ltd.     

Message delay analysis for TDMA schemes using contiguous-slotassignments

Under a TDMA (time-division multiple-access) scheme, a station shares a multiple-access communications channel by transmitting its messages during its dedicated time slots. An exact result concerning the queue-sizer and message delay analysis of TDMA systems in which a station is allocated multiple consecutive slots per frame is presented. The generating function of the system queue-size for a general independent arrival message process is obtained. Messages consist of a random number of packets, following a geometric distribution. An exact result for the generating function of the message delay for various common message arrival processes and light bounds for the mean message queue-size and delay are then derived. The results are compared to previously derived approximations. It is also proved that a slot allocation scheme which distributes station slots uniformly over the frame yields a message-delay lower bound. The results also apply to the analysis of time-shared reservation schemes