Bounds on the mean system-size and delay for a movable-boundaryintegrated circuit and packet switched communications channel

A multiaccess communications channel which is shared on an integrated circuited-switched and packet-switched basis is considered. Applications include time-division multiplexing and demand-assigned/TDMA communication channels which are shared by circuit-switched services (such as isochronous voice) and packet-switched services (such as data). Frames of fixed duration are established and divided into two parts: one part is used for voice transmissions while the other part serves to accommodate packetized data traffic. These two parts are separated by a movable boundary, so that data traffic can use the frame capacity which is temporarily unoccupied by voice transmissions. An exact expression is derived for the mean packet queue-size and delay. To reduce the computational complexity involved in using this expression, simpler expressions for upper and lower bounds are derived for the mean packet queue-size and packet delay. Examples demonstrating the tightness of these numerically efficient bounds are presented     

The Tradeoff Between Delay and TASI Advantage in a Packetized Speech Multiplexer

A packetized speech multiplexer differs from a circuitswitched TASI system in that the presence of a packet buffer allows a tradeoff where the TASI advantage can be increased at a cost in packet delay. This tradeoff is investigated via a simulation. Results are presented to show the relations between TASI advantage and delay, for both an average delay criterion and a maximum delay criterion. It is shown that, particularly for the case where small numbers of talkers are multiplexed, the packetized system offers significant improvements in TASI advantage over the conventional circuit-switched multiplexer, at modest costs in packet delay.     

Design and performance analysis for data transmission in GSM/GPRSsystem with voice activity detection

To efficiently utilize the bandwidth of cellular mobile systems and offer service of high quality to both voice and data users, we propose a protocol to integrate packet-switched data traffic into current time-division multiple-access (TDMA)-type circuit-switched digital voice systems. We analyze the performance of the proposed system, which transmits data packets in the silent periods of a conversation with voice activity detection and adapts itself to the GSM/GPRS system, which uses the idle channels to provide data services. We show that the proposed protocol can increase the bandwidth utilization efficiency and improve the throughput/delay performance of the data transmission while minimizing the impact on the current GSM/GPRS service     

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.     

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.     

Power control and capacity analysis for a packetized indoormultimedia DS-CDMA network

This paper proposes a packetized indoor wireless system using direct-sequence code-division multiple-access (DS-CDMA) protocol. The indoor radio environment is characterized by slow Rayleigh fading with or without lognormal shadowing. The system supports multimedia services with various transmission rates and quality of service (QoS) requirements and allows for seamless interfacing to asynchronous transfer mode (ATM) broadband networks. All packets are transmitted with forward error correction (FEC) using convolutional code for voice packets and Bose-Chaudhuri-Hocquenghem (BCH) code for data packets with an automatic retransmission request (ARQ) protocol and for video packets without ARQ. A queueing model is used for servicing data transmission requests. A power control algorithm is proposed for the system, which combines closed-loop power control with channel estimation to give the best performance. The cell capacity of each traffic type and various multimedia traffic configurations in both single-cell and multiple-cell networks are evaluated theoretically under the assumption of perfect power control. The effect of power control imperfection on the capacity using the proposed power control algorithm is investigated by computer simulation     

Performance evaluation of a packet voice system

A study is made of statistical multiplexing of voice packets from a number of packetized voice sources onto a single channel. Each source alternates between talkspurt (active period) and silence, and packets are generated during active periods only. The packets are buffered (in a finite size buffer) when transmission capacity is not available. An embedded Markov chain model is adopted to analyze the system and a numerical technique is presented to compute system performance. Simulation results validate the analysis     

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     

Delay analysis for forward signaling channels in wireless cellularnetwork

We consider connection-oriented wireless cellular networks. Such second generation systems are circuit-switched digital networks which employ dedicated radio channels for the transmission of signaling information. A forward signaling channel is a common signaling channel assigned to carry the multiplexed stream of paging and channel-allocation packets from a base station to the mobile stations. Similarly, for ATM wireless networks, paging and virtual-circuit-allocation packets are multiplexed across the forward signaling channels as part of the virtual-circuit set-up phase. The delay levels experienced by paging and channel-allocation packets are critical factors in determining the efficient utilization of the limited radio channel capacity. A multiplexing scheme operating in a “slotted mode” can lead to reduced power consumption at the handsets, but may in turn induce an increase in packet delays. In this paper, focusing on forward signaling channels, we present schemes for multiplexing paging and channel-allocation packets across these channels, based on channelization plans, access priority assignments and paging group arrangements. For such multiplexing schemes, we develop analytical methods for the calculation of the delay characteristics exhibited by paging and channel-allocation packets. The resulting models and formulas provide for the design and analysis of forward signaling channels for wireless network systems. This revised version was published online in July 2006 with corrections to the Cover Date.     

Performance Evaluation of a Packetized Voice System--Simulation Study

Introduction of the packet switching technique into digitized voice communication may afford great advantages in efficient use of the channel, compared to both circuit-switched and DSI systems. Detailed characteristics, however, have not been obtained because of difficulty in the exact analysis. Hence, simalation models are developed in this paper for the packetized voice transmission system, and various characteristics such as tranmission delays and loss probability of voice packets are obtained. We further evaluate three types of voice packet reassembly strategy at the receiving terminal, and obtain the optimal packet length, which keeps both overall packet transmission delay and packet loss probabilty less than a certain permissible value. Comparison among three strategies is also stated.     

An efficient multiplexing technique for packet-switched voice/datanetworks

The authors propose a multiplexing frame structure that makes it possible to transmit voice messages synchronously without loss or clipping of contents. This scheme has discrete delay characteristics, and provides a simple play-out method for reproduction of voice signal. The authors investigate its performance by obtaining the cumulative distribution of delay of voice packets and the mean waiting time of data packets. It is concluded that this synchronous frame structure can easily be applied to enhance services with various transmission rates, such as flow control of message streams, node congestion control, and service-class or throughput-class negotiation of channels without significant degradation of trunk utilization     

Slot allocation for an integrated voice/data TDMA mobile radiosystem with a finite population of buffered users

Slot allocation for voice and data in an integrated TDMA mobile radio system is investigated. In the proposed system, voice traffic is circuit-switched and data traffic is packet-switched using slotted ALOHA for channel access; the data traffic model is practically assumed to have a finite number of users with finite buffer capacity. The authors apply an equilibrium point analysis (EPA) technique to analyze the data performance and present a heuristic performance criterion to obtain an optimal slot allocation for voice and data in the integrated TDMA mobile radio system     

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 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.     

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     

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 Model for Evaluating the Performance of an Integrated Circuit- and Packet-Switched Multiplex Structure

Recent papers have introduced a multiplexing structure for mixing voice and data traffic in an integrated telecommunications system. This structure utilizes a master frame format of a time division statistical multiplex facility. A certain portion of the frame is allocated to voice calls, and data traffic is assigned to the remaining frame capacity. To achieve a high transmission utilization, data are allowed to use any residual voice capacity momentarily available due to statistical variations in the voice traffic. The voice traffic is treated as a loss system and data packets are buffered. In this paper we derive exact analytical expressions for the key system perfomance measures, the probability of loss for voice calls, and the expected waiting time for data packets. Actually, two cases are considered, the one discussed above, called the movable boundary case, and one where the boundary is fixed; i.e., data are not allowed to utilize the residual voice capacity. The computational aspects of calculating actual numbers are discussed in some detail, and results are presented for typical cases.     

SEAMA: a source encoding assisted multiple access protocol forwireless communications

This paper presents SEAMA, a source encoding assisted multiple access (MAC) protocol, to integrate voice and data traffic in a wireless network. SEAMA exploits the time variations of the speech coding rate, through statistical multiplexing, to efficiently use the available bandwidth and to increase the link utilization. In each frame, SEAMA allocates bandwidth among calls as needed. Ongoing calls are always assigned some minimum bandwidth to allow for coding of the background noise during silence periods. An embedded voice encoding scheme is employed to allow the network to control the rate of the calls during congestion by selectively dropping some of the less significant packets, thus causing a graceful degradation of quality. It is shown that by employing an appropriate voice coding scheme and exploiting the characteristics of the source encoder in the MAC protocol, SEAMA almost doubles the capacity of the voice section compared to a circuit-switched network, while practically maintaining the quality of voice traffic     

Performance analysis of integrated-services load-adaptive/TDMA satellite networks

A load-adaptive/TDMA multiple-access communications system which serves to interconnect broad-band multimedia packet streams is considered. In particular, the use of a satellite backbone communications link whose channels are dynamically assigned to network stations is investigated. Each station supports packetized voice and data message streams. Incoming streams to a station are statistically multiplexed by the station across the backbone channels currently allocated to this station. To enhance the multiplexing process, a variable bit-rate packet-voice encoding scheme is also employed. Stations periodically issue requests for backbone channel allocations, based upon their estimated loading status. We introduce two distinct multiple-access algorithms for allocating the shared backbone channels to the stations. We develop analytical methods for the analysis and design of such integrated multiplexing/multiple-access networks. Performance measures include voice and data packet delays and packet blocking probabilities. Voice stream performance is also characterized by the average number of bits per sample used by the voice encoding scheme. The effects of the propagation delay across the backbone link are especially demonstrated. Also illustrated are the performance improvements attained due to the use of the load-adaptive/TDMA scheme. Under the example of the ’all-voice’ traffic loading, an LA/TDMA scheme exhibits no obvious performance improvement over a fix-assigned scheme. However, as the burstiness of the traffic loading increases in the example of the ’data-voice’ traffic loading, a significant amount of improvement (36 per cent bandwidth savings) is realized by a LA/TDMA scheme.     

An integrated voice/data system for mobile indoor radio networks

A centralized, integrated voice/data radio network for fading multipath indoor radio channels is proposed and analyzed. The packets of voice and data are integrated through a movable boundary method. The uplink channel access uses a framed-polling protocol whereas the downlink uses a time-division multiple-access (TDMA) scheme. This system dynamically switches between two transmission rates and uses multiple antennas to maximize the throughput in the fading multipath indoor environment. Throughput and delay characteristics of the system are analyzed using four different techniques. The results are compared with those of Monte Carlo computer simulations. A simple relationship between the number of voice terminals and the throughput of the data traffic are derived for an upper bound of 10-ms delay for the data packets