Design approaches and performance criteria for integrated voice/data switching

This paper examines some of the fundamental problems associated with the design and performance of integrated systems and networks that switch both voice and data. Specifically, the need for an integrated approach to the switching and transmission of voice and data is explored and alternative design considerations are discussed. One approach, described in detail, utilizes a distributed architecture to implement variable width channel allocations for the dynamic union of voice and data. Key performance criteria which aid the systems designer in evaluating the merits of a proposed unified design are identified. Examples are illustrated and supportive material is provided by a comprehensive bibliography.     

基于脉冲调制的GSM语音信道数据传输方法

针对全球移动通信系统 (GSM) 的安全机制不能实现端到端安全通信问题,提出了一种低复杂度基于脉冲调制的数据传输方案及优化方法。分别设计了基于脉冲位置调制及脉位结合极性调制的类语音信号,给出了基于跳时脉冲的帧同步方法。搭建了基于脉冲调制的GSM语音信道数据传输仿真平台,分析比较了不同调制阶数和不同自适应多速率编码 (AMR) 码率下系统的性能。为了提高系统可靠性,引入卷积码进一步降低了系统误码率。仿真结果表明,4阶脉冲调制最适用于语音信道,所提脉冲调制方法与传统低复杂度频移键控 (FSK) 调制方式误码率近似,但传输速率高,且提出的脉位结合极性调制进一步提高了传输速率,总提高比例达到36%。     

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     

Design and performance analysis of an advanced, narrowbandintegrated voice/data mobile radio system

Private land mobile communications have experienced significant demand for increased capacity and for new services, but congestion within the currently allocated 450 and 800 MHz bands has severely limited the capability of present generation systems to meet this demand. This paper proposes a narrowband integrated voice/data mobile radio system which triples current voice traffic capacity and simultaneously provides capacity for mobile data communications such as mobile data terminals, automatic vehicle location, and mobile facsimile by utilizing the silence gaps inherent in speech. The system is designed to fit within the narrowband 25 kHz channels in the 450 and 800 MHz frequency bands, and thus the system can replace existing private land mobile radio systems with minimum impact. The centerpiece of the system design is an evolutionary multiplexing and access control technique known as Packetized Data, Voice Dedicated (PDVD) Burst Switching which allows transmission of data within the silence gaps inherent in speech. Simulation results for various voice and data traffic loads show the flexibility and efficiency of the proposed narrowband integrated voice/data mobile radio system     

Semidynamic and dynamic transmission algorithms for multiplexing voice with data

The design of hybrid transmission algorithms for the multiplexing of voice and data over a common digital channel is of interest to various communication networks, including cellular radio and high-speed topologies. In environments where the characteristics of the voice and data traffics may vary dynamically, the issue is the deployment of hybrid multiplexing algorithms (HMAs) which satisfy the constraints imposed by the voice traffic, while they simultaneously attain high channel utilization and induce low implementation overhead. In this paper, we propose, evaluate, and compare two HMAs: a semidynamic and a dynamic. The former induces lower implementation overhead than the latter, but it is applicable only to environments where the rate of the voice traffic may vary relatively slowly and its statistics are parametrically known. The semidynamic HMA induces frame structures, where the capacity allocation per frame, for the voice versus data traffic, is dictated by a superimposed traffic monitoring algorithm. The dynamic HMA, on the other hand, assigns each channel slot to voice versus data packets dynamically; it requires no statistical knowledge about the voice traffic, at the expense of significantly increased implementation overhead.     

A failure detection method for integrated voice/data packetnetworks

A failure detection strategy at the data-link layer, implementable between frame-relay network nodes, is described for integrated voice/data packet networks. This strategy makes it possible to respond to link failures early enough to maintain data-session continuity. A simple, two-phase approach is proposed. In the first phase, the presence of a bit error rate violation along the packet link during short, fixed-length time segments is determined. In the second phase, the failure decision is made on the basis of a sequential algorithm. A simple table lookup implementation of the algorithm that requires no real-time computation is described     

Call admission control for voice/data integrated cellular networks: performance analysis and comparative study

In this paper, we propose a new call admission control scheme called dual threshold bandwidth reservation, or DTBR scheme. The main novelty is that it builds upon a complete sharing approach, in which the channels in each cell are shared among the different traffic types and multiple thresholds are used to meet the specific quality-of-service (QoS) requirements. We present a detailed comparative study based on mathematical and simulation models, and quantitatively demonstrate that the DTBR is capable of providing the QoS guarantee for each type of traffic, while at the same time leading to much better channel efficiency. We further show that the DTBR scheme with elastic data service can offer both service guarantee and service differentiation for voice and data services, and enhance the bandwidth utilization.     

Capacity allocation for integrated voice/data transmission at apacket switched TOM

The integrated transmission of voice and data at a time-division multiplexer (TDM) is discussed and analyzed. The system operates in a frame format and the channel capacity is governed by the frame size. The allocation of channel capacity for the transmission of voice and data is performed by a controller. Digital speech interpolation (DSI) and embedded coding techniques are used to enhance the transmission efficiency and to facilitate the implementation of multiplexing. Using a dynamic programming approach, a capacity allocation policy which jointly optimizes the voice/data performance is introduced. Numerical results indicate that the aggregate throughput of the system can be improved with a slight degradation in voice quality     

基于PC/104总线的语音数据同传Modem的设计

话带Modem技术已经发展成熟，但还有一些功能未被了解和使用。文中介绍一种利用专业Modem芯片实现ASVD(模拟语音和数据同时传输)的方法，这在需要点对点传输数据和语音的通信中是一种很好的选择。利用RCV336ACF／SP芯片设计实现数据和语音同时传输的Modem，电路设计简洁，数据传输稳定，语音清晰，可在非电话线路上实现呼叫和自动应答。     

Adaptive subcarrier allocation and bit loading for voice/data transmission in multiuser OFDM systems

A new algorithm of adaptive subcarrier allocation and bit loading (A‐SABL) is proposed for simultaneous voice and data transmission in multiuser OFDM systems. The algorithm takes advantage of the frequency diversity and the voice/data transmission requirements to dynamically assign the number of subcarriers and bits/per symbol on each subcarrier for each user in a single cell. Due to the strict delay requirement of voice service, the subcarriers with low channel gains are assigned for voice transmission with a small number of bits per symbol to guarantee its required bit‐error‐rate (BER) and transmission rate. Based on the remaining subcarriers with high channel gains and the transmission power, the throughput of data transmission is then maximized by loading as many bits as possible on each subcarrier to achieve the required transmission bit rate and BER. Theoretical analysis and simulation on the proposed algorithm show that a better performance is obtained than previously reported schemes. Copyright © 2008 John Wiley & Sons, Ltd.     

Comparative performance of voice/data local area networks

Using simulation, a network-independent framework compares the performance of contention-based Ethernet and two contention-free round-robin schemes, namely Expressnet and the IEEE 802.4 token bus. Two priority mechanisms for voice/data traffic on round-robin networks are studied: the alternating-rounds mechanism of the Expressnet, and the token rotation timer mechanism of the token bus, which restricts access rights based on the time taken for a token to make one round. It is shown that the deterministic schemes almost always perform better than the contention-based scheme. Design issues such as the choice of minimum voice packet length, priority parameters, and voice encoding rate are investigated. An important aspect that is noted is the accurate characterization of performance over a wide region of the design space of voice/data networks     

GSM数据传输技术及其在野外实时数据采集系统中的应用

利用GSM网络传输野外数据采集系统数据具有非常显著的优点。本文介绍采用点对点方式进行数据采集和传输的野外数据采集系统结构、性能及GSM模块的编程控制方法。     

Self-subrating for voice/data in PCS

Bellcore Personal Access Communications System (PACS) provides the feature of using a single time-slot for two independent calls by the same user. This feature allows a simultaneous voice and data call from a single subscriber unit. This paper compares the performance of a PCS system with voice/data self-subrating (SSR) with a system without self-subrating (NSSR). We show that for the ranges of the input parameters we study, the blocking probability for NSSR is 76% higher than for SSR. For a PCS system engineered at 1% blocking probability, SSR carries 15.3% more offered load than NSSR.     

Performance analysis for voice/data integration on a finite-buffermobile system

Personal communication service (PCS) networks offer mobile users diverse telecommunication applications, such as voice, data, and image, with different bandwidth and quality-of-service (QoS) requirements. This paper proposes an analytical model to investigate the performance of an integrated voice/data mobile network with finite data buffer in terms of voice-call blocking probability, data loss probability, and mean data delay. The model is based on the movable-boundary scheme that dynamically adjusts the number of channels for voice and data traffic. With the movable-boundary scheme, the bandwidth can be utilized efficiently while satisfying the QoS requirements for voice and data traffic. Using our model, the impact of hot-spot traffic in the heterogeneous PCS networks, in which the parameters (e.g., number of channels, voice, and data arrival rates) of cells can be varied, can be effectively analyzed. In addition, an iterative algorithm based on our model is proposed to determine the handoff traffic, which computes the system performance in polynomial-bounded time. The analytical model is validated by simulation     

一种高效数话复用传输方式的实现

在多路数话复用中,提出了数话综合组帧方案,且运用缓存及设置占空指示的方法．提高了数据传输效率。利用FPGA和单片机实现了数据信号复分接、同异步信号处理、话音信令处理及数据速率自适应。     

Performance analysis of voice/data cellular CDMA with SIR-based admission control

In this paper, we analyze the performance of a signal-to-interference ratio (SIR)-based admission control strategy on the uplink in cellular code-division multiple-access (CDMA) systems with voice and data traffic. Most studies in the current literature to estimate CDMA system capacity with both voice and data traffic do not take into account admission control based on SIR constraints. Here, we present an analytical approach to evaluate the outage probability for voice traffic, the average system throughput, and the mean delay for data traffic in a voice/data CDMA system, which employs an SIR-based admission control. We make two main approximations in the voice call outage analysis-one based on the central limit theorem (CLT) and the other based on the Fenton's method. We apply the Fenton's method approximation to compute the retransmission probability and the mean delay for data traffic, and the average system throughput. We show that for a voice-only system, a capacity improvement of about 30% is achieved with the SIR-based admission control as compared with the code availability-based admission control. For a mixed voice/data system with 10 Erlangs of voice traffic, an improvement of about 40% in the mean delay for data is shown to be achieved. Also, for a mean delay of 50 ms with 10 Erlangs of voice traffic, the data Erlang capacity improves by about 50%.     

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 WIMA/SAD protocol for integrated voice/data wireless networks

In this paper, the “Wireless Integrated Multiple Access (WIMA) with Speech Activity Detector (SAD)” protocol for Time Division Duplex (TDD) and Frequency Division Duplex (FDD) hierarchies is proposed and analyzed. This scheme is based on a mixture of movable-boundary WIMA protocol and speech activity process. Both voice and data traffic are handled on a packet reservation basis. The access slot of every uplink frame is allocated on the last slot to save the waiting time of queuing data. The expected data-packet delay for fixed-boundary WIMA, movable-boundary WIMA, and WIMA/SAD protocols are evaluated. Numerical results illustrate the dependence of performance on the system parameters, and demonstrate that the WIMA/SAD protocol provides a lower expected data-packet delay than the movable-boundary WIMA protocol for the values of voice-call completion probability ( μ _v) less than 0.1. As μ _v increases, the expected data-packet delay of the WIMA/SAD protocol approaches to movable-boundary WIMA protocol. The maximum data throughput of WIMA/SAD protocol has smaller variation than that of the movable-boundary WIMA protocol when voice-call completion ratio is changed. This revised version was published online in June 2006 with corrections to the Cover Date.     

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.     

Adaptive hierarchical modulation for simultaneous voice and multiclass data transmission over fading channels

In this paper, a new technique for simultaneous voice and multiclass data transmission over fading channels using adaptive hierarchical modulation is proposed. According to the link quality, the proposed scheme changes the constellation size as well as the priority parameters of the hierarchical signal constellations and assigns available subchannels (i.e., different bit positions) to different kinds of bits. Specifically, for very bad channel conditions, it only transmits voice with binary phase-shift keying (BPSK). As the channel condition improves, a variable-rate adaptive hierarchical M-ary quadrature amplitude modulation (M-QAM) is used to increase the data throughput. The voice bits are always transmitted in the lowest priority subchannel (i.e., the least significant bit (LSB) position) of the quadrature (Q) channel of the hierarchical M-QAM. The remaining (log/sub 2/M-1) subchannels, called data subchannels, are assigned to two different classes of data according to the selected priority parameters. Closed-form expressions as well as numerical results for outage probability, achievable spectral efficiency, and average bit error rate (BER) for voice and data transmission over Nakagami-m fading channels are presented. The adaptive techniques employing hybrid binary shift keying (BPSK)/M-ary AM (M-AM) and uniform M-QAM for simultaneous voice and two different classes of data transmission are also extended. Compared to the extended schemes, the new proposed scheme is spectrally more efficient for data transmission, while keeping the same outage probability for voice and data (both classes) as the scheme employing BPSK/M-AM. The new scheme also provides, as a by-product, a spectrally efficient way of transmitting voice and a single-class data.