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.     

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     

Voice and Data on a CATV Network

A technique for transmitting voice and data on a conventional subsplit CATV network is described. The technique uses a variation on the carrier sense multiple access/collision detection protocol used in local area networks. A variation on this protocol, called movable slot TDM allows periodic voice Sources to share this network without distorting the voice signal. A scheme for sharing frequency spectrum and space is outlined that removes the distance and transmission rate constraints associated with such networks. A transmission strategy is described which allows these protocols to be used on a subsplit CATV system. The system is thus a candidate for local/metropolitan area networks Which support digital voice and data services along with analog (mostly one-way) video.     

VoIP transmission mechanism based on TCP

As the widespread employment of firewalls on the Internet, user datagram protocol (UDP) based voice over Internet protocol (VoIP) system will be unable to transmit voice data. This paper proposed a novel method to transmit voice data based on transmission control protocol (TCP). The method adopts a disorder TCP transmission strategy, which allows discontinuous data packets in TCP queues read by application layer directly without waiting for the retransmission of lost data packets. A byte stream data boundary identification algorithm based on consistent overhead byte stuffing algorithm is designed to efficiently identify complete voice data packets from disordered TCP packets arrived so as to transmit the data to the audio processing module timely. Then, by implementing the prototype system and testing, we verified that the proposed algorithm can solve the high time delay, jitter and discontinuity problems in standard TCP protocol when transmitting voice data packets, which caused by its error control and retransmission mechanism. We proved that the method proposed in this paper is effective and practical.     

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     

Internet telephony: going like crazy

Interet telephony is possibly the fastest-growing part of communications today. This article discusses what exactly it is, who needs it, and how it works. Internet telephony, or voice over Internet protocol (VoIP), is the provision of phone service over the Internet. But in sharp contrast with conventional telephony, it carries voice traffic as data packets over a packet-switched data network instead of as a synchronous stream of binary data over a circuit-switched, time-division multiplexed (TDM) voice network. There are some substantial benefits (as well as some sticky problems) to the scheme, which is why companies and individuals are finding it increasingly attractive     

Voice/data channel access integration in TDMA digital cellularnetworks

The integration of digital data capabilities in the soon to be widely deployed digital cellular networks, which were primarily designed for voice communications, offers a low-cost way to capture the large and ever growing market for mobile data services. The authors propose and evaluate a multiaccess protocol for integrating data traffic in the E(nhanced)-TDMA voice system with digital speech interpolation, which is an enhancement of the emerging North American digital cellular standard. The proposed, protocol combines random access with slot reservation mechanisms to statistically multiplex data packets with speech spurt packets over the shared terminal-to-base air channel. The integrated protocol requires no modification in the voice access protocol used in the E-TDMA system, and can attain performance close to that of an ideal voice/data multiplexer. Furthermore, the protocol may enable multislot assignment per TDMA frame to match the throughput needs of individual data terminals, and can accommodate application-dependent data transmission priorities     

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     

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     

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     

An integrated CDMA code assignment model for voice and data traffic in wireless communications

Wireless personal communication requires a provision of integrated services of multimedia traffic, such as voice and data, over the radio link. The multiple access protocols of code-division multiple-access (CDMA) techniques have been widely investigated in the recent literature. This paper presents an innovative multiple access protocol for CDMA-based wireless communication systems by fully utilizing the characteristics of voice and data traffic. In other words, a voice terminal can reserve a spreading code to transmit packets in multiple talk spurts, while a data terminal can transmit packets by either using the unassigned codes or borrowing the codes from the voice terminals during their silent periods. We build mathematical models for voice and data subsystems, respectively. Two performance parameters, the average dropping probability for voice packets and the average transmission delay for data packets, are derived based on the equilibrium point analysis. The effects of the two performance parameters on the system performance are discussed by varying the code reservation intervals of the voice terminals.     

A Priority Based Opportunistic Routing Mechanism for Real-Time Voice Service in Mobile Ad hoc Networks

Traditional routing protocols send traffic along pre-determined paths and have been shown ineffective in coping with unreliable and unpredictable wireless medium which is caused by the multi-path fading. The most difference between the opportunistic routing and the traditional routing mechanism is that the opportunistic routing mechanism can use several lossy broadcast links to support reliable transmission. In this paper, an opportunistic routing mechanism for real-time voice service is proposed. This mechanism is based on the dynamic source routing (DSR) protocol with some modifications, the routing messages of DSR are used to construct the forwarder list, which guides the data packets forwarding process. The forwarder nodes have priorities to restrict the number of duplicated packets. Simultaneous flows can be supported well by our mechanism. Simulations show that our mechanism can effectively decrease the data packets transmission times and the amount of the control messages and reduce the end-to-end delay for real-time voice service, the quality of service can be supported well over the unstable wireless channel.     

TDM over IP的发展趋势及其应用

论述了TDM over IP技术出现的背景、实现过程以及实际应用方案.利用伪线路仿真技术实现在以太网上传送实时业务,将语音、图像等信息直接装入以太网(或IP网)数据包,然后依照IP路由进行简单高效的传送.此技术可广泛应用于PON、基站回传、接口转换以及传统业务线路的改造等方面.     

Voice over ATM using AAL2 and bit dropping: performance and calladmission control

Asynchronous transfer mode (ATM) adaptation layer 2 (AAL2) has been designed for efficient transport of voice, fax, and voiceband data (VBD) traffic over an ATM virtual circuit. The protocol helps achieve low latency and high bandwidth efficiency while applying suitable compression methods on voice/VBD/fax calls and silence elimination on voice calls. We analyze the performance and capacity of an ATM multiplexer based on AAL2 adaptation. We assume that embedded adaptive differential pulse code modulation (ADPCM) is used to compress voice, and silence elimination is used to achieve statistical multiplexing gain. The embedded ADPCM coding scheme allows selective dropping of less significant bits of voice during congestion in the ATM/AAL2 multiplexer. We compare the call capacities of voice multiplexers with and without bit dropping (BD). The performance models and results presented are based on fairly general assumptions and can be used for traffic engineering and call admission control in land-line or wireless ATM systems for a variety of voice/voiceband compression algorithms. A generalized algorithm for call admission control is also described     

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     

An investigation of reservation random access algorithms for voice-data integration in microcellular wireless environments

We present the results of a simulation study that explores the performance of two promising reservation random access (RRA) protocols for transmitting voice packets over a common radio broadcast channel in a microcellular radio environment. We examine two inherently stable RRA voice protocols, RRA three cell and RRA two cell, with respect to voice transmissions under ideal and adverse channel conditions. In addition, we investigate the ability of both protocols to support efficient voice-data integration within the system. The RRA two-cell and RRA three-cell algorithms clearly mark the end of the voice contention period, thereby enabling all of the terminals within the microcell to differentiate between available voice and available data slots. Separating the two distinct types of transmissions and resolving the contending voice packets first thus enforces the priority of the voice traffic. In addition, each protocol can be combined with efficient, easy to implement, collision resolution random access protocols for transmitting data packets. Such a voice-data integration mechanism eliminates the potential voice degradation caused by competition between voice and data terminals for available slots. Our results show that the protocols provide stable and robust performance under adverse channel conditions and that they can be employed to sustain voice-data integration under heavy system loading.     

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 packet reservation multiple access (PRMA)-based algorithm for multimedia wireless system

The major issue in the wireless multimedia system design is the selection of a suitable channel sharing media access control (MAC) protocol. The design challenge is to identify a wireless "multimedia capable" MAC protocol that provides a sufficient degree of transparency for many different kinds of services. This protocol should guarantee different quality of service (QoS) parameters for different types of traffic while in the same time achieving high throughput. In this paper a MAC protocol to serve different kinds of traffic, namely voice, data, and, real time variable bit rate (rt-VBR) video is proposed. The transmission time scale is divided into frames. Each frame is subdivided into N time slots. In this protocol, a fixed number of slots M out of 150 time slots are reserved at the beginning of every frame to transmit some of the video packets arriving during the frame interval. The rest of the video packets contend with the voice and data packets for the remaining time slots of this frame as in normal packet reservation multiple access (PRMA). One objective of this paper is to find the optimum value of M allowing the maximum number of voice and data users to share the RF channel with one video user. Another objective is to find the optimum permission probabilities of sending contending voice, data, and video packets allowing the maximum number of users sharing the RF channel. The dropping probability requirement for video is examined.     

支持话音/数据分组并传的UPMA多址接入协议

本文提出了一种新的、支持数据/话音业务并传的多址接入协议——根据用户数目妥善安排分组传输的多址接入(User-dependent Perfect-scheduling Multiple Access——UPMA)协议,它根据实际需求对上、下行带宽资源实行动态分配.UPMA协议对不同的业务类型赋予不同的优先级,并用轮询方式妥善地安排节点的分组传输;同时,它采用独特的帧结构,使话音业务总是能够得到优先传输.本文还提出了一种高效的竞争接入算法,以保证激活的节点能够快速接入信道.最后,对UPMA协议的性能进行了仿真并与MPRMA协议的性能进行了比较,结果证明UPMA协议有更好的性能.     

A collisionless multiple access protocol for a wavelength divisionmultiplexed star-coupled configuration: architecture and performanceanalysis

A collisionless wavelength-division multiple-access (WDMA) protocol for a passive star-coupled photonic network is introduced and shown to possess significant performance and flexibility advantages. A performance modeling technique based on a semi-Markov analytic model, which eliminates many of the unrealistic assumptions of past approaches, is introduced. The performance of the protocol is analyzed using this model and discrete-event simulation. Control channel access arbitration is achieved through time-division multiplexing (TDM), enabling all active nodes to transmit once every control cycle. The long synchronization delays typical of TDM systems are significantly reduced, because the control cycle length is proportional to the control packet size rather than the data packet size. The protocol eliminates packet collision and variable-sized data packets are supported without utilization degradation