An approximate analysis of TDMA out-of-slot random access protocols for microcellular mobile communications

In this paper, out-of-slot random access protocols for voice services that operate in microcellular environment are studied and simulated. The bearer service is assumed to be structured as time division multiple access/frequency division multiple access/frequency division duplex (TDMA/ FDMA/FDD). According to a stratification of information flow ascall, talkspurt, andpacket, the protocols are implemented at the talkspurt level. During a call, talkspurts generate a stream of packets. Each talkspurt has to reserve a voice time slot with a special control packet sent in a dedicate control slot (out of slot signaling). After a successful access, a voice slot is assigned for the duration of the talkspurt. This work concentrates on the out of slot random access method. When a transition from the idle state to the active state occurs, a voice terminal starts generating a talkspurt. Access for a voice slotV is then initiated via a dedicated control slotC. The time spent in gaining aV slot depends on the kind of random access protocol used in theC slots. Once the access reservation phase is successful, the talkspurt starts the second phase of information transmission in a freeV slot. If allV slots are occupied by other talkspurts, the new talkspurt is queued until aV slot becomes free. If the sum of the access and queueing times exceeds a thresh-old, a portion of the talkspurt is clipped. In our work we define an analytical model to evaluate the percentage of clipped voice packets. Simulations validate the analytical model.The second version of this work was rewritten while the author was a visiting scholar at WINLABThe IS-54 standard itself has the TDMA/FDMA structure. The ETDMA enhancement appears to be very much like what is described in this paper.     

A demand assignment system for mobile users in a community ofinterest

The paper presents a high performance wireless access and switching system for interconnecting mobile users in a community of interest. Radio channel and time slot assignments are made on user demand, while the switch operations are controlled by a scheduling algorithm designed to maximize utilization of system resources and optimize performance. User requests and assignments are carried over a low-capacity control channel, while user information is transmitted over the traffic channels. The proposed system resolves both the multiple access and the switching problems and allows a direct connection between the mobile end users. The system also provides integration of voice and data traffic in both the access link and the switching equipment. The “movable boundary” approach is used to achieve dynamic sharing of the channel capacity between the voice calls and the data packets. Performance analysis based on a discrete time Markov model, carried out for the case of optimum scheduling yields call blocking probabilities and data packet delays. Performance results indicate that data packets may be routed via the exchange node with limited delays, even with heavy load of voice calls. Also the authors have proposed scheduling algorithms that may be used in implementing this system     

Packet dropping analysis for reservation voice TDMA systems

We analyze the packet dropping performance in an ideal reservation time-division multiple-access (TDMA) multiplexing voice system with the focus on the probability distribution of the number of packets dropped in a particular talkspurt. Our analysis method is based on the discrete three-state Markov speech model which corresponds to the voice source equipped with a fast speech activity detector (SAD). The numerical results for a system of 14 slots/frame reveal that the probability of losing several packets in a talkspurt is much higher than expected (by geometric distribution) and thus not negligible     

System Capacity and Access Control for Multimedia TDMA/SS Networks

In this paper, we study the system capacity and access control for the TDMA/SS (time division multiple access with spread spectrum) cellular networks supporting multimedia services. In the TDMA/SS system, time is divided into frames and each frame is further divided into slots. Only one user is allowed to transmit in a slot and spread spectrum technique is adopted to combat inter-cell interference. A packet can occupy more than one slot, depending on the user's data rate and quality of service requirement. We derive a necessary and sufficient condition for a group of users to be admissible for the TDMA/SS system and prove that its admission region contains that of the TDMA/CDMA system. In the TDMA/CDMA system, time is also divided into frames and each frame consists of several slots. The difference is that every packet occupies exactly one slot and multiple users can transmit their packets in the same slot. Numerical results show that the admission region of the TDMA/SS system can be significantly larger than that of the TDMA/CDMA system. To further increase bandwidth utilization and guarnatee delay bound requirements, several access control schemes are proposed. Simulation results are obtained for these access control schemes.     

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.     

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.     

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     

Admission control in T/CDMA systems supporting voice and dataapplications

This paper analyses the behavior of two hybrid time-code division multiple-access (T/CDMA) architectures on the up-link of a macrocellular mobile radio system. For the examined schemes two categories of users-voice and data-share the domain of available resources, made up of time slots and codewords, through two alternative assignment strategies. Both solutions attribute voice users one single resource pair, i.e., one time slot and a single code to employ on that time slot, but differ in the way data users requests are accommodated: they are either simultaneously granted several codes over the same time slot or are assigned a single code over several distinct time slots. Call admission control is performed by a channel assignment algorithm which dynamically attributes resources only if specified levels of transmission quality are met on the radio channels. The blocking and the outage probability of the two classes of users are determined and compared, showing that one of the proposed schemes exhibits better performance and allows to satisfyingly serve a significant percentage of data users     

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.     

PROTON: a media access control protocol for optical networks withstar topology

A new multiple access protocol called PROTON (PROTocol for Optical Networks) is developed for optical local area networks based on a passive star topology. PROTON uses wavelength division multiplexing (WDM) and is highly bandwidth-efficient. One of the available wavelengths is used as a control channel. Time is divided into fixed-sized slots. The size of the slots is the same for the control and the data channels. Before transmitting a packet, a station must compete with others for a slot in a data wavelength, using a collision-free procedure. Transmitting stations and the corresponding wavelengths for their data transmissions are determined at each station by a simple arbitration scheme. The protocol is suitable for networks where the number of users can be much larger than the number of available data channels. In addition to propagation delays, it is considered that transmitter and receiver tuning times as well as the times required to process control packets are not negligible. Whenever possible, and to maximize the throughput of the network, tuning and processing times of transmitters and receivers are overlapped with each other and with data transmission times. Also, data slot requests and packet transmissions are scheduled in a pipeline fashion, thus reducing the detrimental effects on throughput and packet delay of long propagation delays. The paper includes an analysis of the maximum throughput characteristics of PROTON. An analytical model is developed, and several performance measures are obtained     

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.     

Design and performance evaluation of a MAC protocol for wireless local area networks

We propose and analyze, from a performance viewpoint, a Medium Access Control (MAC) protocol for Wireless Local Area Networks (WLANs). The protocol, named Prioritized-Access with Centralized-Control (PACC), supports integrated traffics by guaranteeing an almost complete utilization of network resources. The proposed protocol combines random access for signalling, with collision-free access to the transmission channel. The transmission channel is assumed to be slotted, with slots grouped into frames. Access to transmission slots is controlled by a centralized scheduler which manages a multiclass queue containing the users' requests to access the transmission channel. Three classes of users are assumed: voice traffic (voice), data traffic with real-time constraints (high-priority data), and classical data traffic (low-priority data). A priority mechanism ensures that speech users have the highest priority in accessing the idle slots, since speech packets have a more demanding delay constraint. The remaining channel bandwidth is shared fairly among the high-priority data terminals. The low-priority data terminals use the slots left empty by the other classes. Specifically, access to transmission slots is controlled by the centralized scheduler by managing a transmission cycle for each class of terminals. The voice-terminals cycle has a constant length equal to one frame, while the lengths of the data-terminals cycles are random variables which depend on the number of active voice and data terminals. In this paper we show that the proposed scheme can support the same maximum number of voice terminals as an ideal scheduler, while guaranteeing an almost complete utilization of network capacity. In addition, via a performance analysis, we verify that by limiting the number of real-time data terminals in the network this class of traffic can be statistically guaranteed access delays in the order of 200–300 msec. Hence, the QoS the network gives to the real-time data terminals makes this service suitable for real-time applications such as alarms or low bit rate video. This revised version was published online in June 2006 with corrections to the Cover Date.     

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     

分布式TDMA网络的时隙设计技术

时隙分配是分布式TDMA组网的一个基础性工作,直接影响到Adhoc网络的性能。为支持低延迟数据业务,通常选择TDMA组网技术,但常规的固定时隙TDMA网络对时隙的利用率较低。提出了一种基于跳频的固定时隙分配和动态时隙分配相结合的TDMA时隙资源管理技术,有效地解决了TDMA网络的不同类型用户信息传输时隙分配问题。     

TDMA卫星通信系统中一种话音业务统计复用算法

本文提出了一种用于计算统计复用ＴＤＭＡ卫星通信系统中话音业务应申请时隙数的算法，给出了计算公式，并进行了分析、计算和模拟。结果证明，采用这种时隙计算算法可在用户能接受的信元丢失概率下，达到比传统ＴＤＭＡ／ＤＡ方案高许多的时隙利用率，同时具有差不多的时延性能。     

Characterization of Arrival Statistics of Multiplexed Voice Packets

An assumption that voice packets arrive by a Poisson arrival (or an exponential interarrival) distribution has not widely been accepted by analysts from an obvious observation that voice packets are generated at regular intervals in a talkspurt. Through a simulation, supported by an approximate analysis, this paper claims that when a sufficiently large number of voice sources are multiplexed, packets observe an exponential or hypoexponential interarrival distribution. When the number of packets arriving during a single or a multiple slot is considered instead, a more liberal claim is made that regardless of the number of voice sources multiplexed, a Poisson process can be assumed with reasonable accuracy.     

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 voice and data multiple-access scheme for a land-mobile satellite system

A Land-Mobile Satellite System (LMSS) is a satellite-based communications network which provides voice and data communications to mobile users in a vast geographical area. By placing a "relay tower" at a height of 22300 mi, an LMSS can provide ubiquitous radio communication to vehicles roaming in remote or thinly populated area. LMSS is capable of supporting a variety of services, such as two-way alphanumeric service, paging service, full-duplex voice service, and half-duplex dispatch service. A Network Management Center (NMC) will handle the channel requests, channel assignments, and in general the network control functions. A pool of channels is managed at the NMC to be shared by all mobile users. An integrated demand-assigned multiple-access protocol has been developed for the experimental LMSS. The pool of channels is divided into reservation channels and information channels. The information channels can be assigned by the NMC to be either voice channels or data channels. Each mobile user must send a request through one of the reservation channels to the NMC via the ALOHA random-access scheme. Once the request is received and processed, the NMC will examine the current traffic condition and assign an information channel to the user. NMC will periodically update the partitions between the reservation channels, voice channels, and data channels to optimize system performance. Data channel requests are queued at the NMC while voice channel requests are blocked calls cleared. Various operational scenarios have been investigated. Tradeoffs between the data and voice users for a given delay requirement and a given voice call blocking probability have been studied. In addition, performance impacts of such technological advancements as satellite on-board switching and variable bandwidth assignment are discussed.     

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.     

Interaction of TCP and data access control in an integrated voice/data CDMA system

This paper considers the interaction between a proposed data access control scheme and the standardized error recovery schemes on the radio link of a voice/data CDMA system. A data access control scheme for combined voice-data CDMA systems has been proposed and studied in previous literature. The scheme aims to maintain a certain target voice signal to interference ratio (SIR); this is achieved by controlling the data load according to the measured voice SIR. The data users are allowed to transmit in a radio-link time slot with a certain permission probability, which is determined by the base station based on the measured voice SIR in the previous slot. As per the IS-99 standards, however, data transmission operates under the framework of TCP, which is a higher level end-to-end protocol. The TCP data unit, called a segment, is typically equivalent to several tens of physical layer frames; hence, a segment transmission takes up several tens of slots. Due to changes in the number of voice users in talkspurt (which occur on a time scale shorter than a segment transmission time), the slot level data access control scheme can introduce significant variability in the segment transmission time. The effect of such variability on the TCP timers, which operate at the segment level, is of interest. In this paper, an approximate upper bound on the data throughput, taking the presence of TCP into account, is computed. The results provide one with an insight into the interaction of the access control scheme with TCP; they also give practical pointers as to choosing suitable parameters and operating points for the scheme. This revised version was published online in June 2006 with corrections to the Cover Date.