Incremental time-slot assignment in SS/TDMA satellite systems

The heterogeneous traffic in this environment can be categorized into a rapidly changing type composed of packet switched data traffic and a relatively static type composed of circuit switched voice traffic. From the time-slot assignment viewpoint, the problem is to construct an efficient TDMA frame that permits the static voice traffic to be transmitted and, then, on a frame-by-frame basis to attempt to insert the data packets into the slots that are unused by the voice traffic. It is proved that the problem is NP-complete, even for very simple traffic configurations. Several suboptimal fast heuristic algorithms are presented and empirically compared by experiments on randomly generated traffic patterns. The experiments reveal that, on the average, the algorithms give close to the optimal performance     

Efficiency of the GSM-GPRS Air Interface for Real-Time IP Traffic Flows With and Without Packet Dropping

One IP terminal can occupy a single slot or a multiple number of slots within time frames in the GSM and GPRS, respectively. A limited number of radio resources (slots) are allocated in a base station for such IP terminals. If one IP terminal can occupy only one slot discontinuously in a time frame, there is one possibility resorting to all IP terminals to preserve active mode at a time. Thus, the number of accepted call in the GSM is the same as that of the radio resource. Similarly, if one terminal can occupy a multiple number of slots discontinuously/dynamically in a time frame, the number of accepted calls is obtained by dividing the number of radio resources during that time by the maximum allowed number of slots per IP terminal. A burstiness factor is defined for the IP traffic over GSM-GPRS air interface. Traffic channel efficiency with a bursty real-time IP traffic is unacceptably low, especially with the range of acceptable call loss probabilities pertaining to a lower burstiness factor. The channel efficiency can be enhanced and the call loss probability can be suppressed significantly if a higher maximum number of calls is accepted. Allocated radio resources are less than the maximum number of packet transmissions at a time. Therefore, some packets could be dropped from the real-time transmission system. A complete analysis for the real-time IP packet transmission over the single slot GSM and dynamically variable multislot GPRS air interface without packet dropping, and with packet dropping that increases the channel efficiency is executed. Results show that the channel efficiency as well as the packet dropping probability increases with increasing call intensity, maximum number of admitted IP calls and the burstiness factor.     

话音/数据综合的多时隙预约多址协议及其性能分析

本文提出了一种综合话音和数据的多时隙预约多址协议.该协议在保证话音终端的优先权的情况下,允许数据终端在报文的传输期间在连续多个帧中预约多个信息时隙.文中对协议进行了理论分析,并推导出了协议的重要性能指标(如话音分组丢失率、数据报文平均接入时延、系统平均吞吐率等)的解析表达式.研究表明,该协议可以支持比IPRMA、NC-IPRMA更高的等效数据终端速率,而且系统平均吞吐率在很大的负载范围内接近最大值.     

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.     

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 reservation protocol for packet voice and data integration inunidirectional bus networks

A movable boundary protocol is proposed for integrating packet voice and data in unidirectional bus networks. The head station on the bus learns the number of ready-to-transmit voice stations by reading a `request' bit in the header of the received packets and allocates the exact number of voice slots needed in each frame. The protocol guarantees that the maximum delay to transmit a voice packet will be less than the round-trip propagation delay at the head station plus twice the time needed to form the packet. The average data packet delay is evaluated via approximate analysis and simulation, for the case in which the voice-reserved slots in a frame are contiguous and for the case in which they are evenly distributed     

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.     

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     

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.     

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     

Expressnet: A High-Performance Integrated-Services Local Area Network

Expressnet is a local area communication network comprising an inbound channel and an outbound channel to which the stations are connected. Stations transmit on the outbound channel and receive on the inbound channel. The inbound channel is connected to the outbound channel so that all signals transmitted on the outbound channel are duplicated on the inbound channel, thus achieving broadcast communication among the stations. In order to transmit on the bus, the stations utilize a distributed access protocol which achieves a conflict-free round-robin scheduling. This protocol is more efficient than existing round-robin Schemes as the time required to switch control from one active user to the next in a round is minimized (on the order of a carrier detection time), and is independent of the end-to-end network propagation delay. This improvement is particularly significant when the channel data rate is so high, or the end-to-end propagation delay is so large, Or the packet size is so small as to render the end-to-end propagation delay a significant fraction of, or larger than, the transmission time of a packet. Moreover, some features of Expressnet make it particularly suitable for voice applications. In view of integrating voice and data, a simple access protocol is described which meets the bandwidth requirement and maximum packet delay constraint for voice communication at all times, while guaranteeing a minimum bandwidth requirement for data traffic. Finally, it is noted that the voice/data access protocol constitutes a highly adaptive allocation scheme of channel bandwidth, which allows data users to recover the bandwidth unused by the voice application. It can be easily extended to accommodate any number of applications, each with its specific requirements.     

Integrated video and voice transmission using packet reservation multiple access

This paper proposes a new protocol for the integration of voice and video transmission over the packet reservation multiple access (PRMA) system that is a modification of reservation‐ALOHA protocol. We focus on low bit‐rate video applications like video conferencing and visual telephony for wireless communications. The ITU–T H.263 standard provides a solution to the need for low bit‐rate video compression under 64 kbytes/s. The proposed protocol assumes that each voice terminal follows a traffic pattern of talk spurts and silent gaps with fixed permission probability (p=0.3), and each video terminal has the higher permission probability (p=1) to access the available slot based on ITU–T H.263 standard. Again, we present a ‘pseudo‐reservation’ scheme to release slots reserved by video terminals according to the contents of each video transmission buffer, and the active voice terminals can temporarily access the additional slots to improve the performance without sacrificing the video capacity of the system. The packet dropping probability of the active voice terminals and bandwidth utilization of the system are superior to the original PRMA, as indicated in simulation results. Copyright © 2001 John Wiley & Sons, Ltd.     

Performance of a Dynamic TDMA System for Mobile Wireless Access

A dynamic TDMA system can utilize voice activityand allow the integration of voice and data traffic.This can be achieved by allocating frequency channelsand time slots on demand. In this approach, upon the arrival of a talkspurt or a data packet,the base station is requested to assign a time slot foreach transmission. Message requests and assignments ofmobile users are carried over a Control channel, while the voice and traffic are transmittedover a Traffic channel. Time slot assignments are madefrom a pool of Traffic channels. A numberof slots in the pool will be shared by voice and data, with voice having priority over data, andthe remaining will be used by data only. Voice slots arereserved for the duration of the talkspurt whereas datapackets are assigned on a per-slot basis. Data packets can be buffered whereas voicetraffic can only tolerate limited delay beyond whichtalkspurts will be clipped off. The Control channeluplink access is based on Slotted Aloha so that mobile users have autonomous access to base stations.This paper presents the performance of the dynamic TDMAsystem outlined here. The analysis aims at assessing thecapacity gained by using voice activity and voice/data integration, in terms of theimpairments introduced to voice quality (e.g., speechclipping and/or delay) and the delays to data packets.The analysis has been based on a discrete time Markov model operating on a frame-by-frame basis thatprovides the joint distribution of the number of activevoice and data users in the system. The analysis alsoevaluates the delays of message requests via the uplink control channel. In evaluating theclipping probability, we combine the impact of both theaccess delays at the control channel as well as theunavailability of time slots in the pool. Performance results indicate that the capacity gain mayexceed 80% and the speech clipping can be kept below 1%.Also, data packets may be transmitted with limiteddelays even when all capacity is allocated for voice users. The proposed approach may be used toenhance the capacity of the existing TDMA cellularsystems and to provide integration of voice and dataservices.     

Performance Evaluation of an Integrated Access Scheme in a Satellite Communication Channel

A method for realizing a circuit and packet integrated access scheme in a satellite communication channel is considered. Two kinds of terminals are assumed, namely, bursty terminals for handling bursty traffic and heavily loaded terminals for long-holdingtime message traffic. In this method, the channel frame is divided into two subframes: one is for bursty terminals, and the other is for heavily loaded terminals. The subframe for heavily loaded terminals is further divided into two subchannels, a reservation subchannel (consisting of small slots) and a message subchannel. The bursty terminals transmit their packets in their dedicated subframes on the slotted ALOHA protocol. The heavily loaded terminal having a message transmits, first of all, a reservation packet in a randomly selected small slot of the reservation subchannel to reserve slots in the coming message subchannels. One slot in the same position of each of the succeeding message subchannels is reserved for the terminal until the end-of-use flag, transmitted from the terminal, is received by the satellite. Mean transmission delays for both kinds of traffic in this method are analytically obtained. We show that there exists an optimal frame length which minimizes mean transmission delay for one kind of traffic while keeping mean transmission delay for the other kind under some permissible value.     

Highly Efficient Voice—Data Integration over Medium and High Capacity Wireless TDMA Channels

A new medium access control (MAC) protocol for mobile wireless communications is presented and investigated. We explore, via an extensive simulation study, the performance of the protocol when integrating voice and data traffic over two wireless channels, one of medium capacity (referring mostly to outdoor microcellular environments) and one of high capacity (referring to an indoor microcellular environment). Data message arrivals are assumed to occur according to a Poisson process and to vary in length according to a geometric distribution. We evaluate the voice packet dropping probability and access delay, as well as the data packet access and data message transmission delays for various voice and data load conditions. By combining two novel ideas of ours with two useful ideas which have been proposed in other MAC schemes, we are able to remarkably improve the efficiency of a previously proposed MAC scheme [5], and obtain very high voice sources multiplexing results along with most satisfactory voice and data performance and quality of service (QoS) requirements servicing. Our two novel ideas are the sharing of certain request slots among voice and data terminals with priority given to voice, and the use of a fully dynamic low-voice-load mechanism.     

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.     

Block Reservation Time Division Multiple Access (BRTDMA) Protocol for a High Capacity Wireless Network

Next generation high capacity wireless networks need to support various types of traffic, including voice, video and data, each of which have different Quality of Service (QoS) requirements for successful transmission. This paper presents an advanced reservation packet access protocol BRTDMA (Block Reservation Time Division Multiple Access) that can accommodate voice and data traffic with equal efficiency in a wireless network. The proposed BRTDMA protocol has been designed to operate in a dynamic fashion by allocating resources according to the QoS criteria of voice and data traffic. Most of the existing reservation protocols offers reservation to voice traffic while data packets are transmitted using contention mode. In this paper we propose a block reservation technique to reserve transmission slots for data traffic for a short duration, which minimizes the speech packet loss and reduce the end-to-end delay for wireless data traffic. The optimum block reservation length for data traffic has been studied in a cellular mobile radio environment using a simulation model. Simulation results show that the BRTDMA protocol offers higher traffic capacity than standard PRMA protocol for integrated voice and data traffic and offers flexibility in accommodating multimedia traffic.     

Time slot switching for integrated services in fiber optic PBX/LAN

A medium-access protocol called time-slot switching (TSS) is proposed for use in optical-fiber local area networks. This protocol incorporates features of time division, space division, and time compression for users to share a common medium. Very-large-integration (VLSI) CMOS electric crosspoints are used to switch traffic within individual time slots. With these features, data, voice, and video services can all be combined in a single network. In addition, the speed of the electronics can be maximized to match the available optical bandwidth. Operational principles of the TSS protocol are explained. A performance analysis is presented to show the tradeoffs among traffic capacity, frame guard time, blocking probability, and the results show that TSS is more attractive than broadcast protocols for voice traffic or constant-rate data traffic. An approach to integrating voice, data, and video traffic within TSS is also described     

Admission policies for integrated voice and data traffic in CDMApacket radio networks

The authors derive optimal admission policies for integrated voice and data traffic in packet radio networks employing code division multiple access (CDMA) with direct-sequence spread spectrum (DS/SS) signaling. The network performance is measured in terms of the average blocking probability of voice calls and the average delay and packet loss probability of data messages. The admission scheme determines the number of newly arrived voice users that are accepted in the network so that the long-term blocking probability of voice calls is minimized. In addition, new data arrivals are rejected if the mean delay or the packet loss probability of data exceeds a desirable prespecified level. A semi-Markov decision process (SMDP) is used to model the system operation. Then, a value iteration algorithm is used to derive the optimal admission control. Two models for the other-user interference of the CDMA system are considered: one based on thresholds and another based on the graceful degradation of the CDMA system performance, and their performance is compared. These admission policies find application in emerging commercial CDMA packet radio networks including cellular networks, personal communication networks, and networks of LEO satellites for global communications     

Variable Bit Rate Voice Using Hysteresis Thresholds

In this work we address the problem of statistically multiplexing a variable number of telephone calls via a limited number of channels. Terminals operate with voice activity and silence detectors, and the speech is encoded to a bit rate which is system state dependent. Growing from zero, calls in the system are admitted with a maximum bit rate (maximum quality) until the cutoff fraction of talkspurt (a front end clipping) reaches a certain threshold. At this point the voice bit rate of all transmitting terminals is reduced. If due to traffic fluctuations, the number of calls in progress decreases, then all voice terminals are allowed to operate at the maximum bit rate again. In order to avoid annoying effects to listeners, both the percentage of voice information that is lost and the mean number of changes in the bit rate per mean call holding time are constrained. The first constraint is strongly dependent on the encoding bit rate, and the second one is controlled by using a hysteresis threshold when switching from one bit rate to another. In this work we have used three encoding bit rates, high, medium and low. A birth–death Markov process is used to model the system, which provided exact numerical evaluations for the percentage of time the system operates in each encoding bit rate and for the mean number of changes in the bit rate. Metrics are defined to measure the percentage of both types of voice information that are lost (not transmitted), cutoff or front end clipping and uniform dropping. Finally, an illustrative example is reported.