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.     

Rapid Access Protocols for Discontinuous Transmission in DS-CDMA Systems

In this paper, three reverse link access protocols (AP) enabling discontinuous transmission (DTX) in DS-CDMA personal communication systems are introduced and discussed. The first protocol is synchronous (SDTX-AP), and it uses a so-called synchronous reservation channel to accommodate access requests in a time slotted frame structure. The second uses an asynchronous approach (ADTX-AP) with a spread slotted ALOHA protocol for access requests; access request messages consist of a synchronizing preamble and a user identifier appendix. The third one employs a synchronous structure of overlapping slots, offset in time by a minimum interval (mini-slot) necessary to enable resolution of overlapping access probes sent by different mobile users (MSDTX-AP). Instead of using different spreading codes for different mobile transmitters, all transmitters are assigned the same spreading code to send their access request messages on the access reservation channel. Analysis considers the mean access delay and throughput of the protocols in a multipath fading channel. Numerical results indicate that at high values of offered traffic of access requests mean access delays of ADTX-AP and SDTX-AP are comparable. The mean access delay for the MSDTX-AP depends on the number of minislots or users within each frame, but its throughput is much higher than that of either SDTX-AP or ADTX-AP.     

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.     

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.     

Performance of token schemes supporting delay-constrained prioritytraffic streams

A symmetric priority-based token network is considered. Messages are divided into two priority classes. High-priority messages are assumed to require tight delay constraints. As a result, each station is allowed to establish, at any time, at most a single real-time high-priority access concentration. High-priority messages are guaranteed access onto the channel within a prescribed limited period. In turn, regular priority messages are only served when the system determines, through the repetitive use of circulating tokens (as used by the IEEE 802.5 token-ring-type protocol), that no high-priority messages are currently waiting in the system. Two token schemes employing different service disciplines are used to provide network access. Exact and approximate mean delay formulas for both message classes are derived. Numerical results are then exhibited to illustrate the network performance under various traffic conditions     

The RRA‐ISA multiple access protocol with and without simple priority schemes for real‐time and data traffic in wireless cellular systems

A multiple access protocol, based on a Reservation Random Access (RRA) scheme, is derived for a wireless cellular network carrying real-time and data traffic. Given a TDMA framed channel and a cellular structure, the aim of the protocol is that of maximizing the one-step throughput over an entire frame. This is achieved by deciding on the access rights at the cell base station, which then broadcasts this information at the beginning of the frame. The decision is made on the basis of binary channel feedback information (collision/no collision) over the previous frames, as well as of long term averages of packet generation rates at the mobile stations, assuming independence in the presence of packets at the latter. The resulting protocol has therefore been termed Independent Stations Algorithm (ISA), and the overall scheme RRA-ISA. As in other RRA protocols, time constrained (e.g., voice) traffic operates in a dynamic reservation mode, by contending for a slot in the frame with the first packet of a burst, and then keeping the eventually accessed slot for the duration of the burst; packets of the time constrained traffic unable to access a slot within a maximum delay are dropped from the input buffer. No such constraint is imposed on data traffic. Together with the “basic” version of the access algorithm, three other variants are presented, which exploit three simple different priority schemes in the RRA-ISA “basic” structure, in order to give a prominence to the voice service. The aim of these variants is to improve the performance in terms of the maximum number of stations acceptable in the system, by slightly increasing the data packets delay. All the proposed schemes are analyzed by simulation in the presence of voice and data traffic. Several comparisons show a relevant performance improvement (in terms of data delay and maximum number of voice stations acceptable within a cell) over other protocols that use ALOHA as a reservation mechanism (RRA-ALOHA or PRMA schemes). This revised version was published online in June 2006 with corrections to the Cover Date.     

Wireless Adaptive Framed Pseudo-Bayesian Aloha (AFPBA) Algorithm with Priorities

In this paper, we propose a new priority algorithm to control the access to the wireless ATM MAC uplink frame, for multimedia traffic like wireless ATM, similar to the Pseudo-Bayesian algorithm presented in [1]. The adaptive framed Pseudo-Bayesian Aloha (AFPBA) algorithm ensures minimum access delay for high priority traffic classes with small delay degradation to low priority traffic classes. Control packets are transmitted in each slot according to transmission probabilities based on the history of the channel and in contention with other packets of the same priority class. The number of contention slots assigned for each priority class, on a given frame, changes adaptively according to its priority index and the estimated arrival rate on each frame using an adaptive slot assignment mechanism. Finally, the throughput analysis of the algorithm is presented and the delay performance is evaluated by simulation on a wireless channel in the presence of shadowing, Rayleigh fading and capture. Results show that the wireless channel offers significant delay improvements to all priority packets, especially in the presence of fast fading.     

A Distributed Channel Access Scheme with Guaranteed Priority and Enhanced Fairness

Although the IEEE 802.11e enhanced distributed channel access (EDCA) can differentiate high priority traffic such as real-time voice from low priority traffic such as delay- tolerant data, it can only provide statistical priority, and is characterized by inherent short-term unfairness. In this paper, we propose a new distributed channel access scheme through minor modifications to EDCA. Guaranteed priority is provided to real time voice traffic over data traffic, while a certain service time and short-term fairness enhancement are provided to data traffic. We also present analytical models to calculate the percentage of time to serve voice traffic and the achieved data throughput. Both analysis and simulation demonstrate the effectiveness of our proposed scheme.     

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.     

Packet Voice on a Local Area Network with Round Robin Service

In this paper, we propose a combined voice/data protocol suitable for multiple access broadcast networks that provide round robin service to the stations. Such networks are well suited to the integration of voice and data since they guarantee bounded delay and provide high utilization even for high bandwidth channels. Using one such network proposal-namely Expressnet-as a representative scheme, we examine the characteristics of the service that voice traffic experiences under the voice/data protocol. We show that the access protocol is able to utilize the channel efficiently to support a large population of voice sources while maintaining low packet delay and guaranteeing some prespecified minimum bandwidth for data traffic. In addition, we show the advantages of silence suppression, i.e., discarding speech that constitutes silent periods, and we examine the cost of overloading the network in terms of the amount of speech discarded.     

AN EFFICIENT MEDIUM ACCESS SCHEME TO SUPPORT VBR TRAFFIC ON A SLOTTED RING

This paper describes a proposed efficient integrated medium access control scheme for a high-speed unidirectional slotted ring which employs destination release. The mechanism uses separate cycles of access for synchronous and asynchronous traffic and is suitable for variable bit rate (VBR) traffic. The paper includes a comparison with the Orwell protocol under various simulated traffic loads. The use of separate cycles of access was found to be more efficient than the sharing of a single access cycle, at the cost of two bits in the cell header for each cycle, an insertion buffer for each cycle apart from the highest priority cycle, and associated node complexity. The proposed mechanism was able to meet delay requirements for synchronous traffic while maintaining high levels of overall utilization, thus making it suitable for use in a ring-based asynchronous transfer mode (ATM)-compatible local area network (LAN).     

On Multiple Traffic Type Integration over Wireless TDMA Channels with Adjustable Request Bandwidth

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, video and data packet traffic over a wireless channel of high capacity (referring to an indoor microcellular environment). Depending on the number of video users admitted into the system, our protocol varies: a) the request bandwidth dedicated to resolving the voice users contention, and b) the probability with which the base station grants information slots to voice users, in order to preserve full priority for video traffic. We evaluate the maximum voice capacity and mean access delay, as well as the aggregate channel throughput, for various voice and video load conditions, and the maximum voice capacity, aggregate channel throughput and average data message delays, for various video, voice and data load conditions. As proven by the comparison with a recently introduced efficient MAC scheme (DPRMA), when integrating voice and video traffic our scheme obtains higher voice capacity and aggregate channel throughput. When integrating all three traffic types, our scheme achieves high aggregate channel throughput in all cases of traffic load.     

Discrete-Time Analysis of Integrated Voice/Data Multiplexers With and Without Speech Activity Detectors

Discrete-time analysis of two schemes for multiplexing voice and data is presented. In each scheme voice and data are multiplexed using the movable boundary frame allocation scheme. In the first scheme, speech activity detectors (SAD's) are not used, and hence, the variations in the voice traffic are only due to the on/off characteristics of voice. In the second scheme, SAD's are employed so that talker silences can he utilized for transmission of additional voice and/or data. In this scheme, the multiplexer performs digital speech interpolation as well as movable boundary frame allocation. The performance measures considered are probability of loss for voice calls, probability of speech clipping, speech packet rejection ratio, and the expected data message delay. In the case of the multiplexer with SAD, a tradeoff exists between data message delay and speech interpolation advantage. Some numerical examples are presented which illustrate the performance of the two multiplexers.     

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.     

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 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 evaluation of packet data services over cellular voice networks

In this paper we develop a Markov chain modeling framework for throughput/delay analysis of data services over cellular voice networks, using the dynamic channel stealing method. Effective approximation techniques are also proposed and verified for simplification of modeling analysis. Our study identifies the average voice call holding time as the dominant factor to affect data delay performance. Especially in heavy load conditions, namely when the number of free voice channels becomes momentarily less, the data users will experience large network access delay in the range of several minutes or longer on average. The study also reveals that the data delay performance deteriorates as the number of voice channels increases at a fixed voice call blocking probability, due to increased voice trunking efficiency. We also examine the data performance improvement by using the priority data access scheme and speech silence detection technique.     

Theoretical analyses of data communications integrated intocordless voice channels

Dynamic integration of data into voice channels of second-generation cordless systems provides effective channel utilization. This paper proposes and theoretically examines an inhibit and random multiple access (IRMA) protocol for data terminals in an integrated voice and data system by assuming that real-time voice traffic has priority over data. Analytical expressions are derived to quantify the effect of data inhibition on data performance, i.e., throughput and delay for both infinite and finite population models for data terminals. In order to find the possible data throughputs with our channel access scheme for data communication using voice channels while not affecting the voice quality, we investigate the data performance under two extreme situations: 1) no voice load and 2) full voice load. The numerical results indicate that IRMA data performance is comparable to that of slotted ALOHA (S-ALOHA) at lighter loads of less than 0.2 while efficiently sharing the network resources between real-time voice and nonreal-time data traffic. For the data loads above 0.2, IRMA pays a price, but an affordable one, in terms of throughput performance for its ability to accommodate data while assuring quality of voice even when all channels are occupied by voice traffic. An optimum number of terminals and the range of data-transmission probabilities have been deduced as 16 and 0.05-0.15, respectively, in order to achieve maximum throughput with minimum delay while maintaining stable data transmission and voice quality in an integrated system     

A Bandwidth Reservation Multiple Access Protocol for Wireless ATM Local Networks

A bandwidth reservation multiple access scheme(BRMA) is proposed to resolve contention and assignbandwidth among multiple users trying to gain access toa common channel such as in mobile users contending for resources in an ATM-based cellular networkor a wireless local area network (LAN) with shortpropagation delays. The protocol is best suited tosupport variable-bit-rate (VBR) traffic that exhibits high temporal fluctuations. Each mobile user isconnected end-to-end to another user over virtualchannels via the base station that is connected to thewired ATM B-ISDN network. The channel capacity is modeled as a time frame with a fixed duration.Each frame starts with minislots, to resolve contentionand reserve bandwidth, followed by data-transmissionslots. Every contending user places a request for data slots in one of the minislots. If therequest is granted by the base station through adownlink broadcast channel, the user then startstransmission in the assigned slot(s). The number ofassigned slots varies according to the required qualityof service (QoS), such as delay and packet lossprobability. A speech activity detector is utilized inorder to indicate the talkspurts to avoid wastingbandwidth. Due to its asynchronous nature, BRMA is ratherinsensitive to the burstiness of the traffic. Since theassignment of the minislots is deterministic, therequest channels are contention-free and the data channels are collision-free. Hence, in spite ofthe overhead (minislots) in each frame, BRMA provideshigher throughput than Packet Reservation MultipleAccess (PRMA) for the same QoS, especially for high-speed systems. A better delay performance is alsoachieved for data traffic compared to Slotted Alohareservation-type protocol PRMA. In addition, BRMAperforms better in terms of bandwidth efficiency thanthe conventional TDMA or the Dynamic TDMA, wherespeech activity detectors are very difficult toimplement.     

An Integrated Voice/Data System for VHF/UHF Mobile Radio

This paper presents the basic architecture and performance of a mobile radio multiaccess voice/data system. Natural pauses in conversational speech allow bandwidth saving through interleaving of data packets and talkspurts from different voice sources. A speech detector designed specifically for the mobile environment is presented. Blocking and delay performance of the multiaccess uplink is analyzed for voice traffic, assuming no traffic effects from the low priority data packets. Performance results from simulation are then presented for two downlink strategies in a two-hop virtual circuit in which a base station acts as a relay. The results verify also that the uplink analysis is valid for low voice traffic. For the data traffic, simulation results are presented in terms of data packet transmission delay and probability of collision with talkspurts. The results indicate that data flow may be limited by the collision factor. This work concludes that relative to conventional radio telephoning in which two channels are dedicated to each transmitter/receiver pair, a bandwidth reduction of 30-35 percent can be achieved.