An integrated services token-controlled ring network

A ring protocol is proposed that allows voice and data traffic to coexist within the ring. This ring has the following distinct features: (1) it allows synchronous traffic such as voice and video to have a definite access to the channel within each packetization period (or frame); (2) it allows data messages to have a higher channel access priority provided that the synchronous traffic is not delayed by more than one frame; (3) it supports variable rate data circuits. Simulation results show that the data-message delay is much smaller than for other integrated services schemes. Urgent messages can be transmitted with a higher priority over voice. Since the voice packet delay is bounded within one packetization period, no time-stamping is needed and the voice loss can be completely avoided by reserving a sufficient number of slots. Continual speech reception is possible by synchronizing the speech regeneration process to the end of each frame. Since the ring is synchronized, gateway switching to external circuit-switched and packet-switched networks is very simple     

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.     

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.     

A Theoretical Traffic Performance Analysis of an Integrated Voice-Data Virtual Circuit Packet Switch

A time division packet switch capable of concurrently handling both voice and data traffic is proposed, and some of its performance limitations are analyzed. The voice packet traffic is handled at a higher priority level than data traffic, in order to meet stringent timing criteria, and can be shown to be handled just as if it were circuit switched. The data traffic utilizes whatever time slots are not occupied with voice traffic. The principal performance limitations described in this exploratory study are the fraction of time the voice traffic is blocked due to all the available time slots already being used for voice traffic, and an upper bound on the mean delay encountered by the data traffic as it waits to find an available time slot. An illustrative numerical result is the following. If we assume that each voice telephone conversation lasts for a mean of five minutes, and that twenty voice calls are generated over a six hour time span, and each data session lasts for a mean of forty minutes, and that five data calls are generated over a six hour time span, then if separate line switched networks are used for voice and for data with long term blocking probability of one percent, a total of 703 64 kbits links would be required to support 461 voice stations and 882 data terminals. On the other hand, using the integrated voice/data switch described here, and if we assume that the total delay due to the switch alone for data packets cannot exceed a long term mean value of one second, then only 298 64 kbit/s links are required to support 461 voice stations and 882 data terminals, reducing the number of required links by a factor of about two. Moreover, the assumptions leading to this comparison suggest that the packet switch could in fact support significantly more than this number of voice stations and data terminals. This is achieved at the expense of additional buffering for the data in the packet switch approach.     

Output and delay process analysis for slotted CDMA wirelesscommunication networks with integrated voice/data transmission

This paper presents the output and delay process analysis of integrated voice/data slotted code division multiple access (CDMA) network systems with random access protocol for packet radio communications. The system model consists of a finite number of users, and each user can be a source of both voice traffic and data traffic. The allocation of codes to voice calls is given priority over that to data packets, while an admission control, which restricts the maximum number of codes available to voice sources, is considered for voice traffic so as not to monopolize the resource. Such codes allocated exclusively to voice calls are called voice codes. In addition, the system monitoring can distinguish between silent and talkspurt periods of voice sources, so that users with data packets can use the voice codes for transmission if the voice sources are silent. A discrete-time Markov process is used to model the system operation, and an exact analysis is presented to derive the moment generating functions of the probability distributions for packet departures of both voice and data traffic and for the data packet delay. For some cases with different numbers of voice codes, numerical results display the correlation coefficient of the voice and data packet departures and the coefficient of variation of the data packet delay as well as average performance measures, such as the throughput, the average delay of data packets, and the average blocking probability of voice calls     

Design and performance 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.     

On protective buffer policies

Studies buffering policies which provide different loss priorities to packets/cells, while preserving packet ordering (space priority disciplines). These policies are motivated by the possible presence, within the same connection, of packets with different loss probability requirements or guarantees, e.g., voice and video coders or rate control mechanisms. The main contribution of the paper is the identification and evaluation of buffering policies which preserve packet ordering and guarantee high priority packets performance (loss probability), irrespective of the traffic intensity and arrival patterns of low priority packets. Such policies are termed protective policies. The need for such policies arises from the difficulty to accurately characterize and size low priority traffic, which can generate large and unpredictable traffic variations over short periods of time. The authors review previously proposed buffer admission policies and determine if they satisfy such “protection” requirements. Furthermore, they also identify and design new policies, which for a given level of protection maximize low priority throughput     

Integrated voice/data multiplexers with slotted and nongated packetservices

Integrated voice/data multiplexers that provide packet services for both voice and data traffic are discussed. A slotted service is assumed, so that packet transmissions are synchronized to slot boundaries. Nongated service, in which packets are transmitted as soon as the transmission capacity becomes available, is also assumed. The performance of nongated and slotted multiplexers is obtained by analytic and simulation approaches. In particular, a PRIO (head-of-the-line priority to voice packets) and a BVFD (busy-voice, fixed-data) multiplexer are shown to be suitable for such a nongated environment     

A Channel Access Protocol for Integrated Voice/Data Applications

A new multiaccess protocol is proposed for an integrated voice/data application. The protocol, which is a variation of virtual time CSMA (VT-CSMA), takes advantage of the periodicity of voice packets and possesses a number of important features. With this protocol, voice stations appear to have a dedicated time-division multiplexed (TDM) slot, and the delay of a voice packet is bounded by the length of a frame (defined to be the period between two consecutive voice packets from a voice station). Also, the amount of data added to the channel has little effect on the voice traffic. When silence detection is used, many more voice conversations can be supported without losing the dedicated-slot characteristic. This is in contrast to a movingboundary TDM system where the excessive bandwidth saved by silence detection can only be used for data. The protocol requires no global synchronization and is easy to implement. Simulation results are presented to evaluate its performance.     

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.     

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.     

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.     

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     

Comparative performance of voice/data local area networks

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

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     

C-PRMA: a centralized packet reservation multiple access for localwireless communications

Packet-switched technology has been demonstrated as effective in cellular radio systems with short propagation delay, not only for data, but also for voice transmission. In fact, packet voice can efficiently exploit speech on-off activity to improve bandwidth utilization over time division multiple access (TDMA). Such an approach has been first suggested in the packet reservation multiple-access (PRMA) technique, an adaptation of the reservation ALOHA protocol to the cellular environment. However, being PRMA-based on a fixed frame scheme, it cannot thoroughly take advantage of the very short propagation delays encountered in microcellular systems that allow, for instance, the immediate retransmission of packets lost because of the interference noise from adjacent cells. We present the centralized PRMA, a natural enhancement of PRMA, in which the base station (BS) plays a central role in scheduling the transmissions of mobile stations (MSs). As a consequence, the transmission scheduling is very flexible and can account for the different traffic rate and delay constraints that emerge from voice and data integration. A packet retransmission policy to recover corrupted packets can be implemented and operated efficiently to provide an acceptable grade of service, even in a very noisy environment. The simulation results presented show the quantitative improvements of the centralized packet reservation multiple-access (C-PRMA) performance with respect to PRMA     

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.     

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.     

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     

Performance measures and scheduling policies in ring networks

A unidirectional ring network is considered. A node may transmit at most one packet per slot to its downstream neighbor. Potentially all nodes may transmit at the same slot. The achievable performance is studied and policies are proposed for both the evacuation mode and continual operation. In the evacuation mode each node has initially an amount of packets destined for every other node of the ring, and no more packets are generated later. It is shown that the furthest destination first (FDF) policy, that gives priority to the packet with the longest way to go at each node, minimizes the time until every packet reaches its destination. Furthermore it is shown that the closest destination first (CDF) policy, that gives priority to the packet with the shortest way to go at each node, minimizes the average packet delivery time. A formula for the optimal evacuation time is obtained. The continual operation of the ring is considered then where packets are generated according to some arrival process. For any arrival sample path, the PDF maximizes the fraction of the time at which the ring is empty. The performance analysis of individual origin-destination traffic streams under FDF is facilitated based on the following. For each traffic stream, a single server priority queue is identified such that the average sojourn time of the traffic stream in the ring is equal to the aggregate transmission time plus the queueing delay of the low priority stream in the queue. Formulas for the sojourn time are obtained for iid arrivals. The performance of CDF and FIFO in continual operation is studied by simulation. It turns out that the CDF, minimum delay policy for the evacuation, has the worst performance in continual operation