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     

Delay Related Issues in Integrated Voice and Data Networks

This paper focuses on network delays as they apply to voice traffic. First the nature of the delay problem is discussed and this is followed by a review of enhanced circuit, packet, and hybrid switching techniques: these include fast circuit switching (FCS), virtual circuit switching (VCS), buffered speech interpolation (SI), packetized virtual circuit (PVC), cut-through switching (CTS), composite packets, and various frame management strategies for hybrid switching. In particular, the concept of introducing delay to resolve contention in SI is emphasized, and when applied to both voice talkspurts and data messages, forms a basis for a relatively new approach to network design called transparent message switching (TMS). This approach and its potential performance advantages are reviewed in terms of packet structure, multiplexing scheme, network topology, and network protocols. The paper then deals more specifically with the impact of variable delays on voice traffic. In this regard the importance of generating and preserving appropriate length speech talkspurts in order to mitigate the effects of variable network delay is emphasized. The results indicate that a desirable length of talkspurt "hangover" of about 200 ms will accomplish this without unduly affecting speech activity, and that, under these circumstances, the perceptable threshold of variable talkspurt delay can be as high as about 200 ms average. As such, the results provide a useful guideline for integrated services system designers. Finally, suggestions are made for further studies on performance analysis and subjective evaluation of advanced integrated services systems.     

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.     

Queueing analysis of delay constrained voice traffic in a packetswitching system

Unlike data traffic, the voice packet stream from a node has very high correlation between consecutive packets. In addition, in order for the speech to be properly reconstructed, a delay constraint must be satisfied. A queueing model that accurately predicts packet loss probabilities for such a system is presented. Analytical results are obtained from an embedded bivariate Markov chain and are validated by a simulation program. Based on this model, the impact of the delay constraint, talkspurt detection thresholds, and packet size on packet loss are studied. Two schemes, named `instant' and `random', for discarding late packets are considered. Simulation results show that better performance can be obtained by using the latter scheme     

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.     

A Comparison of Measured and Calculated Speech Temporal Parameters Relevant to Speech Activity Detection

This paper deals with the measurement and calculation of various speech temporal parameters of interest in an environment where speech activity detection is employed. In particular it is shown that, based on either a measurement or model of the probability density function (pdf) for silence durations for the case of zero talkspurt "hangover" or "fill-in," that the following temporal parameters can be computed for any value of hangover or fill-in: the mean (and pdf) for silence durations, the mean talkspurt duration, the mean talkspurt rate, and the speech activity. Directly measured values of these parameters and those computed from both measured and fitted versions of the pdf for silence durations are compared and are shown to be in reasonable agreement. The illustrated results are based on measurements of about two minutes of taped male monolog source speech. However, the approach to calculating the above parameters is general in the sense that it can be applied to any measured or modeled pdf for silence durations. The significance of this work lies in the important role that talkspurt hangover plays, for example, in minimizing speech detector induced back-end clipping of talkspurts, reducing exposure to the variable talkspurt delay impairment, and in determining signaling overhead and resource occupancy in various speech interpolation, packet voice, and integrated voice/data systems.     

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     

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.     

Burst switching-an update

Burst switching research in dispersed control and integrated switching is described. Burst transport is integrated in that voice and data are switched through the same switching fabric and transmission media. Burst switching is compared to and distinguished from fast packet, fast circuit, and ATM (asynchronous transfer mode) switching. Misunderstandings about burst transport that have appeared in the literature are corrected, to wit: burst does not immediately clip in case of channel contention; burst switches voice and data in the same way; and a burst switch interfaces naturally to other types of switches. Round-trip delay performance is calculated to be less than 5 ms. The current status of the burst project is described     

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     

Burst transmission algorithm to improve packet level performance in contention-free slotted OBS networks

In contention-free slotted optical burst switching (SOBS) networks, controllers are utilized in order to manage the time-slot assignment, avoiding congestions among multiple burst transmissions. In this network, bursts are never lost at intermediate nodes but packets are lost at an ingress edge node due to a burst transmission algorithm. In addition, packet transmission delay increases depending on the algorithm. In order to improve packet level performance, in this paper, we propose a new burst transmission algorithm. In this method, two different thresholds are used; one is used to send a control packet to a controller and the other is used to assemble a burst. With these thresholds, a time slot can be assigned to a burst in advance and packet level performance can be improved. In order to evaluate its packet level performance and investigate the impact of thresholds, we also propose a queueing model of a finite buffer where a batch of packets are served in a slot of a constant length. Numerical results show that our proposed method can decrease packet loss probability and transmission delay with two thresholds. In addition, we show that our analysis results are effective to investigate the performance of the proposed method when the number of wavelengths is large.     

Techniques for Packet Voice Synchronization

Packet switching has been proposed as an effective technology for integrating voice and data in a single network. An important aspect of packet-switched voice is the reconstruction of a continuous stream of speech from the set of packets that arrive at the destination terminal, each of which may encounter a different amount of buffering delay in the packet network. The magnitude of the variation in delay may range from a few milliseconds in a local area network to hundreds of milliseconds in a long-haul packet voice and data network. This paper discusses several aspects of the packet voice synchronization problem, and techniques that can be used to address it. These techniques estimate in some way the delay encountered by each packet and use the delay estimate to determine how speech is reconstructed. The delay estimates produced by these techniques can be used in managing the flow of information in the packet network to improve overall performance. Interactions of packet voice synchronization techniques with other network design issues are also discussed.     

Accuracy of traffic modeling in fast packet switching

Arrivals of calls, bursts, and packets to a fast packet switching system are governed by different time scales. This feature is used to break down the system performance analysis into layers. The impact of each layer on packet delay and blocking is investigated in isolation by assuming the global equilibrium in the next higher layer and deterministic flow of entities in all lower layers. The one-layer analytical model is developed and used to determine lower and upper estimates of a mean delay and blocking. Numerical results are compared with delays obtained from the multilayer simulation. Results of the analysis indicate that the channel utilization must be kept below a threshold value to avoid overload periods in the burst and call layers. Flow control techniques which can be used for that purpose are briefly 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.     

Study of information loss in packet voice systems

Once a voice buffer is full, it remains full for a certain period, during which many packets are possibly blocked, resulting in consecutive clippings in voice. The packet loss rate during this period changes slowly and has large fluctuations. It is shown that the temporal behavior of packet loss, especially at high rate, is inherently determined by voice correlation and system capacity and is independent of buffer size. Buffering may reduce the occurrence of short blocking periods associated with low rates packet loss but does not affect long ones associated with high packet loss rates. In fact, increasing the buffer size merely extends nonblocking periods, and thereby reduces the overall average packet loss rate, but packet-loss performance within existing blocking periods is not significantly improved. A simple tool is developed for calculating the boundary performance. It is found that it is possible to design a packet-switched voice system without buffering only at the expense of supporting a fewer number of calls. The issue of voice delay allocation between source and network is discussed, and it is shown that it is more effective to keep the network delay short while extending the source delay     

Congestion control for multimedia services

The problem of congestion control in high-speed networks for multimedia traffic, such as voice and video, is considered. It is shown that the performance requirements of high-speed networks involve delay, delay-jitter, and packet loss. A framing congestion control strategy based on a packet admission policy at the edges of the network and on a service discipline called stop-and-go queuing at the switching nodes is described. This strategy provides bounded end-to-end delay and a small and controllable delay-jitter. The strategy is applicable to packet switching networks in general, including fixed cell length asynchronous transfer mode (ATM), as well as networks with variable-size packets     

Performance evaluation of a packet voice system

A study is made of statistical multiplexing of voice packets from a number of packetized voice sources onto a single channel. Each source alternates between talkspurt (active period) and silence, and packets are generated during active periods only. The packets are buffered (in a finite size buffer) when transmission capacity is not available. An embedded Markov chain model is adopted to analyze the system and a numerical technique is presented to compute system performance. Simulation results validate the analysis     

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 dynamic rate control mechanism for source coded traffic in a fastpacket network

To achieve better statistical gain for voice and video traffic and to relieve congestion in fast packet networks, a dynamic rate control mechanism is proposed. An analytical model is developed to evaluate the performance of this control mechanism for voice traffic. The feedback delay for the source node to obtain the network congestion information is represented in the model. The study indicates that significant improvement in statistical gain can be realized for smaller capacity links (e.g., links that can accommodate less than 24 voice calls) with a reasonable feedback time (about 100 ms). The tradeoff for increasing the statistical gain is temporary degradation of voice quality to a lower rate. It is shown that whether the feedback delay is exponentially distributed or constant does not significantly affect performance in terms of fractional packet loss and average received coding rate. It is also shown that using the number of calls in talkspurt or the packet queue length as measures of congestion provides comparable performance     

光突发交换控制卡电路的设计与实现

光突发交换(OBS)是兼顾光电路交换和光分组交换优点的一种折衷方案，是未来光互联网的一种实现形式。在我们即将组建的光突发交换试验网中，IP(因特网协议)数据在边缘节点被组装成突发包，需要使用数据卡和控制卡分别处理数据分组和控制分组。因此，对处理控制分组的控制卡电路的优化设计和实现进行了描述。