Traffic smoothing effects of bit dropping in a packet voicemultiplexer

The performance of a packet voice multiplexer queue in which the less significant bits of voiced packets are dropped during states of congestion in the multiplexer is examined. Using the results of simulation and analytical modeling, it is illustrated that bit dropping of voice packets significantly smooth the burstiness of superposition packet voice traffic by speeding up the packet service rate during critical periods of congestion in the queue. The smoothing effect renders it possible to approximate the superposition by a Poisson process for modeling a packet voice multiplexer with bit dropping. By comparison with a simulation, an analytical model based on the Poisson assumption is shown to produce quite accurate performance predictions. The results indicate that significant capacity and performance advantages are gained in the multiplexer as a result of the bit-dropping scheme     

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     

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     

An integrated CDMA code assignment model for voice and data traffic in wireless communications

Wireless personal communication requires a provision of integrated services of multimedia traffic, such as voice and data, over the radio link. The multiple access protocols of code-division multiple-access (CDMA) techniques have been widely investigated in the recent literature. This paper presents an innovative multiple access protocol for CDMA-based wireless communication systems by fully utilizing the characteristics of voice and data traffic. In other words, a voice terminal can reserve a spreading code to transmit packets in multiple talk spurts, while a data terminal can transmit packets by either using the unassigned codes or borrowing the codes from the voice terminals during their silent periods. We build mathematical models for voice and data subsystems, respectively. Two performance parameters, the average dropping probability for voice packets and the average transmission delay for data packets, are derived based on the equilibrium point analysis. The effects of the two performance parameters on the system performance are discussed by varying the code reservation intervals of the voice terminals.     

Performance Evaluation of a Packetized Voice System--Simulation Study

Introduction of the packet switching technique into digitized voice communication may afford great advantages in efficient use of the channel, compared to both circuit-switched and DSI systems. Detailed characteristics, however, have not been obtained because of difficulty in the exact analysis. Hence, simalation models are developed in this paper for the packetized voice transmission system, and various characteristics such as tranmission delays and loss probability of voice packets are obtained. We further evaluate three types of voice packet reassembly strategy at the receiving terminal, and obtain the optimal packet length, which keeps both overall packet transmission delay and packet loss probabilty less than a certain permissible value. Comparison among three strategies is also stated.     

Implication of dropping packets from the front of a queue

When congestion occurs in a packet queuing system, packets can be dropped from the rear or the front of the queue. It is demonstrated that the probability of a packet being dropped is the same in systems with rear and front packet dropping. It is shown that the probability of a packet being delayed longer than a given value in a system with front dropping is less than or equal to that in a system with rear dropping. It is further illustrated that front dropping not only improves the delay performance on an internodal link, but also provides the overall loss performance for time constrained traffic such as packet voice     

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     

Admission control and buffer management of wireless communication systems with mobile stations and integrated voice and data services

This study presents models for management of voice and data traffic and new algorithms, which use call admission control as well as buffer management to optimise the performance of single channel systems such as wireless local area networks in the presence of mobile stations. Unlike existing studies, the new approach queues incoming voice packets as well as data packets, and uses a new pre-emption algorithm in order to keep the response time of voice requests at certain levels while the blocking of data requests is minimised. A new performance metric is introduced to provide uncorrelated handling of integrated services. Queueing related issues such as overall queue capacity, individual capacities for voice and data requests, the probability of blocking, and effects of waiting time on overall quality of service are considered in detail. Analytical models are presented and the results obtained from the analytical models were validated using discrete event simulations.     

数据流的活动队列管理算法:MBLUE

MBLUE(Modified BLUE)是一种面向数据流的活动队列管理算法.它不是使用平均队列长度指示缓冲区拥塞状态,而是使用数据报丢弃的频率和队列空闲程度来管理网络拥塞.探测瓶颈连接早期的拥塞信息,通过数据报的丢弃和标记避免拥塞.它只维护一个先进先出队列,以较少的数据流状态信息,在不同流之间公平的分配网络带宽.能够适应瞬时的猝发流,能合理控制非TCP数据流,又能够保持较短的平均队列长度,从而控制、减轻网络拥塞.通过TCP/IP网络的模拟,证实算法在公平的分配网络带宽和降低数据报的丢失率上具有较好的鲁棒性.     

Optimal Cross-Layer Scheduling of Transmissions Over a Fading Multiaccess Channel

We consider the problem of several users transmitting packets to a base station, and study an optimal scheduling formulation involving three communication layers, namely, the medium access control, link, and physical layers. We assume Markov models for the packet arrival processes and the channel gain processes. Perfect channel state information is assumed to be available at the transmitter and the receiver. The transmissions are subject to a long-run average transmitter power constraint. The control problem is to assign power and rate dynamically as a function of the fading and the queue lengths so as to minimize a weighted sum of long run average packet transmission delays.     

End-to-end performance models for variable bit rate voice overtandem links in packet networks

Analytical models are presented for computing the end-to-end voice call performance in a packet network that drops the less significant bits in voice packets during periods of congestion. These models provide information about the end-to-end quality likely to be experienced in future packet-switched integrated services networks. An existing single-node bit-dropping model is modified to include the situation resulting when the overall arrival process at an internal node consists of a mix of packets of different sizes due to bit dropping at previous nodes. A detailed model to capture bit-dropping effects in a tandem connection of nodes is presented. The model includes the effect of load fluctuations at each node, and also takes into account the dependencies in bit dropping experienced by a voice packet at successive nodes in a tandem connection. The model also incorporates the internodal dependence when reductions in packet service times occur at intermediate nodes due to bit dropping at previous nodes. Two approximation procedures are discussed that serve as upper and lower bounds. In particular, the upper bound is shown to be very tight for a practical range of loads, and hence serves as a good approximation with significant computational simplicity     

Voice packet loss: destination versus internodal links

Voice packet loss behavior at both the destination and internodal links in a packet-switched network is investigated. The fractional loss and blocking time periods for both are derived using a bivariate Markov model. The numerical results show that blocking due to the delay constraint at the destination can result in long periods of consecutive packet loss, which seriously degrade voice quality. The authors' work indicates that packets with excessive delay should be discarded at the internodal links, instead of blocking them at the destination. The relation between the internodal link buffer size and end-to-end permissible queueing delay is established     

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.     

On the Large Deviations of Resequencing Queue Size: 2-M/M/1 Case

In data communication networks, packets that arrive at the receiving host may be disordered for reasons such as retransmission of dropped packets or multipath routing. Reliable protocols such as the Transmission Control Protocol (TCP) require packets to be accepted, i.e., delivered to the receiving application, in the order they are transmitted at the sender. In order to do so, the receiver's transport layer is responsible for temporarily buffering out-of-order packets and resequencing them as more packets arrive. In this paper, we analyze a model where the disordering is caused by multipath routing. Packets are generated according to a Poisson process. Then, they arrive at a disordering network (DN) modeled by two parallel M/M/1 queues, and are routed to each of the queues according to an independent Bernoulli process. A resequencing buffer follows the DN. In such a model, the packet resequencing delay is known. However, the size of the resequencing queue (RSQ) is unknown. We derive the probability for the large deviations of the queue size.      

一种新的基于虚拟队列的无线多播网络编码调度策略

网络编码由于其传输效率高的特性,近年来在无线多播网络中得到广泛的应用。针对无线多播网络中丢包自动重传效率低的问题,该文提出一种新的基于虚拟队列中数据包到达时间的编码调度策略(CSAT)。在CSAT策略中,为了提高编码效率,采用虚拟队列来存放初始以及未被所有接收者接收到的数据包。考虑到队列的稳定性,CSAT策略按照一定的比率从主次队列选择发送;在次队列发送数据包时,结合了编码和非编码两种方式,根据数据包到达队列的先后,选取能够使较多数据包参与编码的方式发送。仿真结果表明,该文所提的CSAT编码调度策略在有效提高了数据包传输效率的同时,提高了网络的吞吐量并降低了平均等待时延。

     

Hybrid channel assignment scheme for accommodating voice/datatraffic in DS-CDMA cellular systems

A hybrid channel assignment (HCA) scheme in direct sequence-code division multiple access (DS-CDMA) systems for accommodating integrated voice/data traffic is proposed and the required power levels of voice and data traffic are derived. These levels can be used to maintain the minimum required link qualities of all calls. In the proposed scheme, delay-sensitive voice traffic is accommodated in circuit mode and delay-nonsensitive data traffic is accommodated in packet mode. The capacity region is derived and it can be used for controlling voice call admission and scheduling data packets. The proposed scheme can achieve a high link efficiency with reduced control overhead by statistically multiplexing voice and data traffic     

Performance of a nonblocking space-division packet switch withcorrelated input traffic

This work studies the performance of a nonblocking space-division packet switch in a correlated input traffic environment. In constructing the input traffic model, the author considers that each input is a time division multiaccess (TDM) link connecting to multiple sources. Every source on a link supports one call at a time. Each call experiences the alternation of ON and OFF periods, and generates packets periodically while in ON period. The stochastic property of each call does not have to be identical. Packets from each individual call are destined to the same output. The output address of each call is assumed to be uniformly assigned at random. The author derives both upper and lower bounds of the maximum throughput at system saturation. His study indicates that, if the source access rate is substantially lower than the link transmission rate, the effect of input traffic correlation on the output contentions can generally be ignored. Also, the analysis of each input queue becomes separable from the rest of the switch. The same study is carried out with nonuniform call address assignment     

The information-theoretic capacity of discrete-time queues

The information-theoretic capacity of continuous-time queues was analyzed recently by Anantharam and Verdu (see ibid. vol.42, p.4-18, 1996). Along similar lines, we analyze the information-theoretic capacity of two models of discrete-time queues. The first model has single packet arrivals and departures in a time slot and independent packet service times, and is the discrete-time analog of the continuous-time model analyzed by Anantharam and Verdu. We show that in this model, the geometric service time distribution plays a role analogous to that of the exponential distribution in continuous-time queues, in that, among all queues in this model with a given mean service time, the queue with geometric service time distribution has the least capacity. The second model allows multiple arrivals in each slot, and the queue is modeled as serving an independent random number of packets in each slot. We obtain upper and lower bounds on the capacity of queues with an arbitrary service distribution within this model, and show that the bounds coincide in the case of the queue that serves a geometrically distributed number of packets in each slot. We also discuss the extremal nature of the geometric service distribution within this model     

Congestion control for packet voice by selective packet discarding

In order to reduce the time delays as well as multiplexer memory requirements in packet voice systems, a family of congestion control schemes is proposed. They are all based on the selective discarding of packets whose loss will produce the least degradation in quality of the reconstructed voice signal. A mathematical model of the system is analyzed and queue length distributions are derived. These are used to compute performance measures, including mean waiting time and fractional packet loss. Performance curves for some typical systems are presented, and it is shown that the control procedures can achieve significant improvement over uncontrolled systems, reducing the mean waiting time and total packet loss (at transmitting and receiving ends). Congestion control with a resume level is also analyzed, showing that without increasing the fractional packet loss, the mean and variance of the queue can be reduced by selecting an appropriate resume level. The performance improvements are confirmed by the results of some informal subjective testing     

A cross-layer optimized adaptive modulation and coding scheme for transmission of streaming media over wireless links

In this paper, a novel cross-layer Adaptive Modulation and Coding scheme that optimizes the overall packet loss (by both transmission errors and excessive delays) probability under a given arrival process is developed. To this end, an improved Large Deviations approximation for the fraction of packets that suffer from excessive queuing delay is proposed. This approximation is valid for G/G/1 queues with infinite buffers that are driven by stationary arrival and service processes which satisfy certain conditions. Such models can capture the time correlations in the amount of traffic generated by streaming media sources and the time varying service capacity of a wireless link. Through numerical examples, the proposed AMC policy is shown to achieve a significant reduction in the overall packet loss rate compared to previously proposed schemes. This algorithmic performance gain can be translated into a sizeable decrease in the required transmit power or an analogous increase in the rate of the arrival process, subject to a given maximum packet loss rate Quality of Service constraint. Furthermore, the proposed AMC policy can be combined with ARQ in order to achieve an even lower overall packet loss probability.