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.     

Optimal resource allocation and adaptive call admission control for voice/data integrated cellular networks

Resource allocation and call admission control (CAC) are key management functions in future cellular networks, in order to provide multimedia applications to mobiles users with quality of service (QoS) guarantees and efficient resource utilization. In this paper, we propose and analyze a priority based resource sharing scheme for voice/data integrated cellular networks. The unique features of the proposed scheme are that 1) the maximum resource utilization can be achieved, since all the leftover capacity after serving the high priority voice traffic can be utilized by the data traffic; 2) a Markovian model for the proposed scheme is established, which takes account of the complex interaction of voice and data traffic sharing the total resources; 3) optimal CAC parameters for both voice and data calls are determined, from the perspective of minimizing resource requirement and maximizing new call admission rate, respectively; 4) load adaption and bandwidth allocation adjustment policies are proposed for adaptive CAC to cope with traffic load variations in a wireless mobile environment. Numerical results demonstrate that the proposed CAC scheme is able to simultaneously provide satisfactory QoS to both voice and data users and maintain a relatively high resource utilization in a dynamic traffic load environment. The recent measurement-based modeling shows that the Internet data file size follows a lognormal distribution, instead of the exponential distribution used in our analysis. We use computer simulations to demonstrate that the impact of the lognormal distribution can be compensated for by conservatively applying the Markovian analysis results.     

Performance analysis for voice/data integration on a finite-buffermobile system

Personal communication service (PCS) networks offer mobile users diverse telecommunication applications, such as voice, data, and image, with different bandwidth and quality-of-service (QoS) requirements. This paper proposes an analytical model to investigate the performance of an integrated voice/data mobile network with finite data buffer in terms of voice-call blocking probability, data loss probability, and mean data delay. The model is based on the movable-boundary scheme that dynamically adjusts the number of channels for voice and data traffic. With the movable-boundary scheme, the bandwidth can be utilized efficiently while satisfying the QoS requirements for voice and data traffic. Using our model, the impact of hot-spot traffic in the heterogeneous PCS networks, in which the parameters (e.g., number of channels, voice, and data arrival rates) of cells can be varied, can be effectively analyzed. In addition, an iterative algorithm based on our model is proposed to determine the handoff traffic, which computes the system performance in polynomial-bounded time. The analytical model is validated by simulation     

Voice over ATM using AAL2 and bit dropping: performance and calladmission control

Asynchronous transfer mode (ATM) adaptation layer 2 (AAL2) has been designed for efficient transport of voice, fax, and voiceband data (VBD) traffic over an ATM virtual circuit. The protocol helps achieve low latency and high bandwidth efficiency while applying suitable compression methods on voice/VBD/fax calls and silence elimination on voice calls. We analyze the performance and capacity of an ATM multiplexer based on AAL2 adaptation. We assume that embedded adaptive differential pulse code modulation (ADPCM) is used to compress voice, and silence elimination is used to achieve statistical multiplexing gain. The embedded ADPCM coding scheme allows selective dropping of less significant bits of voice during congestion in the ATM/AAL2 multiplexer. We compare the call capacities of voice multiplexers with and without bit dropping (BD). The performance models and results presented are based on fairly general assumptions and can be used for traffic engineering and call admission control in land-line or wireless ATM systems for a variety of voice/voiceband compression algorithms. A generalized algorithm for call admission control is also described     

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     

Dynamic bandwidth allocation on a slotted ring with integratedservices

The authors discuss what they consider the fundamental issue of bandwidth allocation on an integrated local area network. An approach is introduced for dynamic bandwidth allocation which is based on traffic prediction concepts. It is especially well suited for real-time services such as video and voice. Using a control model two allocation schemes are proposed: the first is based on an analytical model of the traffic flow; the second is a simpler version that can be easily implemented on very high-speed systems. The results of simulation studies indicate a marked improvement in performance. The presented approach is especially effective when used in systems with large transmission path latencies as the network performance does not deteriorate with increasing latency. This is very useful if the network is to be used as a metropolitan area network     

Call admission control for voice/data integration, in.broadband, wireless networks

This paper addresses bandwidth allocation for an integrated voice/data broadband mobile wireless network. Specifically, we propose a new admission control scheme called EFGC, which is an extension of the well-known fractional guard channel scheme proposed for cellular networks supporting voice traffic. The main idea is to use two acceptance ratios, one for voice calls and the other for data calls in order to maintain the proportional service quality for voice and data traffic while guaranteeing a target handoff failure probability for voice calls. We describe two variations of the proposed scheme: EFGC-REST, a conservative approach which aims at preserving the proportional service quality by sacrificing the bandwidth utilization, and EFGC-UTIL, a greedy approach which achieves higher bandwidth utilization at the expense of increasing the handoff failure probability for voice calls. Extensive simulation results show that our schemes satisfy the hard constraints on handoff failure probability and service differentiation while maintaining a high bandwidth utilization.     

AAL2分组话音复接器带宽分配算法的研究

本文根据分组话音业务的特点,结合分组话音业务服务质量的要求,特别是分组丢弃概率和平均分组排队时延的要求,研究AAL2分组话音复接器带宽分配算法.得出结论:对于无比特丢弃的AAL2分组话音复接器,按平均速率分配带宽基本上可以满足分组话音业务服务质量的要求;如果适当降低ATM VC的带宽利用率ρ(例如:令ρ=0.9),则可以进一步提高话音质量,获得令人满意的话音;对于带比特丢弃的AAL2分组话音复接器,按平均速率分配带宽,可以很好地满足分组话音业务服务质量要求,获得较高质量的话音.计算机仿真证实了上述结论是正确的.     

A heuristic approach of bandwidth management for video sources in ATM networks

One of the most important properties in the ATM network is that the resource of the network, including buffer and bandwidth, can be flexibly managed according to different demands of various applications. The network bandwidth can be effectively allocated and utilized if the data volume of the arrival traffic can be predicted precisely. In this paper, we study the bandwidth management schemes for variable bit rate (VBR) pre‐coded MPEG video sources. The proposed bandwidth allocation method, which predicts the bandwidth by the frame correlation, demonstrates a quite good performance when comparing with a previous scheme, especially for the video scenes with the combination of intraframes and interframes. Bandwidth allocation of a multiplexer connected to several video sources is also studied by using heuristic information. The experimental results show that the proposed method is much better than that of the fixed bandwidth allocation and is suitable for the application of MPEG video services. Copyright © 2000 John Wiley & Sons, Ltd.     

Performance management issues of currently operated ATM enterprisenetworks

The author proposes a solution for the allocation and balancing of resources to maximize available bandwidth shared among corporate users. Currently established broadband virtual private networks (BVPNs) based on asynchronous transfer mode (ATM) technology comprise ATM cross-connects (ATM-CCs) and a lot of intelligent customer premises equipment (CPE). The CPE, an intelligent ATM service switcher or ATM multiplexer, enables the corporate user to connect routers, private branch exchanges (PBXs), or codecs onto the ATM network. One fundamental characteristic of CPE is that it is capable of accumulating asynchronous and synchronous traffic which may belong to different corporate users' sites. A typical example given of a BVPN configuration serving two corporate network users with four user sites each. In general, each user site needs to exchange asynchronous (connectionless) data streams for the inter-local area network (LAN) communication and synchronous (connection-oriented) data streams with constant bit rates for video/voice communication. The configuration and the performance aspects of inter-LAN communications employing a connectionless server (CLS) are discussed. The bandwidth allocation aspects of the BVPN having to convey synchronous and asynchronous traffic in an ATM environment without a CLS are discussed, including the bandwidth allocation algorithm. The important characteristics of the proposed algorithm is also summarised     

Simulation analysis of a communication link with statisticallymultiplexed bursty noise sources

Variable bit rate (VBR) coding techniques have received great research interest as very promising tools for transmitting bursty multimedia traffic with low bandwidth requirements over a communication link. Statistically multiplexing the multimedia bursty traffic is a very efficient method of maximizing the utilization of the link capacity. The application of computer simulation techniques in analyzing a rate-based access control scheme for multimedia traffic such as voice traffic is discussed. The control scheme regulates the packetized bursty traffic at the user network interface of the link. Using a suitable congestion measure, namely, the multiplexer buffer length, the scheme dynamically controls the arrival rate by switching the coder to a different compression ratio (i.e., changing the coding rate). VBR coding methods can be adaptively adjusted to transmit at a lower rate with very little degradation in the voice quality. Reported results prove that the scheme greatly improves the link performance, in terms of reducing the probability of call blocking and enhancing the statistical multiplexing gain     

An Efficient and Adaptive Bandwidth Allocation Scheme for Mobile Wireless Networks Using an On-Line Local Estimation Technique

The next generation of mobile wireless networks has to provide the quality-of-service (QoS) for a variety of applications. One of the key generic QoS parameters is the call dropping probability, which has to be maintained at a predefined level independent of the traffic condition. In the presence of bursty data and the emerging multimedia traffic, an adaptive and dynamic bandwidth allocation is essential in ensuring this QoS. The paradox, however, is that all existing dynamic bandwidth allocation schemes require the prior knowledge of all traffic parameters or/and user mobility parameters. In addition, most proposals require extensive status information exchange among cells in order to dynamically readjust the control parameters, thus making them difficult to be used in actual deployment.In this paper, we introduce a novel adaptive bandwidth allocation scheme which estimates dynamically the changing traffic parameters through local on-line estimation. Such estimations are restricted to each individual cell, thus completely eliminating the signaling overhead for information exchange among cells. Furthermore, we propose the use of a probabilistic control policy, which achieves a high channel utilization, and leads to an effective and stable control. Through simulations, we show that our proposed adaptive bandwidth allocation scheme can guarantee the predetermined call dropping probability under changing traffic conditions while at the same time achieving a high channel utilization.     

Bandwidth Reallocation for Bandwidth Asymmetry Wireless Networks Based on Distributed Multiservice Admission Control

This paper addresses when and how to adjust bandwidth allocation on uplink and downlink in a multi-service mobile wireless network under dynamic traffic load conditions. Our design objective is to improve system bandwidth utilization while satisfying call level QoS requirements of various call classes. We first develop a new threshold-based multi-service admission control scheme (DMS-AC) as the study base for bandwidth re-allocation. When the traffic load brought by some specific classes under dynamic traffic conditions in a system exceeds the control range of DMS-AC, the QoS of some call classes may not be guaranteed. In such a situation, bandwidth re-allocation process is activated and the admission control scheme will try to meet the QoS requirements under the adjusted bandwidth allocation. We explore the relationship between admission thresholds and bandwidth allocation by identifying certain constraints for verifying the feasibility of the adjusted bandwidth allocation. We conduct extensive simulation experiments to validate the effectiveness of the proposed bandwidth re-allocation scheme. Numerical results show that when traffic pattern with certain bandwidth asymmetry between uplink and downlink changes, the system can re-allocate the bandwidth on uplink and downlink adaptively and at the same time improve the system performance significantly.     

A dynamic packet reservation multiple access scheme for wireless ATM

The dynamic packet reservation multiple access (DPRMA) scheme, a medium access control protocol for wireless multimedia applications, is proposed and investigated. DPRMA allows the integration of multiple traffic types through a single access control mechanism that permits users to specify their immediate bandwidth requirements. The primary feature of DPRMA is the dynamic matching of the traffic source generation rates with the assigned portion of the channel capacity. This is accomplished by a control algorithm that regulates the actual amount of channel capacity assigned to users. To support multimedia communication, channel capacity assignments are prioritized by traffic type. The performance of the scheme is evaluated and the scheme is shown to perform well in a system with voice, video conferencing, and data users present. It is also shown to provide improved performance over a system with a modified version of the packet reservation multiple access (PRMA) scheme. Furthermore, several system parameters are studied and optimized.     

Radio Resource Allocation for Scalable Video Services Over Wireless Cellular Networks

Good quality video services always require higher bandwidth. Hence, to provide the video services e.g., multicast/broadcast services (MBSs) and unicast services along with the existing voice, internet, and other background traffic services over the wireless cellular networks, it is required to efficiently manage the wireless resources in order to reduce the overall forced call termination probability, to maximize the overall service quality, and to maximize the revenue. Fixed bandwidth allocation for the MBS sessions either reduces the quality of the MBS videos and bandwidth utilization or increases the overall forced call termination probability and of course the handover call dropping probability as well. Scalable video coding (SVC) technique allows the variable bit rate allocation for the video services. In this paper, we propose a bandwidth allocation scheme that efficiently allocates bandwidth among the MBS sessions and the non-MBS traffic calls (e.g., voice, unicast, internet, and other background traffic). The proposed scheme reduces the bandwidth allocation for the MBS sessions during the congested traffic condition only to accommodate more calls in the system. Instead of allocating fixed bandwidths for the MBS sessions and the non-MBS traffic, our scheme allocates variable bandwidths for them. However, the minimum quality of the videos is guaranteed by allocating minimum bandwidth for them. Using the mathematical and numerical analyses, we show that the proposed scheme maximizes the bandwidth utilization and significantly reduces the overall forced call termination probability as well as the handover call dropping probability.     

PRNN/ERLS-based predictive QoS-promoted DBA scheme for upstream transmission in EPON

This article proposes a PRNN/ERLS-based predictive QoS-promoted dynamic bandwidth allocation (PQ-DBA) scheme for upstream transmission in Ethernet passive optical network (EPON) systems. The proposed PQ-DBA scheme originally divides incoming packets of voice, video, data service traffic into six priorities, where packets having less room before QoS requirements violation or being in starvation situation will be dynamically promoted to high priority cycle-by-cycle. It predicts packets arriving at prediction interval for ONUs using pipeline recurrent neural network (PRNN)/extended recursive least squares (ERLS) so that the bandwidth allocation can be more up-to-date and then accurate. Simulation results show that the proposed PQ-DBA scheme achieves higher system utilization and lower average voice, video, data packet delay time than the DBAM scheme [Luo and Ansari, OSA J Opt Netw 4(9):561–572] by 4, and 21, 90, 43%, respectively, and the PQ-DBA scheme but without prediction by 2, and 26, 29, 34%, respectively.     

一种基于无线多媒体业务的可升降级QoS动态带宽分配与优化策略

无线多媒体网络中的业务包括话音、流媒体、交互类和背景类业务4种,除话音业务外其余3种业务都是可变比特速率业务。对该网络用户资源分配(主要是带宽的分配)若采用传统的固定分配方法,必定陷入资源利用率低下和用户QoS得不到保障的两难境地。该文提出了一种基于无线多媒体业务的动态带宽分配与优化策略,在保证用户QoS的前提下,尽可能提高资源利用率。该文分别从网络和用户两个角度考虑,通过系统容量、业务阻塞率、数据延迟、流媒体的实际传输比和VBR业务综合服务等级等参数,对可升降级QoS无线多媒体网络进行了仿真分析,结果表明,对比传统的网络资源管理策略,该策略大大改善了系统的性能,提高了系统资源利用率。     

多信源ATM分组语音图像流的分析

分析了ATM复用器中的分组语音图像流.为研究多信源、大流量条件下的流量特性,我们将语音源近似为马尔柯夫泊松过程MMPP(2).2个泊松过程的平均到达率由生灭链得出,转移率则通过极限定理,由高斯过程的矩匹配MMPP(2)相应参数而获得.同时,我们运用流体流法来解决复用缓冲性能.同样的方法适用于图像源及语音图像合成源的情况.