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.     

Fair Delay Optimization-based Resource Allocation Algorithm for Video Traffic over Wireless Multimedia System

The major issue related to the realization of wireless multimedia system is the design of suitable medium access control (MAC) protocol. The design challenge is to maximize the utilization of the limited wireless resources while guaranteeing the various quality of service requirements for all traffic classes especially for the stringent real-time constraint of real time variable bit rate (rt-VBR) video service. In this paper a novel resource allocation algorithm for video traffic is proposed. The proposed allocation algorithm aims to provide fair delay for video packets by minimizing the delay difference among transmitted video packets. At the same time it adaptively controls the allocated resources (bandwidth) for video traffic around the corresponding average bit rate, and has the ability of controlling the quality of service (QoS) offered for video traffic in terms of packet loss probability and average delay. A minimized control overhead of only two bits is needed to increase the utilization efficiency. Simulation results show that the proposed algorithm achieves very high utilization and provides nearly fair delay among video packets. Its efficiency is also investigated under traffic integration condition with voice and data traffic to show that the QoS offered to video traffic does not change in the presence of the highest priority voice traffic while data traffic increases the channel utilization to 98% by using the remaining bandwidth after voice and video traffic while a good QoS is offered to voice and data traffic.

Integrating Voice, Video, and E-mail Data Packet Traffic over Wireless TDMA Channels with Errors

In this paper, we explore, via an extensive simulation study, the performance of a new medium access-control (MAC) protocol when integrating voice, video, and e-mail data packet traffic over a wireless channel of high capacity, with errors. 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 voice and video packet-dropping probabilities for various voice and video load conditions and the average e-mail data message delays. Our scheme achieves high aggregate channel throughput in all cases of traffic load despite the introduction of errors in the system.     

QoS support for integrated services over CATV

Cable TV has emerged as a promising access network infrastructure for the delivery of voice, video, and high-speed data traffic. A central issue in the design of protocols for CATV networks is to support different levels of QoS for diverse user applications. While CATV service providers and equipment have standardized, in the so-called MCNS protocol, the basic network architecture and interfaces, issues in the MAC layer for QoS support are likely to be left for differentiation in vendor products. This article first presents an overview of the basic CATV network architectural assumptions and the set of QoS requirements for supporting integrated services over CATV. It then discusses a MAC layer scheduling protocol that can efficiently multiplex constant bit rate traffic, such as voice over IP with guaranteed delay bound, and best-effort traffic, such as data services with minimum bit rate guarantee, while achieving fairness on any excess available bandwidth. The performance of this algorithm is illustrated by simulation results using Opnet. We also discuss a dynamic polling mechanism that enhances the link utilization while preserving delay bounds for latency-critical traffic     

An adaptive reservation scheme for VBR voice traffic in wireless ATM networks

A reservation scheme, named dynamic hybrid partitioning, is proposed for the Medium Access Control (MAC) protocol of wireless ATM (WATM) networks operating in Time Division Duplex (TDD) mode. The goal is to improve the performance of the real-time Variable Bit Rate (VBR) voice traffic in networks with mixed voice/data traffic. In most proposed MAC protocols for WATM networks, the reservation phase treats all traffic equally, whether delay-sensitive or not. Hence, delay-sensitive VBR traffic sources have to compete for reservation each time they wake up from idle mode. This causes large and variable channel access delays, and increases the delay and delay variation (jitter) experienced by ATM cells of VBR traffic. In the proposed scheme, the reservation phase of the MAC protocol is dynamically divided into a contention-free partition for delay-sensitive idle VBR traffic, and a contention partition for other traffic. Adaptive algorithms dynamically adjust the partition sizes to minimize the channel bandwidth overhead. Simulation results show that the delay performance of delay-sensitive VBR traffic is improved while minimizing the overhead.     

Block Reservation Time Division Multiple Access (BRTDMA) Protocol for a High Capacity Wireless Network

Next generation high capacity wireless networks need to support various types of traffic, including voice, video and data, each of which have different Quality of Service (QoS) requirements for successful transmission. This paper presents an advanced reservation packet access protocol BRTDMA (Block Reservation Time Division Multiple Access) that can accommodate voice and data traffic with equal efficiency in a wireless network. The proposed BRTDMA protocol has been designed to operate in a dynamic fashion by allocating resources according to the QoS criteria of voice and data traffic. Most of the existing reservation protocols offers reservation to voice traffic while data packets are transmitted using contention mode. In this paper we propose a block reservation technique to reserve transmission slots for data traffic for a short duration, which minimizes the speech packet loss and reduce the end-to-end delay for wireless data traffic. The optimum block reservation length for data traffic has been studied in a cellular mobile radio environment using a simulation model. Simulation results show that the BRTDMA protocol offers higher traffic capacity than standard PRMA protocol for integrated voice and data traffic and offers flexibility in accommodating multimedia traffic.     

An efficient traffic control scheme for store‐and‐forward video‐on‐demand services

According to the MPEG‐1/2 standard, full motion video can be compressed and stored in an information warehouse. Multiplexing with normal voice calls, it is retrieved and delivered to the customer's local BISDN central office via 155 Mb/s trunks. These voice calls have higher priority than video‐on‐demand (VOD) so that normal voice services will not be influenced by VOD transmission. The number of voice calls always fluctuates in real‐time. Thus, an efficient traffic control scheme is highly desired to guarantee a given level of performance and achieve as great as possible use of available bandwidth. Based on the fact that the number of real‐time voice calls is variable, and the use of available capacity is less than 100 per cent, this paper will present an efficient traffic control scheme for store‐and‐forward VOD services, and demonstrate the scheme by various simulation results. Copyright © 2001 John Wiley & Sons, Ltd.     

An uplink CDMA system architecture with diverse QoS guarantees forheterogeneous traffic

The uplink access control problems for cellular code-division multiple-access (CDMA) systems that service heterogeneous traffic with various types of quality-of-service (QoS) and use multicode CDMA to support variable bit rates are addressed. Considering its distinct QoS requirements, class-I real-time traffic (e.g., voice and video) is differentiated from class-II non-real-time traffic (e.g., data). Connection-oriented transmission is achieved by assigning mobile-oriented code channels for class-I traffic, where each corresponding mobile needs to pass an admission test. Class-II traffic is transmitted in a best-effort manner through a transmission-rate request access scheme which utilizes the bandwidth left unused by class-I traffic. Whenever a mobile has class-II messages to transmit, the mobile requests code channels via a base station-oriented transmission-request code channel, then, according to the base station scheduling, the transmission is scheduled and permitted. Addressed are the admission test for class-I connections, transmission power allocation, and how to maximize the aggregate throughput for class-II traffic. The admission region of voice and video connections and the optimum target signal-to-interference ratio of class-II traffic are derived numerically. The performance of class-II traffic transmissions in terms of average delay is also evaluated and discussed     

Time slot switching for integrated services in fiber optic PBX/LAN

A medium-access protocol called time-slot switching (TSS) is proposed for use in optical-fiber local area networks. This protocol incorporates features of time division, space division, and time compression for users to share a common medium. Very-large-integration (VLSI) CMOS electric crosspoints are used to switch traffic within individual time slots. With these features, data, voice, and video services can all be combined in a single network. In addition, the speed of the electronics can be maximized to match the available optical bandwidth. Operational principles of the TSS protocol are explained. A performance analysis is presented to show the tradeoffs among traffic capacity, frame guard time, blocking probability, and the results show that TSS is more attractive than broadcast protocols for voice traffic or constant-rate data traffic. An approach to integrating voice, data, and video traffic within TSS is also described     

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     

Dynamic Code Assignment Medium Access (DCAMA) Protocol for Wireless Integrated Services Networks

A medium access control (MAC) protocol for wireless mobile networks that supports integrated services and provides quality of service (QoS) support is presented and evaluated via simulation. A controlled random access protocol which allows all terminals to dynamically share a group of spread spectrum spreading codes is used. The protocol provides mobile terminals the access control required for efficient transfer of integrated traffic with QoS guarantees. Two service classes are provided; "best-effort" service, with priority queueing, and reserved bandwidth circuit service. The performance of the protocol is evaluated via simulation for traffic consisting of integrated voice, data and compressed video. The performance assessment measure is packet delay.     

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.     

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     

4G蜂窝网中移动互联网IP电视服务的优化研究

4G蜂窝网的频谱分配效率难以负担日益增长的移动IPTV服务与大视频流数据,为此提出了蜂窝网支持移动IPTV服务的优化方案,在保持传统语音电话质量的前提下,提高了移动IPTV服务的性能.首先,为语音电话分配较高的频率分配优先级以维护语音电话的QoS(Quality of Service);然后,采用软频谱复用技术来增强频谱的效率,并降低相邻蜂窝之间的频谱干扰;最终,设计了动态频谱分配算法,对相邻蜂窝间边缘频带的频谱分配进行协调与优化,进一步地提高了频谱利用率.仿真结果显示,本频谱分配算法在不降低语音电话服务质量的前提下,明显地提高了移动IPTV服务的性能.     

Joint source and sending rate modeling in adaptive video streaming

This work addresses the modeling of traffic generated by a video source operating in the context of adaptive streaming services. Traffic modeling is a key in several network design issues, such as dimensioning of core and access network resources, developing pricing procedures, carrying out cost-revenue studies. The actual traffic generated during a video streaming session depends on both the video source and the bandwidth variations imposed by lower communication layers. We propose a new traffic model that jointly encompasses these two effects. Specifically, we consider the modeling of the sequence of frame sizes generated by a video streaming source that dynamically adapts its rate to the available communication channel bandwidth using bitstream switching techniques. In order to represent the source rate adaptation to the random network bandwidth variations on the communication channel, we resort to a framework based on Hidden Markov Processes (HMPs). Our HMP model represents the first joint source and sending rate model in adaptive streaming literature. Thanks to effective modeling assumptions on the frame size probability density function (pdf), the HMP parameters can be estimated by means of the Expectation Maximization algorithm. The traffic model is validated by numerical simulations of a mobile adaptive video streaming scenario. We study the model's ability to predict several traffic statistics, including the traffic load of a video streaming source in different network points. Besides, we evaluate the model accuracy in characterizing aggregate video traffic resulting from multiplexing various video sources. In all experiments, we show that the proposed model is able to accurately capture the traffic characteristics.     

A rate‐based overload control method for the radio channel in PCN

Third‐generation wireless digital communication systems, currently being developed, are intended to integrate all the existing wireless systems and cover a wide range of services, including voice, video and multimedia. A difficult problem towards this direction is the efficient use of the limited available bandwidth. Although considerable improvements have been made recently in transmitter and receiver technology, the capacity of the air interface is still considerably smaller compared to other media such as fiber optics. Accordingly, traffic congestion is an important problem, especially for bandwidth demanding applications (e.g., video), leading to poor quality‐of‐service (QoS). This paper presents an overload control method, for TDMA systems, to temporarily reduce the source rate requirements to a sustainable level, in order to avoid a sudden degradation in QoS. The control is activated when the aggregate rate crosses a predefined threshold that identifies congestion. To ensure fairness, the selection of the sources whose rate will be reduced is performed in co‐operation with a priority‐based scheduling technique. The performance of the system under the proposed method is analyzed and system parameter values are optimized. It is shown that the method attains considerable improvement in the loss probability performance. This revised version was published online in June 2006 with corrections to the Cover Date.     

QoS provisioning for multimedia wireless networks

As we move towards IP-based multimedia wireless networks with voice, video and data convergence, quality of service (QoS) provisioning will become an increasingly challenging task. One implication is that greater emphasis on managing the call admission and overall network resources will be needed. This paper presents a conservative and adaptive quality of service (CAQoS) framework for provisioning the QoS for both real-time and non-real-time traffic in a multimedia wireless network. Unlike most conventional schemes, which gradually scale down the bandwidth of ongoing connections to accommodate new connection/hand-off requests, CAQoS introduces an early scaling-down of bandwidth for new connections based on a designated provisioning model. The performance of a CAQoS system is evaluated through simulations of a realistic wireless environment. Simulation results show that CAQoS meets our design goals and outperforms conventional schemes.     

MPEG4视频流量预测自适应算法的研究

为了保证用户的服务质量(QOS),宽带分组网在传送视频信息时需要进行动态带宽分配,而视频流量预测在动态带宽分配中发挥着重要的作用。本文从自相关性、自相似性的Hurst参数两个方面,阐明GOP时间尺度上的流量能够体现原始帧序列的流量特性,并在固定步长的LMS自适应算法(FSSA)的基础上提出的一种新的可变步长自适应算法(VSSA),在GOP的大时间尺度上预测MPEG4视频流量,通过大量的仿真实验表明,VSSA算法可以明显地改善预测性能。     

变码率MPEG-2视频编码的统计特性及数据分割可分级的新方法

本文研究了变码率ＭＰＥＧ－２视频编码的统计特性，认为基于条带的统计分析方法能更好地描述视频序列的特性。本文同时提出了一种按照图像编码类型的可分级数据分割新方法，该方法降低了基本层流量的变化，减少了带宽需求，并能提供良好的主观图像质量。     

Optimal resource allocation for heterogeneous traffic in multipath networks

Multipath networks allow each source to send packets from it to its destination over multiple paths, which increases the available bandwidth and throughput for source‐destination pairs. Recently, a variety of flow control schemes have been presented for multipath networks to achieve optimal resource allocation. Unfortunately, much of the investigation focused on elastic traffic with controllable packet injection rates. Networks have witnessed an increase in real‐time traffic (voice and video), which are inelastic. We consider resource allocation for heterogeneous traffic in multipath networks and formulate an optimization problem, which is intrinsically difficult nonconvex. In order to address the aforementioned issue and obtain the optimum, we approximate an equivalent problem of the original optimization problem to a strictly convex problem and present a primal‐dual resource allocation algorithm for the approximation problem, which converges to an optimal solution satisfying the Karush‐Kuhn‐Tucker conditions of the original problem. We evaluate its convergence performance through theoretical analysis and illustrate it with numerical examples. Copyright © 2014 John Wiley & Sons, Ltd.