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.     

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     

Providing statistical QoS guarantee for voice over IP in the IEEE 802.11 wireless LANs

Recent years have seen greatly increasing interests in voice over IP in wireless LANs, in which the IEEE 802.11 distributed coordination function protocol or enhanced DCF protocol is used. However, since both DCF and EDCF are contention-based medium access control protocols, it is difficult for them to support the strict QoS requirement for VoIP. Therefore, in this article we propose a novel call admission control scheme that runs at the MAC layer to support VoIP services. The call admission control mechanism regulates voice traffic to efficiently coordinate medium contention among voice sources. The rate control mechanism regulates non-voice traffic to control its impact on the performance of voice traffic. Extensive simulations demonstrate that the proposed schemes can well support statistical QoS guarantees for voice traffic and maintain stable high throughput for non-voice traffic at the same time.     

IP switching-ATM under IP

Internet protocol (IP) traffic on the Internet and private enterprise networks has been growing exponentially for some time. This growth is beginning to stress the traditional processor-based design of current-day routers. Switching technology offers much higher aggregate bandwidth, but presently only offers a layer-2 bridging solution. Various proposals are under way to support IP routing over an asynchronous transfer mode (ATM) network. However, these proposals hide the real network topology from the IP layer by treating the data-link layer as a large opaque network cloud. We argue that this leads to complexity, inefficiency, and duplication of functionality in the resulting network. We propose an alternative in which we discard the end-to-end ATM connection and integrate fast ATM hardware directly with IP, preserving the connectionless nature of IP. We use the soft-state in the ATM hardware to cache the IP forwarding decision. This enables further traffic on the same IP flow to be switched by the ATM hardware rather than forwarded by IP software. We claim that this approach combines the simplicity, scalability, and robustness of IP, with the speed, capacity, and multiservice traffic capabilities of ATM     

A quantitative comparison of multiple access control protocols forwireless ATM

The multiple access control (MAC) problem in a wireless network has intrigued researchers for years. For a broad-band wireless network such as wireless ATM, an effective MAC protocol is very much desired because efficient allocation of channel bandwidth is imperative in accommodating a large user population with satisfactory quality of service. Indeed, MAC protocols for a wireless ATM network in which user traffic requirements are highly heterogeneous (classified into CBR, VBR, and ABR), are even more intricate to design. Considerable research efforts expended in tackling the problem have resulted in a myriad of MAC protocols. While each protocol is individually shown to be effective by the respective designers, it is unclear how time different protocols compare against each other on a unified basis. In this paper, we quantitatively compare seven previously proposed TDMA-based MAC protocols for integrated wireless data and voice services. We first propose a taxonomy of TDMA-based protocols, from which we carefully select seven protocols, namely SCAMA, DTDMA/VR, DTDMA/PR, DQRUMA, DPRMMA, DSA++, and PRMA/DA, such that they are devised based on rather orthogonal design philosophies. The objective of our comparison is to highlight the merits and demerits of different protocol designs     

Data communication in ATM networks

An increasing number of customers require LAN access with high bandwidth and low delay over long distances. To satisfy these needs, several high-speed network techniques have been developed. Asynchronous transfer mode (ATM) is superior compared to other networking technologies, as it offers high bandwidth and is scalable in the sense that the bandwidth capacity of an ATM system is not fundamentally limited to the technology itself. Initial ATM installations will operate as subnetworks of existing networks and MAC layer protocols. One of the main challenges in ATM is the transparent support of existing connectionless LAN services. Several activities have been launched within international standard bodies and forums to specify ways of providing data communication services over ATM. Most notable examples are Switched Multimegabit Data Service (SMDS) and the similar Connectionless Broadband Data Service (CBDS) supported mostly by public network service providers. But also other approaches such as IP over ATM, and LAN emulation that show more adherence to the existing local and campus area networking paradigms. The article presents the requirements and architecture of the LAN emulation service. It describes the alternative methods for carrying IP packets over ATM, a public broadband service architecture and CBDS. The traffic management aspects of the data communication services are also discussed     

SEAMA: a source encoding assisted multiple access protocol forwireless communications

This paper presents SEAMA, a source encoding assisted multiple access (MAC) protocol, to integrate voice and data traffic in a wireless network. SEAMA exploits the time variations of the speech coding rate, through statistical multiplexing, to efficiently use the available bandwidth and to increase the link utilization. In each frame, SEAMA allocates bandwidth among calls as needed. Ongoing calls are always assigned some minimum bandwidth to allow for coding of the background noise during silence periods. An embedded voice encoding scheme is employed to allow the network to control the rate of the calls during congestion by selectively dropping some of the less significant packets, thus causing a graceful degradation of quality. It is shown that by employing an appropriate voice coding scheme and exploiting the characteristics of the source encoder in the MAC protocol, SEAMA almost doubles the capacity of the voice section compared to a circuit-switched network, while practically maintaining the quality of voice traffic     

无线ATM通信网多址访问协议与信道动态分配算法研究

本文为无线ATM通信网提出了可支持话音、数据和图像业务的多址访问控制协议(MAC)和信道动态分配(DCA)算法.所提出的正交码预约多址访问协议(ORMA)可避免各终端在预约竞争时发生碰撞,提高信道利用率.同时,还提出了一种突发业务信道动态分配算法(DCA-BT),ORMA与DCA-BT相结合能有效地提高系统内多媒体业务的质量和信道利用率,增大系统容量,支持多种业务在无线ATM通信网中的应用.     

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.     

An opportunistic cross‐layer architecture for voice in multi‐hop wireless LANs

We propose an opportunistic cross‐layer architecture for adaptive support of Voice over IP in multi‐hop wireless LANs. As opposed to providing high call quality, we target emergencies where it is important to communicate, even if at low quality, no matter the harshness of the network conditions. With the importance of delay on voice quality in mind, we select adaptation parameters that control the ratio of real‐time traffic load to available bandwidth. This is achieved in two ways: minimizing the load and maximizing the bandwidth. The PHY/MAC interaction improves the use of the spectral resources by opportunistically exploiting rate‐control and packet bursts, while the MAC/application interaction controls the demand per source through voice compression. The objective is to maximize the number of calls admitted that satisfy the end‐to‐end delay budget. The performance of the protocol is studied extensively in the ns‐2 network simulator. Results indicate that call quality degrades as load increases and overlonger paths, and a larger packet size improves performance. For long paths having low‐quality channels, forward error correction, header compression, and relaxing the delay budget of the system are required to maintain call admission and quality. The proposed adaptive protocol achieves high performance improvements over the traditional, non‐adaptive approach. Copyright © 2008 John Wiley & Sons, Ltd.     

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.     

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.     

Overview of implementing ATM based enterprise local area networkfor desktop multimedia computing

The authors discuss important implementation issues in an ATM-based enterprise network, and propose possible migration strategies for the smooth introduction of ATM into the desktop computing environment. They present the ATM traffic service classes and the associated traffic management functions. The authors cover the latest ATM forum standardization efforts on traffic management functions and LAN emulation. Finally, they discuss how to seamlessly support the existing transport control protocol (TCP)/Internet protocol (IP) in an ATM environment     

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     

100+ VoIP Calls on 802.11b: The Power of Combining Voice Frame Aggregation and Uplink-Downlink Bandwidth Control in Wireless LANs

The bandwidth efficiency of voice over IP (VoIP) traffic on the IEEE 802.11 WLAN is notoriously low. VoIP over 802.11 incurs high bandwidth cost for voice frame packetization and MAC/PHY framing, which is aggravated by channel access overhead. For instance, 10 calls with the G.729 codec can barely be supported on 802.11b with acceptable QoS - less than 2% efficiency. As WLANs and VoIP services become increasingly widespread, this inefficiency must be overcome. This paper proposes a solution that boosts the efficiency high enough to support a significantly larger number of calls than existing schemes, with fair call quality. The solution comes in two parts: adaptive frame aggregation and uplink/downlink bandwidth equalization. The former reduces the absolute number of MAC frames according to the link congestion level, and the latter balances the bandwidth usage between the access point (AP) and wireless stations. When used in combination, they yield superior performance, for instance, supporting more than 100 VoIP calls over an IEEE 802.11b link. The authors demonstrate the performance of the proposed approach through extensive simulation, and validate the simulation through analysis.     

Analysis of a polling system for telephony traffic with application to wireless LANs

Recently, polling has been included as a resource sharing mechanism in the medium access control (MAC) protocol of several communication systems, such as the IEEE 802.11 wireless local area network, primarily to support real-time traffic. Furthermore, to allow these communication systems to support multimedia traffic, the polling scheme often coexists with other MAC schemes such as random access. Motivated by these systems, we develop a model for a polling system with vacations, where the vacations represent the time periods in which the resource sharing mechanism used is a non-polling mode. The real-time traffic served by the polling mode in our study is telephony. We use an on-off Markov modulated fluid (MMF) model to characterize telephony sources. Our analytical study and a counterpart validating simulation study show the following. Since voice codec rates are much smaller than link transmission rates, the queueing delay that arises from waiting for a poll dominates the total delay experienced by a voice packet. To keep delays low, the number of telephone calls that can be admitted must be chosen carefully according to delay tolerance, loss tolerance, codec rates, protocol overheads and the amount of bandwidth allocated to the polling mode. The effect of statistical multiplexing gain obtained by exploiting the on-off characteristics of telephony traffic is more noticeable when the impact of polling overhead is small.     

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.     

Neural bandwidth allocation function (NBAF) control scheme at WiMAX MAC layer interface

The paper proposes a bandwidth allocation scheme to be applied at the interface between upper layers (IP, in this paper) and Medium Access Control (MAC) layer over IEEE 802.16 protocol stack. The aim is to optimally tune the resource allocation to match objective QoS (Quality of Service) requirements. Traffic flows characterized by different performance requirements at the IP layer are conveyed to the IEEE 802.16 MAC layer. This process leads to the need for providing the necessary bandwidth at the MAC layer so that the traffic flow can receive the requested QoS. The proposed control algorithm is based on real measures processed by a neural network and it is studied within the framework of optimal bandwidth allocation and Call Admission Control in the presence of statistically heterogeneous flows. Specific implementation details are provided to match the application of the control algorithm by using the existing features of 802.16 request–grant protocol acting at MAC layer. The performance evaluation reported in the paper shows the quick reaction of the bandwidth allocation scheme to traffic variations and the advantage provided in the number of accepted calls. Copyright © 2006 John Wiley & Sons, Ltd.     

VMcMAC: Voice Call Support over Wireless Multi-Channel MAC Protocol

This paper presents a new scheme to support voice calls over a wireless multi-channel MAC protocol (VMcMAC). We increase the voice capacity of wireless networks by reducing protocol overhead and interference between voice traffic and data traffic. Voice calls are allocated to specific reserved channels in a distributed TDMA fashion. Each voice node visits the voice channel with a fixed frequency and then transmits a voice frame without sending control messages. Simulation results show a significant improvement in the voice capacity of wireless ad-hoc networks     

LMDS/LMCS hub interconnection alternatives and multiple access issues

LMDS/LMCS is a broadband wireless local loop, millimeter‐wave alternative to emerging integrated multiservice access networks. Significantly large amounts of bandwidth – in the order of one GHz of spectrum – are made available to residential subscribers or supported business users respectively that employ highly directional antennas and signal polarization to establish communication with a central hub. Besides the requirement for dynamic bandwidth allocation capabilities, these networks should be able to guarantee negotiated quality of service (QoS) levels to a number of constant‐length (ATM) – and possibly variable length (TCP/IP) – packet streams. In this context, we analyze the performance of contention, polling/probing and piggybacking mechanisms that will be used by the LMDS MAC protocol for the dynamic support of both real‐time and non‐real‐time traffic streams. More specifically, we focus on the end‐to‐end performance of a real‐time variable bit rate connection for which the LMDS link is only the access component of a multi‐link path through an ATM network. Results are presented on maximum end‐to‐end cell delays under a Weighted Round Robin service discipline and buffer requirements are calculated for no‐loss conditions. In parallel, we also consider the case in which variable length IP packet traffic is supported as such by the same wireless access network. Backbone interconnection alternatives of LMDS hubs, multiple access proposals and scheduling algorithms are addressed in this framework. This revised version was published online in July 2006 with corrections to the Cover Date.