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.     

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.     

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.     

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.     

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     

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     

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.     

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.     

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.     

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     

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     

A novel self‐adaptive transmission scheme over an IEEE 802.11 WLAN for supporting multi‐service

The IEEE 802.11 wireless local area network (WLAN) media access control (MAC) specification is a hybrid protocol of random access and polling when both distributed coordination function (DCF) and point coordination function (PCF) are used. Data traffic is transmitted with the DCF, while voice transmission is carried out with the PCF. Based on the performance analysis of the MAC protocol for integrated data and voice transmission by simulation, this paper puts forward a self‐adaptive transmission scheme to support multi‐service over the IEEE 802.11 WLAN. The simulation results show that, on the premise of satisfying the maximum allowable delay of packet voice, the self‐adaptive transmission scheme can improve the data traffic performance and increase the WLAN capacity through dynamic and appropriate adjustment of the protocol parameters. Especially, voice traffic is sensitive to delay jitter, and the self‐adaptive scheme can effectively decrease it. Finally, it is worth noting that the adaptive scheme is easy to be realized, whereas no change in the MAC protocol is needed. Copyright © 2006 John Wiley & Sons, Ltd.     

Quality-of-service provisioning system for multimedia transmission in IEEE 802.11 wireless LANs

IEEE 802.11, the standard of wireless local area networks (WLANs), allows the coexistence of asynchronous and time-bounded traffic using the distributed coordination function (DCF) and point coordination function (PCF) modes of operations, respectively. In spite of its increasing popularity in real-world applications, the protocol suffers from the lack of any priority and access control policy to cope with various types of multimedia traffic, as well as user mobility. To expand support for applications with quality-of-service (QoS) requirements, the 802.11E task group was formed to enhance the original IEEE 802.11 medium access control (MAC) protocol. However, the problem of choosing the right set of MAC parameters and QoS mechanism to provide predictable QoS in IEEE 802.11 networks remains unsolved. In this paper, we propose a polling with nonpreemptive priority-based access control scheme for the IEEE 802.11 protocol. Under such a scheme, modifying the DCF access method in the contention period supports multiple levels of priorities such that user handoff calls can be supported in wireless LANs. The proposed transmit-permission policy and adaptive bandwidth allocation scheme derive sufficient conditions such that all the time-bounded traffic sources satisfy their time constraints to provide various QoS guarantees in the contention free period, while maintaining efficient bandwidth utilization at the same time. In addition, our proposed scheme is provably optimal for voice traffic in that it gives minimum average waiting time for voice packets. In addition to theoretical analysis, simulations are conducted to evaluate the performance of the proposed scheme. As it turns out, our design indeed provides a good performance in the IEEE 802.11 WLAN's environment, and can be easily incorporated into the hybrid coordination function (HCF) access scheme in the IEEE 802.11e standard.     

ATM traffic management in an LMDS wireless access network

We investigate the capacity of LMDS to support ATM services in the local loop. In particular, we evaluate the performance of a MAC protocol for this system when transporting voice and IP traffic using the VBR and GFR service categories of ATM, respectively. Our results show that the MAC protocol is well suited for voice traffic but in general lacks efficient bandwidth management mechanisms to support the more dynamic bandwidth requirements of IP traffic     

Packet Voice on a Local Area Network with Round Robin Service

In this paper, we propose a combined voice/data protocol suitable for multiple access broadcast networks that provide round robin service to the stations. Such networks are well suited to the integration of voice and data since they guarantee bounded delay and provide high utilization even for high bandwidth channels. Using one such network proposal-namely Expressnet-as a representative scheme, we examine the characteristics of the service that voice traffic experiences under the voice/data protocol. We show that the access protocol is able to utilize the channel efficiently to support a large population of voice sources while maintaining low packet delay and guaranteeing some prespecified minimum bandwidth for data traffic. In addition, we show the advantages of silence suppression, i.e., discarding speech that constitutes silent periods, and we examine the cost of overloading the network in terms of the amount of speech discarded.     

Performance analysis of a PRMA protocol suitable for voice and datatransmissions in low Earth orbit mobile satellite systems

Mobile satellite systems (MSSs) are expected to play a significant role in providing users with communication services worldwide. In such a context, low Earth orbit (LEO) satellite constellations seem to be a good solution to attain a global coverage and to allow the use of low-power lightweight mobile terminals. This paper analyzes the performance of a novel medium access control (MAC) scheme suitable for applications in LEO-MSSs, named packet reservation multiple access with hindering states (PRMA-HS), that has been derived by proper modifications of the well-known PRMA protocol. We envisage a mixed traffic with voice sources and data sources with different quality of service (QoS) requirements. The good behavior of the proposed PRMA-HS scheme is validated by extensive comparisons with the classical PRMA protocol. Finally, it is shown that PRMA-HS efficiently supports integrated voice and data traffic in LEO-MSSs     

WiMAX自适应健壮性报头压缩算法

由于WiMAX通信传输协议框架中各层协议的报头含有太多的冗余信息,导致了无线信道带宽的利用率较低。同时由于分组过长也会引起误码率的增加。为了解决WiMAX中报头开销过大问题,提高无线信道带宽利用率,结合ROHC压缩算法和WiMAX MAC协议,提出了WiMAX自适应健壮性报头压缩方案。通过分析W-LSB编码,给出了自适应健壮性报头压缩算法在WiMAX的实现过程。仿真结果表明：该算法可以将60个字节的RTP/UDP/IPv6报头压缩到1～3个字节;可以适应WiMAX通信系统链路特性经常变化的无线信道,能够在报头压缩率和抗差错鲁棒性之间获得较好的平衡性。     

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     

An AWG-based WDM-PON architecture employing WDM/TDMA transmission for upstream traffic with dynamic bandwidth allocation

In this article, we examine a candidate architecture for wavelength-division multiplexed passive optical networks (WDM-PONs) employing multiple stages of arrayed-waveguide gratings (AWGs). The network architecture provides efficient bandwidth utilization by using WDM for downstream transmission and by combining WDM with time-division multiple access (TDMA) for upstream transmission. In such WDM-PONs, collisions may occur among upstream data packets transmitted simultaneously from different optical networking units (ONUs) sharing the same wavelength. The proposed MAC protocol avoids such collisions using a request/permit-based multipoint control protocol, and employs a dynamic TDMA-based bandwidth allocation scheme for upstream traffic, called minimum-guaranteed maximum request first (MG-MRF), ensuring a reasonable fairness among the ONUs. The entire MAC protocol is simulated using OPNET and its performance is evaluated in terms of queuing delay and bandwidth utilization under uniform as well as non-uniform traffic distributions. The simulation results demonstrate that the proposed bandwidth allocation scheme (MG-MRF) is able to provide high bandwidth utilization with a moderately low delay in presence of non-uniform traffic demands from ONUs.     

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.