A call admission control framework for voice over WLANs

In this article a call admission control framework is presented for voice over wireless local area networks (WLANs). The framework, called WLAN voice manager, manages admission control for voice over IP (VoIP) calls with WLANs as the access networks. WLAN voice manager interacts with WLAN medium access control (MAC) layer protocols, soft-switches (VoIP call agents), routers, and other network devices to perform end-to-end (ETE) quality of service (QoS) provisioning and control for VoIP calls originated from WLANs. By implementing the proposed WLAN voice manager in the WLAN access network, a two-level ETE VoIP QoS control mechanism can be achieved: level 1 QoS for voice traffic over WLAN medium access and level 2 QoS for ETE VoIP services in the networks with WLANs as the local access. The implementation challenges of this framework are discussed for both level 1 and level 2. Possible solutions to the implementation issues are proposed and other remaining open issues are also addressed.     

基于WLAN的VoIP终端移动性管理研究

目前,随着WLAN的发展,VoIP(Voice over Internet Protocol)技术在其中的应用也越来越广泛。但VoWLAN仍然面临着移动终端切换时延过长、话音容量低、安全隐患以及QoS保证等技术上的挑战。文章简要介绍了VoIP技术,重点研究了VoWLAN终端的移动性管理,在现有支持移动性管理协议的基础上,融合各层协议提出了一种改进的移动性管理新方案。     

Performance Optimizations for Deploying VoIP Services in Mesh Networks

In the recent past, there has been a tremendous increase in the popularity of VoIP services as a result of huge growth in broadband access. The same voice-over-Internet protocol (VoIP) service poses new challenges when deployed over a wireless mesh network, while enabling users to make voice calls using WiFi phones. Packet losses and delay due to interference in a multiple-hop mesh network with limited capacity can significantly degrade the end-to-end VoIP call quality. In this work, we discuss the basic requirements for efficient deployment of VoIP services over a mesh network. We present and evaluate practical optimizing techniques that can enhance the network capacity, maintain the VoIP quality and handle user mobility efficiently. Extensive experiments conducted on a real testbed and ns-2 provide insights into the performance issues and demonstrate the level of improvement that can be obtained by the proposed techniques. Specifically, we find that packet aggregation along with header compression can increase the number of supported VoIP calls in a multihop network by 2-3 times. The proposed fast path switching is highly effective in maintaining the VoIP quality. Our fast handoff scheme achieves almost negligible disruption during calls to roaming clients     

3G手机VoIP话音的QoS实现技术研究

在对3G手机VoIP话音QoS的主要实现技术进行分析的基础上,提出了3G手机VoIP话音QoS新的实现技术。文中通过对实时传输控制协议（RTCP协议）的详细研究,同时根据3G系统无线信道的具体特点,说明了实时传输控制协议运用于3G手机VoIP话音的QoS控制中的缺陷,并阐述了相应的控制解决方法。在基于Android的3G智能手机的VoIP客户端软件中,综合运用VoIP话音QoS的主要成熟实现技术,同时结合文中提出的VoIP话音QoS的解决思路,实现了对VoIP话音的QoS的控制。基于Android的3G智能手机的VolP客户端软件通过在不同的网络环境条件下的测试,VoIP话音质量良好,说明文中提出的3G手机VoIP话音QoS新的实现技术具有一定的实用价值。     

一种支持VoIP语音高优先级传输的WLAN系统设计

针对无线局域网（WLAN）多址接入的特点以及VoIP业务自身的质量要求，提出了一种设计方法，将上下行数据发送分离，支持实时业务的上下行对等传输，同时通过一种改进的早期随机检测（RED）算法，实现基于语音抖动参数的流量控制，为语音业务提供QoS保证。     

Seamless SIP-based mobility for multimedia applications

Application-level protocol abstraction is required to support seamless mobility in next-generation heterogeneous wireless networks. Session initiation protocol (SIP) provides the required abstraction for mobility support for multimedia applications in such networks. However, the handoff procedure with SIP suffers from undesirable delay and hence packet loss in some cases, which is detrimental to applications like voice over IP (VoIP) or streaming video that demand stringent quality of service (QoS) requirements. In this article we present a SIP-based architecture that supports soft handoff for IP-centric wireless networks. Soft handoff ensures that there is no packet loss and that the end-to-end delay jitter is kept under control.     

VoIP语音时延的分析和研究

文章介绍了VoIP(IP网络上传送语音)语音质量的测试方法,分析了影响VoIP语音质量的主要因素:延迟、抖动、丢包率和时延.利用E模型定量地分析了语音质量与端到端时延的关系,通过建立数学模型,指出了VoIP 系统中主要的时延分量,并研究了这些时延分量产生的机理和影响它们的参数.在设计实际的VoIP系统时,可以通过优化影响时延分量的主要参数,改善VoIP系统的时延.     

VoIP业务QoS性能及其优化研究

简要介绍了VoIP传输的基本原理,对影响VoIP业务QoS性能的3个主要因素(时延、抖动和丢包)进行分析,提出了利用MPLSdiffserv awareTE(流量工程)集成模型进行端到端QoS性能优化的方法。MPLSdiffserv awareTE能够感知CoS(服务等级),并根据CoS细粒度来预留资源,在每个CoS级别提供MPLS容错机制,能够为VoIP业务提供低丢失、低延迟、低抖动以及确定的带宽服务,很好地满足服务质量要求。     

Contention‐free approaches for WiFi MAC design for VoIP services: performance analysis and comparison

The exponential growth in the demand of voice over internet protocol (VoIP) services along with the increasing demand for mobility in VoIP services has attracted great research efforts towards provisioning of VoIP services in IEEE 802.11‐based Wireless LANs (WiFi networks). We address one of the important research problems, namely, the quality of service (QoS)‐aware efficient silence suppression in the bursty voice traffic, for provisioning VoIP services in WiFi networks. The research works in the recent literature on silence suppression in voice calls have been surveyed categorising them on how the activity arrival is notified to the access point (AP). In most of the recent schemes, notification of uplink activity arrival is done through contention based medium access mechanisms such as the distributed coordination function (DCF). Contention‐based medium access causes non‐deterministic delays, therefore such schemes are not suited to voice traffic which require strict delay bound guarantees. This paper focuses on the schemes which do not use contention based approaches for silence suppression in voice traffic. Analytical performance evaluation and comparison of such schemes is carried out. Two very important performance metrics are modelled mathematically. One is the expected polling overhead time that the schedulers in these schemes can save per voice call during one voice activity cycle as compared to that in the round‐robin polling scheduler. The other is the expected unnecessary wireless channel access delay that a typical first talk‐spurt frame experiences due to the specific design of each scheme. The numerical results of this evaluation lead us to the conclusion whether or not and to what extent each of these schemes is viable. Copyright © 2008 John Wiley & Sons, Ltd.     

一种动态时分窄带多业务接入新方案

如何在窄带低比特率链路上进行高效的语音数据等多业务综合接入,并保证语音等实时业务的质量是目前多业务接入的一个重点问题.本文提出的动态时分多业务接入方案解决了Digital Data Network (DDN)专线上窄带压缩语音,ADPCM语音,传真,以及数据的同时接入问题,有效地解决了DDN专线上多业务接入的质量保证问题,提高了链路利用率.本文对于该方案的性能进行了分析,并与目前基于IP的多业务接入方案进行了比较.本方案能够提供目前的VoIP方案下所无法提供的语音,传真业务的质量保证,在多业务的支持上比VoIP更加简单,更有吸引力.     

Quality of Service in VoIP Communication

The focus of this paper lies in the practical aspects of voice over IP communication. VoIP configurations in the H.323 standard will be presented briefly. Following that, the fundamental protocol procedure of H.323 communication will be briefly explained. A further part of the paper will address the subject QoS (quality of service), and present the common measurement methods used in QoS. Results gained from experiments conducted in a VoIP environment will then follow. The investigations concentrate primarily on the load behavior of voice packets in relation to important parameters of this service. The results obtained are presented and evaluated in diagrams. The paper concludes with a summary.     

Performance evaluation of packet aggregation scheme for VoIP service in wireless multi-hop network

In a wireless multi-hop network environment, energy consumption of mobile nodes is an important factor for the performance evaluation of network life-time. In Voice over IP (VoIP) service, the redundant data size of a VoIP packet such as TCP/IP headers is much larger than the voice data size of a VoIP packet. Such an inefficient structure of VoIP packet causes heavy energy waste in mobile nodes. In order to alleviate the effect of VoIP packet transmission on energy consumption, a packet aggregation algorithm that transmits one large VoIP packet by combining multiple small VoIP packets has been studied. However, when excessively many VoIP packets are combined, it may cause deterioration of the QoS of VoIP service, especially for end-to-end delay. In this paper, we analyze the effect of the packet aggregation algorithm on both VoIP service quality and the energy consumption of mobile nodes in a wireless multi-hop environment. We build the cost function that describes the degree of trade-off between the QoS of VoIP services and the energy consumption of a mobile node. By using this cost function, we get the optimum number of VoIP packets to be combined in the packet aggregation scheme under various wireless channel conditions. We expect this study to contribute to providing guidance on balancing the QoS of VoIP service and energy consumption of a mobile node when the packet aggregation algorithm is applied to VoIP service in a wireless multi-hop networks.     

Performance analysis of macrodiversity, voice/data CDMA systems

A performance analysis is presented for macrodiversity integrated voice/data CDMA systems. Macrodiversity with maximal ratio combining (MMRC) is ideal for such voice/data systems since it allows robust resource sharing between the two users while removing uncertainties in estimating the system capacity. Our analytical model allows the systems to dynamically allocate additional channels to data users to increase throughput when the necessary resource is available. The data users are characterized by arrival rates and average data size, and the resulting data user quality of service (QoS) performances are evaluated by using a simple birth-and-death Markov process. Our analytical results are fully verified by computer simulation. We show how various system QoS measures such as blocking and outage probabilities can be obtained and used in call admission control (CAC) decisions.     

An analytical simulator for deploying IP telephony

Deploying IP telephony or voice over IP (VoIP) is a major and challenging task. This paper describes an analytical design and planning simulator to assess the readiness of existing IP networks for the deployment of VoIP. The analytical simulator utilizes techniques used for network flows and queuing network analysis to compute two key performance bounds for VoIP: delay and bandwidth. The simulator is GUI‐based and has an interface with drag‐and‐drop features to easily construct any generic network topology. The simulator has an engine that automates and implements the analytical techniques. The engine determines the number of VoIP calls that can be sustained by the constructed network while satisfying VoIP QoS requirements and leaving adequate capacity for future growth. As a case study, the paper illustrates how the simulator can be utilized to assess the readiness to deploy VoIP for a typical network of a small enterprise. We have made the analytical simulator publicly available in order to improve and ease the process of VoIP deployment. Copyright © 2008 John Wiley & Sons, Ltd.     

在MPLS网络中如何保证VoIP的QoS

在以IP为基础的网络中,IP网络上传送语音(VoIP)成为新一代语音系统的代表,但其业务质量因由IP网络承载而受到限制.多协议标签交换(MPLS)是当前被普遍看好的高速骨干网络技术,通过MPLS,第三层的路由可以得到第二层技术的很好补充,以保证端到端的服务质量(QoS).将MPLS的QoS特性与区分服务(DiffServ)相结合,可实现更优化的网络业务QoS.     

VoIP capacity analysis in cognitive radio system

In this letter, we analyze a voice over IP (VoIP) capacity in a cognitive radio system. We formulate the system as a two-dimensional discrete time Markov chain (DTMC). The VoIP traffic and wireless channel in the cognitive radio system are described as a Markov modulated Poisson process (MMPP) model and a Markov channel model, respectively. We demonstrate various numerical and simulation results, such as packet dropping probability and VoIP capacity.     

Voice over the internet: A tutorial discussing problems and solutions associated with alternative transport

This article provides a tutorial overview of voice over the Internet, examining the effects of moving voice traffic over the packet switched Internet and comparing this with the effects of moving voice over the more traditional circuit-switched telephone system. The emphasis of this document is on areas of concern to a backbone service provider implementing Voice over IP (VoIP). We begin by providing overviews of the Plain Old Telephone Service (POTS) and VoIP. We then discuss techniques service providers can use to help preserve service quality on their VoIP networks. Next, we briefly discuss Voice over ATM (VoATM) as an alternative to VoIP. Finally, we offer some conclusions.     

Quality assurance of voice over WLANs (VoWLANs) with differentiated services

Several technical issues make commercial and large voice over wireless local area network (VoWLAN) services difficult to provide. The most challenging issue when voice over Internet Protocol (VoIP) services are ran over IEEE 802.11-based WLANs is the bandwidth inefficiency due to the considerable overhead associated with WLAN packet transmission. In this work, we propose a session-based quality-of-service management architecture (SQoSMA) to overcome the low number of VoIP calls in IEEE 802.11 Wireless LANs and the negative effect of new call addition when the WLAN reaches its capacity. The SQoSMA combines data and control planes to detect VoWLAN QoS degradations and performs either an adaptive audio codec switching or a call stopping to fix VoWLAN issues in a differentiated services manner. In addition, our solution deals with user sessions information, by considering user priority (from its agreement) to guarantee a certain level of its multimedia applications. Performance evaluation using a real test-bed shows that call codec change and call stopping techniques can easily assure high-priority calls with acceptable call blocking probability.     

Efficient silence suppression and call admission control through contention-free medium access for VoIP in WiFi networks

In view of the rapidly growing trend of migrating customers from traditional wired phones to mobile phones and then to VoIP services in the recent past, there is a tremendous demand for wireless technologies to support VoIP, specially on WiFi technologies which have already matured commercially. This has put forth great research challenges in the area of wireless VoIP. In this article we have addressed two core issues, efficient silence suppression and call admission control, in QoS provisioning for VoIP services in WiFi networks. In this connection we present a QoS-aware wireless MAC protocol called hybrid contention-free access (H-CFA) and a VoIP call admission control technique called the traffic stream admission control (TS-AC) algorithm. The H-CFA protocol is based on a novel idea that combines two contention-free wireless medium access approaches, round-robin polling and TDMA-like time slot assignment, and provides substantial multiplexing capacity gain through silence suppression of voice calls. The TS-AC algorithm ensures efficient admission control for consistent delay bound guarantees and further maximizes the capacity through exploiting the voice characteristic so that it can tolerate some level of non-consecutive packet loss. We expose the benefits of our schemes through numerical results obtained from simulations.     

Supporting multimedia applications through network redesign

Recent evolutions in high‐performance computing and high speed broadband Internet access have paved a way to enterprise‐wide multimedia applications, which require stern QoS from the underlying networks. In this paper, we have explored threefold studies on existing enterprise network, whereby we proposed an analytical approach to evaluate the performance of the existing network; we have examined the feasibility of existing enterprise networks to accommodate voice over Internet protocol (VoIP) services with acceptable QoS, and we have redesigned the enterprise network to accommodate VoIP services to comply with the user defined QoS. The network performance is evaluated by number of VoIP calls sustained by the network, bandwidth utilization, loss rate and latency through Network Simulation (NS‐2) tool. We have derived a cost model to show the cost‐effectiveness of VoIP services over telephonic network. For a medium‐size enterprise network of 200 clients and 9 servers, our simulation results show that the redesign improves the network performance by increasing the number of VoIP calls by 57% and decreasing bandwidth utilization and packet loss rate by 20% and 7%, respectively. Moreover, the proposed network redesign demonstrates that the network can be scalable and it can handle up to 4% increased voice calls in the future maintaining QoS standards. Copyright © 2012 John Wiley & Sons, Ltd.