NCTU‐VT: a freeware for wireless VoIP performance measurement

Voice over IP (VoIP) is a promising low‐cost voice communication over the wireless IP network. To provide satisfactory VoIP services, the Quality of Service (QoS) of the wireless network should be guaranteed. This paper proposes a VoIP performance measurement freeware called NCTU VoIP Testing Tool (NCTU‐VT). We compare NCTU‐VT with two commercial tools SmartVoIPQoS and IxChariot in terms of packet loss, latency, and Mean Opinion Score (MOS) of the VoIP sessions in Wi‐Fi network. Our study indicates that these three tools can accurately measure VoIP performance in Wi‐Fi environment. Copyright © 2010 John Wiley & Sons, Ltd.     

A framework for network quality monitoring in the VoIP environment

Monitoring speech quality in Voice over IP (VoIP) networks is important to ensure a minimal acceptable level of speech quality for IP calls running through a managed network. Information such as packet loss, codec type, jitter, end‐to‐end delay and overall speech quality enables the network manager to verify and accurately tune parameters in order to adjust network problems. The present article proposes the deployment of a monitoring architecture that collects, stores and displays speech quality information about concluded voice calls. This architecture is based on our proposed MIB (Management Information Base) VOIPQOS, deployed for speech quality monitoring purposes. Currently, the architecture is totally implemented, but under adjustment and validation tests. In the future, the VOIPQOS MIB can be expanded to automatically analyze collected data and control VoIP clients and network parameters for tuning the overall speech quality of ongoing calls. Copyright © 2006 John Wiley & Sons, Ltd.     

iPTT: peer‐to‐peer push‐to‐talk for VoIP

This paper proposes iPTT, a peer‐to‐peer (P2P) Push‐to‐Talk (PTT) service for Voice over IP (VoIP). In iPTT, a distributed and mobile‐operator independent network architecture is presented to accelerate the deployment of the PTT service. Based on the serverless architecture, we develop two mechanisms, that is, flooding‐based floor control mechanism (FFC) and tree‐based floor control mechanism (TFC), for real‐time talk‐burst determination. The determination algorithms and the corresponding message flows for these two mechanisms are designed to show the feasibility of FFC and TFC. The performance of FFC and TFC is investigated through our analytical and simulation models in terms of the determination latency and the number of floor‐control message exchanges. Copyright © 2008 John Wiley & Sons, Ltd.     

An analysis of VoIP service using 1/spl times/EV-DO revision a system

While voice-over-Internet protocol (VoIP) on wireline network is maturing, VoIP on wireless mobile network is still in its infancy. This disparity is due to the fact that the wireline bandwidth is abundant and can be traded off for delay performance and overhead, whereas bandwidth in wireless mobile network is still a scarce resource. With the deployment of 1/spl times/EV-DO revision 0 (DOr0) worldwide, the spectrum efficiency has been significantly improved. However, DOr0 still lacks of features essential for VoIP. For this reason, 1/spl times/EV-DO revision A (DOrA) has been standardized in the 3GPP2 with many improvements favorable for VoIP implementation. In this paper, we identify challenges and explore the feasibility of implementing VoIP using DOrA. We develop both analytical and simulation models to evaluate the VoIP capacity and delay performance over the air interface.     

Active networks for efficient distributed network management

The emerging next generation of routers exhibit both high performance and rich functionality, such as support for virtual private networks and QoS. To achieve this, per-flow queuing and fast IP filtering are incorporated into the router hardware. The management of a network comprising such devices and efficient use of the new functionality introduce new challenges. A truly distributed network management system is an attractive candidate to address these challenges. We describe how active network techniques can be used to allow fast and easy deployment of distributed network management applications in IP networks. We describe a prototype system where legacy routers are enhanced with an adjunct active engine, which enables the safe execution and rapid deployment of new distributed management applications in the network layer. This system can gradually be integrated in today's IP network, and allows smooth migration from IP to programmable networks. This is done with an emphasis on efficient use of network resources, which is somewhat obscure by many of today's high-level solutions     

RIDE Replacement — A Hybrid PSTN/VoIP Application Solution

This paper, co-authored by BT and Telspec, describes the architecture, development and deployment of the recorded information distribution equipment (RIDE) replacement platform. The platform delivers large volumes of announcements into the PSTN yet has an internal architecture built on a multicast QoS-enabled VoIP network and provides a ready-made evolution path to terminate calls originated from IP networks. It is therefore seen as one of the first platforms to be deployed by BT to deliver consistent functionality into both the legacy PSTN and the evolving IP world.     

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     

Estimation of token bucket parameters for aggregated VoIP sources

The paper presents an analytical study aimed to establish a dimensioning procedure for the token bucket algorithm, used as a meter in a Differentiated Services network architecture, when a stochastic model for the multiplexed traffic is available. In the work, we propose an equivalent queueing system method to ‘on line’ estimate the linear bounded arrival processes (LBAP) parameters when a non‐zero probability of non‐conforming packets is accepted. Then, we validate our approach considering an aggregation of fluidic On‐Off processes with exponentially distributed sojourn time in each state, used to model telephone sources with voice activity detection (VAD). To test the goodness of our analytical results, we employ discrete event simulations, which have highlighted the accuracy of the proposed dimensioning procedure in a voice over IP (VoIP) scenario. Moreover, we investigate the multiplexing gain and the effect of different parameters on the LBAP characterization of the VoIP traffic, taking into account a non‐zero probability of non‐conforming packets. Copyright © 2002 John Wiley & Sons, Ltd.     

Evaluating VoIP Technology — Technology Push/Market Pull

With many raised expectations of what it will offer, voice over IP has enjoyed a high profile in the communications industry. However, in common with any emerging technology, the reality of the functionality that available equipment can deliver is often at variance with such expectations. This paper highlights the role of technology assessment in refining VoIP products and highlights the value of such activities from a number of perspectives. It describes an approach to evaluate VoIP solutions, the problems associated with the testing of VoIP networks and network components, and the tools and techniques employed to do so. The paper also outlines the capabilities of the BTexaCT VoIP technology evaluation test bed, and example case studies of detailed evaluation and testing.     

Measurement Scheme for One‐Way Delay Variation with Detection and Removal of Clock Skew

One‐way delay variation (OWDV) has become increasingly of interest to researchers as a way to evaluate network state and service quality, especially for real‐time and streaming services such as voice‐over‐Internet‐protocol (VoIP) and video. Many schemes for OWDV measurement require clock synchronization through the global‐positioning system (GPS) or network time protocol. In clock‐synchronized approaches, the accuracy of OWDV measurement depends on the accuracy of the clock synchronization. GPS provides highly accurate clock synchronization. However, the deployment of GPS on legacy network equipment might be slow and costly. This paper proposes a method for measuring OWDV that dispenses with clock synchronization. The clock synchronization problem is mainly caused by clock skew. The proposed approach is based on the measurement of inter‐packet delay and accumulated OWDV. This paper shows the performance of the proposed scheme via simulations and through experiments in a VoIP network. The presented simulation and measurement results indicate that clock skew can be efficiently measured and removed and that OWDV can be measured without requiring clock synchronization.     

WLAN中基于MIPv6的移动VoIP切换性能仿真分析

移动IP是一个在Internet上基于网络层提供移动性支持功能的要求较高的VoIP业务,切换延迟将直接影响到话音质量,严重时甚至会中断正在进行的会话.文章借助ns2网络模拟器仿真分析了WLAN中基于MIPv6的移动VoIP切换性能.结果表明,MIPv6及其扩展协议的切换性能优劣顺序依次为:F-HMIPv6、FMIPv6、HMIPv6、MIPv6.尤其是F-HMIPv6协议,无论端到端延迟还是切换延迟,都得到了最大的改善.所得结论能为网络切换性能的进一步优化提供重要依据.     

Mobility protocols and RSVP performance in wireless IPv6 networks: shortcomings and solutions

Resource reservation protocol (RSVP) is a network‐control protocol used to guarantee Quality‐of‐Service (QoS) requirements for real‐time applications such as Voice‐over‐IP (VoIP) or Video‐over‐IP (VIP). However, RSVP was designed for end‐systems whose IP addresses do not change. Once mobility of an end‐system is allowed, the dynamically changing mobile IP address inevitably impacts on RSVP performance. Our study aims to first quantify the significance of this impact, and then propose a modified RSVP mechanism that provides improved performance during handoffs. Our simulations reveal that the deployment of standard RSVP over Mobile IPv6 (MIPv6) does not yield a satisfactory result, particularly in the case of VIP traffic. Fast Handovers for Mobile IPv6 (FMIPv6) was found to be providing the best performance in all tested scenarios, followed by Hierarchical Mobile IPv6 (HMIPv6) with a single exception: during low handoff rates with VoIP traffic, MIPv6 outperformed HMIPv6. We then designed a new RSVP mechanism, and tested it against standard RSVP. We found that the proposed approach provides a significant improvement of 54.1% in the Total Interruption in QoS (TI_QoS) when deployed over a MIPv6 wireless network. For HMIPv6, performance depended primarily on the number of hierarchical levels in the network, with no improvement in TI_QoS for single‐level hierarchy and up to 37% for a 5‐level hierarchy. FMIPv6 on the other hand, provided no room for improvement due to pre‐handoff signaling and the tunneling mechanism used to ensure a mobile node (MN)'s connectivity during a handoff, regardless of the RSVP mechanism used. Copyright © 2007 John Wiley & Sons, Ltd.     

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.     

An energy‐efficient transmission mechanism for VoIP over IEEE 802.11 WLAN

Voice over IP (VoIP) over WLAN (VoWLAN) is an important application for public and private WLANs. However, VoWLAN systems suffer from several technical challenges such as power consumption of a WLAN station (STA) and service capacity of an access point (AP), making the commercial deployment of a large‐scale VoWLAN service problematic. This study presents a cross‐layer and energy‐efficient mechanism for transmitting VoIP packets over IEEE 802.11 WLAN. The proposed mechanism considers the characteristics of voice packets that can tolerate certain loss, and dynamically disables the medium access control (MAC) layer acknowledgement for voice packets. In doing so, the time and energy consumed to transmit and receive voice packets for an STA can be reduced. Simulation results demonstrate that the mechanism improves the energy efficiency of a VoWLAN STA and WLAN utilization without sacrificing voice qualities. Copyright © 2009 John Wiley & Sons, Ltd.     

Joint Routing and Playout Buffering of IP Telephony Flows in MANETs

The low-cost of packet-based networking technologies with respect to traditional circuit-switched ones and the reliability of the current (wired) IP networks have brought to a considerable employment of the VoIP (Voice over IP) technologies in the voice services market. This success is expected to happen also in mobile ad hoc networks (MANETs), which may offer a good platform for the fast deployment of VoIP mobile networks. However, efforts must be made to improve performance before MANETs can be used for this purpose. One of the main limitations is related to the highly variability of the network topology and channel behavior, which heavily influences the service quality due to route losses and significant delay variations. In this paper, we propose a strategy where these impairments are jointly addressed. The source is responsible for jointly selecting the transmission paths and adjusting the playout delay, with an adaptive inter-talkspurt approach. These tasks are accomplished on the basis of historical data on network connectivity and transmission delays, and are driven by a quality-based approach. The collection of statistics of the network status relies on the QOLSR routing algorithm, whereas the voice quality is measured by means of the ITU-T E-Model.     

VoIP Performance over HSPA with Different VoIP Clients

High Speed Packet Access (HSPA) Holma H, Toskala A (in HSDPA/HSUPA for UMTS, 2006) is expected to provide enough bandwidth for voice over IP (VoIP) service. In this article we assess the performance of VoIP over HSPA with different VoIP clients and voice codecs. The simulations results show that VoIP can have a good voice quality over HSPA if a proper VoIP client and codec is used. However it is possible that the delay can increase with early HSPA implementations (mobile, network).     

Video‐on‐Demand (VoD) deployment over hospitality networks

Video‐on‐Demand (VoD) deployment over existing IP networks has recently gained significant popularity. Typically, the deployment of VoD is done in an arbitrary manner, without utilizing a proper engineering approach. In this paper, we present an engineering approach to deploy VoD services over IP‐based hospitality networks, such as those networks seen in hotels and hospitals. In particular, our approach aims to determine the total number of VoD sessions that can be sustained by an existing hospitality network, while satisfying the QoS requirements of all network services, and at the same time leaving adequate capacity for future growth. We gauge the capacity of the hospitality network to sustain VoD services using both analysis and simulation. The capacity is gauged considering VoD quality of service requirements of throughput and delay constraints. Our analysis utilizes the principles of queuing theory, and our simulation is performed using OPNET simulation. Copyright © 2011 John Wiley & Sons, Ltd.     

基于H.323的VoIP通信系统的安全性的研究

基于IP网络的语音传输(VoIP)技术目前已经发展成为一种专门的通信技术,随着IP电话技术的成熟和应用的普及,IP电话安全成为越来越迫切需要解决的问题。本文简要介绍了基于H．323协议的VoIP的基本原理及体系结构,全面的分析了其在安全方面存在的漏洞与不足,并从IP网络和IP电话产品本身两个方面提出了相应的安全解决方案,可以明显的改善其安全性能。     

Voice over Internet protocol (VoIP)

During the Internet stock bubble, articles in the trade press frequently said that, in the near future, telephone traffic would be just another application running over the Internet. Such statements gloss over many engineering details that preclude voice from being just another Internet application. This paper deals with the technical aspects of implementing voice over Internet protocol (VoIP), without speculating on the timetable for convergence. First, the paper discusses the factors involved in making a high-quality VoIP call and the engineering tradeoffs that must be made between delay and the efficient use of bandwidth. After a discussion of codec selection and the delay budget, there is a discussion of various techniques to achieve network quality of service. Since call setup is very important, the paper next gives an overview of several VoIP call signaling protocols, including H.323, SIP, MGCP, and Megaco/H.248. There is a section on telephony routing over IP (TRIP). Finally, the paper explains some VoIP issues with network address translation and firewalls