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.     

基于RTP协议的IP电话QoS监测及提高策略

介绍了多媒体通信的发展趋势和当前存在的问题,对基于RTP协议的网络电话中音频数据传输技术进行了研究,对影响实时传输质量QoS的典型因素进行了分析。在局域网的环境下进行了语音包分析实验,探讨了基于RTP协议的QoS动态监测方法,并提出可行的改进方案。     

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.     

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.     

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.     

Assessing the quality of voice communications over Internet backbones

As the Internet evolves into a ubiquitous communication infrastructure and provides various services including telephony, it will be expected to meet the quality standards achieved in the public switched telephone network. Our objective in this paper is to assess to what extent today's Internet meets this expectation. Our assessment is based on delay and loss measurements taken over wide-area backbone networks and uses subjective voice quality measures capturing the various impairments incurred. First, we compile the results of various studies into a single model for assessing the voice-over-IP (VoIP) quality. Then, we identify different types of typical Internet paths and study their VoIP performance. For each type of path, we identify those characteristics that affect the VoIP perceived quality. Such characteristics include the network loss and the delay variability that should be appropriately handled by the playout scheduling at the receiver. Our findings indicate that although voice services can be adequately provided by some ISPs, a significant number of Internet backbone paths lead to poor performance.     

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.     

Improving the R-score of an adaptive VoIP codec in IEEE 802.16 networks

Worldwide Interoperability for Microwave Access (WiMAX) technology, which is based on the IEEE 802.16 standard, supports different quality of service (QoS) for different services. WiMAX is expected to support QoS in real-time applications such as Voice over Internet Protocol (VoIP). When network congestion occurs, the VoIP bit rate needs to be adjusted to achieve the best speech quality. In this study, we propose a new scheme called Adaptive VoIP Level Coding (AVLC). This scheme takes into consideration network conditions (packet delay and packet loss) and a connection’s modulation scheme. The amount of data that can be transmitted increases with the speed of the modulation scheme. When network congestion occurs, AVLC scheme prioritizes reducing the bit rate of a connection that has a slower modulation scheme to mitigate congestion. Depending on network conditions, such as modulation scheme, packet delay, packet loss, and residual time slot, we use the G.722.2 codec to adjust each connection’s bit rate. Simulations are conducted to test the performance (network delay, packet loss, number of modulation symbols, and R-score) of the proposed scheme. The simulation results indicate that speech quality is improved by the use of AVLC.     

QoS evaluation of sender-based loss-recovery techniques for VoIP

Voice over Internet protocol (VoIP) is a technology that transports voice data packets across packet-switched networks using the Internet protocol (IP). Losing packets in the network is inevitable, and losing voice packets degrades audio quality. There are many loss-recovery techniques that designers can use to mitigate the undesired effects of packet loss. Some of these loss-recovery techniques use sender-based procedures, and others use receiver-based procedures. We examine several well-known sender-based loss-recovery techniques and evaluate the feasibility and effectiveness of each one in real-time interactive VoIP applications. We analyze the bandwidth requirements, buffering delays, and perceptual sound qualities of these techniques. We study the effectiveness of these approaches under various packet-loss conditions, and we also compare the effectiveness of these techniques against a speech codec that has high degree of packet-loss robustness     

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.     

The effect of packet dispersion on voice applications in IP networks

Delivery of real time streaming applications, such as voice and video over IP, in packet switched networks is based on dividing the stream into packets and shipping each of the packets on an individual basis to the destination through the network. The basic implicit assumption on these applications is that shipping all the packets of an application is done, most of the time,over a single path along the network. In this work, we present a model in which packets of a certain session are dispersed over multiple paths, in contrast to the traditional approach. The dispersion may be performed by network nodes for various reasons such as load-balancing, or implemented as a mechanism to improve quality, as will be presented in this work. To study the effect of packet dispersion on the quality of voice over IP (VoIP) applications,we focus on the effect of the network loss on the applications, where we propose to use the Noticeable Loss Rate (NLR) as a measure (negatively) correlated with the voice quality. We analyze the NLR for various packet dispersion strategies over paths experiencing memoryless (Bernoulli) or bursty (Gilbert model) losses,and compare them to each other. Our analysis reveals that in many situations the use of packet dispersion reduces the NLR and thus improves session quality. The results suggest that the use of packet dispersion can be quite beneficial for these applications.     

QoS requirements for a voice-over-IP PSTN

Voice over IP is already widespread in enterprise private networks and is growing in public switched voice networks as manufacturers withdraw support for earlier technologies. Packet transmission of voice can introduce new impairments, including packet loss, extra sources of delay, and the use of compressed speech coding, all of which may affect voice quality delivered to the user. Factors affecting the quality of a voice telephony connection are described, concentrating on those which are changed by the move to packet transmission, including the complex area of delay. We outline subjective testing based on users’ opinions of fragments of recorded audio material or of connections realised in a laboratory, and describe the abstraction of these results into transmission planning models to assist with design of networks and their QoS mechanisms. QoS requirements are stated for a packet technology to support a PSTN and ISDN service in the UK telecommunications environment.     

A multiplexing scheme for H.323 voice-over-IP applications

Voice communications such as telephony are delay sensitive. Existing voice-over-IP (VoIP) applications transmit voice data in packets of very small size to minimize packetization delay, causing very inefficient use of network bandwidth. This paper proposes a multiplexing scheme for improving the bandwidth efficiency of existing VoIP applications. By installing a multiplexer in an H.323 proxy, voice packets from multiple sources are combined into one IP packet for transmission. A demultiplexer at the receiver-end proxy restores the original voice packets before delivering them to the end-user applications. Results show that the multiplexing scheme can increase bandwidth efficiency by as much as 300%. The multiplexing scheme is fully compatible with existing H.323-compliant VoIP applications and can be readily deployed.     

Mobility Management for VoIP on Heterogeneous Networks: Evaluation of Adaptive Schemes

The introduction of the IP multimedia subsystem on 3G cellular networks and the integration with other widely deployed wireless networks based on the IEEE 802.11 protocol family require support for both mobility and quality of service. When mobile systems move across heterogeneous networks, ongoing real-time sessions are affected not only by handoff delay but also by different packet delay and bit rate. In this paper, we propose a cross-layer mechanism that takes into account mobility at different layers of the network stack in order to yield better quality for VoIP, videoconferencing, and other real-time applications. We describe our cross-layer architecture, adaptation techniques, a prototype implementation, and experimental results.     

A perceptual quality model intended for adaptive VoIP applications

Quality models predict the perceptual quality of services as they calculate subjective ratings from measured parameters. In this article, we present a new quality model that evaluates Voice over IP (VoIP) telephone calls. In addition to packet loss rate, coding mode and delay, it takes into account the impairments due to changes in the transmission configuration (e.g. switching the coding mode or re‐scheduling the playout time). Moreover, this model can be used at run time to control the transmission of such calls. It is also computationally efficient and open source. To demonstrate the potential of our model, we apply it to select the ideal coding and packet rate in bandwidth‐limited environments. Furthermore, we decide, based on model predictions, whether to delay the playout of speech frames after delay spikes. Delay spikes often occur after congestion and cause packets to arrive too late. We show a considerable improvement in perceptual speech quality if our model is applied to control VoIP transmissions. Copyright © 2005 John Wiley & Sons, Ltd.     

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     

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.     

Voice/Data Integration Using Circuit Switched Networks

This paper assesses the impact of integrating voice and data over circuit switched networks. Three main types of circuit switching are considered: 1) traditional circuit switching, 2)fast circuit switchingemploying advanced switching speeds, and 3) enhanced circuit switchingemploying time assigned speech interpolation (TASI) and adaptive data multiplexing (ADM) techniques. The circuit switching networks are evaluated in terms of two main network performance parameters: transmission efficiency and delay. In addition, an evaluation is made of such things as protocol and error control, precedence and preemption, routing and flow control, synchronization, voice continuity, probability of error or loss, and classmarking flexibility. One of the main conclusions of this paper is that circuit switching technologies have several deficiencies associated with providing integrated voice/data service and that the future lies in the effective use of packet and hybrid (circuit/packet) switching technologies.