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.     

一种语音信号重建算法

为了减轻因信包丢失而造成的语音失真,提出了一种基于双边线性预测的信包丢失隐藏算法。这种方法利用丢失信包的前一信包或邻接信包(在后一信包可获得的情况下)预测丢失信包,通过线性加权双边线性预测的样点获得最终的重建信号,使用重叠相加和幅度调整操作平滑重建信号和真实信号之间的边界。经过非正式试听和ITU-T P.862协议所推荐的PESQ算法测试,该算法的重建语音信号质量与其他四种流行重建算法相比,有了较为明显的改善。     

一种新的语音流队列管理调度机制

本文提出一种新的语音流队列管理调度机制,结合随机早期探测(RED)和主动丢包调度算法实现因特网语音流的队列管理和调度.采用仿真方法分析了新机制的性能特征,并与RED做了性能对比.当网络拥塞时,该算法可有效改善包转发的性能.语音质量测试表明新机制是可行的和有效的.     

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.     

VoIP电话的安全问题及防护措施

VoIP电话是综合了传统电信技术与计算机网络技术的一种新型应用。VoIP电话在传输时将信号压缩后封装成IP包，在IP网络上传输，这种传输方式存在着各种安全隐患。讨论了VoIP电话多种可能存在的安全问题，并就这些安全问题作了详细的分析，同时提出了防护措施，以最大限度地保障VoIP电话的安全。     

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     

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.     

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 adaptive algorithm for connecting mobile ad hoc network to Internet with unidirectional links supported

The performance of Internet connectivity is degraded dramatically by blind rebroadcast of gateway discovery packets when obvious unidirectional links are ignored. To overcome this problem, an effective adaptive gateway discovery algorithm for connecting mobile ad hoc network to Internet with unidirectional links supported is proposed in this paper. On the basis of modification for ad hoc on-demand distance vector (AODV), through rebroadcasting the gateway discovery messages and gateway advertisement messages, both of which are extended with local connection information, unidirectional links are successfully removed from global route computations and Internet connectivity is simultaneously enhanced. Furthermore, an adaptive Internet working scheme is adopted to provide the best coverage of gateway advertisement according to dynamically adjusting broadcast range and sending interval of gateway advertisement messages in terms of network conditions. Simulation results show that the proposed hybrid Internet connectivity algorithm can provide better connectivity performance due to avoiding unidirectional links effectively with reasonable overhead.     

New voice over Internet protocol technique with hierarchical datasecurity protection

The authors propose a voice over Internet protocol (VoIP) technique with a new hierarchical data security protection (HDSP) scheme. The proposed HDSP scheme can maintain the voice quality degraded from packet loss and preserve high data security. It performs both the data inter-leaving on the inter-frame of voice for achieving better error recovery of voices suffering from continuous packet loss, and the data encryption on the intra-frame of voice for achieving high data security, which are controlled by a random bit-string sequence generated from a chaotic system. To demonstrate the performance of the proposed HDSP scheme, we have successfully verified and analysed the proposed approach through software simulation and statistical measures on several test voices     

IP电话在HFC上的实现

符合ＭＧＣＰ体系的ＨＦＣ上的ＩＰ电话系统充分利用了ＣＡＴＶ网络资源，从根本上满足了大规模ＩＰ电话服务的需要，在对ＨＦＣ上的ＩＰ电话系统进行讨论的基础上，提出了能同时处理两路语音信号的基于Ｃａｂｌｅ Ｍｏｄｅｍ的ＩＰ电话适配卡设计方案。     

海事卫星通信系统VoIP技术的应用分析

随着VoIP技术的发展,VoIP技术结合卫星通信网络的应用越来越广泛。Inmarsat卫星系统是地球同步轨道系统,网络传播时延大,卫星VoIP电话的语音通信是否可行值得研究。结合VoIP关键技术和海事卫星通信语音通信应用场景,探讨了基于Inmarsat卫星网络实现VoIP技术的方案,并分析出此方案下VoIP系统通话过程的单向时延为350 ms,低于ITU G.114的400 ms的要求。在实际使用环境中进行了测试和验证,结果表明,基于Inmarsat网络下实现VoIP的方案是可行的。该方案实现复杂度低,可以方便地实现Inmarsat网络与地面电话网之间的互联互通,也可以为我国自主研制的宽带卫星通信系统实现VoIP技术提供参考。     

卫星物联网混合随机接入技术

在卫星物联网(IoT)场景中,随着终端数量不断增加,频谱资源日益紧张。传统的随机接入技术频谱利用率较低,使得传统随机接入协议不适用于未来卫星IoT的高并发业务需求。同时,卫星通信链路长,开放性强,难以保证特种终端信号的安全性。对此,本文提出一种适用于卫星IoT的混合随机接入方案。该方案引入重叠传输的容量提升与安全性优势,利用扩频码对瞬时功率谱密度的控制能力,构造功率域非正交接入条件,并通过接收端的迭代分离实现稳健接收。对本文所提方案的吞吐量性能进行闭式解推导分析与计算机仿真,结果表明,与传统的随机接入协议相比,所提方案可提高系统吞吐量。同时,相较于常用信号隐藏方法,所提方法利用常规接入数据包的功率优势,强化了波形隐藏效果,提升了特种信息接入的安全性。     

Optimum VoIP packet routing using a communications processor

This article presents a new architecture for the VoIP media gateway using only a communications processor and digital signal processors. The new architecture can be used by telecommunications equipment manufactures to replace a network processor and a general-purpose processor with a single communications processor, thereby can reduce the system cost, power consumption, printed circuit board (PCB) area, software complexity and time to market. In the new architecture the modules are interconnected via Ethernet interfaces, which make voice packet encapsulation possible in digital signal processors. This relieves the network processor, which in voice over IP (VoIP) media gateways is most commonly used for the routing of VoIP packets and voice-packet encapsulation, and means it can be replaced by a communications processor. The presented media gateway architecture makes it possible to combine the data- and control-plane application on a single-communications processor, but only in the case of a properly optimized program code and an optimized Ethernet driver. Therefore, the main part of the article is dedicated to a presentation of the methodology for the analysis and optimization of the presented systems. In order to support this methodology, a new tool named performance monitor (PM) was developed. The PM tool is presented here, and was used for optimizing the Ethernet driver. The Ethernet driver was optimized and modified in such a way as to put a minimal load on the microprocessor core of the communications processor when routing the VoIP packets to digital signal processors and back. The article ends with a presentation of the experimental optimization results, which were acquired from a real telecommunications system.     

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     

A Low Latency Handoff Algorithm for Voice over IP Traffics in the Next Generation Packet-Based Cellular Networks: Cellular Mobile IPv6

In the mobile communication environments, Mobile IP is defined to provide users roaming everywhere and transmit information freely. It integrates communication and network systems into Internet. The Mobile IPv6 concepts are similar to Mobile IP, and some new functions of IPv6 bring new features and schemes for mobility support. Two major problems in mobile environments are packet loss and handoff. To solve those problems, a mobile management scheme – the cellular mobile IPv6 (CMIv6) is proposed. Our approach isbased on the Internet Protocol version 6 and is compatible with the Mobile IPv6 standard. Besides, it also combines with the cellular technologies which is an inevitable architecture for the future Personal Communication Service system (PCS). In this paper, {Cellular Mobile IPv6 (CMIv6)}, a new solutionmigrated from Mobile IPv6, is proposed for mobile nodes moving among small wireless cells at high speed. This is important for future mobile communication trends. CMIv6 can solve the problems of communication break off within smaller cellular coverage during high-speed movement when packet-switched data or the real-time voice messages are transmitted. Voice over IP (VoIP) packets were chosen to verify this system. The G.723.1 Codec scheme was selected because it has better jitter resistance than GSM and G729 in a packet-based cellular network. Simulation results using OPNET show smooth and non-breaking handoffs during high-speed movement.     

Performance Analysis of an Enhanced PRMA-HS Protocol for LEO Satellite Communication

The packet reservation multiple access with hindering state （PRMA-HS） is a protocol suitable for LEO satellite mobile communication. Although working well with light system payload （amount of user terminals）, the protocol imposes high channel congestion on system with heavy payload, thus degrades the system＇s quality of service. To controlling the channel congestion, a scheme of enhanced PRMA-HS protocol is proposed, which aims to reduce the collision of voice packets by adopting a mechanism of access control. Through theoretic analysis, the system＇s mathematic model is presented and the packet drop probability of the scheme is deduced. To testify the performance of the scheme, a simulation is performed and the results support our analysis.     

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.     

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.     

Type-2 fuzzy logic control of PQoS driven adaptive VoIP scheme

Adaptive VoIP schemes have potentially suboptimal performance owing to imprecision in the metrics used to infer network state. An interval Type-2 fuzzy logic controlled scheme for VoIP services is presented. It infers network state from average delivered perceived quality of service and its degradation due to network congestion and updates an AMR codec mode to match voice quality to available network bandwidth. Tests showed that the scheme maximised delivered voice quality and outperformed an existing adaptive scheme. The scheme achieves robust performance in the presence of input imprecision and can be implemented in VoIP terminals, and the fuzzy rule base is easy to understand and change by non-experts because of its similarity to the human decision-making process.