民航甚高频VoIP新技术的应用研究与展望

VoIP(Voice over Internet Protocol)即网络话音通信,其工作原理是将模拟的声音数字化,经过压缩与封包之后,以数据包形式在IP网络实时传输。VoIP也叫互联网语音通信或IP电话。甚高频VoIP语音通信使用IP技术,在IP网络上布置支持数字音频的甚高频电台和内话系统,传输话音,区别于基于PCM(Pulse-code modulation,即脉冲编码调制)技术的传统数字基带信号传输。VoIP兼具操作功能性及灵活性,这是基于TDM的传统系统所不具备的。     

Steganalysis of transcoding steganography

Transcoding steganography (TranSteg) is a fairly new IP telephony steganographic method that functions by compressing overt (voice) data to make space for the steganogram by means of transcoding. It offers high steganographic bandwidth, retains good voice quality, and is generally harder to detect than other existing VoIP steganographic methods. In TranSteg, after the steganogram reaches the receiver, the hidden information is extracted, and the speech data is practically restored to what was originally sent. This is a huge advantage compared with other existing VoIP steganographic methods, where the hidden data can be extracted and removed, but the original data cannot be restored because it was previously erased due to a hidden data insertion process. In this paper, we address the issue of steganalysis of TranSteg. Various TranSteg scenarios and possibilities of warden(s) localization are analyzed with regards to the TranSteg detection. A novel steganalysis method based on Gaussian mixture models and mel-frequency cepstral coefficients was developed and tested for various overt/covert codec pairs in a single warden scenario with double transcoding. The proposed method allowed for efficient detection of some codec pairs (e.g., G.711/G.729), while some others remained more resistant to detection (e.g., iLBC/AMR).     

Implementing VoIP: a voice transmission performance progress report

Aiming to introduce voice over IP networks and services in ways that satisfy the voice quality expectations of our customers, we have been conducting laboratory studies of how VoIP transmission affects voice quality while also carefully monitoring and managing several field implementations of VoIP. This article summarizes much of what we have learned in this work, and we hope it provides a useful progress report on the industry's evolution to VoIP. We review our data on the voice quality effects of packet loss, delay, speech coders, packet loss concealment algorithms, and the compression option of suppressing transmission during silence. Because the familiar problem of echo has emerged repeatedly in the VoIP environment, we review this issue in some detail. Packet loss and delay variation measurements made on private VoIP networks are reviewed, and the data here are encouraging. We finish by making our case that the network planning tool known as the E-model is currently an inexact predictor of VoIP network performance.     

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.     

An Approach to Integrating SIP in Converged Multimodal/Multimedia Communication Services

In this paper, we present an approach of integrating SIP (Session Initiation Protocol) in converged multimodal/multimedia communication services. An extensible VoIPTeleserver for VoIP in SIP environment is described. It is based on the concept of dialogue system and Web convergence that separates the channel dependent media resources from the application dependent service creation and hosting environment. It supports XML based service applications for multiple channels including voice, DTMF, IM and chat over IP. The loosely coupled open architecture in our approach is highly extensible. We describe the concept and structure of VoIPTeleServer used in our approach in detail, which interfaces to the VoIP world through SIP signaling and works as a broker between the VoIP SIP environment and MTIP to deliver converged communication services. A prototype of VoIPTeleServer was implemented, and services and applications based on SIP and MTIP convergence are constructed. Special attention is given to the adverse effect of delay, jitter and packet loss for voice portal services over IP. In particular, case studies of DTMF service in voice portal under adverse channel conditions are performed. The compounding effects of multiple channel impairments to DTMF in voice portal services over IP are characterized. The potential high error rate of the DTMF service indicates that the data redundancy method as proposed in RFC 2198 is needed for DTMF in order to achieve reliable voice portal services over IP.     

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.     

Performance analysis of wireless multi-user VoIP system with adaptive modulation and coding

In this paper, we propose a new framework to analyze performance considering finite-length queuing and adaptive modulation and coding for multi-user Voice over IP (VoIP) services in wireless communication systems. We formulate an uplink VoIP system as a two-dimensional discrete-time Markov chain (DTMC) based on a Markov modulated Poisson process traffic model for VoIP services and modulation and coding scheme (MCS)-level set transition reflecting users’ channel variations. We extend the transition modeling of the MCS-level for a single-user to the transition modeling of the MCS-level set for multiple users. Since the users can have various MCS combinations in the case of a multi-user system, the MCS-level set transitions are more complicated than the MCS-level transitions of the single-user case. Throughout our DTMC formulation, we present various performance metrics, such as average queue-length, average throughput, packet dropping probability, packet loss probability, and so on. By using the results of the packet loss probability, we can find an optimum packet error rate value that minimizes the total packet loss probability.     

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.     

The 1974 origins of VoIP

This paper brings light on the digital signal processing (DSP) roots of a modern concept, voice over IP (VoIP). An example is also provided in which developments in DSP - speech coding, in particular - had a profound impact on the early development of the ARPANET, the ancestor of the Internet. The author shows how packet speech, recently rediscovered and made popular as VoIP, was first successfully demonstrated in 1974 on the ARPANET and how the Internet protocol (IP) emerged largely as a result of that effort.     

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     

短波地空语音组网中VoIP语音质量评估

短波地空话音组网是短波话音通信发展的必然趋势,组网后VoIP技术的引入会为短波语音通信带来很大的便利,但同时语音质量会受到影响,其中VoIP编码技术是影响语音质量的主要因素之一。通过Matlab对采用不同编码技术重构后的语音,经过基于Watterson模型的短波信道传输后到达接收端的质量进行了PESQ评估,并仿真分析了丢包率对不同编码语音质量的影响,得出了不同编码的优劣性。     

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.     

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.     

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.     

可堆叠VoIP语音板卡的固件程序设计

本文提出了基于口可堆叠式的VoIP通信应用系统的系统架构,重点介绍了面向IP可堆叠式的VoIP语音板卡的固件程序设计.每块VoIP语音板卡支持8路语音,通过自定义的通信协议可使不同的VoIP语音板卡独立地通过IP互联,实现基于IP可堆叠.自定义通信协议实现了VoIP语音板卡中芯片内部通道之间、VoIP语音板卡上芯片之间、不同VoIP语音板卡之间,以及VoIP语音板卡与管理PC间的通信.VoIP语音板卡控制软件以内核模块方式运行,并在内核模块方式下由VINETIC-2CPE语音芯片中断服务程序激活回调函数,提高了实时性.     

基于Android的语音编码算法研究与应用

针对基于Android的VoIP(Voice over IP)在通信过程中,具有语音数据量大的特点.对目前存在的多种语音编码方案进行分析研究,编写基于Android的VoIP系统对各种方案的语音压缩率、编码速率以及MOS(Mean Opinion Score)值进行测试.结果表明,Speex(一种基于码激励线性预测编码而设计的音频编码压缩格式)语音编码方案在综合考虑多方面因素的情况下,在基于Android的VoIP通信系统中具有相对优势.     

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.     

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     

M—DCF：在单区网中实现QoS的媒体访问协议

本文结合IP通信的特点,提出了一种适用于WLAN单区网的改进型媒体访问控制协议M-DCF,并进行了计算机仿真。仿真结果表明,M-DCF可以降低系统总的丢包率,话音业务的丢包率,降低话音帧的时延,从而保证话音业务的服务质量,同时也降低了数据帧的时延。     

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.