Voice Applications Research Articles

A scalable multimedia QoS architecture for ad hoc networks

Communication demands have grown from separate data and voice to integrated multimedia, paving the way to converging fixed, mobile and IP networks. Supporting Multimedia is a challenging task for wireless ad hoc network designers. Multimedia forms high data rate traffic with stringent Quality of Service (QoS) requirements. Wireless ad hoc networks are characterized by frequent topology changes, unreliable wireless channel, network congestion and resource contention. Providing scalable QoS is the most important challenge for multimedia delivery over ad hoc networks. We introduce here a provisioning and routing architecture for ad hoc networks which scales well while provisioning QoS. The proposed architecture is analysed using a mix of HTTP, voice and video streaming applications over 54 Mbps 802.11 g-based ad hoc networks. The architecture is simulated and compared to well-known routing protocols using the OPNET Modeller. The results show that our architecture scales well with increase in the network size, and outperforms well-known routing protocols.

Quality of experience (QoE) driven adaptation scheme for voice/video over IP

Network quality of service (NQoS) of IP networks is unpredictable and impacts the quality of networked multimedia services. Adaptive voice and video schemes are therefore vital for the provision of voice over IP (VoIP) services for optimised quality of experience (QoE). Traditional adaptation schemes based on NQoS do not take perceived quality into consideration even though the user is the best judge of quality. Additionally, uncertainties inherent in NQoS parameter measurements make the design of adaptation schemes difficult and their performance suboptimal. This paper presents a QoE-driven adaptation scheme for voice and video over IP to solve the optimisation problem to provide optimal QoE for networked voice and video applications. The adaptive VoIP architecture was implemented and tested both in NS2 and in an Open IMS Core network to allow extensive simulation and test-bed evaluation. Results show that the scheme was optimally responsive to available network bandwidth and congestion for both voice and video and optimised delivered QoE for different network conditions, and is friendly to TCP traffic.

Variable and shiftable acoustic output signal limiter

Systems and methods for variably and/or shiftably limiting signal output are disclosed. The system may include an attenuation circuit having an attenuation network activated by a source signal crossing a threshold to attenuate the source signal and a resistive element such as resistors bridging between the attenuation circuit input and output for variably attenuating the source signal. The attenuation circuit may be activated by a switch such as transistors. To shift the limiting threshold, the system may include a circuit to shift the activating threshold, the circuit having a control voltage source such as DC voltage source coupled to the attenuation network. The threshold shifting circuit may be configured to selectively increase the threshold during music listening and to be inactive during voice applications. The system may be incorporated in a headset adapter or a headset receiver.

A Converged Charging Framework for Mobile Voice and Data Applications

The evolution of an integrated mobile communication environment enables service deployment and content delivery offered by independent providers through heterogeneous network infrastructures. In a few months, a mobile user will be able to access value added services through several wireless public access networks (i.e., 3G cellular, Wireless LAN or WAN) belonging to the same or different operators and simultaneously use multiple different services at the same time. In this scenario there is a strong need for a common charging and billing scheme enhancing access flexibility and encouraging services and network convergence. We propose, starting from an AAAbased 3G-WLAN roaming/interworking paradigm, a unified “one stop” charging scheme, according to which the customers will receive a single itemized bill for all consumed services regardless of the providing party and access technology. Such a platform can become very helpful for the homogeneous development of charging functionalities for new-generation integrated mobile services.

Analysis of security threats to MPLS virtual private networks

Multiprotocol label switching (MPLS) based virtual private networking is one of the fastest-growing network technologies. It provides corporate and government customers with flexible, inexpensive “autobahns” that seamlessly connect multiple, geographically-dispersed sites, enabling voice, video, data and other high-bandwidth applications. The technology is also attractive to service providers because it enables them to flexibly provision resources for a variety of classes of service and applications with excellent quality of service at low cost. This paper analyzes the principal security threats to MPLS virtual private networks (VPNs). Because BGP is crucial to implementing MPLS VPNs, special attention is directed at the protocol and its multiprotocol extensions. This paper describes three classes of exploits on MPLS VPNs: route modification, traffic injection and denial-of-service attacks. It also discusses mitigation strategies that can be implemented by service providers and MPLS VPN customers.

A Framework for Intelligent Voice-Enabled E-Education Systems

Although the Internet has received significant attention in recent years, voice is still the most convenient and natural way of communicating between human to human or human to computer. In voice applications, users may have different needs which will require the ability of the system to reason, make decisions, be flexible and adapt to requests during interaction. These needs have placed new requirements in voice application development such as use of advanced models, techniques and methodologies which take into account the needs of different users and environments. The ability of a system to behave close to human reasoning is often mentioned as one of the major requirements for the development of voice applications. In this paper, we present a framework for an intelligent voice-enabled e-Education application and an adaptation of the framework for the development of a prototype Course Registration and Examination (CourseRegExamOnline) module. This study is a preliminary report of an ongoing e-Education project containing the following modules: enrollment, course registration and examination, enquiries/information, messaging/collaboration, e-Learning and library. The CourseRegExamOnline module was developed using VoiceXML for the voice user interface(VUI), PHP for the web user interface (WUI), Apache as the middle-ware and MySQL database as back-end. The system would offer dual access modes using the VUI and WUI. The framework would serve as a reference model for developing voice-based eEducation applications. The e-Education system when fully developed would meet the needs of students who are normal users and those with certain forms of disabilities such as visual impairment, repetitive strain injury (RSI), etc, that make reading and writing difficult.

A novel network mobility handoff scheme using SIP and SCTP for multimedia applications

In a heterogeneous wireless environment, seamless mobility is the basis of network support with which mobile users who roam between or among various wireless access networks are able to fully enjoy uninterrupted wireless services. When users are in a mass transportation vehicle, e.g., a bus or a train that provides network service, the vehicle can be regarded as a network which is serving users as it moves from one location to another. The movement of a network is called network mobility (NEMO). The network mobility protocol based on Mobile IPv6 as proposed by the Internet Engineering Task Force (IETF) in 2005 has some fundamental drawbacks, such as header overhead and the pinball problem. In this paper, we propose a novel hybrid method for network mobility called Hybrid-NEMO, which provides a soft handoff scheme at the transport layer basically utilizing SIP and SCTP protocols to ensure a lossless packet-transmission environment and less handoff-delay variation, which are critical in providing QoS voice and multimedia applications. Experimental validation and performance evaluation were also conducted in this study.

Coarticulated concatenated speech

Described are methods and systems for reducing the audible gap in concatenated recorded speech, resulting in more natural sounding speech in voice applications. The sound of concatenated, recorded speech is improved by also coarticulating the recorded speech. The resulting message is smooth, natural sounding and lifelike. Existing libraries of regularly recorded bulk prompts can be used by coarticulating the user interface prompt occurring just before the bulk prompt. Applications include phone-based applications as well as non-phone-based applications.

Voice transmission over ad hoc network adapting optimum approaches to maximize the performance

Although a lot of research and implementations have been performed for voice transmission over regular network and internet, a little is done for mobile ad hoc network (MANET). Like other computer networks, in MANET voice transmission is also very much demanding and necessary. Since the wireless links in an ad hoc network are highly error prone and may go down frequently because of node mobility, interference, channel fading, and the lack of infrastructure, it is very difficult and challenging to implement voice transmission over MANET. In this research, to maximize the performance of MANET during voice transmission we choose some parameters and methodological approaches e.g. Method of media access, selection of audio codec, selection of routing protocol etc in efficient and optimum way. Since voice applications consume more energy than typical applications, we use an energy aware routing protocol known as WEAC for the study. During the selection process of routing protocol we have a performance comparison study between Improved DSR and WEAC, where WEAC performs better consuming less energy in case of larger network. We also have a comparative study among several audio codecs (G.711, G.729 and G.723.1) where by simulation we show that the G.729 codec is more suitable to use for voice transmission over MANET in terms of latency. Finally, we show that it is possible to launch voice transmission with acceptable quality and throughput over MANET using G.729 and WEAC protocol.

Mobility in an RF-isolated test platform

Understanding the practical impact of mobility in a wireless network is essential for the wireless networks of tomorrow. Mobility influences the network performance, behavior, and ability to provide seamless service across a wide area. Testing and evaluating the real impact of mobility is difficult in the field with so many variables such as interference, fading, and so on. Experimental wireless evaluation in a test environment must correspond to an actual deployment. Furthermore, it is important to achieve repeatability without sacrificing realism. This study presents a functional test platform that uses real IEEE 802.11 equipment, providing repeatability and reliability. The platform is used to test the practical impact of device movement in a WLAN cell while voice and data applications are running. The mobility characteristics of wireless devices are based on individual models used by researchers with the addition of real aspects of mobility from empirical studies to improve realism.

Adaptive FEC-Based Packet Loss Resilience Scheme for Supporting Voice Communication over Ad hoc Wireless Networks

Providing real-time voice support over multihop ad hoc wireless networks is a challenging task. To make a voice application feasible over ad hoc wireless networks, the perceived voice quality must be improved while not significantly increasing the packet overhead. We suggest packet-level media-dependent adaptive forward error correction (FEC) at the application layer in tandem with multipath transport for improving the voice quality. Since adaptive FEC masks packet losses in the network, at the medium access control (MAC) layer, we avoid retransmissions in order to reduce the control overhead and end-to-end delay. Further, we exploit the combined strengths of layered coding and multiple description (MD) coding for supporting error resilient voice communication in ad hoc wireless networks. We propose an efficient packetization scheme in which the important sub-stream of the voice stream is protected adaptively with FEC depending on the loss rate present in the network and is transmitted over two maximally node-disjoint paths. Our scheme achieves significant gains in terms of reduced frame loss rate, reduced control overhead, and minimum end-to-end delay and almost doubles the perceived voice quality compared to the existing approaches.

An Architectural Model to Provide QoS in a Home Network and its Evaluation in a Real Testbed

Home networks are an evolution of office local area networks, answering the need for residential distribution of broadband services. The design of a home network must strive to keep low costs, easy installation and maintenance while at the same time providing a large variety of services, from voice and data applications to multimedia streaming. A successful platform for home networking must put together simple interfaces for the users and sophisticated mechanisms for managing the distribution of the desired broadband services. In this paper we propose mechanisms to support flows with different Quality of Service (QoS) requirements in a home network whose architectural model has been defined in the framework of the European IST ePerSpace project. The model is based on a centralized Residential Gateway (RG), which connects and coordinates different network technologies to support home services. The proposed mechanisms are simple to be implemented since they can be installed on top of existing home network technologies, without modifying them. We implement the whole model and our mechanisms in a real testbed and we perform an extensive performance evaluation over it. The performance evaluation allows us to understand the effect of different settings both on the wired and on the wireless interfaces of the RG, to comprehend which are the key aspects that influence the overall performance and to propose some RG configurations which turn out to be very effective in achieving the desired performance. Special attention is devoted to the wireless support of home services on IEEE 802.11b devices.

Respiratory muscle strength training applications

To provide an overview of respiratory muscle strength training applications for voice and speech disorders. Little examination of respiratory muscle training for the rehabilitation of voice and speech has occurred. Less than a handful of studies discuss use of strengthening techniques for the voice-disordered population and of those, inspiratory muscle strength training shows promise for use with upper airway disorders including abductor vocal fold paralysis and paradoxical vocal cord dysfunction. Case study results of Lance Adams syndrome indicated a positive outcome for maximum phonation time and improved intelligibility ratings and communication effectiveness. Work with the multiple sclerosis population showed no change in qualitative ratings of patient voice handicap and no significant improvement in speech production variables. Current trials with Parkinson's disease are ongoing with no published data currently available with regard to voice and speech production. Respiratory muscle strengthening programs are either inspiratory or expiratory and are typically accomplished using handheld devices. Training durations typically examined have been 4-8 weeks. Training trials usually take 20-30 min per day. Training frequency varies across studies. Detraining outcomes are important to examine as part of clinical trials. Defining detraining outcomes has major implications for the development of a complete rehabilitation program.

Ovid's Heroides, Drayton and the articulation of the feminine in the English Renaissance

ABSTRACTThis essay argues that Michael Drayton's Englands Heroicall Epistles engages in a highly productive and transformative relationship with Ovid's Heroides, one which focuses on the utility and application of the female voice. By adapting the Ovidian model to a distinctively patterned version of English history, Drayton manages to invest the female voice with both authority and political meaning, as the various female figures outline the symbolic importance of the feminine to the public sphere and to dynastic stability. The means by which Drayton does this are heavily indebted to the practices of imitation and invention, yet in the process, Drayton produces heroines who critique and comment on male conduct, particularly sexual conduct, whilst manipulating early modern culture's norms for women's textual production. At the same time, Drayton also attempts to enhance his own poetic authority by his overt competition with Ovid.

A low complexity double-talk detector based on the signal envelope

A new algorithm for double-talk detection, intended for use in the acoustic echo canceller for voice communication applications, is proposed. The communication system developed by the authors required the use of a double-talk detection algorithm with low complexity and good accuracy. The authors propose an approach to double-talk detection based on the signal envelopes. For each of three signals: the far-end speech, the microphone signal and the echo estimate, an envelope is detected. Next, using these envelopes, a detection function is determined and compared to the threshold. Additionally, a dynamic threshold is introduced in order to improve the accuracy of the algorithm. The results of the simulations presented in the paper proved that the accuracy of double-talk detection obtained using the proposed algorithm is higher than in the Geigel algorithm and comparable to the correlation-based methods, while the computational complexity of the proposed method remains at an acceptable level. The double-talk detection algorithm presented here may be used in voice communication systems having limited resources, allowing for accurate double-talk detection and, as a consequence, efficient acoustic echo cancellation.

Next-generation access network architectures for video, voice, interactive gaming, and other emerging applications: challenges and directions

Internet access was the main service supported in legacy access networks via asynchronous transfer mode (ATM)-based digital subscriber line access multiplexers (DSLAMs) and aggregation networks. The broadband remote access server (BRAS) was the Internet Protocol (IP) edge router, and it provided various services including service routing, Point-to-Point Protocol (PPP) handling, shaping, scheduling, and policy-based controls. This architecture is being increasingly replaced by an Ethernet-based aggregation network with potentially multiple broadband network gateways (BNGs) replacing the single BRAS. The migration to Ethernet-based networks is partly driven by the high bandwidth needs of applications such as video on demand (VoD), IP television (IPTV), and the ability to support these applications at cost-effective points. Other drivers that are emerging to further shape the next generation of access networks are examined in this paper. In particular, we focus closely on emerging video and voice applications and the new challenges and requirements they bring. The set of applications and issues that we consider is by no means comprehensive. They have been selected as they pose some interesting challenges and have significant impact on the access network functionality and architecture. For video traffic, we examine the challenges for providing high video quality, fast response for IPTV channel change, and content distribution. For voice, we examine the challenges of emergency standalone operation. Finally, we examine aspects of quality of service (QoS) management, resource control, and network resiliency mechanisms for these emerging applications. We see that the access network needs to become extremely application aware and dynamic and move beyond the traditional IP layer 2/3 forwarding functions in order to meet these challenges effectively.

A novel method for the float-point to fixed-point conversion from statistical perspective

Float-point to fixed-point conversion is required by many communication, voice and image applications, in order to achieve acceptable levels of performance and cost. This is particularly crucial in real-time applications that have strict timeline requirement on processing throughput and latency. Therefore, fixed-point conversion from float-point computation brings great challenges with the improvement at the same time: How to evaluate the accuracy of the fixed-point system quickly? This paper proposes a novel method for the float-point to fixed-point conversion and evaluates the precision of complex signal systems from statistical perspective. The proposed approach greatly reduces the complexity of the evaluation, and achieves optimum fixed-point implementation in a short time. The experiments in MP3 and the AAC decoding algorithm demonstrate that the accuracy of the fixed-point system is greatly improved by this method.

An MDP-Based Vertical Handoff Decision Algorithm for Heterogeneous Wireless Networks

The architecture for the Beyond 3rd Generation (B3G) or 4th Generation (4G) wireless networks aims at integrating various heterogeneous wireless access networks. One of the major design issues is the support of vertical handoff. Vertical handoff occurs when a mobile terminal switches from one network to another (e.g., from wireless local area network to code-division multiple-access 1x radio transmission technology). The objective of this paper is to determine the conditions under which vertical handoff should be performed. The problem is formulated as a Markov decision process with the objective of maximizing the total expected reward per connection. The network resources that are utilized by the connection are captured by a link reward function. A signaling cost is used to model the signaling and processing load incurred on the network when vertical handoff is performed. The value iteration algorithm is used to compute a stationary deterministic policy. For performance evaluation, voice and data applications are considered. The numerical results show that our proposed scheme performs better than other vertical handoff decision algorithms, namely, simple additive weighting, the technique for order preference by similarity to ideal solution, and Grey relational analysis.

Quality assessment of interactive voice applications

The conversational quality of a VoIP communication is dependent on several factors such as the coding process used, the network conditions and the type of error correction or concealment employed. Furthermore, the quality perceived by the users is also dependent on the characteristics of the conversation itself. Assessing this kind of communication is a very difficult problem, and most of the studies available in the literature simplify the issue by restricting the analysis to only one or two parameters. However, the number of potentially affecting factors is typically higher, and their joint effect on quality is complex. In this paper we study the combined effects of bit rate, forward error correction , loss rate, loss distribution , delay and jitter on the perceived conversational quality. In order to achieve this we use the pseudo-subjective quality assessment (PSQA) technique, which allows us to obtain accurate, subjective-like assessments, in real time if necessary. Our contributions are thus twofold: firstly, we offer a detailed analysis of the impact of these parameters and their interactions on the perceived conversational quality. Secondly, we show how the PSQA methodology can be used to provide accurate conversational quality estimations.

Apparatus and method for providing server state and attribute management for voice enabled web applications

A unified web-based voice messaging system provides voice application control between a web browser and an application server via an hypertext transport protocol (HTTP) connection on an Internet Protocol (IP) network. The application server generates and maintains a server-side data record, also referred to as a “brownie”, that includes application state information and user attribute information for an identified user session with the web browser. The application server, in response to receiving a new web page request from the browser, initiates a web application instance to begin a transient application session with the browser. The brownie also includes a session identifier that uniquely identifies the session with the user of the application session. The application server stores the brownie in a memory resident within the server side of the network, and sends to the browser the session identifier and the corresponding web page requested by the web browser.