Transmission Of Voice Signals Research Articles

Design of an asterisk-based VoIP system and the implementation of security solution across the VoIP network

Voice over Internet Protocol (VoIP) is a rapidly advancing technology that facilitates the transmission of voice and audio signals over the Internet or an IP-based network in real-time. This technology has seen a significant rise in demand due to its advantages over traditional circuit-switched telephony, including lower call rates, reduced operational costs, easier management, and enhanced call features. However, the growth in VoIP usage has also increased the potential for various security threats and attacks, jeopardizing the privacy, confidentiality, and integrity of transmitted data. This paper presents the design of an Asterisk-based VoIP system and the implementation of a comprehensive security solution across the VoIP network. The study involves an in-depth analysis of VoIP technology, identifying its vulnerabilities and addressing potential threats. A security framework is proposed and implemented to safeguard the VoIP network. The designed system and security solutions are rigorously tested and evaluated to ensure robustness and effectiveness. The findings highlight critical security measures necessary for protecting VoIP infrastructures and provide a framework for future research and development in securing VoIP networks.

RETRACTED ARTICLE: High-performance cloud computing–oriented corpus and English teaching speech abnormal classification

High-performance cloud computing is closely related to scientific computer programs. It is an interface for transmitting information, including input and output files that need to be stored for a long time, a large number of intermediate result files, and check files for error recovery. The corpus has a wide range of sources, objective data, and multi-sample characteristics. At present, it can assist English teaching and can also enable students to learn independently. As a carrier of specific information, voice signal transmission has become an important means of collecting and disseminating information in people’s social life. In speech processing, speech classification is the basic task of its research, and it is also an arduous task. English is a widely used language in international communication, and its actual usage rate is relatively high. Therefore, in China, it is regarded as a compulsory basic course for all kinds of education. Improving students’ English level is the main challenge for teaching English in Chinese universities. How to change the traditional English teaching methods in the classroom and improve students’ English level is a top priority for college English teaching. Through continuous exploration of English teaching, it is found that the multi-dimensional interactive English teaching mode can help reform and innovate the traditional teaching mode, and improve the learners’ English level. This article develops a new multi-dimensional interactive based on the original English teaching mode. This method can train English talents to meet the needs of personal and professional development, and make learning more meaningful, knowledgeable, interesting, and practical.

Blind Source Separation Algorithm for Chaotic Masking Multipath Signals Based on Spectral Peak Search Counter Permutation

Since voice communication is the main method of information transmission, it is important to ensure the safety and efficiency of voice communication. In this paper, a more complex multipath channel model in a wireless environment is considered, and chaotic masking technology is used to ensure the security of voice signal transmission. Based on this, a multipath blind source separation algorithm based on cross-correlation permutation of spectral peak search counters is proposed. First, chaotic masking is performed between multiple voice signals and chaotic signal, and an impulse response filter is used to simulate the multipath transmission process. Next, the short-time Fourier transform is employed to convert the observation signal into a frequency-domain signal. Then, the joint approximate diagonalization of eigen-matrix algorithm is used to perform linear instantaneous blind source separation at each frequency. And to solve the permutation uncertainty problem in the frequency domain method, a cross-correlation permutation algorithm based on a spectral peak search counter is proposed. Finally, the permuted signal is transformed back to the time domain to get the estimated source signal by the inverse short-time Fourier transform. And simulation experiments show that the algorithm greatly reduces the calculation amount of the autocorrelation coefficient in the traditional permutation algorithm. Moreover, the permutation algorithm has stability and accuracy, and provides a solution for the secure transmission and effective separation of voice signals in multipath transmission channels.

Smart Sensor Network Based Fire Rescue System Design using LabVIEW

This paper enhances fire rescue system for reducing the number of victims in case of fire accidents. This system has been developed and implemented using the smart sensors and LabVIEW technology. It will automatically identify the fire accident and inform it to the rescue team (loco pilot for trains, security in buildings) through voice signal transmission. Servo motors are connected to water sprinklers to control fire. As an initial step, fire will be suppressed and intimation about the accident will be sent through short message service to the local fire stations, ambulance. This short message service consists of the status of fire accident accurately by transmitting the physical parameters. This also display the particular area code based upon that global system for mobile network. The complete details about the location (buildings, trains) could be intimated. With simple camera the number of members entered into the room can be known. So that directly saves the people instead of searching every room. As a whole this system ensures that it will reduce the human death ratio due to accidents.

The design of analog signal communication system based on visible light

With the rapid development of science and technology, communication mode has made a great leap forward, and visible optical communication has also been developed around the word. This paper mainly studies the audio signal transmission of LED visible light indoor communication. The system uses audio chip to generate audio signal. We have not only established a channel model, but also received optical signal through BH1750FVI module. In addition, we adopt the direct pulse counting frequency discrimination method for modulation and demodulation, and finally recover the original signal, achieving the purpose of voice signal transmission.

A Smartphone-Based Multi-Functional Hearing Assistive System to Facilitate Speech Recognition in the Classroom

In this paper, we propose a smartphone-based hearing assistive system (termed SmartHear) to facilitate speech recognition for various target users, who could benefit from enhanced listening clarity in the classroom. The SmartHear system consists of transmitter and receiver devices (e.g., smartphone and Bluetooth headset) for voice transmission, and an Android mobile application that controls and connects the different devices via Bluetooth or WiFi technology. The wireless transmission of voice signals between devices overcomes the reverberation and ambient noise effects in the classroom. The main functionalities of SmartHear include: 1) configurable transmitter/receiver assignment, to allow flexible designation of transmitter/receiver roles; 2) advanced noise-reduction techniques; 3) audio recording; and 4) voice-to-text conversion, to give students visual text aid. All the functions are implemented as a mobile application with an easy-to-navigate user interface. Experiments show the effectiveness of the noise-reduction schemes at low signal-to-noise ratios in terms of standard speech perception and quality indices, and show the effectiveness of SmartHear in maintaining voice-to-text conversion accuracy regardless of the distance between the speaker and listener. Future applications of SmartHear are also discussed.

The Design and Implementation of Infrared Communication System

As a kind of wireless communication, Infrared wireless data communication has been developing rapidly and has been widely used in the close wireless data communication. In this paper, there is the design and achievement of infrared communication system, which has realized the short distance transmission of voice and digital signal (temperature signal). The modem part of the system is implemented by the phase locked circuit composed of LM567. At the same time, with the single chip processor as the core, the temperature information which has been encoded is added to the audio signal and then to be transmitted to the receiver to be decoded to restore the temperature information. After the test of implementation, this system can transmit the speech signal and digital signal directly and the voice signal received has no obvious distortion. This system is simple in making and has a good application prospect.

Design of Time Division MUX and DEMUX for Large Optical Transceiver

The working principle of large optical transceiver is introduced in this article, and the time division MUX and the time division DEMUX are designed by the online system programmable technology and the FPGA (Field Programmable Logic Array), including hardware design, software design, and testing. The transmitting design and the receiving design method of time division MUX are introduced in detail, and a kind of time division MUX method based on synchronic control character is designed to realize the reliable transmission of VGA signals, PAL signals, voice signals, digital IO, USB data, and network signals, which is helpful and practical to enhance the capacity of transmission channel and save the fiber resources.

Echo Delay Estimation Using Algorithms Based on Cross-correlation

The following paper introduces a study on correlation-based time delay estimation algorithms. Its main purpose is to suggest a number of algorithms, which are the most suitable for real-time processing of speech signals. Time delay estimation is an important topic during transmission of voice signals over packet-switching telecommunication systems. An echo delay problem associated with IPbased transport networks is discussed in the text. An experimental comparison of the performance of several methods based on cross-correlation, normalized cross-correlation and generalized crosscorrelation to estimate the true time delay between two speech signals is presented, too.

Time Delay Estimation Algoritms for Echo Cancellation

The following case study describes how to eliminate echo in a VoIP network using delay estimation algorithms. It is known that echo with long transmission delays becomes more noticeable to users. Thus, time delay estimation, as a part of echo cancellation, is an important topic during transmission of voice signals over packetswitching telecommunication systems. An echo delay problem associated with IP-based transport networks is discussed in the following text. The paper introduces the comparative study of time delay estimation algorithm, used for estimation of the true time delay between two speech signals. Experimental results of MATLab simulations that describe the performance of several methods based on cross-correlation, normalized crosscorrelation and generalized cross-correlation are also presented in the paper.

ОценкА показателеЙ пОфОнемнОгО спектральнОгО АнАлизА речевыХ фрагментоВ В IP–сетях

The integral scalar quality characteristic of transmission of voice signals in package networks with the losses, based on phoneme spectrum analysis is offered. On the basis of the gated in index dependence of transmission quality of speech on network parameters is researched.

Ocultamiento del Mensaje de Señales de Voz usando la Transformada de Ondita Haar Discreta

This work introduces a message concealment method for voice signals transmission using codification through multi-resolution filter banks implementation for analysis and synthesis for the Discrete Haar Wavelet Transform. Through these multi-resolution filter banks of Haar wavelet the voice signal can be split in its components, and later a reversible distortion of the synthesis of its multi-resolution component signals is made, obtaining an output signal that, although maintains the same information of the original input signal, hides the message of the carried voice signal in itself. This method can be applied in codified transmission of voice, analog or digital signals, and due to the simplicity of the method it’s suitable to be implemented on digital electronic systems like FPGA, or ASIC.

Speech Quality Transmitted by Circuit Multiplication Equipment Optimized for IP-Based Networks (IP-CME)

We present a quantitative evaluation of speech quality using the multiplexing scheme for the efficient transmission of voice signals in order to reduce the number of the IP packets carrying voice signals (called VoIP packets) transferred. The multiplexing scheme is applicable to a variety of media gateways controlling the bulk of voice streams over IP-based networks, based on VoIP technology. We speculated that the multiplexing scheme would reduce the degradation of speech quality due to packet loss since it also has a similar effect to interleaving the voice signal streams. However, the interleaving effect for maintaining speech quality in the scheme characterized by the feature of IP-based multiplication is not quantitatively clear. Through our end-to-end quality evaluation results of speech, as transmitted via the multiplexing scheme using dedicated hardware, we confirm the advantages of the multiplexing scheme from the perspective of achieving improved speech quality without increasing the processing delay when considering practical packet loss conditions within an IP-based network.

Opening up powerful lines of communication

Power line carrier communications (PLCC) is a technology that utilizes electricity power lines for the transmission of process, data, and voice signals. The basic principle behind this communication technique is that the signal is modulated with a high-frequency carrier signal; then this high-frequency signal is coupled to the power line for communication. This system enables easy implementation of concepts like prepaid electricity. PLCC is slowly gaining acceptance in the field of home controlling and remote metering as they do not require high data rates.

Speech coding research at Bell Laboratories

The field of speech coding is now over 70 years old. It started from the desire to transmit voice signals over telegraph cables. The availability of digital computers in the mid 1960s made it possible to test complex speech coding algorithms rapidly. The introduction of linear predictive coding (LPC) started a new era in speech coding. The fundamental philosophy of speech coding went through a major shift, resulting in a new generation of low bit rate speech coders, such as multi-pulse and code-excited LPC. The semiconductor revolution produced faster and faster DSP chips and made linear predictive coding practical. Code-excited LPC has become the method of choice for low bit rate speech coding applications and is used in most voice transmission standards for cell phones. Digital speech communication is rapidly evolving from circuit-switched to packet-switched networks to provide integrated transmission of voice, data, and video signals. The new communication environment is also moving the focus of speech coding research from compression to low cost, reliable, and secure transmission of voice signals on digital networks, and provides the motivation for creating a new class of speech coders suitable for future applications.

Improved ADPCM voice signal transmission employing click-noise detection scheme for TDMA-TDD personal communication systems

In personal communication systems, better voice quality and lower terminal price are essential requirements, and adaptive differential pulse-code modulation (ADPCM) codecs are one of the voice codec candidates that have good voice quality and reasonable cost. However, in slow fading environments, burst errors induce click noise in ADPCM voice signal transmission. This paper proposes a novel error-concealment scheme to improve the ADPCM voice signal transmission quality of time division multiple access time division duplex (TDMA-TDD) personal communication systems in slow fading environments. The proposed scheme employs an improved error-detection technique with PCM differential detection and overflow detection coupled with a cyclic redundancy check (CRC) code. To compensate for residual click noise, an expanded CRC error-detection signal is used. Erroneous PCM signals are adaptively suppressed according to the click-noise detection information. Hardware simulation results show that these techniques significantly improve ADPCM voice signal transmission quality by 0.5 points and 1.0 compared to that of the conventional ADPCM muting scheme and no processing, respectively, with the mean opinion scoring (MOS) test in the slow fading environments typical of personal communication systems.

Novel satellite digital video TDMA system for business video communications

The authors propose a satellite video signal transmission system that uses a time division multiple access (TDMA) scheme for digital video signal transmission and a code division multiple access (CDMA) scheme for voice signal transmission from a video signal reception-only earth station (backward voice signal transmission). The adoption of a TDMA scheme makes it possible to transmit video signals from various places and to communicate in multipoint-to-(multi)point mode. For backward voice channel transmission from video signal reception-only earth stations, a superposed CDMA voice signal transmission over TDMA video signals by using the same transponder is proposed. The employment of high coding gain forward error correction and development of a cochannel interference cancellation technique have made it possible for the proposed system to transmit a practical number of voice channels. The performance of the proposed system has been experimentally evaluated and the results show the proposed scheme can transmit about 20 CDMA voice channels simultaneously. >

Optical Fiber: The 1990s Adaptive Television Transmission Medium

Within the “television industry,” a constantly increasing array of studio interface and recording standards for analog component, digital component and digital composite video signal waveforms now exists, supplementing the 1953 FCC-approved NTSC analog composite broadcasting standard. SMPTE 240-M is a studio interface standard for HDTV production and post-production. The “computer industry” is spawning its own array of standards for high resolution image generation. — There is an expanding demand for a terrestrial infrastructure of local and long haul fiber circuits to transmit these signals between and among a variety of unrelated business enterprise facilities. This demand in total comes from broadcasting and cable, plus business, financial, education, health care and government enterprises involved in distance learning, videoconferencing, advanced television (ATV, which includes HDTV) programming and new image transmission services. — However, the “telephone industry's” fiber infrastructure currently operates on its own hierarchy of digital standards for voice and data signal transmission. Terrestrial circuits for analog video transmission are scarce and costly. The demand for these urgently needed fiber transmission media will not become real, therefore, until the standards setters of the television and telephone industries begin to counsel together.

Door Phone Camera Transmission System Using Simple PVC Pair Cable

A residential door phone camera system usually employs the use of a multi-wired cable, including a coaxial one, to connect the indoor and outdoor units. Concurrently, interphone systems generally use a pair of PVC (Poly Vinyl Chloride) insulated wires. Therefore, for the purpose of converting this interphone system cable (which is already installed) into cable for a TV door phone system, a method using cable composed of PVC-coated parallel wires (each 0.5 mm in diameter and 100 m in length) has been examined under various conditions. This enables the multiplexed transmission of two-way simultaneous voice and picture signals, which are superposed on DC power, to be fed to an outdoor unit comprised, of a TV camera and processors.

Diversity improvement of voice signal transmission using postdetection selection combining in land mobile radio

This paper describes the effect of postdetection selection combining diversity on voice signal transmission for wide-band PM and narrow-band PM in a Rayleigh fading environment. First, the baseband average SNR versus the average CNR for a two-branch selection diversity with the syllabic compandor is derived in a closed form including the effect of the envelope correlation coefficient. Then, the dual selection diversity receiver, which shares an envelope detector with two branches in time division, and its performance are described. Finally, laboratory tests using a Rayleigh fading simulator are carried out with the result that an effective diversity improvement is obtained for a cross correlation coefficient in the range from 0 to 0.5.