Real-time Voice Communication Research Articles

Real-Time Intelligent Video Surveillance System using Recurrent Neural Network

Security is a significant concern at all locations where CCTV cameras are installed. Security is a top priority; you must invest considerable time and effort to keep track of everything. Shortly, developments in computer vision may substantially impact video surveillance systems. To measure the video feed in real-time and detect any abnormal behavior without human intervention., such as violence or theft. The real-time video was captured as data with regular and strange events, and it has been trained (21 videos) and tested (16 videos) using a deep learning neural network. The captured video has been converted as frames using a Spatio-temporal autoencoder and 3D convolution network with the help of the RNN algorithm with a threshold value of 0.006 to detect the events. The RNN algorithm analyzed the captured video to see the possibilities with higher accuracy, 96%, than others. A data processing model for event detection will be constructed using deep-learning neural network technologies. A sophisticated monitoring system also transmits real-time video and voice communications to the web.

Research on Holographic Visualization Verification Platform for Construction Machinery Based on Mixed Reality Technology

As “Industry 4.0” progresses, construction machinery is evolving toward large-scale, automation, and integration, resulting in the equipment becoming increasingly sophisticated, and the designs more difficult. Labor costs, transportation, and time will be huge challenges for construction machinery, and mixed reality technology is one of several possible ways to solve this challenge. The research presented in this paper develops a holographic visual verification platform for a digital prototype of construction machinery based on virtual terminal equipment, through investigating the synchronous remote collaboration of multiple terminal devices in a mixed reality scenario. These included semi-physical virtual-real fusion assembly, multi-person real-time voice communication, dynamic loading of MR model based on a cloud server, virtual imitation control, interface design, and human-computer interaction. The effectiveness of this paper’s method is demonstrated through remote collaborative design cases. These included a double drum roller, loader, and milling planer welding production line, as well as tractor modeling review and virtual simulation manipulation of an aerial work platform. The experimental results show that this visual verification platform is a feasible, low-cost and scalable solution, which brings a qualitative breakthrough to the design, research and development, production and other stages in the field of construction machinery.

Zigbee-Based Low Power Consumption Wearables Device for Voice Data Transmission

Short-range wireless technologies can transmit real-time voice, audio, picture, and video communications. Such networks’ energy usage and transmission reach are crucial, especially for portable and power autonomous devices. Voice over Zigbee technology’s goal is to create a competitive service that excels in these performance factors. IEEE standard 802.15.4 specifies two layers for low-rate WPANs: the physical layer and the media access control layer, with a data rate limit of 250 kbps. This partnership used low-level PHY and MAC layers as the foundation for developing the Zigbee network protocol, security, and applications. To overcome the problem of lowering costs and power usage to make this technology more accessible to the general public. This system displays the network’s operational data, such as wireless voice communication. This effort was made to minimize the overall cost of manufacturing such gadgets that contribute to home automation systems and related projects. The primary goal of the research was to communicate voice through low-power microcontrollers such as 8-bit microcontrollers using Zigbee. In this study, we developed a wireless sensor network (WSN) communication system based on Zigbee to securely transfer speech data over a medium-range distance (medium) and lower the equipment cost and communication expenses by 30% without compromising the data integrity. A testbed has been established to validate the fundamental functionality of the proposed ZigBee-based home network system. The suggested system’s network coverage capabilities were evaluated, as well as its packet loss and energy savings in long-term wireless network monitoring duties.

Segregating Signaling and Media Planes into Different containers

Scalability is an important aspect of communication networks. With the ascent of SIP and associated modern real-time protocols, IP telephony has become a revolutionary technology in connecting users through real-time voice communications with enabled video and instant messaging capabilities. B2BUA is a SIP server that provides call management and authentication functionality by reformulating the request and routing the traffic to other user agents in the network. It comprises signaling and media entities that handle all control signaling messages and real-time data(media) information respectively. The signaling and entities run as different processes in the same container. Such an architecture encounters a large CPU utilization after a specific number of maximum calls due to increase traffic flowing within the same node. Further Packet processing is CPU intensive and there is a need for an architecture that scales well with increasing traffic without hitting the CPU performance. The paper presents the design of decoupled architecture for Signaling and Media entities by running both the processes in different containers. With such an approach, one Signaling entity can communicate with multiple Media entities or vice versa thereby providing a suitable scalable solution to deal with the increased traffic and further maintaining the system efficiency. The paper is concluded by highlighting the difference between Kubernetes and OpenStack for the proposed architecture

A Noise-Tolerant Audio Encryption Framework Designed by the Application of S8 Symmetric Group and Chaotic Systems

The recent decade has witnessed an exponential surge of digital content, especially multimedia and its applications. The security requirements of these innovative platforms necessitate the significance of enhancing advanced encryption schemes. In this paper, a novel encryption scheme is presented for real-time audio applications. The framework of the proposed scheme is grounded on the principles of confusion and diffusion. The confusion incorporates nonlinearity by the application of Mordell elliptic curves (MEC) and a symmetric group of permutations S8. The endurance of the proposed scheme is further enriched through the application of chaotic maps. The proposed scheme is intended to cater requirements of real-time voice communications in defense applications particularly warzones. The adoption of a modular design and fusion of chaotic maps makes the algorithm viable for numerous real-time audio applications. The security can further be enriched by incorporating additional rounds and number of S-boxes in the algorithm. The security and resistance of the algorithm against various attacks are gaged through performance evaluation and security measurements. The audio encryption scheme has the ability to tolerate noise triggered by a channel or induced by an invader. The decryption was successful and the resultant output was audible for noisy data. The overall results depict that the proposed audio encryption scheme contains an excellent cryptographic forte with the minimum computational load. These characteristics allow the algorithm to be a hotspot for modern robust applications.

Design of Marine Two-Way Voice Communication System Based on Human-Computer Interaction

Zhu, T. and Zhang, F., 2020. Design of marine two-way voice communication system based on human-computer interaction. In: Malvarez, G. and Navas, F. (eds.), Global Coastal Issues of 2020. Journal of Coastal Research, Special Issue No. 95, pp. 1389-1394. Coconut Creek (Florida), ISSN 0749-0208.In order to improve the long-distance transmission and speech receiving and receiving ability of marine bidirectional voice communication, the marine bi-directional voice communication transmission technology based on human-computer interaction is proposed in this paper. The channel equalization model of marine bi-directional speech communication is designed to modulate the marine bi-directional speech communication signal in the space-disguised coherent multi-path channel of marine bi-directional speech information transmission, and the linear equalization controller is used to filter the interference. The fractionally spaced equalizer is designed to deal with the anti-interference of the marine bidirectional speech communication signal, to improve the real-time performance of the marine bi-directional speech transmission, and to design the receiver at the output terminal to realize the accurate feedback of the marine bi-directional speech communication signal. The hardware modularization design of marine bi-directional speech communication system based on man-machine interaction model is carried out. The main function modules of the system include data receiving module, speech information processing module, human-computer interaction module and receiver conversion module. The simulation results show that the proposed method has low bit error rate (BER) for long-distance transmission of marine bi-directional speech communication signals, and can realize real-time monitoring and voice communication of marine bi-directional speech communication.

Research and Design of LED Light Suppression Flicker Technology in Visible Light Communication Technology

“Lighting communication integration” is a key technology research for the future application of visible light communication(VLC). However, the bandwidth characteristics, dynamic modulation range and linear modulation performance of existing white LED still do not meet the requirements of high-speed communication. After adding the dimming function, the naked eye It is noticeable that flicker is felt. In order to solve the problem of LED flicker caused by high-speed communication, this paper proposes a “lighting and communication integration” technology scheme based on visible light communication(VLC), which studies from the aspects of suppressing flickering VLC scheme, LED communication dimming, and high-efficiency modulation driving circuit design. A PWM dimming control technology based on optical OFDM system is used for VLC and LED dimming scheme. Finally, the technology is applied to the typical application scenario of the power grid—cable tunnel, which realizes the two-way real-time voice communication of cable tunnel inspection personnel. And accurate positioning to ensure the personal safety of the inspectors and the safety of the grid production.

Survey on Detection Techniques for Denial-of-Service Attacks in SIP-Based Voice Over IP Networks

VoIP provides real-time voice communications over packet-switched networks. VoIP is widely used as it offers more flexibility when compared to the traditional Public Switched Telephone Network (PSTN). Session Initiation Protocol (SIP) is a protocol used for initiating a session between the caller and the callee in the VoIP system. SIP is a text-oriented protocol and is implemented in a susceptible environment. These features make SIP protocol vulnerable to various security risks such as Denial of Service, Registration hijacking and Eavesdropping. Detection of these attacks plays a vital role. The Survey focuses on different techniques used for detection of DoS attacks and differentiating between the proposed mechanisms for identifying abnormalities and then notifying the end-users about the possible intrusions.

Difference-Set-Based Channel Hopping for Minimum-Delay Blind Rendezvous in Multi-Radio Cognitive Radio Networks

Rendezvous in the blind fashion is known as one of the most robust and effective ways for cognitive users (CUs) to exchange control information under the dynamic interference of primary users (PUs). Earlier, no existing scheme could guarantee CUs to rendezvous blindly in each timeslot, which makes them experience relatively long exchange delay. Based on the mathematical concept of relaxed cyclic difference set ( DS ), this paper proposes two novel channel hopping (CH) schemes for multi-radio CUs with and without a synchronized time clock to rendezvous blindly in every timeslot. This achievement of minimum rendezvous delay is especially useful for such delay-sensitive applications as real-time voice and emergency communications. Moreover, based on a special class of DS, the proposed two schemes can closely approach the theoretical lower bound on the number of radio transceivers for CUs to rendezvous blindly in every timeslot and, hence, minimizes their required hardware complexity to exchange control information at the minimum rendezvous delay. Numerical simulation shows that, under dynamic PU interference, the proposed schemes outperform the existing ones in terms of the average throughput and delay of CUs.

Teknologi Bluetooth dengan Mekanisme Multihop

Bluetooth is a wireless communication technology that is used to exchange information, at a lower cost, use low power, the ability to work or interact between applications through approved protocols is very good, and easy operation. The Bluetooth operating range works on the 2.4 GHz unlicensed ISM (Industrial, Scientific and Medical) frequency band using a frequency hopping tranceiver that is able to provide real-time data and voice communication services between Bluetooth hosts. When Bluetooth devices communicate with each other, they will form a dynamic network called piconet, a network consisting of master and slave, in one piconet of no more than 7 slaves. Bluetooth also supports connections between piconets called scattemets. The mechanism for connecting between one piconet and another is called a hop mechanism.

Modified Iterative Decoding Algorithm for Polar Code Using Maximum Likelihood Estimation

The key challenges in real time voice communication in long term evolution mobile are reduction in complexity and latency. Efficient encoding and decoding algorithms can cater to these. The implementation of such polar code based efficient algorithms is proposed in this paper. The overall latency of 3.8 ms is needed to process 8 bit block length. The novel sub-matrix near to identity matrix is presented. This resulted into minimization of loops among least reliable bits due to iterated parity check matrix. Look-up table based memory mapping is used in encoder to reduce latency while Euclidian decoding technique is used in decoder. The number of iterations is reduced by 50%. The experimentation is performed with additive white Gaussian noise and QPSK modulation. The proposed modified iterative decoding algorithm requires SNR of 5.5 dB and 192 computations for targeted bit error rate of 10−4. The second proposed method needs 9 dB, 2 iterations for 384 computations. The penalty paid is quantization error of 0.63% due to restricting computations to fourth order series of hyperbolic function with same 8 bit block length.

An appropriate system for securing real-time voice communication based on ADPCM coding and chaotic maps

This paper presents a very simple and efficient encryption scheme based on controlled chaotic maps and ADPCM (Adaptive Differential Pulse Code Modulation) coding, in order to secure the real-time voice communication for operating at 16, 24, 32 or 40 kbps. This encryption algorithm adopts three main operations one to generate chaotic values using two chaotic logistic maps starting from independent initial conditions, the second to transform them into binary words using random encoding tables and the third to execute some basic operations and substitutions. It has important properties of randomness that can pass NIST batteries of tests. The evaluation and simulation analysis indicate that our proposal possesses an excellent statistical and cryptographic properties; it provides low correlation between adjacent samples in encryption speech, extremely sensitive encryption keys and has a large key space which is sufficient to protect against brute-force attack. As an illustrative example, an application on a commonly-encoder type used in communications, standard ITU-T G.726, is presented.

A software system for enabling real-time voice communication over existing ZigBee networks

The ZigBee wireless protocol specification, based on the IEEE 802.14.5 standard, is used in many low cost, distributed wireless sensor networks. It is widely used in various embedded applications, such as for industrial, medical, and home automation. Its popularity has spawned an increasing interest in the capability of the protocol to support multimedia applications, including real-time voice transmission. Several vendors have developed hardware codecs for developers to implement 'voice over ZigBee' solutions. However, given the already wide-spread deployment and use of ZigBee sensor nodes, we propose a system for voice transmission over a ZigBee star network using a purely software-based solution, such that it can be applied to existing commercial ZigBee network nodes without additional hardware. A voice detection system optimised for embedded systems is also proposed in order to reduce network load. The proposed system is implemented onboard a common wireless sensor network node, the Freescale MC1322x. The voice quality of the system is measured via a mean opinion score test and compared to other common voice over internet protocol solutions.

Implementing voice over Internet protocol in mobile ad hoc network – analysing its features regarding efficiency, reliability and security

Providing secure and efficient real-time voice communication in mobile ad hoc network (MANET) environment is a challenging problem. Voice over Internet protocol (VoIP) has originally been developed over the past two decades for infrastructure-based networks. There are strict timing constraints for acceptable quality VoIP services, in addition to registration and discovery issues in VoIP end-points. In MANETs, ad hoc nature of networks and multi-hop wireless environment with significant packet loss and delays present formidable challenges to the implementation. Providing a secure real-time VoIP service on MANET is the main design objective of this paper. The authors have successfully developed a prototype system that establishes reliable and efficient VoIP communication and provides an extremely flexible method for voice communication in MANETs. The authors’ cooperative mesh-based MANET implementation can be used for rapidly deployable VoIP communication with survivable and efficient dynamic networking using open source software.

A-LNT: A Wireless Sensor Network Platform for Low-Power Real-Time Voice Communications

Combining wireless sensor networks and voice communication for multidata hybrid wireless network suggests possible applications in numerous fields. However, voice communication and sensor data transmissions have significant differences, Meanwhile, high-speed massive real-time voice data processing poses challenges for hardware design, protocol design, and especially power management. In this paper, we present a wireless audio sensor network platform A-LNT and study and discuss key elements for systematic design and implementation: node hardware design, low-power voice codec and processing, wireless network topology, hybrid MAC protocol design based on superframe, radio channel allocation, and clock synchronization. Furthermore, we discuss energy management methods such as address filtering and efficient power management in detail. The experimental and simulation results show that A-LNT is a lightweight, low-power, low-speed, and high-performance wireless sensor network platform for multichannel real-time voice communications.

An Improved Algorithm of Adaptive Multi-Pulse AMR-WB PLC

Packet loss is common in internet, when use IMS mobile terminal accessed by wireless to do real-time voice communication, the quality of speech is affected a lot. This text summarizes the basic principles of AMR-WB encoding and decoding algorithm and main technology of PLC. Based on AMR-WB encoding and decoding algorithm, it improves an algorithm of multi-plus PLC. The algorithm described in this text can speed up the recovery of adaptive-codebook by small bandwidth cost and effectively prevent the spread of errors, improve the quality of speech.

소규모 그룹에서의 음성 통신을 위한 TDMA 기반의 릴레이 프로토콜

Since the wireless communications have a limited transmission, the devices just around a master node can exchange data. Though Bluetooth and Zigbee support ad hoc, they are not appropriate for real-time voice communications. In this paper, we present a TDMA-based relay protocol for several users to communicate simultaneously. The proposed protocol can relay data or voice to other nodes in real-time by the multi-hop transmission method using TDMA. And the proposed protocol improves the network performance by allocating different frequencies to the slaves depending on the routing path scheduled by the routing table. NS-2 simulation shows that the performance of the proposed protocol is good in terms of the transmission delay and pecket loss probability in the real-time voice transmission.

VoIP Receiver Structure for Enhancing Speech Quality Based on Telematics

The quality of real-time voice communication over Internet Protocol networks based on telematics is affected by network impairments such as delays, jitters, and packet loss. To resolve this issue, this paper proposes a receiver-based enhancing method of VoIP speech quality. The proposed method enables users to deliver high-quality voice using playout control and signal reconstruction, which consists of concealment of lost packets, adaptive playout-buffer scheduling using active jitter estimation, and smooth interpolation between two signals in a transition region. The proposed algorithm achieves higher Perceptual Evaluation of Speech Quality (PESQ) values and low buffering delay than the reference algorithm.

Performance Comparison of Noise Reduction Algorithms for Enhancing Voice Quality based on Telematics

To provide high voice quality of real-time voice communication based on telematics exposed to various noise environments, the noise reduction algorithm with low computing load is required to effectively remove the noise. In this paper, we propose a noise reduction algorithm based on Mel-Filter and illustrate the proposed algorithm comparing with conventional noise reduction algorithms. As a experimental result that evaluates the performance of the noise reduction algorithms under the car and babble noise environments, the proposed noise reduction algorithm has the lower computing load with the similar PESQ score compared to the conventional noise reduction algorithms. It proves that the proposed noise reduction algorithm can efficiently remove the noise in mobile telematics.

ANALISA PERANCANGAN SERVER VOIP (VOICE INTERNET PROTOCOL) DENGAN OPENSOURCE ASTERISK DAN VPN (VIRTUAL PRIVATE NETWORK) SEBAGAI PENGAMAN JARINGAN ANTAR CLIENT

Abstrack Voice over Internet Protocol (VoIP) is a technology that utilizes the Internet Protocol to provide real-time voice communication. VoIP technology is a today telecommunication technology, where the costs of the technology infrastructure is much cheaper than the telecommunications technology that is commonly used today. Forming elements are VoIP User Agent, Proxy, Protocol and Coder-Decoder (CODEC). Asterisk is a softswicth to operate a proxy, which is based on session initiation protocol (SIP). 10.10 Ubuntu operating system as a VoIP server is flexible to support a package of performance Asterisk. The goal of this research is to build Asterisk-based VoIP server, that can be developed in further research as needed. The methodology of research conducted, is devided by two, the study of literature and experimental. The research was conducted at the installation that has been built before the Internet network. VoIP so here functioned as maximizing existing internet network is to reduce expenses communication needs. Services provided in this study form with voice and video call services client to server, client to client call, video call conferencing, video conferencingKey Word : Voice over Internet Protocol (VoIP), Asterisk, Session Initiaton Protocol (SIP), VPN