GenRTC: Generative Real-Time Video Conferencing via Joint Adaptive Coding and Bandwidth Estimation

Real-time communication (RTC) is a new standard and industry-wide effort that expand the web browsing model, allowing access to information in areas like social media, chat, video conferencing, and television over the internet, and unified communication. These systems users can view, record, remark, or edit video and audio content flows using time-critical cloud infrastructures that enforce the quality of services. However, there are many proprietary protocols and codecs available that are not easily interoperable and scalable to implement multipoint video-conference systems. WebRTC (Web Real-Time Communication) is a State-of-the-Art open technology that makes real-time communication capabilities in audio, video, and data transmission possible in real-time communication through web browsers using JavaScript APIs (Application Programming Interfaces) without plug-ins. This paper aims to introduce the P2P video conferencing system based on Web Real-Time Communication (WebRTC). In this paper, we have proposed a web-based peer-to-peer real-time communication system using the Mozilla Firefox together with the ScaleDrone service that enables users to communicate with high-speed data transmission over the communication channel using WebRTC technology, HTML5 and use Node.js server address. Our experiments show that WebRTC is a capable building block for scalable live video conferencing within a web browser.We also study a thermal camera for the videoconference system to identify the temperature body results regarding the COVID 19 crisis.

In this paper, we proposed a reference-free (RF) objective measurement algorithm for the digital video signals of multimedia communication. With the statistical analysis, we utilized the important feature vectors (FVs) which have high relationships with the subjective perceptual quality of digital video sequences. After considering the fundamentality and force of these FVs, we combined them with different weighted coefficients to build an ideal video quality measurement algorithm. With our simulation results, the proposed method exhibits its significant function for video quality measurement (VQM). It can give a high correlation with the subjective perceptual video quality score. And because of its low computational complexity, it can be used to the real-time multimedia communication systems.

Coiled tubing (CT) is one of the most common and effective methods employed to mill out composite bridge plugs (CBPs) that are used to isolate zones during completion. Typically, the zones are normally pressured or over pressured, i.e., each interval approximately the same pressure. Recently, in a New Zealand field, a plan was set to mill out CBPs that had been set between zones that were tight gas and conventional with as much as 3,780 psi pressure differential between them. One major challenge relates to the CT CBP milling. While milling the plug that isolates the over-pressured zones, the differential pressure can exert significant forces on the milling assembly, raising the potential for buckling/failure of the CT. To reduce this risk it was decided that a bridge plug with an equalizing path would be used to eliminate the large pressure differential and potential upward forces. Another challenge was the issue of cross flow between the zones after milling was completed, and the potential for well control issues due to some under-pressured zones that could not support a full fluid column, causing loss of hydrostatic pressure. For the under-pressured zones, nitrogen was used to equalize the pressure across the plugs prior to milling them. To minimize the risk of getting stuck and potential CT failure, the operation was closely monitored and optimized by using a CT real-time downhole communication system. The real-time downhole pressure captured by the system during plug milling provided the firm evidence that equalization had occurred. This paper reviews the selection of plugs/mills and the milling strategy for safely milling CBPs that have been set across zones with significant differential pressures. The benefits of using a CT real-time downhole communication system is discussed by illustrating the case study of successfully removing 25 CBPs with CT in the three wells within a field.

In these years, streaming media have become very popular. In order to protect the legal copyright of streaming media, DRM (Digital Right Management) techniques are utilized widely. However, encryption, the most representative DRM technique, is difficult to install into real-time communication systems. In addition, existing DRM techniques for real-time communication are not supported by several widely-used web browsers. Scrambling, as a substitute to encryption, is suitable for real-time communication. In this paper, we propose a scrambling technique embedding soundtracks into videos. In the proposed technique, we embed audio data into quantized discrete cosine transform (qDCT) coefficients of the video data and generate scrambling at the same time during the video encoding process. After transmission, the streaming media data can be recovered during decoding process. We conducted experiments to investigate proper parameters to make the technique effective. Using the determined parameters, we conducted verification experiments to prove the effectiveness of the proposed technique.

Abstract: Human interaction is important in today's internet-based world. To effortlessly approach each other to communicate people can use the new technology i.e. web real-time communication (WebRTC). Our real-time communication system will provide virtual voice rooms to users to audio chat in groups or peer to peer using technologies like WebRTC, Node.js, Web Sockets and React.js. On the basis of a range of topics, users can create public/private rooms and join them as well. The implications of this study are better than the previous projects

Synchronous e-training is emerging as an alternative for developing human resources training plans in large organizations. Real-time communications are used to emulate face-to-face interaction that occurs in on-campus learning environments. However, the security concerns that a synchronous e-training platform must face may compromise the integrity, availability and confidentiality of corporate information, which may lead to serious economic and legal consequences. The disclosure of corporate information or the unauthorized participation in e-training activities must be prevented. In this paper, the security issues in synchronous e-training are identified, and the threats to a real e-training platform are analyzed. The platform is organized into four virtual networks with different security requirements and vulnerabilities. The platform assumes that multicast communications are available in the underlying corporate network. The threats affecting each element of the platform and their impact on e-training activities are discussed. Finally, a security scheme is proposed fixing the aforementioned vulnerabilities. Digital certificates and encryption algorithms solve most of the vulnerabilities, but other techniques such as access control lists and user skills on security basics are essential. Most of the proposed scheme is applicable to other real-time communication systems, since the e-training platform is built using standard technologies commonly used in voice over IP systems.

To improve the performance of data processing in radar testing, a new design is proposed to realize the serial real-time communication. This design is based on ZYNQ, a kind of all programmable SoC produced by Xilinx. The ZYNQ SoC resource is composed of two parts: the processing system (PS) and the programmable logic (PL). Compared with the conventional communication and processing systems based on FPGA and PXI bus, this new design has several advantages and compatibility. The hardware of this design consist of core board with ZYNQ, PXI-bus interface, serial communication ports such as RS-485, RS-422, and RS-232, configuration ports such as USB, Ethernet, and UART. Meanwhile, the software of this design is divided into two parts: upper visual software and embedded software in the PS. To realize the key 10ms real-time communication demand, an accurate timing module is designed in the PL. The experimental results show that the new design only cost about 4 ms to operate the whole communication and data calculation. In addition, the complex serial communication protocols and the control commands are configurable with UI.

The widespread availability of convenient wireless network connection and video capture have fueled the development of multi-party realtime video streaming (MRVS) services, such as Zoom or Microsoft Teams. These services have transformed the generation and distribution of realtime streaming content and offer a new way of online communication, striving to provide high Quality-of-Experience (QoE) for individuals. However, delivering high QoE in MRVS is more challenging than in traditional video scenarios due to the stringent delay requirements and complex multi-party interactive architectures. In this paper, we propose DSJA, a distributed server-driven multi-party realtime video streaming framework that conquers the challenges. We first design an appropriate QoE model for MRVS services to capture the interplay among perceptual quality, variations, bitrate mismatch, loss damage, and streaming delay. We then model the QoE maximization problem in MRVS as a route scheduling and streaming adaptation problem. Afterward, we design DSJA which seamlessly integrates multiple selective forwarding units (SFU) architecture and server-driven approaches based on a two-step solution of route scheduling and streaming adaptation. DSJA first determines the most suitable SFU and streaming routes for each video session based on SFUs' job queuing delay and path latency. Then, the server conducts joint loss and bitrate adaptation decisions to optimize the streaming configuration of all clients, considering network conditions and QoE preferences. Our evaluations show that our framework outperforms state-of-the-art solutions by <inline-formula xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink"><tex-math notation="LaTeX">$23.1\% \sim 41.7\%$</tex-math></inline-formula> from the perspective of QoE, and reduces the backbone network transmission by <inline-formula xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink"><tex-math notation="LaTeX">$14.0\% \sim 36.6\%$</tex-math></inline-formula> .

Motion control is a sub-field of automation, in which the position and/or velocity of machines are controlled using some type of device. In motion control the position, velocity, force, pressure, etc., profiles are designed in such a way that the different mechanical parts work as an harmonious whole in which a perfect synchronization must be achieved. The real-time exchange of information in the distributed system that is nowadays an industrial plant plays an important role in order to achieve always better performance, better effectiveness and better safety. The network for connecting field devices such as sensors, actuators, field controllers such as PLCs, regulators, drive controller etc., and man-machine interfaces is commonly called fieldbus. Since the motion transmission is now task of the communication system, and not more of kinematic chains as in the past, the communication protocol must assure that the desired profiles, and their properties, are correctly transmitted to the axes then reproduced or else the synchronization among the different parts is lost with all the resulting consequences. In this thesis, the problem of trajectory reconstruction in the case of an event-triggered communication system is faced. The most important feature that a real-time communication system must have is the preservation of the following temporal and spatial properties: absolute temporal consistency, relative temporal consistency, spatial consistency. Starting from the basic system composed by one master and one slave and passing through systems made up by many slaves and one master or many masters and one slave, the problems in the profile reconstruction and temporal properties preservation, and subsequently the synchronization of different profiles in network adopting an event-triggered communication system, have been shown. These networks are characterized by the fact that a common knowledge of the global time is not available. Therefore they are non-deterministic networks. Each topology is analyzed and the proposed solution based on phase-locked loops adopted for the basic master-slave case has been improved to face with the other configurations.

In this paper, we propose four methods for equalizing the silence periods experienced by users in a multi-party VoIP conversation in order to improve their perceived conversational quality. To mitigate the unbalanced silence periods caused by delay disparities in Internet connections, the playout scheduler at the receiver of each client equalizes the silence periods experienced. Our limited subjective tests show that we can improve the perceptual quality when the network connections are lossy and have large delay disparities. Because it is impossible to conduct subjective tests under all possible conditions, we have developed a classifier that learns to select the best equalization algorithm using learning examples derived from subjective tests under limited network and conversational conditions. Our experimental results show that our classifier can consistently pick the best algorithm with the highest subjective conversational quality under unseen conditions, and that our system has better perceptual quality when compared to that of Skype (Version 3.6.0.244).

&lt;span lang="EN-US"&gt;Real-time communication is important in control network. In real-time communication, message need to be delivered from source to destination within specification. Embedded Ethernet and Controller Area Network (CAN) protocol can be used in control network to achieve hard real-time communication. For embedded Ethernet protocol, Carrier Sense Multiple Access with Collision Detection (CSMA/CD) is the media access control (MAC) used to control data transmission between nodes in network. Back-off algorithm in CSMA/CD is used to handle packet collisions and retransmission. For CAN protocol, it is communication protocol developed mainly for automotive application. It has priority arbitration to handle collisions and retransmission. In this project, embedded Ethernet network models and CAN network models are developed and simulated in MATLAB Simulink software. Several back-off algorithms, which are Binary Exponential Backoff (BEB), Linear Back-off Algorithm, Exponential-Linear back-off Algorithm and Logarithm Back-off Algorithm are proposed and implemented into Embedded Ethernet network model to evaluate the performance. Both embedded Ethernet and CAN network models are extended to 3 nodes, 10 nodes, and 15 nodes to evaluate performance at different network condition. The performance criteria evaluated and discussed are average delay and jitter of packets. The results show that in network with high number of nodes, Linear Back-off Algorithm and Exponential-Linear back-off Algorithm shows improvement in packets delay and jitter. For CAN network, the packet jitter is relatively low.&lt;/span&gt;

One of the stated aims of 3rd generation mobile communications systems is to provide enhanced multimedia services to the user. Such services will include an integration of video, high-quality audio and data services. This paper describes a real-time mobile video communications system which employs ETSI’s GPRS mobile packet system as a mobile access medium. Subjective performance tests are carried out for the error-resilient MPEG-4 video decoder using a real-time GPRS radio access emulator

The ICASSP 2023 Speech Signal Improvement (SSI) Challenge concentrates on improving the speech signal quality of real-time communication (RTC) systems. In this paper, we introduce the speech signal improvement network (SSI-Net) submitted to the ICASSP 2023 SSI Challenge, which satisfies the real-time condition. The proposed SSI-Net has a multi-stage architecture. We present the time-domain restoration generative adversarial network (TRGAN) in the first restoration stage for speech restoration. Regarding the second enhancement stage, we employ a lightweight multi-scale temporal frequency convolutional network with axial self-attention (MTFAA-Net) called MTFAA-Lite to enhance the fullband speech. In the subjective test on the SSI Challenge blind test set, our proposed SSI-Net yields a P.835 overall mean opinion score (MOS) of 3.190 and a P.804 overall MOS of 3.178, which eventually takes the 3rd place in tracks 1&2.

Next-generation Wireless Sensor Networks (WSNs), including 5G and Beyond-5G networks, have made significant progress. However, their development requires re-evaluating intelligent communication approaches to meet the growing demands for higher data transmission rates, more efficient spectral utilization, and reduced energy consumption. Scalability issues related to energy efficiency remain a critical concern for WSNs, which remain integral to the digital revolution. This paper investigates the role of Deep Learning (DL) methods in improving energy efficiency for routing and clustering in 5G WSNs. It is pertinent to note that Deep Q-Networks (DQNs) and their variants have significantly enhanced Cluster Heads (CHs) selection schemes and routing techniques, thereby extending network lifetime by optimizing power consumption. Device-to-Device (D2D) communication is a key enabler of 5G WSNs, and recent studies have shown that combining DL with D2D can achieve significant performance gains. These approaches are critical to ensuring the implementation of intelligent, energy-efficient 5G WSNs. Future research should continue to support the development of hybrid models that integrate DL approaches and optimization techniques to adapt to dynamic network conditions in real-time communication systems.

Rapid growth of mobile wireless communication system has increased for numerous real-time applications. This causes heavy usage of radio spectrum in computing services. However, the system provides high speed communication is unable to serve the required performance with wrong use of resources and spectrum. In order to overcome this problem, Cognitive Radio Network (CRN) architecture is recommended in real time communication systems. CRN based system experiences lack of resource allocation, spectrum sensing and consumption of power. In this paper, we recommend game theory approach for allocation of resources, sharing spectrum cooperatively and ensures maxumized payoff. The extensive simulation study results in optimizing the backlog performance and reduces the consumption of energy.