Multipoint Control Unit Research Articles

Design of Personalised English Distance Teaching Platform Based on Artificial Intelligence

In the application of the existing distance education platform, the server load is often unbalanced, which increases the response time of the platform. This paper designs a personalised English distance teaching platform based on artificial intelligence. Put forward the overall architecture of the platform, and design the platform on this basis. In order to realise the personalised course resource recommendation function, a collaborative filtering personalised recommendation algorithm is designed to get the best course recommendation results. Using artificial intelligence technology to set up multipoint control unit, the task balance is allocated to multiple operation units, and the reasonable allocation of curriculum resources is realised. According to the functional requirements of the platform design, the composition of functional modules is further set to realise the scheduling and management of curriculum resources. The experimental results show that, under the condition of the same number of concurrent users, the average response time of this design platform is less than that of the existing distance teaching platform, which shows that it has certain advantages in server load balancing and resource scheduling, which can improve the rapid response ability of the platform and enhance the stability of the platform.

Techno-pedagogical Solution to Support the Improvement of the Quality of Education in Technical and Vocational Training in Mauritania

E-learning is the most promising and fastest growing activity since the advent of the COVID-19 pandemic. Although the pandemic seems to have been eradicated in several countries of the world, it is worth mentioning that some positive cases of the covid-19 variant have been detected. This could accelerate the rate of infection again. Hence the interest generate in reinforcing the quality of distance learning platforms. Technical and vocational training (TVT) in Mauritania is based on science, technology, engineering, and mathematics (STEM) disciplines. Unfortunately, the expansion of the COVID-19 pandemic has negatively impacted the quality of education with a halt in teaching affecting 8000 students. Yet, the quality of education in these disciplines is a key factor in meeting the demands of emergence and economic growth. This paper advocates the mixed pedagogical model by proposing a techno-pedagogical solution to improve the quality of teaching and learning processes. The proposed solution combines the use of technologies such as Modular Object-Oriented Dynamic Learning Environment (Moodle) and Web Real Time Communication (WebRTC) to provide pedagogical services in a context with a limited Internet connection. In addition, we set up a signaling system to maintain direct communication between the pairs, Application Programming Interface (API) of Multipoint Control Unit (MCU) to ensure simultaneous collaboration in a peer-to-peer context, used implementations of security protocols such as Datagram Transport Layer Security (DTLS) and Secure Real-time Transport Protocol (SRTP) to secure data transport.

Adaptive multicast streaming for videoconferences on software-defined networks

Real-time applications, such as video conferences, have strong Quality of Service requirements for ensuring a decent Quality of Experience. Nowadays, most of these conferences are performed over wireless devices. Thus, an appropriate management of both heterogeneous mobile devices and network dynamics is necessary. Software Defined Networking enables the use of multicasting and stream layering inside the network nodes, two techniques able to enhance the quality of live video streams. In this paper, we propose two algorithms for building and maintaining multicast sessions in a software-defined network. The first algorithm sets up the initial multicast trees for a given call. It optimally places the stream layer adaptation function inside the core network in order to minimize the bandwidth consumption. This algorithm has two versions: the first one, based on shortest path trees is minimizing the latency, while the second one, based on spanning trees is minimizing the bandwidth consumption. The second algorithm adapts the multicast trees according to the network changes occurring during a call. It does not recompute the trees, but only relocates the stream layer adaptation functions. It requires very low computation at the controller, thus making our proposal fast and highly reactive. Extensive simulation results confirm the efficiency of our solution in terms of processing time and bandwidth savings compared to existing solutions such as multiple unicast connections, Multipoint Control Unit solutions and application layer multicast.

A Novel Multi-Level Security Technique Based on IRIS Image Encoding

Providing highly secured access to restricted or private areas has become highly required these days, mainly due to terrorism threats. One method of security is no longer sufficient, hence the term and technology of a “multi-level” security system was developed by integrating more than one security procedure, on both hardware and software levels. This research provides a simple, cheap, easily achievable, yet highly secured multi-level security system to control access through doors. The system integrates IRIS scan authentication, innovative IRIS image encoding, encrypted of mobile communication, and multipoint Control Unit (MCU) Security as main procedures of security. The system’s novelty shows in the encoding and encryption of IRIS image data that is acquired by a mobile phone before it is sent to the authentication site, where it is decrypted by a cheap and fast MCU to retrieve the IRIS image that is fed into a Neural Network in order to grant authorization to the user.

Video conference in the fog: an economical approach based on enterprise desktop grid

There exist two classical and well-understood approaches to video-processing tasks (such as mixing or trans-coding) for videoconferencing. The first one is using a centralized multipoint control unit (MCU), hardware- or software-based, deployed on-premises or in the cloud. In the second approach, the video-processing tasks are directly handled in endpoints (i.e., equipment such as PCs, laptops, and tablets that are involved in the video session). Performance is then restricted by device characteristics, especially in the case of mobile devices. In this paper, we propose a third alternative approach. It has been shown that there exist significant computational resources in user equipment deployed in enterprises, which are under-utilized most of the time. In this paper, we propose a system, which distributes real-time video-processing tasks on these available resources. A dedicated multi-attribute decision-making (MADM) method is designed in order to take into account the variety of attributes impacting Quality of Experience. We enumerate a comprehensive list of events, which may cause distribution or redistribution of video-processing activities and provide simple algorithms to tackle all these cases. We then test the MADM algorithm by means of simulations in order to study the impact of the main critical parameters.

Adaptive Speech Streaming Based on Packet Loss Prediction Using Support Vector Machine for Software-Based Multipoint Control Unit over IP Networks

Adaptive Speech Streaming Based on Packet Loss Prediction Using Support Vector Machine for Software-Based Multipoint Control Unit over IP Networks

Audio Mixer Algorithm for Enhancing Speech Quality of Multi-party Audio Telephony

The speech quality of multi-party audio telephony between two, three or more participants is decreased by audio volume imbalance, audio volume saturation and noise level increase. To solve this issue, this paper proposes an advanced audio mixing algorithm for software-based multi-point control unit. Our approach is based on the combined voice activity detection and gain control technique that consists of a set of algorithms that classify audio signals, estimate audio volumes, adjust gain factors and mix audio signals of all channels. The proposed audio mixing algorithm is computationally efficient, delivers high-quality speech, and is suitable for use in any practical multi-party audio telephony.

Software-based serverless endpoint video combiner architecture for high-definition multiparty video conferencing

This paper proposes an endpoint video combiner architecture in a multipoint control unit (MCU) system for high definition multiparty video conferencing. The proposed architecture addresses the current reliability, computational and quality drawbacks of a conventional centralized based video combiner architecture. This is achieved by redesigning the MCU video to move away the video combiner from the bridge and into the client endpoints. Moreover, the proposed architecture represents a serverless system and is able to scale a large number of clients at high resolutions in a multipoint video conferencing session. In order to realize this design, this paper also proposes a custom robust sustainable session management protocol which allows a dynamic multi-port management between the MCU video and client endpoints. In addition, the proposed custom session management protocol includes recommendation for a session protection structure. Experimental results suggest that the proposed architecture exhibits significant computational frame rate performance gains of up to 762.95% in comparison with the conventional centralized video combiner architecture based on a series of four and eight high definition combined video assessments. Moreover, reliability analysis suggests that the proposed architecture is also able to consistently sustain a high frame rate performance within a long duration high definition multipoint video conferencing session.

A study of H.263 traffic modeling in multipoint videoconference sessions over IP networks

This manuscript is a contribution on the modeling of H.263 traffic in multipoint videoconference sessions over IP Networks. Our study includes analysis and modeling assessment of extensive data gathered during realistic videoconference sessions between commercial H.263-compliant terminal clients (with different videoconference software packages installed). All terminal clients were communicating through a Multipoint Control Unit (software or hardware MCU) at ‘switched presence’ mode and for comparative purposes the same typical videoconference content (a person speaking, with mild movement and occasional zoom/span) was used. The analysis of the H.263 data at the frame level suggests that the traffic from the different terminals to the MCU can be represented by a stationary stochastic process with an AutoCorrelation Function (ACF) rapidly decaying to zero and a Gamma formed marginal frame-size Probability Distribution Function (PDF). An accurate analysis of the H.263 traffic from all terminals (with the same visual content and different videoconference software used) shows indicative differences in the ACF and PDF of different terminals' traffic and insights that no generic traffic model can be applied for all cases. Aiming at a realistic, reusable and simple H.263 traffic model, conservative enough for queueing analysis and network estimation, this study discusses methods for calculating the appropriate model parameters from the observed traffic data and proposes a new technique for unconventional fitting of the PDF. The presented modeling and queueing results indicate the suitability of the proposed models for H.263 traffic modeling in IP networks.

A world first development of a multipoint videophone system over 3G-324M protocol

This paper describes a world first development of a multipoint videophone system over third generation wireless cellular phone services using the 3G-324M circuit-based international standard protocol. A multipoint TV phone system comprises a control server and a multipoint control unit (MCU), and enables multipoint conferencing systems over 3G-324M. The picture quality is 176 by 144 picture size format (QCIF) and 10fps at 64kbps constant bitrate. The control server unit controls the conference reservations from the users, and MCU controls audio/visual processing of mixing audio and video data. Up to eight people can join the same meeting using 3G-visual terminals over 3G-324M; also, simultaneously, 60 3G-visual terminals can be connected to the system, and the number of terminals can be extensible according to market demand. In this paper, the system architecture is described. Since this system enables audio-visual communications anywhere and at anytime, audio-visual communications are expected to penetrate the market.

Measurement and Statistical Analysis of Asymmetric Multipoint Videoconference Traffic in IP Networks

The paper contributes results on the modeling of videoconferencing traffic over IP networks. The study is based on extensive data, gathered by tracing the actual packet exchange during a comprehensive set of realistic teleconferencing sessions over an asymmetric platform, in which commercial H.261-compliant terminal clients were communicating through a Multipoint Control Unit (MCU) at `continuous presence' mode. Analysis of the data suggests that the video traffic from the client terminals can always be represented at the frame level as a stationary stochastic process with an autocorrelation function of exponentially fast decay and a marginal frame size distribution of approximately Gamma form. The video traffic from the MCU to the clients is again stationary and with exponentially decaying correlations, while the corresponding marginal frame-size PDF has the form of an appropriately weighted sum of Gamma components, the number of terms in the sum always being equal to the number of conferring terminals. The paper discusses methods for correctly matching the parameters of the modeling components to the data and for combining these components into complete traffic models that have been proposed in the literature.

Real-Time Multimedia Applications in a Web-Based Robotic Telecare System

In this paper, real-time multimedia applications are integrated into a Web-based robotic telecare system. Using Java media framework (JMF), a video/audio conference system is developed to improve the interaction among caregivers, and switches remote robot manipulating privilege with the help of a centralized management server. This video/audio conference system is established on a similar profile of H.323 multipoint control unit (MCU). Access control is utilized to guarantee confidentiality of telecare users. Different telecare services are provided to meet requirements of various users. The whole telecare system is verified in a campus network.

A Standard-Compliant Virtual Meeting System with Active Video Object Tracking

This paper presents an H.323 standard compliant virtual video conferencing system. The proposed system not only serves as a multipoint control unit (MCU) for multipoint connection but also provides a gateway function between the H.323 LAN (local-area network) and the H.324 WAN (wide-area network) users. The proposed virtual video conferencing system provides user-friendly object compositing and manipulation features including 2D video object scaling, repositioning, rotation, and dynamic bit-allocation in a 3D virtual environment. A reliable, and accurate scheme based on background image mosaics is proposed for real-time extracting and tracking foreground video objects from the video captured with an active camera. Chroma-key insertion is used to facilitate video objects extraction and manipulation.We have implemented a prototype of the virtual conference system with an integrated graphical user interface to demonstrate the feasibility of the proposed methods.

멀티미디어 회의 서비스의 직접모델 방식에 의한 H.323 게이트키퍼의 설계 및 구현

고속통신망과 컴퓨터 기술의 발전과 함께 다양한 멀티미디어 응용서비스가 개발되고 있다. 그 중 인터넷망에서의 영상회의 시스템은 활용도가 매우 뛰어나며 많은 관심을 받고 있으며, 이를 위하여 ITU-T 에서는 H.323으로 표준안을 제정하고 있다. H.323 표준안은 단말기, 다지점 제어장치, 게이트키퍼, 게이트웨이의 4개의 구성 요소로 구성되어 있다. 이 중 게이트키퍼의 역할은 영역의 관리로 상대방의 이름주소를 IP 주소로 변환해 주는 주소변환 서비스, 회의의 시작과 종료를 제어해 주는 회의승인 서비스, 그리고 H.323 단말기들의 대역폭을 관리해 주는 대역폭관리 서비스 등을 들 수 있다. 본 논문에서는 인터넷 환경에서의 영상회의 요소들의 효율적인 관리를 위한 게이트키퍼의 구현에 대하여 기술한다. 실험결과, 대표적인 H.323 단말기인 CUSeeMe와 NetMeeting 이용하여 구현된 게이트키퍼가 표준안을 만족함을 입증한다. A various multimedia application services should be developed with techniques of high speed networks and computer. Among these, video-conference system over Internet is useful and important, and the standardization for it should be showed in ITU-T H.323. H.323 standardization consists of four components such as Terminal, MCU(Multipoint Control Unit), Gatekeeper, and Gateway. Among these, the functions of Gatekeeper are as follows, firstly the address translation service to translate the alias address into the IP address, secondary conference admission control service to control of conference start and termination, thirdly bandwidth management service for H.323 terminals. In this paper, we implemented the Gatekeeper for an efficient management of video-conference components in Internet environment, and will introduce our system. As the experimental results with CUSeeMe and Netmeeting which are well-known H.323 terminal, it is known that our gatekeeper should be satisfied with H.323 standardization.

Bit-rate control for uniform picture quality within composed video sequence

In a continuous presence multipoint control unit, the multiple video sequences of the high spatial resolution are decoded and downconversed, then composed into an output video sequence. When the bandwidth to transmit the composed sequence is fixed, if the bit rate for each low spatial resolution sequence within the composed sequence is independently allocated, then the degradation of subjective picture quality among the low spatial resolution sequences is severe due to the different video characteristics. An adaptive bit-rate control method to maintain the uniform picture quality among the low spatial resolution sequences is presented. We first find a relationship of bit rate and quantizer parameter for the video manipulation and composition environments. Then we obtain the bit rate for each low spatial resolution sequence to have same quantizer parameter among the sequences by adaptively using the coded results of low or high spatial resolution sequences while considering the scene change of each sequence. We show by simulation that the proposed adaptive bit-rate control method can maintain almost uniform picture quality among the low spatial resolution sequences and almost constant picture quality within the composed sequence in comparison to an independent bit-rate control method.