• KIISE Transactions on Computing Practices
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• v.23 no.4
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• pp.238-243
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• 2017

This paper presents a study regarding a systemic structure that forms a link between heterogeneous terminals and the WebRTC (Web real-time communication) based video-conferencing systems. This system consists of a WebRTC-based terminal, a signal server for the relaying of the necessary signals for media negotiations, an MCU provides an interworking interface for the heterogeneous terminal, and the heterogeneous terminals that interwork with the MCU. This technology enables the interoperability between the WebRTC-based video-conferencing solutions and the legacy devices. In addition to reducing the replacement costs regarding new-technology-based solutions, the existing legacy equipment enables the use of multiparty-conference, thereby increasing the use ability of the conferencing systems.

• Proceedings of the Korean Society of Computer Information Conference
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• 2022.07a
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• pp.101-103
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• 2022

본 논문에서는 오프라인 컨퍼런스의 한계점인 접근성 문제와 온라인 컨퍼런스의 한계점인 참여자간 상호 소통과 네트워킹 문제점을 해결하기 위한 솔루션으로 WebRTC 기반 메타버스 컨퍼런스 시스템을 제안한다. 해당 솔루션은 WebRTC를 활용한 실시간 화상채팅 및 멀티접속을 구현하고 WebVR을 기반으로 하는 A-Frame 프레임워크를 활용하여 메타버스 컨퍼런스 시스템을 구현하였다.

• International Journal of Computer Science & Network Security
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• v.24 no.9
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• pp.186-194
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• 2024

WebRTC (Web Real-Time Communication) is a technology that enables browser-to-browser communication. Therefore, a signalling mechanism must be negotiated to create a connection between peers. The main aim of this paper is to create and implement a WebRTC hybrid signalling mechanism named (WebNSM) for video conferencing based on the Socket.io (API) mechanism. WebNSM was designed over different topologies such as simplex, star and mesh. Therefore it offers several communications at the same time such as one-to-one (unidirectional/bidirectional), one-to-many (unidirectional) and many-to-many (bi-directional) without any downloading or installation. In this paper, WebRTC video conferencing was accomplished via LAN and WAN networks, including the evaluation of resources in WebRTC like bandwidth consumption, CPU performance, memory usage, Quality of Experience (QoE) and maximum links and RTPs calculation. This paper presents a novel signalling mechanism among different users, devices and networks to offer video conferencing using various topologies at the same time, as well as other typical features such as using the same server, determining room initiator, keeping the communication active even if the initiator or another peer leaves, etc. This scenario highlights the limitations of CPU performance, bandwidth consumption and the use of different topologies for WebRTC video conferencing.

• Proceedings of the Korean Society of Computer Information Conference
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• 2016.07a
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• pp.209-210
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• 2016

본 논문은 WebRTC를 통해 웹 브라우저 안에서 플러그인의 도움 없이 peer-to-peer로 실시간 화상회의 서비스를 구현하였다. 영상, 음성과 채팅 서비스를 지원하며 WebRTC, Javascript, HTML, Node.js, CSS 기반으로 구현하였다. 시그널링 관련 STUN과 TURN 서버의 사용을 위해서는 ICE 프레임워크를 사용하였다. PC와 스마트폰에서의 일 대 일 화상회의 서비스의 작동 결과를 확인하였다.

• Journal of the Korean Institute of Intelligent Systems
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• v.23 no.6
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• pp.500-504
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• 2013

In this paper, we have developed the application of a remote multi-lingual video guide that allows interaction with Set Top Box of a thin client by using the WebRTC, an web browser-based communication method. A server accesses a web camera in conjunction with digital information display of a thin client connected remotely and is exchanged guide information with user. This server can be played in thin client STB using virtualization based the high-quality video compression technology. Further, it is not dependent on the performance of the STB and provides a video guide remotely.

• Journal of Advanced Navigation Technology
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• v.27 no.1
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• pp.139-145
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• 2023

When a doctor examines a patient in a hospital, the doctor directly checks the patient's condition and conducts a face-to-face diagnosis through dialogue with the patient. However, it is often difficult for doctors to directly treat patients. Recently, several types of telemedicine systems have been developed. However, the systems have lack of capabilities to observe heart disease, neck condition, skin condition, inside ear condition, etc. To solve this problem, in this paper, an interactive telemedicine robot system with autonomous driving in a room capable of visual examination and auscultation of patients is developed. The developed robot can be controlled remotely through the WebRTC platform to move toward the patient and check a patient's condition under the doctor's observation using the multi-joint robot arm. The video information, audio information, patient's heart sound, and other data obtained remotely from patients can be transmitted to a doctor through the web RTC platform. The developed system can be applied to the various places where doctors are not possible to attend.

• Review of Korea Contents Association
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• v.14 no.1
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• pp.34-37
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• 2016

H/W, S/W 기술과 통신 기술이 발전함에 따라 기술융합이 혁신적으로 이루어지고 있으며, 다양한 미디어 통신을 위한 기술 또한, 쉽고, 빠르게 진화하고 있다. 미디어 통신은 데이터 유형이 다양하고, 미디어 스트림의 경우 전송양이 방대하며, 전송기술 또한 다양하게 적용된다. 최근에는 웹을 활용한 전송기술을 제공하고 있으며, 이에 대한 표준기술로 webRTC 기술을 제공하고 있다. 무비용의 빠르고 쉽게 미디어 전송을 가능하게 하며, 다양한 스마트 및 미디어기기에 동일한 플랫폼이 제공 가능함으로써 개발비용을 단축시킬 수 있다. WebRTC 기술발전 과정 및 제공되는 플랫폼과 프로토콜 스택을 살펴보고, 멀티플랫폼이 제공 가능한 하이브리드 스마트 앱을 제공한다.

• Journal of Digital Contents Society
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• v.17 no.6
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• pp.581-592
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• 2016

WebRTC has quickly grown to be the world's advanced real-time communication in several platforms such as web and mobile. In spite of the advantage, the current technology in WebRTC does not handle a big-streaming efficiently between peers and a large amount request of users on the Signaling server. Therefore, in this paper, we put our work to handle the problem by delivering the flow of data with dynamical load balancing algorithms. We analyze the request source users and direct those streaming requests to a load balancing component. More specifically, the component determines an amount of the requested resource and available resource on the response server, then it delivers streaming data to the requesting user parallel or alternately. To show how the method works, we firstly demonstrate the load-balancing algorithm by using a network simulation tool OPNET, then, we seek to implement the method into an Ubuntu server. In addition, we compare the result of our work and the original implementation of WebRTC, it shows that the method performs efficiently and dynamically than the origin.

• Annual Conference of KIPS
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• 2014.04a
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• pp.210-211
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• 2014

WebRTC(Web Real-Time Communication)는 플러그인의 도움 없이 간단한 웹 표준 API(Application Programming Interface)를 이용하여 웹 브라우저 환경에서 Peer-to-Peer 실시간 통신을 가능케 하고자 하는 기술이다. 실시간 통신으로 오디오, 비디오 스트림은 물론이고 데이터 스트림을 포함한다. 해당 기술은 브라우저간에 직접적으로 Peer-to-Peer 세션을 형성하여 스트리밍을 수행하므로 중계 서버를 이용하는 통신에 비하여 향상된 네트위크 성능을 보이지만 연결을 구성하기 위한 보조 기술들이 요구된다. WebRTC 는 웹 기반 기술이기 때문에 기존의 웹 어플리케이션이 가지는 장단점을 공유한다. 본 논문에서는 WebRTC 기술과 해당 기술을 구성하고 있는 세부 기술들의 구조 및 기능을 분석한다. 또한 해당 기술을 이용하여 간단한 컨퍼런스 시스템을 설계하고 구현한 예제를 보인다.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.20 no.8
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• pp.1478-1486
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• 2016

This paper introduces technical methods to improve the accessibility and usability of a WebRTC video conference (VC) application. Simplified login is essential, by applying such as single sign-on (SSO) to improve the accessibility of VC applications. High usability and manageability are also necessary to attract more users, enhance user experiences, and save service management cost. The proposed VC application leverages SAML-based federated identity management (FIM) to enable higher service accessibility. Users can access the application with their organizational ID and SSO authentication. The FIM eases user ID management and indirectly strengthens privacy information protection. Proposed web application has high usability and manageability because users and/or administrators can easily create, join, monitor, or tear down VC sessions through RESTful web service (REST API). We verify the feasibility of the VC application after illustrating the SAML-based identity federation and the designed REST API.