• 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.

• 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.

• Proceedings of the Korea Information Processing Society Conference
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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.

• 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.

• The Journal of the Korea institute of electronic communication sciences
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• v.5 no.2
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• pp.214-220
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• 2010

In this paper, we developed component for constituent of mobile robot system using RTC which is adapted as an international standardization among the robot middleware technology. We implemented one system by connecting the f elements of mobile robot constituent to each other and performed performance test on mobile robot system using real mobile robot NRLAB 02 that were implemented.

• Clinics in Shoulder and Elbow
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• v.26 no.1
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• pp.32-40
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• 2023

Background: The purpose of this study was to identify predictors of the time from initial presentation to total shoulder arthroplasty (TSA) in patients with primary glenohumeral osteoarthritis (OA) and rotator cuff (RTC) arthropathy who were conservatively managed with corticosteroid injections. Methods: We conducted a retrospective cohort study of patients who underwent TSA from 2010 to 2021. Kaplan-Meier survival analysis was used to estimate median time to TSA for primary OA and RTC arthropathy patients. The Cox proportional hazards model was used to identify significant predictors of time to TSA and to calculate hazard ratios (HRs) with 95% confidence intervals (CIs). Statistical significance was set at P<0.05. Results: The cohort included 160 patients with primary OA and 92 with RTC arthropathy. In the primary OA group, median time to TSA was 15 months. Significant predictors of shorter time to TSA were older age at presentation (HR, 1.02; 95% CI, 1.00-1.04; P=0.03) and presence of moderate or severe acromioclavicular joint arthritis (HR, 1.45; 95% CI, 1.05-2.01; P=0.03). In the RTC arthropathy group, median time to TSA was 14 months, and increased number of corticosteroid injections was associated with longer time to TSA (HR, 0.87; 95% CI, 0.80-0.95; P=0.003). Conclusions: There are distinct prognostic factors for progression to TSA between primary OA patients and RTC arthropathy patients managed with corticosteroid injections. Multiple corticosteroid injections are associated with delayed time to TSA in RTC arthropathy patients.

• Proceedings of the Korea Information Processing Society Conference
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• 2010.11a
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• pp.852-854
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• 2010

MCM-ERC32 는 우주 환경에서 동작하는 시스템에 사용할 목적으로 유럽에서 개발된 집약 프로세서 모듈이다. MCM (Multi Chip Module)은 크게 ERC32 single chip 과 VASI (Very Advanced Sparc Interface) 및 6MByte 의 SRAM, 32MByte 의 DRAM 으로 구성되어 있다. VASI 의 경우 각종 I/O 처리 및 timer 의 기능을 수행하며 특히 VASI RTC 의 경우 VASI cycle, slot 을 이용하여 다양한 형태의 timer 구현이 가능하다. Timer 의 경우 각종 태스크의 관리와 스케줄링에 사용되는 가장 기본적이며 매우 중요한 요소이다. 위성의 고유 시간 역시 timer 를 활용하여 설계하게 되는데 이 부분이 잘 구현 되어야 정확한 임무 수행 및 위성의 제어가 가능하다. 본 논문에서는 VASI RTC 의 구조와 기능에 대해 설명하고 이를 위성의 고유 시간 운영에 적용하는 방법에 대해 다루도록 하겠다.

• 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.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2015.10a
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• pp.1035-1038
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• 2015

The video conference system which enables multiple users even at places away from each other proceed with the work at the same time is required to establish a convenient and efficient smart work environment. The conventional S/W-based video conference system is developed by the different implementation methods using provider's own technology and as a separate program for each OS type of user. On the other hand, the video conference system using HTML5 and WebRTC provides features of video conference system through a web browser that support HTML5 and WebRTC functions without installing any programs In this paper, in order to activate a HTML5-based video conference system in the domestic market, we propose the standard for the interworking functions that need to be supported in each system to ensure the interoperability among different HTML5-based video conference systems.

• Journal of Broadcast Engineering
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• v.24 no.2
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• pp.251-263
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• 2019

Recently the huge demand for Internet media streaming has dramatically increased the cost of the CDN (Content Delivery Network) and the need for a means to reduce it is increasing day by day. In this situation, a P2P-assisted DASH technology has recently emerged, which uses P2P (Peer-to-Peer) communications based on WebRTC (Web Real-Time Communication) standards to reduce the CDN cost. This paper proposes an algorithm that can significantly improve CDN cost savings in this technology by selecting peers based on the transport history. Also we implemented this algorithm in an experimental system and, after setting experimental conditions that emulate the actual mobile network environment, we measured the performance of the experimental system. As a result, we demonstrated that the proposed algorithm can achieve higher CDN cost savings compared to the conventional algorithm where peers are selected at random.