• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.23 no.3
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• pp.59-64
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• 2023

Fire and disaster broadcasting systems are divided into analog, digital, and network-based digital public address systems, and important specifications in network-based digital public address systems are low-latency audio, high sampling rate, and multi-channel input and output. In the past, it has been widely used to the AoE method for distinguishing based on the MAC address of the data link layer. However, this method has a problem of increasing complexity and cost. This proposal is an AoIP/UDP method, which allows communication to be easily distinguished by IP address without the need for a separate redundant network, so that the network can be freely used and configured, and cost can be reduced by reducing complexity. After implementing the AoIP/UDP method, the experimental results showed that the cost was improved with the equivalent performance with 2.66ms latency.

• Proceedings of the Korea Contents Association Conference
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• 2006.11a
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• pp.845-848
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• 2006

In this paper, a ETI management system for a variety of media services based on Terrestrial-Digital Multimedia Broadcasting(T-DMB) is introduced. This system consists of DMB I/O board and DMB multiplexer. The system transforms media data stream over IP(TCP/IP or UDP/IP) for these services to ETI frame(G.703) pushed to T-DMB transmitter.

• Journal of Internet Computing and Services
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• v.13 no.1
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• pp.57-63
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• 2012

Recently, Development of high-speed internet networks can be a real-time video conference system. SD-quality still images using the software codec system exists, and two-way system with mobile systems, but in most video only send and receive video system. In this study, HD web-cam using the two-way live broadcasts simultaneously in the education system and the mobile system has developed an audible system. In the study using the H.264 standard video compression techniques were applied to high-bandwidth compression technology, smart phones, to develop a streaming server for broadcasting were applied. Video sharing, document, web-sharing was made possible in real time. In addition, the private IP for WiFi zone, available anytime, anywhere that can be converted to a public IP technology, IP tunneling technology applied.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2012.07a
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• pp.339-342
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• 2012

증강방송 서비스는 하이브리드 방송(Hybrid Broadcasting) 환경 하에서 IP망을 통해 전송받은 증강 콘텐츠를 방송 프로그램과 같이 보여줌으로써 스마트TV와 같은 단말을 통해 증강 현실을 제공하는 서비스이다. 증강방송 서비스를 제공하기 위해서는 증강 콘텐츠의 위치(서버), 증강 콘텐츠가 표시될 화면상의 위치 및 시간 정보 등의 증강 콘텐츠 연동정보가 필요하다. 본 논문에서는 MPEG-2 SYSTEM: Transport Stream의 서술자(Descriptor)를 통해 증강 콘텐츠 연동정보를 전송하는 방법 및 IP망의 증강 콘텐츠를 이용한 증강방송시스템을 제안한다.

• ETRI Journal
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• v.38 no.5
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• pp.838-846
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• 2016

The objective of digital broadcasting has evolved from providing a plain video service to offering a realistic visual experience. Technologies such as 3DTV and UHDTV have been suggested to achieve this new objective by providing an immersive and stereoscopic visual experience. However, owing to the high bandwidth requirements of such services, the broadcasting industry has faced a challenge to find a new transport mechanism for overcoming the bandwidth limitation. The standardization organizations, the Advanced Television Systems Committee, Digital Video Broadcasting, and Telecommunications Technology Association, have been working on the integration of broadcasting and a broadband network (IP) to resolve the bandwidth issue of realistic video services. This paper introduces a frame-level timeline synchronization and transport system target decoder model for providing a stable 3DTV service over a hybrid network. The experimental results indicate that the proposed technologies can be successfully adopted as a reference model in a broadcast-broadband hybrid 3DTV service and other IP-associated hybrid broadcasting services.

• Journal of Advanced Navigation Technology
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• v.21 no.1
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• pp.113-118
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• 2017

As interest in mobile devices and networks has increased recently, voice over internet protocol (VoIP) service, which is a technology for transmitting voice data using an existing internet protocol (IP) network, has rapidly spread, Cheap voice call service has become possible. As the digital broadcasting service becomes popular, hybrid fiber coaxial (HFC) network technology, which uses broadband cable network through fusion of broadcasting and communication, utilizes existing communication system and network equipment to provide various new services such as interactive broadcasting service. Therefore, if UGS-AD is applied to VoCM and RTPS is applied to MTA in order to guarantee the quality of voice data in actual HFC Internet service network, it is possible to smoothly perform voice data transmission in narrow upstream band which is a problem in actual commercial HFC network We also proposed a method to improve VoIP service by improving QoS of voice data in HFC Internet service network.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2005.11a
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• pp.155-158
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• 2005

본 논문에서는 대화형 기능을 가지는 MPEG-4기반의 멀티미디어를 IP 네트워크를 통해 전송할 수 있는 대화형 멀티미디어 스트리밍 시스템을 제안한다. 제안한 대화형 멀티미디어 시스템은 MPEG-4 콘텐츠를 분석, 전송할 수 있는 스트리밍 서버와 전송되는 MPEG-4 콘텐츠를 수신, 재생할 수 있는 클라이언트, 그리고 효율적인 멀티미디어의 전송을 위해 도입한 오버레이 네트워크로 구성된다. 오버레이 네트워크는 실제 IP 네트워크 상에 콘텐츠를 중계하는 릴레이 노드와 릴레이 노드들을 관리하는 네트워크 분배 관리자로서 구성된다.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.15 no.4
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• pp.99-104
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• 2015

Service platform which can offer various multimedia communication as the video, audio, voice and data is IMS(IP Multi-Media Subsystem). It is effective in the company is introducing only such convergence IMS services to be required to provide various multimedia services at the lowest cost and existing communication environment while keeping the maximum Therefore, in this study, we had developed IMS 코어 main system that not more than 1,000 employees of companies can effectively establish IMS solutions. This system is located at the middle between IMS terminal and CSCF(Call Session Control Function) in line with IMS services and provides CSCF in response to the IMS terminal and IMS terminal in response to the CSCF. As well, corded telephone and SIP phone which were used as terminal is linked with gateway.

• International journal of advanced smart convergence
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• v.6 no.2
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• pp.1-8
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• 2017

Recently, the development of the information communication industry has made many changes in the industrial acoustic industry. Especially, it has a great influence on the change of system and equipment of acoustic system. Analog equipment is changing to digital equipment, and integrated control equipment makes it easier to operate and manage the sound system. However, the integrated control system currently on the market is only controllable for some devices. In this paper, we propose a new AoIP - based system configuration method, which enables the operation status monitoring, unmanned operation and self - diagnosis of equipment. As a result of the study, it is confirmed that the proposed system can be operated, monitored, and self - diagnosed at remote sites. It is expected that an AoIP- based sound system will be the industry standard in the future.

• International journal of advanced smart convergence
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• v.6 no.2
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• pp.51-58
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• 2017

Recently, rapid advances of Ethernet and IP technology have brought many changes in the sound industry. In addition, due to AoIP-based audio transmission technology, various problems of the acoustic system (sound quality deterioration due to long distance transmission, complicated wiring) have improved dramatically. However, when many distributed audio systems are connected with AoIP equipment, if there is a problem in the equipment, it is impossible to operate the connected system. AoIP equipment only can transmit audio signals but cannot adjust the system for acoustic environment. In this paper, AoIP equipment is to be installed with sound equipment on a one-to-one basis, so that various existing problems can be solved and adjustment of sound quality (reverberation, echo, delay and EQ) can be possible by AoIP-based OAC (On-site Audio Center) with built-in DSP function. As a result, uncompressed real-time transmission by distributed transmission/receipt module in OAC (On-site Audio Center) and high quality sound by adjustment of sound quality with built-in DSP can be acquired. It is expected that OAC based sound system will be the industry standard in ubiquitous environment.