• Journal of the Korea Institute of Information and Communication Engineering
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• v.18 no.12
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• pp.2871-2877
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• 2014

Real-time communication and diversification of multimedia application services using high-capacity data over the Internet demand high speed connection, and real-time and high-speed access internet service such as multimedia make Internet traffic increase rapidly. Meanwhile, QoS (Quality of Service) of internet service is not satisfied. In this situation, ISP (Internet Service Provider) has been required to improve service quality and extend network according to the user's needs. This request includes issue of extending network focused on increasing router and the number of routing table as well as simply extending bandwidth.

• 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 Advanced Marine Engineering and Technology
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• v.27 no.1
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• pp.65-75
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• 2003

The internet telephony service is one of the successful internet application services. VoIP is the key technology for the service to come true. VoIP uses H.323 or SIP as the standard protocol for the distributed multimedia services over the internet environment, in which QoS is not guaranteed. VoIP carries the packetized voice by using the RTP/UDP/IP protocol stack. The UDP-based internet services cause the data transmission problem to the users behind the internet firewall. So does the internet telephony service. The users are not able to listen the voices of the counter-parts on the public internet or PSTN. It makes the problem more difficult that the internet telephony service addressed in this paper uses only one UDP port number to send the voice data of all sessions from gateway to terminal node. In this paper, two schemes including the usage of dummy UDP datagrams, and the protocol conversion are suggested. The implementation of one of the schemes, the protocol conversion, and the performance evaluation are described in detail.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.14 no.4
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• pp.39-44
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• 2014

IMS(IP Multi-Media Subsystem) base on the IP service platform which can offer multimedia as the voice, audio, video, and data is service platform. In 3G mobile communication in the early day, IMS had a suggestion for supporting to multimedia service in the 3GPP. But now It is broadly substituting in the IPTV, wire phone company and it is substituted in internet platform base on the soft-switch in currently. Especially nowadays, 4G LTE in a mobile communication company is rapidly growing in market. Therefore, in this study, we had designed to the main prosser that can admit to 1,000 user over and SIP gateway which can link the IMS 코어 that can link SIP Device which adopt the standard protocol on the SIP.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.39B no.7
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• pp.467-476
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• 2014

Recently, various services which use the fused form of broadcasting and communication are being offered over the Internet globally. To provide various interactive services through connected smart TV on wired or wireless network, the seamless multimedia service is key feature. In this paper, Dynamic Adaptive Streaming over HTTP (DASH)is adopted to develop cost-effective adaptive streaming system for Open Hybrid TV(OHTV) service. This system can provide seamless adaptive streaming service for IP VOD, Video Clips, Ad insertion which are defined in OHTV.

• The Journal of the Korea institute of electronic communication sciences
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• v.4 no.2
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• pp.93-100
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• 2009

As the Internet Protocol be widely/rapidly used in packet communication, common carriers are providing the multimedia service(Both direction real-time voice, video conference, remote educational etc.)on the Internet. Also the 070 VoIP (Voice over IP) service is provided by the carriers on the packer network. In order to offer VoIP service in Korea, common carrier has to acquire the optimum level for QoS(quality of service). In this paper, we study on CODEC quality to get a higher QoS for VoIP.

• Proceedings of the Korean Information Science Society Conference
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• 2001.10c
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• pp.763-765
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• 2001

현재 활발하게 연구되고 있는 VoIP(Voice Over IP)기술로는 ITU-T에서 제정한 멀티미디어 회의를 위한 H.323 표준기 술과 IETF에서 제정 한 SIP(Session Initiation Protocol)를 이용한 기술이 있다. 본 논문에서는 기존의 인터넷 기반의 환경에 그대로 접목될 수 있고, 기능 및 부가 서비스의 제공이 용이한 IETF의 SIP를 설계 구현하고, ITU-T에서 제안한 H.323과 비교해 보았다.

• The KIPS Transactions:PartC
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• v.8C no.1
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• pp.41-50
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• 2001

With growth of Internet, real time multimedia application services require guarantee of QoS (Quality of Service). However, the existing Internet based on best-effort service is not enough to support those services, so MARS model to support IP multicast over ATM networks and RSVP to provide heterogeneous QoS by resource reservation scheme are proposed. As a simulation result, it is proved that processing delay of MARS is increased and success of resource reservation is decreased as a number of QoS change request of a receiver is increased. Therefore, it is important to provide heterogeneous QoS that distribute a load of MARS caused by convergence of messages and efficiently manage the resource of network as to find out scale of receivers and frequency of QoS change request.

• Proceedings of the Korea Multimedia Society Conference
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• 2000.04a
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• pp.7-16
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• 2000

초고속 통신의 실현과 다양한 멀티미디어 서비스를 위한 TCP/IP 통신의 전송 메카니즘으로서 ATM이 널리 사용되고 있다. TCP층과 ATM층의 상호작용이 TCP층 뿐만 아니라 ATM층에도 영향을 미쳐 처리율과 공정성의 성능이 저하되므로, ATM상의 TCP 트래픽 파라미터, 즉 TG(timer granularity)와 윈도우 크기를 적절하게 조정함으로써 TCP 흐름제어가 ABR 율제어와 상호작용하지 않는다. 그 결과로 상호작용의 영향을 최소화할 수 있으며, TCP의성능향상을 가져올 수 있다. 본 논문에서는 윈도우 크기와 TG와 같은 TCP 파라미터를 조절하여 VBR서비스상에서의 TCP 트랙픽의 성능평가를 통해 높은 처리율과 공정성을 나타내었다.

• Proceedings of the Korea Multimedia Society Conference
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• 2002.05c
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• pp.410-415
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• 2002

최근 초고속인터넷과 다채널 방송, VOD(Video on Demand), HDTV 등의 광대역 통합 서비스와 음성, 음악, 원격강시, 화상전화/회의 등의 부가기능을 제공하는 통신 서비스를 동시에 수용하기 위해 VDSL(Very high speed Digital Subscriber Loop)에 대한 연구가 진행중이다. 본 논문에서는 DSL 포럼 및 FS-VDSL 위원회의 기고서와 규격를 바탕으로 ATM과 IP 기반의 VDSL물 이용한 디지털 방송 서비스를 제공하기 위한 종단간 ATM 모델과 ATM-IP 연동 모델에 대한 망의 구성 및 프로토콜 스택을 제시하고 비교 분석한다. 그리고 ATM/VDSL 기반에서 DSM-CC(Digital Storage Media Command and Control)와 IGMP(Internet Group Management Protocol) 프로토콜을 이용한 디지털 방송 서비스 제공 절차를 제안한다.