• The KIPS Transactions:PartC
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• v.10C no.3
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• pp.345-354
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• 2003

IP Multicast is required of more little network resources than one in unicast. Furthermore, reliable multicast has been researched for supporting reliability at IP Multicast mechanism. Although these studies are carried out, they only have focused on general data. In other words, in case that realtime packet, they can not support reliability since they do not consider realtime properties such as dependency of interframe and playback in time. Besides, we also request to support scalability because we are based on Mobile IP network together with internet. Thus, we need a mechanism to guarantee reliability and scalability of realtime stream data. In this paper, we propose PER (Preceding Error Recovery) that reflect characteristics of the realtime data, especially for H.323. PER provides scalable reliability because it is based on tree-based multicast basically and helps to support scalable relibility as reducing control packet and recovers stream buffer space from underflow status as soon as possible. PER shows much better scalable and reliable than existing works.

• Journal of Broadcast Engineering
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• v.13 no.6
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• pp.930-940
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• 2008

This paper proposes a prototype realization of multi-view HD video transport system with synchronized multiplexing over IP networks. The proposed synchronized multiplexing considers the synchronization during video acquisition and the multiplexing for the interactive view-selection during transport. For the synchronized acquisition from multiple HDV camcorders through IEEE 1394 interface, we estimate the timeline differences among MPEG-2 compressed video streams by using global time of network between the cameras and a server and correct timelines of video streams by changing the time stamp of the MPEG-2 system stream. Also, we multiplex a selected number of acquired HD views at the MPEG-2 TS (transport stream) level for the interactive view-selection during transport. Thus, with the proposed synchronized multiplexing scheme, we can display synchronized HD view.

• Korean Management Science Review
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• v.23 no.3
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• pp.133-156
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• 2006

The ubiquitous technology, leading the stream of change in telecommunication market, makes VoIP a foundation technology to enter NGN era. With this trend in mind, VoIP providers try to satisfy customer needs and invest on R&D in order to survive in this emerging market. The critical issues to VoIP providers are two fold the first one is to find out the quality attributes of Internet telephony service; and the second one is to assess the impact they have on customer satisfaction, customer loyalty and customer performance. In this study, we attempt to empirically answer the above two issues employing SEM (structural equation modeling) and AHP (analytic hierarchy process), and suggest strategic guidelines for successful deployment of Internet telephony service.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.20 no.9
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• pp.7-13
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• 2019

Unlike a typical video wall system, video wall systems used for integrated monitoring in smart city environments should be able to display various videos, images, and texts simultaneously. In this paper, we propose an Internet Protocol (IP)-based video wall system that has no limit on the number of videos that can be monitored simultaneously, and that can arrange the monitor screen layout without restrictions. The proposed system is composed of multiple display servers, a wall controller, and video source providers, and they communicate with each other through an IP network. Since the display server receives and decodes the video stream directly from the video source devices, and displays it on the attached monitor screens, more videos can be simultaneously displayed on the entire video wall. When one video is displayed over several screens attached to multiple display servers, only one display server receives the video stream and transmits it to the other display servers by using IP multicast communications, thereby reducing the network load and synchronizing the video frames. Experiments show that as the number of videos increases, a system consisting of more display servers shows better decoding and rendering performance, and there is no performance degradation, even if the display server continues to be expanded.

• Journal of Ubiquitous Convergence Technology
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• v.2 no.2
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• pp.88-96
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• 2008

IPTV Content delivery systems over wired networks confront scalability problems due to their high network bandwidth requirement for real-time services. Especially, VoD service provides Trick Mode features such as pause, fast forward and similar operations. However, Trick Mode services are delivered by the method of unicast only for controlling of the stream. With a point of views, this paper propose a new real-time multimedia streaming architecture over IP Networks, which tries to achieve bandwidth efficiency and supporting for mass clients better than traditional unicast services. The proposed methods divide the Transport Stream into a series of segments. After that, this divided partial Transport Stream makes multicast streaming periodically. Meanwhile Set-top Box of a client makes a rearrangement orderly by using Presentation Time Stamp field from the served Transport Stream packets. While the current Transport Stream segment is playing, it should be guaranteed that the next segment is downloaded on time. Consequently, the original video content can be played out continuously. The detail introduction of a new real-time multimedia streaming system with analysis and simulation follows as below.

• 대한공업교육학회지
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• v.33 no.1
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• pp.233-248
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• 2008

Stream Control Transmission Protocol(SCTP) is a new transport protocol that is known to provide improved performance than Transmission Control Protocol(TCP) in multi-homing environment that is having two and more IP addresses. But currently single-homed computer is used primarily that is having one IP address. To identify whether mean transfer time for SCTP is faster that for TCP in single-homed environment considering packet loss, we make up real testbed regulating the bandwidth, delay time and packet loss rate on router and observe the transfer time. We write server and client applications to measure SCTP and TCP mean transfer time by C language. Analysis of these experimental results from the testbed implementation shows that mean transfer time of SCTP is not better than performance of TCP in single homed environment exceptional case. Main reasons of performance are that SCTP compared to TCP stops transmitting data by timeout and data transmission is often delayed when SACK congestion happens. The result of study shows that elaborate performance tuning is required in developing a new SCTP module or using a implemented SCTP module.

• The Journal of the Korea institute of electronic communication sciences
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• v.15 no.6
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• pp.1045-1054
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• 2020

In this paper, we designed a TCP/IP based ring buffer system that can stably transfer bulk data streams in the unstable network environments. In the scheme we proposed, The observation data stream generated and output by each radio observatory's backend system as a UDP frame is stored as a UDP packet in a large capacity ring buffer via a socket buffer in the client system. Thereafter, for stable transmission to the remote destination, the packets are processed in TCP and transmitted to the socket buffer of server system in the correlation center, which packets are stored in a large capacity ring buffer if there is no problem with the packets. In case of errors such as loss, duplication, and out of order delivery, the packets are retransmitted through TCP flow control, and we guaranteed that the reliability of data arriving at the correlation center. When congestion avoidance occurs due to network performance instability, we also suggest that performance degradation can be minimized by applying parallel streams.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.8 no.6
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• pp.1076-1085
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• 2004

Recently, the demands for real-time service and multimedia data are rapidly increasing. There are significant redundancies between header fields both within the same packet header and in consecutive packets belonging to the same packet stream. And there are many overheads in using the current UDP/IP protocol. Header compression is considered to enhance the transmission efficiency for the payload of small size. By sending the static field information only once initially and by utilizing dependencies and predictability for other fields, the header size can be significantly reduced for most packets. This work describes an implementation for header compression of the headers of IP/UDP protocols to reduce the overhead on Ethernet network. Typical UDP/IP Header packets can be compressed down to 7 bytes and the header compression system is designed and implemented in Linux environment. Using the Header compression system designed between a server and clients provides have the advantage of effective data throughput in network. Since the minimum packet size in Ethernet is 64 bytes, the amount of reduction by header compression in practical chatting environment was 6.6 bytes.

• Proceedings of the Korea Information Processing Society Conference
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• 2011.04a
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• pp.1188-1191
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• 2011

IPTV 방송에서 인터넷 기반 멀티캐스트는 일대다 또는 다대다 통신을 위한 차세대 중요한 서비스로 주목 받고 있다. 멀티캐스트는 네트워크 또는 애플리케이션 레벨에서 서비스할 수 있다. IP 멀티캐스트는 소스노드에서 라우터로 데이터그램을 보내면 라우터가 이를 복제하여 수신노드들에게 전달해 주는 네트워크레벨 서비스로 네트워크 자원을 효율적 사용할 수 있다. 그러나 네트워크에 IP 멀티캐스트 라우터가 설치되어야 하는 등 여러 문제로 인해 널리 사용되지 못하고 있다. 따라서 대안으로 애플리케이션 레벨에서의 오버레이 멀티캐스트가 주목 받고 있다. 오버레이 멀티캐스트는 종단호스트가 라우터처럼 동작하는 것으로 비록 IP 멀티캐스트에 비해서 링크 사용율과 지연 값이 높아질 수 있지만, IP 멀티캐스트의 현실적인 적용의 어려움을 해결할 수 있는 장점을 가지고 있다. 본 논문에서는 IP 멀티캐스트가 제공되지 않는 네트워크에서 효율적인 SVC(Scalable Video Coding) 멀티미디어 방송서비스와 SSENS(SVC Stream Exchange Network Server) 라우팅을 위한 MST(Minimum Spanning tree)를 목적으로 하는 오버레이 멀티캐스트 트리를 구현하는 알고리즘을 설계한다. 제안된 알고리즘의 성능 분석을 위해 시뮬레이션을 통해 패킷 손실 측면에서 Prim 알고리즘에 비해 평균 패킷 손실율이 40% 가까이 향상됨을 증명하였다.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2011.11a
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• pp.356-359
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• 2011

최근 방송 시스템의 디지털화, 통신기술과 가전제품의 발전으로 스마트 TV 시대가 열리게 되었다. 이제 시청자들은 스마트 TV를 통해 기존의 디지털 방송과 IP(Internet Protocol) 통신이 융합한 하이브리드 방송 서비스가 가능해지면서 기존 방송망을 통한 실시간 콘텐츠 서비스와 더불어 IP망을 통한 VOD(Video on Demand) 서비스, 멀티뷰(Multi-view) 서비스의 이용이 가능하게 되었다. 그 중 멀티뷰 서비스는 시청자가 동일한 시간의 영상을 다양한 시점에서 동시에 볼 수 있는 서비스로서, 하이브리드 방송 환경 하에서 현 방송 규격인 MPEG-2 TS(Transport Stream) 콘텐츠와 IP망을 통해 전달되는 실시간/비실시간 콘텐츠가 멀티뷰 서비스가 이루어질 때 콘텐츠 간의 시간정보 규격이 서로 상이하여 동기화에 어려움이 있다. 따라서 본 논문에서 이종망을 기반으로 한 하이브리드 방송 환경내의 다시점 콘텐츠의 동기화 방안을 제안하고자 한다.