• Proceedings of the Korean Information Science Society Conference
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• 2007.06d
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• pp.315-320
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• 2007

본 논문에서는 동시 접속자 수의 제약과 서버 증설에 의한 고비용 문제를 효과적으로 해결할 수 있는 실시간 웹캐스팅 시스템이 설계 구현된다. 기존의 시스템과는 다르게, 다양한 해상도를 지원할 수 있는 IP 카메라로부터 실시간으로 스트리밍 되는 동영상 데이터를 콘텐츠로 하는 라이브 웹캐스팅과, 저장 공간에 저장된 후, 사용자 요구에 따라 파일 데이터를 콘텐츠로 스트리밍 하는 주문형 비디오(VOD) 서비스 모두를 포함하는 total 시스템이다. 압축률이 높은 마이크로소프트사의 WMV(혹은 VC-1) 형식의 동영상 데이터를 사용함으로써 사용 대역폭을 줄여, 많은 Tuner들을 수용할 수 있도록 하였으며, 각 Tuner는 미디어플레이어를 뷰어로써 사용할 수 있도록 하여 시청자들이 별도의 소프트웨어를 설치하지 않도록 하였다. 가능한 모든 시스템 구성 요소들 또한 마이크로소프트에서 제공되는 유틸리티들을 사용할 수 있도록 하여 기존 시스템과의 호환성을 최대화 시켰으며, 특히, Caster, Distributor, 그리고 Tuner, 3 부분으로 나누어, Caster의 규모, 혹은 Tuner의 규모에 따라 Distributor의 확장이 모듈 단위로 가능하도록 하였다.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2015.10a
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• pp.791-793
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• 2015

Recently, UCC (User Created Contents) and VoD (Video on Demand), and multimedia content are growing, IP-TV, Smart TV, OHTV (Open Hybrid TV) various services such as multi platform (Multi-platform) environment, services and QoS issues. To solve this problem, the network efficiently, and improve the quality of content is necessary for the system. In this paper, the network of channels State and transmission of multimedia data based on dynamic resource usage, TCP and UDP, Adaptive dual-streaming system used for design and analysis. In addition, the existing TCP and UDP streaming system using a single protocol for analysis and verification of the effectiveness of the difference between and. This is a disaster, and medical/first aid system will be utilized in the field of feed, are ubiquitous.

• Journal of KIISE:Computer Systems and Theory
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• v.35 no.9_10
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• pp.466-475
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• 2008

Owing to the improved wireless communication technologies, it is possible to provide streaming service of multimedia with PDAs and mobile phones in addition to desktop PCs. Since mobile client devices have low computing power and low network bandwidth due to wireless network, the transcoding technology to adapt media for mobile client devices considering their characteristics is necessary. Transcoding servers transcode the source media to the target media within corresponding grades and provide QoS in real-time. In particular, an effective load balancing policy for transcoding servers is inevitable to support QoS for large scale mobile users. In this paper, the transcoding load estimation algorithm is proposed for load balance on the distributed transcoding environments. The proposed algorithm estimates transcoding time from transcoding server information, movie information and target transcoding bit-rate. The estimated transcoding time is proved based on experiments.

• Journal of the Korean Institute of Intelligent Systems
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• v.14 no.3
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• pp.285-289
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• 2004

Recently, due to the high development of the internet, needs for multimedia streams such as digital audio and video is increasing much more. In case of transmitting multimedia streams using the User Datagram Protocol (UDP), it may cause starvation of TCP traffic on the same transmission path, thus resulting in congestion collapse and enormous delay because UDP does not perform TCP-like congestion control. Because of this problem, diverse researches are being conducted on new transmission schemes and protocols intended to efficiently reduce the transmission delay of real-time multimedia streams and perform congestion control. The TCP-friendly congestion control schemes can be classified into the window-based congestion control, which uses the general congestion window management function, and the rate-based congestion control, which dynamically adjusts transmission rate by using TCP modeling equations and the like. In this paper, we suggest the square-root congestion avoidance algorithm with the hybrid TCP-friendly congestion control scheme which the window-based and rate-based congestion controls are dealt with in a combined way. We apply the proposed algorithm to the existing TEAR. We simulate the performance of the proposed TEAR by using NS, and the result shows that it gives better improvement in the stability needed for providing congestion control than the existing TEAR.

• Journal of KIISE:Computing Practices and Letters
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• v.15 no.9
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• pp.626-636
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• 2009

In Mobile IP-enabled wireless LAN (WLAN), packet flows are corrupted due to the handoff of a mobile node (MN) at the link and network layers, which results in burst packet losses and can cause temporary buffer underflow in a streaming client at the MN. This transient behavior hurts time-sensitive streaming media applications severely. Among many suggestions to address this handoff problem, few studies are concerned with empirical issues regarding the practical validation of handoff options on the time-sensitive streaming media applications. In this paper, targeting seamless streaming over Mobile IP-enabled WLAN, we introduce a seamless media streaming framework that estimates accurate pre-buffering level to compensate the handoff latency. In addition, we propose a link-layer (L2) assisted seamless media streaming system as a preliminary version of this framework. The proposed system is designed to reduce the handoff latency and to overcome the playback disruption from an implementation viewpoint. A packet buffering and forwarding mechanism with L2 trigger is implemented to reduce the handoff latency and to eliminate burst packet losses generated during the handoff. A pre-buffering adjustment is also performed to compensate the handoff latency. The experimental results show that the proposed approach eliminates packet losses during the handoff and thus verify the feasibility of seamless media streaming over Mobile IP-enabled WLAN.

• Proceedings of the Korean Information Science Society Conference
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• 2004.04b
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• pp.535-537
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• 2004

본 논문은 유비쿼터스 환경에서 미디어 서비스를 이용하는 사용자가 위치 이동에 따라 서비스 단말의 전환을 요구하는 경우 005 저하 없이 유연하게 서비스하기 위한 에이전트 기반의 스위칭 서비스 시스템 모델을 제안한다. 사용자는 미디어 서비스를 받기 위하여 기존의 이동단말기 분만 아니라 일반 데스크탑, 그리고 셋탑박스 등의 다양한 형태의 단말환경을 유기적으로 연결하여 사용할 수 있다. 하나의 단말환경에서 제공되던 미디어 서비스는 에이전트에 의해 다른 단말환경에서 사용자의 조작 없이 미디어를 연속하여 제공받을 수 있으며, 사용자는 이러한 서비스를 별도의 설정 없이 이용할 수 있다 논문에서 제안된 시스템은 사용자의 이동에 따라 각기 다른 단말장치로 미디어가 제공되기 위한 세션 관리 센서와 적응형 서비스를 제공할 수 있는 실시간 스트리밍 서버, 단말에서 정보를 교환하기 위한 에이전트로 구성된다. 사용자의 이동 상황에 따라 미디어 서비스를 받기 위한 에이전트 스위칭 서비스 모델을 제안하고 실제 시스템에 적용하기 위한 모델을 구현한다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.10A
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• pp.886-895
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• 2005

As the networked media technology have been grown in recent, there have been many research works to deliver high-quality video such as HDV and HDTV over the Internet. To realize high-quality media service over the Internet, however, the network adaptive streaming scheme is required to adopt to the dynamic fluctuation of underlying networks. In this paper, we design and implement the network-adaptive HD(high definition) MPEG-2 streaming system employing the frame-based prioritized packetization. Delivered video is inputted from the JVC HDV camera to the streaming sewer in real-time. It has a bit-rate of 19.2 Mbps and is multiplexed to the MPEG-2 TS (MPEG-2 MP@HL). For the monitoring of network status, the packet loss rate and the average jitter are measured by using parsing of RTP packet header in the streaming client and they are sent to the streaming server periodically The network adaptation manager in the streaming server estimates the current network status from feedback packets and adaptively adjusts the sending rate by frame dropping. For this, we propose the real-time parsing and the frame-based prioritized packetization of the TS packet. The proposed system is implemented in software and evaluated over the LAN testbed. The experimental results show that the proposed system can enhance the end-to-end QoS of HD video streaming over the best-effort network.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2014.05a
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• pp.571-573
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• 2014

In today's technology streaming service's QoS technologiey is an issue. On quality video streaming service there are some technical issues exist, such as buffering. This submission is "Adaptive dual-streaming system design" which is for the integrity of the data streaming that is sent to TCP and UDP for faster transmission of data to the stream. This system provides real-time incoming video encoding in bitrate of h.264-based H through a process based on the video footage of several server and client-to-TCP and UDP via Adaptive providing streaming services in a network environment. This is an unspecified number of buffers in a network environment and continued through the minimization of various streaming for playback of videos and multimedia will be utilized in the field.

• Journal of KIISE:Information Networking
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• v.36 no.6
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• pp.492-504
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• 2009

The delivery of multimedia that efficiently adapts its bit-rate to changing network characteristics and conditions is one of the important challenging tasks in the design of today's real-time multimedia streaming systems such as IPTV, Mobile IPTV and so on. In these work, the primary focus is on network congestion, to improve network stability and inter-protocol fairness. However, these existing works have problems which do not support QoE (Quality of Experience), because they did not consider essential characteristics of contents playback such as the media continuity. In this paper, we propose a novel rate control scheme for improving the QoE of multimedia streaming service in the Internet congestion, called NCAR (Network and Client-Aware Rate control), which is based on network-aware congestion control and client-aware flow control scheme. Network-aware congestion control of the NCAR offers an improving reliability and fairness of multimedia streaming, and reduces the rate oscillation as well as keeping high link utilization. Client-aware flow control of NCAR offers a removing the media discontinuity and a suitable receiver buffer allocation, and provides a good combination of low playback delay. Simulation results demonstrate the effectiveness of our proposed schemes.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2020.11a
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• pp.190-191
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• 2020

본 논문은 모델 분할 기법과 중복성 제거 기법을 통한 대용량 3차원 메쉬 모델의 고속 압축 기술에 관한 내용이다. 대용량 3차원 메쉬 모델의 비실시간 압축은 실시간 스트리밍 응용 시나리오에서 제약점으로 작용하고 있고, 본 논문에서는 인코딩 시간을 줄이기 위해 경량 메쉬 분할 방법을 통해 대용량 메쉬를 여러 개의 작은 메쉬로 분할하고, 각각의 분할된 메쉬를 병렬적으로 인코딩하여 처리 속도를 개선하였다. 또한, 메쉬 모델 내의 같은 기하학적 정보를 가진 중복된 정점들이 존재할 수 있으며, 중복된 정보를 제거하고 제거된 정점과 삼각형 표면 간의 연결 정보를 갱신하는 과정을 통해 메쉬 모델의 기하학적 정보를 유지하면서 압축 성능을 확보하였다.