• Journal of KIISE
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• v.45 no.2
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• pp.187-194
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• 2018

Recently, Hypertext Transfer Protocol (HTTP) adaptive streaming service is attracting attention. The existing quality adaptive scheme of HTTP adaptive streaming service adjusts the video quality according to the network bandwidth or the client buffer size. However, the problem with the existing quality adaptive scheme is the QoE (Quality of Experience) degradation caused by the unnecessary quality change that occurs due to frequent bandwidth change or fixed buffer threshold. We propose a video quality maintenance scheme that improves average video quality and minimizes unnecessary quality change in order to improve the QoE of HTTP adaptive streaming service in the changing network environment. The proposed scheme maintains high quality for a long time by setting the quality maintenance duration to be long when buffer occupancy and video quality are high. The experimental results show that the proposed scheme improves QoE by improving the average video quality and minimizing the quality change.

• Journal of Broadcast Engineering
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• v.16 no.5
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• pp.722-737
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• 2011

In this paper, we propose an estimation of network jitter that occurs when transmitting TCP packets containing MPEG-2 TS in progressive streaming service over wired or wireless Internet networks. Based on the estimated network jitter size, we can calculate required de-jitter buffering time to absorb the network jitter at the receiver side. For this purpose, by exploiting the PCR timestamp existing in the TS packet header, we create a new timestamp information that is marked in the optional field of TCP packet header to estimate the network jitter. By using the proposed de-jitter buffering scheme, it is possible to employ the conventional T-STD buffer model without any modification in the progressive streaming service over IP networks. The proposed method can be applicable to the recently developed international standard, MPEG DASH (dynamic adaptive streaming over HTTP) technology.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.3B
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• pp.144-151
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• 2008

In this paper, we propose a video streaming server that guarantees a certain level of quality when a server should serve video streaming service to multiple heterogenous clients simultaneously with TCP transport. If each heterogeneous client requests video streaming service in according to its own requirement such as bitrate of content and these requests are accepted by a server, then TCP flows for each video streaming session fairly share limited uplink bandwidth of the server. At this time, because TCP's bandwidth fair-share characteristics can result in bandwidth shrinkage of higher bitrate video streaming session, the client of higher bitrate video may suffer sluggish playback which is related with streaming QoS degradation. To tackle this problem, our proposed server system uses multiple TCP connections adaptively for each video streaming session depending on the anticipated status of the client playout buffer. Simulation results show that our proposed algorithm can successfully reduce the occurrence of playout buffer underrun and enhance streaming quality for whole video clients.

• Journal of KIISE
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• v.41 no.11
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• pp.974-981
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• 2014

These days, use of multimedia streaming service and demand of QoS (Quality of Service) improvement have been increased because of development of network. QoS of streaming service is influenced by a jitter, delay, throughput, and loss rate. For guaranteeing these factors which are influencing QoS, the role of transport layer is very important. But existing TCP which is a typical transport layer protocol increases the size of congestion window slowly and decreases the size of a congestion window drastically. These TCP characteristic have a problem which cannot guarantee the QoS of UHD multimedia streaming service. In this paper, we propose a router buffer based congestion control method for improving the QoS of UHD streaming services. Our proposed scheme applies congestion window growth rate differentially according to a degree of congestion for preventing an excess of available bandwidth and maintaining a high bandwidth occupied. Also, our proposed scheme can control the size of congestion window according to a change of delay. After comparing with other congestion control protocols in the jitter, throughput, and loss rate, we found that our proposed scheme can offer a service which is suitable for the UDH streaming service.

• Journal of KIISE
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• v.45 no.2
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• pp.175-186
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• 2018

Recently, HAS (HTTP Adaptive Streaming) has been the subject of much attention to improve the QoE (Quality of Experience). In a high latency network, HAS degrades the QoE due to the lost RTT cycle since it replies with a response of one segment to the request of one segment. The server-push based HAS schemes of downloading multiple segments in one request cause QoE degradation due to the buffer underflow. In this paper, we propose a VSSDS (Video Streaming Scheme based on Dynamic Server-push) scheme to improve the QoE in a high latency network. The proposed scheme adjust video quality by estimating available bandwidth and determine the number of segments to be downloaded for each segment request cycle. Through the simulation, the proposed scheme not only improves the average video bitrate but also alleviates the buffer underflow.

• Proceedings of the Korean Information Science Society Conference
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• 2001.04a
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• pp.247-249
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• 2001

최근 VOD(Video On Demand)나 AOD(Audio On demand) 서비스가 급증하면서 멀티미디어 스트리밍 기술에 대한 관심이 커지고 있다. 기존의 스트리밍 서비스는 정적으로 멀티미디어를 전송하므로 네트워크 환경의 대역폭(bandwidth) 변화에 대처하는데 한계가 있다. 이러한 문제점을 해결하기 위하여 본 논문에서는 실시간 스트리밍 서비스(RTSS; Real Time Streaming Service) 시스템을 제안한다. RTSS 시스템은 동화상의 끊김없는 전송보장을 위해 미디어 스케일링 방식을 사용하고, 대역폭 변화에 능동적으로 대처할 수 있도록 동적인 멀티미디어 전송을 이용한다. 기존의 스트리밍 서비스의 버퍼링 방식의 단점을 해결하기 위하여 본 논문에서는 프레임 조정방식과 화질 조정방식을 동시에 채택하여 적용함으로써 고화질이 보장되는 전송환경과 느린 회선에서도 끊김없이 안정적인 스트리밍 서비스를 제공받을수 있도록 한다.

• 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.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2018.10a
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• pp.103-104
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• 2018

치과의 모든 정보뿐만 아니라, 공급 사슬관리, 고객의 정보까지 포함하여 통합적으로 관리해야 하므로 특정한 기능을 수행하는 기술개발의 범위보다는 환자의 구강 정보의 원활한 처리를 위한 서버 구축 및 영상의 질 및 촬영의 용이함을 위한 실시간 스트리밍 기술이 필요하다. 영상정보 판독을 통해 치주염의 위험을 보고 환자들의 치과 방문을 유도하는 것이 치료 적기를 놓치지 않기 위해 반드시 필요하다. 이 논문에서는 구강 질환 예방 및 관리를 위한 스트리밍 기반 모바일 촬영 시스템을 제안한다. 특히 환자 자신이 구강 내의 이미지를 클라이언트 앱을 통해 환부를 직접 확인해 가며 조절하여 촬영가능하도록 스트리밍 기법을 제시하며, 이 과정에서 프레임의 버퍼링 및 딜레이 현상을 사용자의 개입없이 자동 조절하는 기능을 제안한다.

• Journal of the Korean Society for Precision Engineering
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• v.21 no.9
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• pp.56-62
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• 2004

Recently, necessity and usage of the teleoperation have been increased in various fields from industrial automation to home application. Also, the internet is considered as a strong candidate far the transmission media of signals. However, it has an irregular transmission time delay and causes critical problems such as instability and poor performance. This paper presents a practical internet teleoperation system with a streaming buffer system which makes a variable time delay fixed. Validity of the proposed system is demonstrated by implementing the embedded system on a HILS(hardware in the loop system) which models a two-wheel mobile robot.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2017.06a
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• pp.197-200
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• 2017

퍼지 논리를 기반으로 한 적응형 스트리밍 기법인 FDASH 적응 알고리즘은 빠르게 변하는 네트워크 상황에서 우수한 콘텐츠의 화질을 보장하면서 끊김 없는 서비스를 제공하는 특성을 보이지만 비디오의 화질이 자주 변하기 때문에 최고의 사용자 체감 품질 (QoE: Quality of Experience)을 제공하지 못 할 수도 있다. 본 논문에서는 제한된 버퍼 크기를 가지고 동일한 콘텐츠의 화질을 보장하면서도 비디오 화질의 변화 횟수를 줄여서 최적의 QoE를 제공할 수 있도록 하는 변환된 퍼지 논리 기반의 DASH 적응 알고리즘을 제안하고자 한다. 제안된 방식은 우선 퍼지 논리 제어부(FLC : Fuzzy Logic Controller)의 수정을 통하여 다음 세그먼트의 비트율에 대해 최적의 판단을 하도록 하였고, 세그먼트 비트율 필터링 모듈 (SBFM: Segment Bitrate Filtering Module)을 추가하여 비디오 화질의 변화 횟수가 최소화 될 수 있도록 하였으며, 스트리밍 서비스 시작 시 SBFM에 의해 일정시간 저화질의 비디오를 시청해야 하는 상황을 막기 위한 Start Mechanism을 추가하였고, 마지막으로 버퍼의 오버플로우를 방지하기 위해 Sleeping Mechanism을 추가하였다. NS-3를 이용한 네트워크 모의실험 결과를 통해 제안된 방식이 FDASH 방식에 비하여 제한된 버퍼크기 상황 하에서도 오버플로우가 발생하지 않으며 점대점(Point to Point) 상황에서는 거의 동일 화질 성능을 보이면서도 비디오 화질 변화 횟수를 50% 이상 줄일 수 있음과 일반 Wifi환경에서는 오히려 17.8%정도 더 뛰어난 비디오 화질 성능을 보이면서 비디오 화질변화 횟수 측면에서는 53.1%정도 줄일 수 있음을 보여준다.