• Journal of KIISE
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• v.44 no.5
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• pp.545-552
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• 2017

Because of the explosive growth of mobile devices and development of network technologies, HTTP (Hypertext Transfer Protocol) adaptive streaming has become a new trend in video delivery to provide efficient multimedia streaming services. As interest in personalized broadcasting grows, the study of interactive multimedia has been actively pursued. The interactive multimedia service is a method of playing media according to a scenario selected by the user. Providing the interactive multimedia service with the existing HTTP adaptive streaming causes switching delay and buffer underflow according to the point in time when the user selects the scenario while the client streams the interactive multimedia and therefore decreases the user QoE (Quality of Experience). In this paper, we propose an adaptive streaming scheme for interactive multimedia service in HTTP adaptive streaming to provide seamless playback. We design the architecture and prefetching scheme for interactive multimedia streaming.

• Journal of KIISE:Information Networking
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• v.34 no.4
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• pp.276-286
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• 2007

Existing streaming mechanisms have no consideration for the characteristics of streaming applications because they only consider network stability. In this paper, in order to overcome limitations of the previous work on video streaming, we propose a new video streaming mechanism called "HAViS(Hybrid Approach for Video Streaming)". The proposed mechanism includes more sophisticated features that consider both network and user requirements. Therefore, the HAViS mechanism improves the network stability by adjusting the sending rate of video stream based on the network state and it also provides the smooth playback by preventing the buffer underflow or overflow. Moreover, it is designed to take into consideration the streaming video content. Through the simulation, we prove that the HAViS mechanism efficiently uses the buffer resources and provides improved network stability and smooth playback.

• Journal of Broadcast Engineering
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• v.16 no.1
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• pp.155-173
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• 2011

Recently, with the digitalization of a broadcasting system and the development of a communication technology, the existing trend of consuming media contents throughout the fixed-displayer and the dedicated channel is being changed. The existing user only could consume media contents under limited time and places because of the fixed-displayer and the dedicated channel. However, with advent of the IP-based terminals and HTTP adaptive streaming which transfer the media sequence according to the user's transmission condition, users become possible to enjoy the media content anytime anywhere. As the result of the alteration of the broadcasting surrounding, users can enjoy the media content while changing his terminals according to their preferences and circumstances. However, in case that users try to consume consecutively the content from last view-point ended in the previous terminal under current HTTP adaptive streaming environment, a user has to remember the last view-point, and then has to apply the view-point to the changed terminal. Thus, for solving this problem, this paper defines "User Profile" for describing the metadata for the chained content consume environment between the terminals. Also, for proving the proposed method, this paper try to demonstrate the proposed method throughout the realization of the system based on Smooth Streaming from Microsoft.

• Journal of Broadcast Engineering
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• v.24 no.6
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• pp.948-955
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• 2019

As interests in 360-degree VR (Virtual Reality) video services enormously grow, compression and streaming technologies for VR video data have been rapidly developed. Quality Emphasized Region (QER) based streaming scheme has been developed as a kind of viewport-adaptive 360-degree video streaming technology for maintaining immersive experience and reducing bandwidth waste. For selecting a QER corresponding to the user's gaze coordinate, QER-based streaming scheme requires the calculation of Quality Emphasis Center (QEC) distance and signaling message delivery for requesting QER switching. QEC distance calculations require high computational complexity because of repeated calculations as many times as the number of QERs. Furthermore, the signaling message interval results in a trade-off relationship between efficient bandwidth usage and flexible QER switching. In this paper, we propose an improved QER selection algorithm based on MMT protocol to solve this problem. The proposed algorithm could achieve computational complexity reduction by using preprocessed QER_ID_MAP. Also, flexible QER switching could be achieved, as well as efficient bandwidth utilization by an adaptive adjustment of the signaling interval.

• The KIPS Transactions:PartA
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• v.12A no.3 s.93
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• pp.197-204
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• 2005

In this paper, we consider the transmission or variable bit rate (VBR) stored video for the distributed video streaming service over Internet. In streaming service, users often suffer from the discontinuity in playback due to the decrease in bandwidth during transmission according to bandwidth renegotiation protocol. We propose a novel transmission technique to overcome this problem for stored variable bit rate video. This scheme uses a priori information of stored VBR video to continue streaming without playback discontinuity. In addition, an approximation scheme for the buffer-bandwidth relation is proposed in order to facilitate the admission control under the proposed scheme.

• Journal of Communications and Networks
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• v.16 no.3
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• pp.344-354
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• 2014

Transmission control protocol friendly rate control (TFRC) is designed to mainly provide optimal service for unicast applications, such as multimedia streaming in the best-effort Internet environment. However, high bandwidth networks with large delays present an environment where TFRC may have a problem in utilizing the full bandwidth. TFRC inherits the slow-start mechanism of TCP Reno, but this is a time-consuming process that may require many round-trip-times (RTTs), until an appropriate sending rate is reached. Another disadvantage inherited from TCP Reno is the RTT-unfairness problem, which severely affects the performance of long-RTT flows. In this paper, we suggest enhanced TFRC for high quality video streaming over high bandwidth delay product networks. First, we propose a fast startup scheme that increases the data rate more aggressively than the slow-start, while mitigating the overshooting problem. Second, we propose a bandwidth estimation method to achieve more equitable bandwidth allocations among streaming flows that compete for the same narrow link with different RTTs. Finally, we improve the responsiveness of TFRC in the presence of severe congestion. Simulation results have shown that our proposal can achieve a fast startup and provide fairness with competing flows compared to the original TFRC.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.10B
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• pp.947-956
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• 2003

The current IP Routing Protocolspacket networks also need to provide the network QoS based of DiffServ, RSVP, MPLStraffic model which is standardized as IETF reference model for NGN. The first topic of this paper is to propose Traffic-Balanced Routing Protocol(TBRP) to process existing best effort traffic. TBRP will process low priority interactive data and background data which is not sensitive to dealy. Secondly Hierarchical Traffic-Traffic-Scheduling Routing Protocol(HTSRP) is also proposed. HTSRP is the hierarchical routing algorithm for backbone and access networkin case of fixed-wireless convergence network. Finally, HTSRP_Q is proposed to meet the QoS requirement when user want interactive or streaming packet service. This protocol will maximize the usage of resources of access layer based on the QoS parameters and process delay-sensitive traffic. Service classes are categorized into 5 types by the user request, such as conversational, streaming, high priority interactive, low priority interactive, and background class. It could be processed efficiently by the routing protocolstraffic model proposed in this paper. The proposed routing protocolstraffic model provides the increase of efficiency and stability of the next generation network thanks to the routing according to the characteristic of the specialized service categories.

• Proceedings of the Korea Information Processing Society Conference
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• 2014.11a
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• pp.160-163
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• 2014

IP 네트워크를 통해 전송되는 실시간 스트라밍은 네트워크 상태에 따라 데이터 손실 및 전송 지연의 문제가 발생 할 수 있다. 이에 멀티미디어 네트워킹 애플리케이션은 RTP(Realtime Transport Protocol)와 RTCP(RTP Control Protocol)를 지원하는 RTSP(Realtime Streaming Protocol)를 사용한다. RTSP를 사용하는 송신단은 패킷 손실과 전송지연이 발생할 경우 끊김 없는 멀티미디어 스티리밍을 제공하기 위해 네트워크 상태에 따라 전송률을 변경한다. 하지만 전송률이 낮아지면 멀티미디어 품질이 떨어지게 된다. 만약 짧은 시간 동안 전송률이 빈번하게 변경된다면 일관성 있는 품질의 서비스를 제공하기 어렵다. 본 논문에서는 SDN 환경에서 실시간 스트리밍을 효과적으로 제어하기 위해 중앙 컨트롤러가 RTCP를 수신하고 송신단의 전송률과 현재 링크의 가용 전송율을 추정한다. 추정된 가용 전송율이 송신단의 전송률보다 낮다며 각 스위치의 활성화된 플로우 엔트리가 낮은 스위치로 라우팅 경로를 선택하는 메커니즘을 제안한다.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.26 no.7
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• pp.1091-1097
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• 2022

In this paper, we implement the eXclusive OR-based Cast (XC) system that is a video streaming system using exclusive OR operations, and measure various performance metrics in wireless local area network (WLAN) environments. In addition, we investigate the performance improvement of the XC system considering various practical video streaming environments, while conventional studies analyzed the performance of XC through computer simulations in limited environments. To this end, we propose new control messages such as STR_REQ_MSG (SRM) that clients transmit to a video streaming server and STR_CON_MSG (SCM) that is used for the video streaming server to control the clients, and develop a new protocol by using the new control messages. According to the various measurement results using the implemented XC system, XC video streaming system can reduce the consumption of network bandwidth by 8.6% on average and up to 25% compared to the conventional video streaming system. In addition, the outage probability can be also reduced up to 76%.

• KIISE Transactions on Computing Practices
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• v.21 no.12
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• pp.780-785
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• 2015

Recently HTTP (Hypertext Transfer Protocol) adaptive streaming services have been the subject of much attention. The video quality control scheme of conventional HTTP adaptive streaming services estimates bandwidth using segment throughput and smooths out the sample of segment throughput. However, the conventional scheme has the problem of QoE (Quality of experience) degradation occurring with buffer underflow and frequent quality change due to the fixed number of samples. In order to solve this problem, we propose a video quality control scheme based on segment throughput and buffer occupancy. The proposed scheme determines the number of samples according to the variation of segment throughput. The proposed scheme also controls video quality based on the threshold of bitrate to keep stable buffer occupancy. The simulation results show that proposed scheme improves QoE by preventing buffer underflow and decreasing quality change when compared with the conventional scheme.