• KSII Transactions on Internet and Information Systems (TIIS)
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• 제12권1호
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• pp.308-324
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• 2018

HTTP adaptive streaming (HAS) is a promising technology for delivering video content over the Internet. HAS-based video streaming solutions rely on bandwidth estimation to select the appropriate video bitrate. Video streaming solutions that consider network conditions provide users with seamless video playback. However, when multiple clients compete for a common bottleneck link, conventional bandwidth estimation schemes that consider only one client overestimate the network bandwidth due to the ON-OFF traffic pattern. The bandwidth overestimation can cause Quality of Experience (QoE) degradation, such as unnecessary changes in video quality, and unfairness of video quality. In this paper, we propose a client-side bandwidth estimation scheme to obtain a better QoE of HAS in the multiple-client environment. The proposed scheme differentiates the client buffer status according to the buffer occupancy, and then estimates the available network bandwidth based on the buffer status and segment throughput. We evaluate the performance of HAS implemented in the ns-3 network simulator. Simulation results show that compared with the conventional schemes, the proposed scheme can enhance the QoE.

• KSII Transactions on Internet and Information Systems (TIIS)
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• 제9권11호
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• pp.4585-4603
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• 2015

Video streaming services make up a large proportion of Internet traffic on both fixed and mobile access throughout the world. Adaptive streaming allows for dynamical adaptation of the bitrate with varying network conditions, to guarantee the best user experience. Adaptive bitrate algorithms face a significant challenge in correctly estimating the throughput as it varies widely over time. In this paper, we first evaluate the throughput estimation techniques and show that the method that we have used offers stable response to throughput fluctuations while maintaining a stable playback buffer. Then, we propose an adaptive bitrate scheme that intelligently selects the video bitrates based on the estimated throughput and buffer occupancy. We show that the proposed scheme improves viewing experience by achieving a high video rate without taking unnecessary risks and by minimizing the frequency of changes in the video quality. Furthermore, we show that it offers a stable response to short-term fluctuations and responds swiftly to large fluctuations. We evaluate our algorithm for both constant bitrate (CBR) and variable bitrate (VBR) video content by taking into account the segment sizes and show that it significantly improves the quality of video streaming.

• KSII Transactions on Internet and Information Systems (TIIS)
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• 제8권11호
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• pp.3775-3788
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• 2014

With the increasing number of the wireless and mobile networks, the way that people use the Internet has changed substantively. Wireless multimedia services, such as wireless video streaming, mobile video game, and mobile voice over IP, will become the main applications of the future wireless Internet. To accommodate the growing volume of wireless data traffic and multimedia services, cognitive radio (CR) and Information-Centric Network (ICN) have been proposed to maximize the utilization of wireless spectrum and improve the network performance. Although CR and ICN have high potential significance for the future wireless Internet, few studies have been conducted on collaborative operations of CR and ICN. Due to the lack of infrastructure support in multi-hop ad hoc CR networks, the problem is more challenging for video streaming services. In this paper, we propose a Cross-layer Video Streaming Mechanism (CLISM) for Cognitive Radio Ad Hoc Information Centric Networks (CRAH-ICNs). The CLISM included two distributed schemes which are designed for the forwarding nodes and receiving nodes in CRAH-ICNs. With the cross-layer approach, the CLISM is able to self-adapt the variation of the link conditions without the central network controller. Experimental results demonstrate that the proposed CLISM efficiently adjust video transmission policy under various network conditions.

• 한국통신학회논문지
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• 제32권12B호
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• pp.735-742
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• 2007

본 논문에서는 DiffServ 네트워크에서 비디오 스트리밍 서비스의 품질을 보장하기 위한 적용방법의 하나로서 채널변화에 적응적인 패킷 사전 마킹 알고리듬을 제시하였다. 전통적인 패킷 마커들이나 기존에 제안된 TMS(Two Marker System, ETBTCM)은 다수의 비디오 소스가 존재하는 네트워크 상황에서는 혼잡현상 발생 시 패킷내에 포함된 비디오 정보의 중요도에 대한 고려 없이 패킷이 드롭되는 문제점을 가지고 있다. 이러한 문제점을 해결하기 위해 본 논문에서는 적응적 사전마킹 알고리듬을 제안하였다. 제안 알고리듬에서는 네트워크의 채널상태와 GOP내에서의 전송 프레임의 중요도를 고려하여 확률적으로 패킷 드롭 우선순위를 결정하여 이를 서버 단에서 사전에 마킹 하게 된다. 이를 통하여 다수의 비디오 트래픽간의 경쟁이 발생한 상황에서도 화질 저하를 최소화 되도록 할 수 있다.

• 대한임베디드공학회논문지
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• 제10권4호
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• pp.241-255
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• 2015

Recent network environment has been rapidly evolved to cloud computing environment based on the development of the Internet technologies. Furthermore there is an increasing demand on mobile cloud computing due to explosive growth of smart devices and wide deployment of LTE-based cellular networks. Thus mobile Desktop-as-a-Service(DaaS) could be a pervasive service for nomadic users. In addition, video streaming traffic is currently more than two-thirds of mobile traffic and ever increasing. All such trends enable that video streaming in mobile DaaS could be an important concern for mobile cloud computing. It should be noted that the performance of the Transmission Control Protocol(TCP) on cloud host servers greatly affects Quality of Service(QoS) of video streams for mobile users. With widely deployed Linux server platforms for cloud computing, in this paper, we conduct an experimental analysis of the twelve Linux TCP variants in mobile DaaS environments. The results of our experiments show that the TCP Illinois outperforms the other TCP variants in terms of wide range of packet loss rate and propagation delay over LTE-based wireless links between cloud servers and mobile devices, even though TCP CUBIC is usually used in default in the current Linux systems.

• 대한전자공학회논문지TC
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• 제41권1호
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• pp.23-34
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• 2004

본 논문에서는 초고속 패킷 스위치 네트워크에서 VoD나 HDTV, VoIP같은 고품질 스트리밍 서비스의 QoS를 보장하는 패킷 스케줄러의 구조 및 제어방법을 제시한다. 스트리밍 서비스는 버스트 데이터 응용서비스보다 더욱 엄격한 QoS(jitter, delay, packet loss)보장을 요구한다 또한 스트리밍 서비스는 다른 플로우들의 동작에 상관없이 끊김 없는 서비스를 제공하기 위해 각 플로우별로 최소 대역 보장과 종단간 지연조건을 보장해야 한다. 이들 요구조건들을 만족하기 위해, 패킷 스케줄러는 플로우들이 다른 플로우의 영향을 받지 않도록 분리하고, 각 플로우들에게 종단간 지연 보장을 제공해야 한다. 그리고 각 플로우들에게 요구되는 최소 대역폭을 할당해야한다 지금까지 많은 벤더들이 10Gbps급 트래픽 관리기 칩을 개발하였지만 대부분 칩들은 고품질 스트리밍 서비스를 지원하지 못하는 단점이 있다. 따라서 본 논문에서는 상용 TM칩들의 단점 및 스트리밍서비스의 트래픽 특성을 조사하고, 제안한 패킷 스케줄러의 하드웨어 구조를 제시한다. 그리고 마지막으로 제안한 스케줄러의 시물레이션 결과를 분석하였다.

• 한국통신학회논문지
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• 제31권7B호
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• pp.608-618
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• 2006

멀티미디어 스트리밍 서비스의 증가는 인터넷 콘텐츠의 새로운 국면으로 나타나고 있다. 특히, 무선이동통신망에서 증가하는 멀티미디어 응용에 대한 QoS 제공은 무엇보다 중요하다. 서비스 제공자는 클라이언트 가까이에 있는 프락시에서 자주 억세스되어지는 멀티미디어 스트림의 초기 세그먼트를 캐슁함으로써 성능을 향상 시킬 수 있다. 프락시는 서버로부터 스트림의 나머지 부분을 요구함과 동시에 클라이언트에 전송을 시작할 수 있다. 본 논문에서는 IETF의 RTSP 환경에서 프리픽스 캐슁 서비스를 무선망에 적용시키고, 무선 상황이나 핸드오프 시에 네트 워크 상황에 적응적으로 대응하고 단절현상을 줄일 수 있는 효과적인 RTSP 핸들링 기법을 제안 한다. 또한 캐슁 프락시의 성능을 향상시키기 위해 트래픽 기반 캐슁 기법(TSLRU)을 제안한다, TSLRU는 트래픽을 세 종류로 분류하여 캐슁하며, 교체 대상 결정시 여러 요소(traffic types, recency, frequency, object size)를 반영함으로써 캐슁 프락시의 성능을 향상 시킨다. 모의실험에서 캐슁 알고리즘은 byte hit Rate와 startup latency에서 높은 성능을 보였으며, 제안한 RTSP 핸들링 기법 역시 throughput에서 좋은 성능을 보였다.

• 한국방송∙미디어공학회:학술대회논문집
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• 한국방송공학회 2009년도 IWAIT
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• pp.49-54
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• 2009

Synchronization between media is an important aspect in the design of multimedia streaming system. This paper proposes a precise media synchronization mechanism for digital video and audio transport over IP networks. To support synchronization between video and audio bitstreams transported over IP networks, RTP/RTCP protocol suite is usually employed. To provide a precise mechanism for media synchronization between video and audio, we suggest an efficient media synchronization algorithm based on NPT (Normal Play Time) which can be derivable from the timestamp information in the header part of RTP packet generated for the transport of video and audio streams. With the proposed method, we do not need to send and process any RTCP SR (sender report) packet which is required for conventional media synchronization scheme, and accordingly could reduce the number of required UDP ports and the amount of control traffic injected into the network.

• International Journal of Computer Science & Network Security
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• 제23권3호
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• pp.81-93
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• 2023

The popularity of short-form video platforms like TikTok has increased recently. Short-form videos are significantly shorter than traditional videos, and viewers regularly switch between different types of content to watch. Therefore, a successful prefetching strategy is essential for this novel type of video. This study provides a resource-effective prefetching technique for streaming short-form videos. The suggested solution dynamically adjusts the quantity of prefetched video data based on user viewing habits and network traffic conditions. The results of the experiments demonstrate that, in comparison to baseline approaches, our method may reduce data waste by 21% to 83%, start-up latency by 50% to 99%, and the total time of Re-buffering by 90% to 99%.

• 한국멀티미디어학회논문지
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• 제19권7호
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• pp.1166-1178
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• 2016

In Internet of Things (IoT) environments, billions of interconnected devices and multimedia sensors generate a huge amount of multimedia traffic. Since the environment are in general deployed as a server-centric architecture wireless sensor networks could be bottlenecks between IoT gateways and IoT devices. The bottleneck causes high power consumption of the device and triggers very heavy network overload by transmission of sensing data. The deterioration could decrease the quality of multimedia streaming service due to delay, loss, and waste of device power. Thus, in this paper, we propose a context-based adaptive multimedia streaming scheme to support enhanced QoS and low power consumption in IoT environments. The goal of the scheme is to increase quality score per voltage of the streaming service, given an adaptation algorithm with context that are classified network and hardware such as throughput, RTT, and CPU usage. From the both context, the quality score per voltage is used in the comparison of a only network context-based adaptive multimedia streaming scheme, a fixed multimedia streaming and our scheme. As a result, we achieves a high improvement that means the quality score per voltage is increased up to about 4, especially in case of resolution change.