• KSII Transactions on Internet and Information Systems (TIIS)
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• v.9 no.11
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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.

• Proceedings of the Korea Information Processing Society Conference
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• 2002.11b
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• pp.1415-1418
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• 2002

최근 정보 통신 기술의 발달에 따라 음성 서비스 등과 같은 단일 서비스를 제공하던 무선 통신망이 음성, 데이터, 고화질 정지화면, 동영상 및 다양한 정보형태의 멀티미디어 서비스를 추구하고 있다. 이처럼 다양한 종류의 멀티미디어 트래픽을 지원하기 위해서는 매체 접근 제어(MAC) 프로토콜이 필요하다. 본 논문에서는 서로 다른 종류의 트래픽이 혼재된 무선 ATM 환경에서 폭주제어 정보를 가지고 각 트래픽 클래스의 QoS를 만족할 수 있는 동적 자원 할당 방식 기반의 MAC 프로토콜을 제안한다. 각 트래픽 클래스의 데이터 전송 스케줄링을 제안함으로서 CBR이나 VBR의 실시간 데이터의 손실을 최소화하고 네트워크 상태에 따른 ABR 트래픽 클래스에 최소의 고정적인 대역폭을 할당함으로서 실시간 비실시간 데이터의 비효율성 문제 해결 방안을 제시한다. 그리고 각각의 트래픽 모델에 대한 허용 테스트 식을 제공함으로서 다른 실시간 또는 비실시간 채널에 영향을 수지 않고 새로운 채널을 설정한 수 있는 방안을 제시한다.

• Proceedings of the Korea Information Processing Society Conference
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• 2000.10b
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• pp.1609-1612
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• 2000

무선망에서 기존의 회선 서비스뿐만 아니라 패킷 서비스 및 영상 서비스 등을 수용하는 데 있어서 고려되어야 할 것들 중 하나는 무선 자원의 활용이다. 특히 패킷 서비스의 경우 가변적인 패킷의 길이와 양으로 효율적인 무선 자원관리를 어렵게 하는 요소를 지니고 있다. 즉, 무선 자원을 효율적으로 운용하기 위해서는 동적인 무선 자원할당 메커니즘이 중요한 역할을 한다. UMTS 표준 문서에서는 계층 2 전송 프로토콜인 RLC (Radio Link Control) 버퍼의 양만을 측정하여 무선 자원 할당이 제어가 되도록 권고하고 있다. 하지만 트래픽 양의 변화가 큰 VBR 트래픽의 경우, 이와 같은 방법은 갑자기 증가하거나 감소하는 트래픽 양에 적응하기 힘들고 이는 무선 자원의 효율성을 저하시키며, QoS 제공도 어렵다. 본 논문에서는 먼저 트래픽 특성을 알 수 있는 토큰 버켓 모델을 RLC에 적용하고, 동적 무선 자원 할당을 위해 RLC 버퍼의 양뿐만 아니라 예상되는 유입 트래픽 양을 측정 및 보고하는 메커니즘을 제안한다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.9A
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• pp.837-843
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• 2006

IEEE802.15.3 and IEEE802.15.4 have defined the hybrid MAC protocols based on TDMA and CSMA where a multi-frame TDMA structure is employed so that multiple data frames can be transmitted within one timeslot to guarantee minimum delay bounds of isochroous traffic. However, TDMA has an intrinsic problem that cannot dynamically allocate optimal length of timeslot to each station. Therefore the idle timeslot can be produced by stations when each transmission queue is instantaneously empty during its timeslot, which would waste lots of timeslots especially in the multi-frame TDMA systems. In this paper, we propose a more flexible multiple-access scheme for the multi-frame TDMA system based on the concept of virtual slot which is accessible by every station with the highest priority for slot owner and lower priority for other stations. Finally, our simulation results from various environments show that proposed scheme can achieve magnitude improvement of total system throughput and average message delay by maximizing channel utilization.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.23 no.8
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• pp.2042-2051
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• 1998

VBR compressed, pre-recorded video is known to exhibit significant, multiple time-scale bit rate variability. To reduce the variability and burstiness of the aggregated transmission rate as low as possible, in this paper, we present an adaptive bandwidth smoothing algorithm, that can be effectively applicalble for VoD services over a shared-medium channel. For these environments where many clients are connected to a single server, by introducing the conventional MVBA(minimum variability bandwidth allocation) algorithm and controlling adaptively the conventional MVBA(minimum variability bandwidth asllocation) algorithm and controlling adaptively the aggregated transmission rate whenever a new clients request is arrived at the server side, the proposed algorithm effectively reduces the burstiness and variability of the aggregated transmission rate. Through computer experiments, it is shown that the proposed method perporms better than the convertional non-aggregated bandwidth smoothing schemes in terms of the peak rate, standard deviation, number of rate changes for the aggregated traffic.

• The Transactions of the Korea Information Processing Society
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• v.4 no.7
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• pp.1770-1780
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• 1997

Our problem is improvement of image quality because it is inevitable cell loss of image data when traffic congestion occurs. If cells are discarded indiscriminately in transmission of MPEG video data, it occurs severe degradation in quality of service(QOS). In this paper, to solve this problem, we propose two method. The first, we analyze the traffic characteristics of an MPEG encoder and generate high priority and low priority data stream. During network congestion, only the least low priority cells are dropped, and this ensures that the high priority cells are successfully transmitted, which, in turn, guarantees satisfactory QoS. In this case, the prioritization scheme for the encoder assigns components of the data stream to each priority level based on the value of a parameter ${\beta}$. The second, Number of high priority cells are increased when value of ${\beta}$ is large. It occurs the loss of high priority cell in the congestion. To prevent it, this paper is regulated to data stream rate as buffer occupancy with UPC controller. Therefore, encoder's bandwidth can be calculated renegotiation of the encoder and networks. In this paper, the encoder's bandwidth requirements are characterized by a usage parameter control (UPC) set consisting of peak rate, burstness, and sustained rate. An adaptive encoder rate control algorithm at the Networks Interface Card(NIC) computes the necessary UPC parameter to maintain the user specified quality of service. Simulation results are given for a rate-controlled VBR video encoder operating through an ATM network interface which supports dynamic UPC. These results show that dynamic bandwidth renegotiation of prioritized data stream could provided bandwidth saving and significant quality gains which guarantee high priority data stream.