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Energy-efficient Video Streaming System with TCP/UDP over Heterogeneous Wireless Networks
Yunmin Go, Hyunmin Noh, Jeung Won Choi, Kyungwoo Kim, Kihun Kim, Jongman Lee, Hyun Park, Hwangjun Song
http://doi.org/10.5626/JOK.2020.47.11.1092
In this paper, we propose a video streaming system with TCP/UDP over heterogeneous wireless networks. Unlike the existing HTTP adaptive streaming system, the proposed system can use both TCP and UDP to overcome the TCP"s performance degradation problem and improve energy efficiency. The proposed system determines the type of transport protocol, the video quality, and the amount of data received through each wireless network according to the network conditions, buffer occupancy, and energy efficiency of the mobile device. When UDP is selected as the transport protocol, the proposed system employs raptor code to provide more reliable data transmission. And raptor parameters, such as symbol size, the amount of redundant data, are also determined to provide energy-efficient transmission. The proposed system is implemented in an android device and verified in real WiFi and LTE networks. Also, it is verified that the proposed system can provide high energy-efficient and high-quality streaming services.
Quality Adaptation Scheme Based on Stability to Improve QoE of HTTP Adaptive Streaming in Wireless Networks
Minsu Kim, Heekwang Kim, Kwangsue Chung
http://doi.org/10.5626/JOK.2019.46.3.268
With the recent development of networks, the demand for video streaming has also been increasing, and more attention has been focused on HTTP adaptive streaming. HTTP adaptive streaming can guarantee QoE (Quality of Experience) because the client is provided with the options of selecting the quality adaptively according to the network state. However, in wireless networks, bandwidth is inaccurately measured due to high delay and packet loss. The existing quality adaptation schemes degrade the QoE because these schemes select quality using the measured bandwidth. In this paper, a quality adaptation scheme based on stability is proposed to improve QoE of HTTP adaptive streaming. The proposed scheme calculates the buffer underflow probability and the instability using the changes in the video quality and buffer and selects the quality according to the defined quality adaptation region. Experimental results established that the proposed scheme improves QoE as compared to the existing schemes by high average video quality and fewer quality changes.
Improved Real-time Transmission Scheme using Temporal Gain in Wireless Sensor Networks
Taehun Yang, Hyunchong Cho, Sangdae Kim, Cheonyong Kim, Sang-Ha Kim
http://doi.org/10.5626/JOK.2017.44.10.1062
Real-time transmission studies in wireless sensor networks propose a mechanism that exploits a node that has a higher delivery speed than the desired delivery speed in order to satisfy real-time requirement. The desired delivery speed cannot guarantee real-time transmission in a congested area in which none of the nodes satisfy the requirement in one hop because the desired delivery speed is fixed until the packet reaches the sink. The feature of this mechanism means that the packet delivery speed increases more than the desired delivery speed as the packet approaches closer to the sink node. That is, the packet can reach the sink node earlier than the desired time. This paper proposes an improved real-time transmission by controlling the delivery speed using the temporal gain which occurs on the packet delivery process. Using the received data from a previous node, a sending node calculates the speed to select the next delivery node. The node then sends a packet to a node that has a higher delivery speed than the recalculated speed. Simulation results show that the proposed mechanism in terms of the real-time transmission success ratio is superior to the existing mechanisms.
Jitter-based Rate Control Scheme for Seamless HTTP Adaptive Streaming in Wireless Networks
Yunho Kim, Jiwoo Park, Kwangsue Chung
http://doi.org/10.5626/JOK.2017.44.6.628
HTTP adaptive streaming is a technique that improves the quality of experience by storing various quality videos on the server and requesting files of the appropriate quality based on network bandwidth. However, it is difficult to measure the actual bandwidth in wireless networks with frequent bandwidth changes and high loss rate. Frequent quality changes and playback interruptions due to bandwidth measurement errors degrade the quality of experience. We propose a technique to estimate the available bandwidth by measuring the jitter, which is the derivation of delay, on a packet basis and assigning a weight according to jitter. The proposed scheme reduces the number of quality changes and mitigates the buffer underflow by reflecting less bandwidth change when high jitter occurs due to rapid bandwidth change. The experimental results show that the proposed scheme improves the quality of experience by mitigating buffer underflow and reducing the number of quality changes in wireless networks.
Modified RTT Estimation Scheme for Improving Throughput of Delay-based TCP in Wireless Networks
Hyunsoo Kang, Jiwoo Park, Kwangsue Chung
In a wireless network, TCP causes the performance degradation because of mistaking packet loss, which is caused by characteristics of wireless link and throughput oscillation due to change of devices connected on a limited bandwidth. Delay based TCP is not affected by packet loss because it controls window size by using the RTT. Therefore, it can solve the problem of unnecessary degradation of the rate caused by misunderstanding reason of packet loss. In this paper, we propose an algorithm for improving the remaining problems by using delay based TCP. The proposed scheme can change throughput adaptively by adding the RTT, which rapidly reflects the network conditions to BaseRTT. It changes the weight of RTT and the increases and decreases window size based on the remaining amount of the buffer. The simulation indicated that proposed scheme can alleviate the throughput oscillation problem, as compared to the legacy TCP Vegas.
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