• Journal of Communications and Networks
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• v.5 no.1
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• pp.73-81
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• 2003

The controlled load service defined within the IETF's Integrated Services architecture for quality-of-service (QoS) in the Internet requires source nodes to regulate their traffic while the network, in combination with an admission control strategy, provides a guarantee of performance equivalent to that achieved in a lightly loaded network. Packets sent in violation of the traffic contract are marked so that the network may assign them a lower priority in the use of bandwidth and buffer resources. In this paper, we define the requirements of a scheduler serving packets belonging to the controlled load service and present a novel scheduler that exactly achieves these requirements. In this set of requirements, besides efficiency and throughput, we include an additional important requirement to bound the additional delay of unmarked packets caused due to the transmission of marked packets while dropping as few marked packets as possible. Without such a bound, unmarked packets that are in compliance with the traffic contract are not likely to experience delays consistent with that in al lightly loaded network. For any given desired bound ${\alpha}$ on this additional delay, we present the CL(${\alpha}$) scheduler which achieves the bound while also achieving a per-packet work complexity of O(1) with respect to the number of flows. We provide an analytical proof of these properties of the CL(${\alpha}$) scheduler, and we also verify this with simulation using real traces of video traffic. The scheduler presented here may be readily adapted for use in scheduling flows with multi-level priorities such as in some real-time video streams, as well as in other emerging service models of the Internet that mark packets to identify drop precedences.

• The KIPS Transactions:PartB
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• v.12B no.7 s.103
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• pp.755-760
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• 2005

To transmit a video bit stream over low bandwith, such as mobile, channels, encoding algorithms for high bit rate like H.263+ are used. In transmitting video bit-streams, packet losses cause severe degradation in image quality. This paper proposes a new algorithm for the recovery of missing or erroneous motion vectors when H.263+ bit-stream is transmitted. Considering that the missing or erroneous motion vectors are closely related with those of neighboring blocks, this paper proposes a temporal-spatial error concealment algorithm. The proposed approach is that missing or erroneous Motion Vectors(MVs) are recovered by clustering the movements of neighboring blocks by their homogeneity. MVs of neighboring blocks we clustered according to ALA(Average Linkage Algorithm) clustering and a representative value for each cluster is determined to obtain the candidate MV set. By computing the distortion of the candidates, a MV with the minimum distortion is selected. Experimental results show that the proposed algorithm exhibits better performance in subjective and objective evaluation than existing methods.

• Journal of the Korea Society of Computer and Information
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• v.26 no.3
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• pp.1-7
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• 2021

In this paper, optimized media data retrieval and transmission based on the Hybrid CDN/P2P architecture and selective storage through user's prediction of requestability enable seamless data transfer to users and reduction of unnecessary traffic. We also propose a new media management method to minimize the possibility of transmission delay and packet loss so that media can be utilized in real time. To this end, we construct each media into logical segments, continuously compute weights for each segment, and determine whether to store segment data based on the calculated weights. We also designate scattered computing nodes on the network as local groups by distance and ensure that storage space is efficiently shared and utilized within those groups. Experiments conducted to verify the efficiency of the proposed technique have shown that the proposed method yields a relatively good performance evaluation compared to the existing methods, which can enable both initial latency reduction and seamless transmission.

• The Journal of Korean Institute of Next Generation Computing
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• v.13 no.2
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• pp.18-25
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• 2017

In computer science, an encoding is a standardization of converting information to one format for audio, video or text. Therefore, the encoding information of the data should be known to open and read it and there are algorithms detecting encoder of the data. However, some informations of data could be disappeared by packet loss when transmitted on network, especially, if the data is snatched by packet sniffing or eavesdropping from wireless communications. In this paper, we improve the performance of encoding detection algorithm of 'uchardet' program for multi-byte encoded data with errors based on bit-shift algorithm. To simulate the performance, we generated Korean and Japanese text data with errors that is removed some random bits at random positions. Then the detection algorithm are tested using the data and 'uchardet-bitshift' showed better performance than 'uchardet'. When Korean texts are used, 'uchardet' could detect perfectly with ≤0.005% errors but it showed 0% detection rate with ≥1% errors while 'uchardet-bitshift' detected perfectly with ≤0.05% errors and it showed correct detection cases with ≥1% errors. Japanese texts with errors tend to report falsely as Chinese encoding because Japanese texts include lots of Chinese characters. As a results, we improved encoding detection algorithms by applying bit shift operation.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.1B
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• pp.1-7
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• 2004

Increase of internet users and new type of applied traffic such as game, news, distributed computing, online image conference, and real time audio and video have leaded to demand for more bandwidth for each application. This algorithm represents a complex optical exchanger having typical wavelength switching function and time-slotted transmission function. Performance assessment of the proposed OXC (Optical Cross connect) sttucture defines LFS (Limit Frame Size) and VFS (Variable Frame Size) for classification by packet type and calculates the channel effect and loss probability depending the demanded bandwidth by access node increase. Optical exchanger in this type of structure can guarantee future network expansion as well as decrease of frame collision resulted from node increase.

• The KIPS Transactions:PartC
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• v.16C no.6
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• pp.717-724
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• 2009

The most important issue for successful deployment of IPTV service is the service quality. In this paper, we first choose critical quality metrics belonging to each layer of IPTV service transmission architecture and then analyze the relation between the quality parameters. For this purpose, we implement a test network that emulates the most practical environment in which IPTV service is provided and repeat measurements on the test network for the chosen metrics. Based on the analysis result, it is shown that packet losses have greater effect on the video quality of IPTV service comparing to delay and jitter.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.10 no.7
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• pp.1214-1221
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• 2006

Header compression scheme is suggested as a solution to reduce the inefficient overhead of general packet stream data. Especially, it is shown that there are more overhead rate for real-time media stream links such as voice because of its short payload size, and it is possible to get higher bandwidth efficiency using the header compression scheme. There are two kinds of error recovery in header compression such as Periodic Header Refresh(PHR) and Header Request(HR) schemes. In this paper, we analyze the performance of these two compression recovery schemes, and some results such as the overhead rate, bandwidth gain and bandwidth efficiency(BE) are presented.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.12 no.2
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• pp.9-17
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• 2012

Because best-effort Internet provides no guarantees on packet delay and loss, transient network congestion may cause negative effects on H.264/SVC streaming. Thus, the congestion control is required to adjust bit rate by dropping enhancement layers of H.264/SVC streams. This paper differentiates the video streams according to different levels of importance and proposes weighted-based congestion control algorithms to use the rate-distortion characteristics of streams. To maximize the weighted sum of PSNR values of all streams on a bandwidth-constrained node, this paper proposes WNS(Weighted Near-Sighted) and WFS(Weighted Far-Sighted) algorithms to control the number of enhancement layers of streams. Through simulation, this paper shows that weighted-based congestion control algorithm can efficiently adapt streams to network conditions and analyzes the characteristics of congestion control algorithms.

• 한국HCI학회:학술대회논문집
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• 2008.02a
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• pp.635-638
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• 2008

Unlike traditional analog broadcasting, digital broadcasting provides users with various additional services that we have never seen before. To receive these kind of services. data broadcasting includes not only audio, video signal, but also additional data associated with the program. In this paper, we present the efficient parsing and caching mechianism for data carousel in digital broadcasting set-top box. In order to speed up the process of parsing, we use the Message Pool that stores elementary_pid syntax of DSM-CC message packets.

• The Journal of the Korea Contents Association
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• v.5 no.1
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• pp.73-79
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• 2005

Recent advances on wireless technology are enabling the design and deployment of real-time traffic in wireless LAN. Point Coordination Function (PCF) mode of Wireless LAN is defined to provide QoS of real time traffic, such as voice, video in wireless LANs. The service polling scheduling plays an important role in real-time traffic services of wireless LAN. This paper proposes a multicast polling scheme to increase the maximum number of conversations by reducing the amount of empty polls in IEEE 802.11 PCF mode. Our simulation studies show that our scheme may improve performance of the PCF in terms of the average transfer delay and packet discard ratio.