• Journal of KIISE:Information Networking
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• v.37 no.5
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• pp.363-373
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• 2010

The existing QoS mechanisms for video streaming are short of the consideration for various user environments and the characteristic of streaming applying programs. In order to overwhelm this problem, studies on the video streaming protocols exploiting scalable video coding (SVC), which provide spatial, temporal, and qualitative scalability in video coding, are progressing actively. However, these protocols also have the problem to deepen network congestion situation, and to lower fairness between other traffics, as they are not equipped with congestion control mechanisms. SVC based streaming protocols also have the problem to overlook the property of videos encoded in SVC, as the protocols transmit the streaming simply by extracting the bitstream which has the maximum bit rate within available bandwidth of a network. To solve these problems, this study suggests TCP-friendly network adaptive SVC streaming(T-NASS) protocol which considers both network status and SVC bitstream property. T-NASS protocol extracts the optimal SVC bitstream by calculating TCP-friendly transmission rate, and by perceiving the network status on the basis of packet loss rate and explicit congestion notification(ECN). Through the performance estimation using an ns-2 network simulator, this study identified T-NASS protocol extracts the optimal bitstream as it uses TCP-friendly transmission property and perceives the network status, and also identified the video image quality transmitted through T-NASS protocol is improved.

• Journal of Korea Multimedia Society
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• v.2 no.3
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• pp.339-346
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• 1999

When a multicast video conference system utilizes RTP (Real Time Protocol) and RTCP (Real Time Control Protocol), the loss rate and the synchronization of transfer in RTCP affect the scalability of the system. The random delay technique introduced to resolve the problems is so simple that leads the network to meet some congestion in synchronizing feedback information when lots of people transfer the feedback information simultaneously, which reduces the scalability of system. In this paper, we propose a new feedback control algorithm that provides priority levels with the RTCP packet, which cuts down the feedback delay and increases the scalability. The criteria of providing priority Based on the decided priority level, Agent forced the session participants to provide much more RTCP packets, positively controlled, and the possible bandwidth can be measured. The simulation on this technique decreases the delay, and the feedback messages are equally distributed on a given time period.

• Journal of Digital Contents Society
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• v.15 no.2
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• pp.147-154
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• 2014

There have been company researches for TCP-Westwood algorithms in wireless TCP environment with high packet loss rate. Because the TCP-Westwood algorithm adjusts the congestion window according to the ABE(Available Bandwidth Estimation), the algorithm has a problem which the accuracy of ABE decreases as the error rate increases. To solve such a problem, the proposed scheme in this paper adopts the existing packet pattern based algorithm that the ABE is ignored when the arriving interval time of ACK is longer than a given interval time and uses new algorithm that the ABE is reallocated to a given allowable ABE when the ABE is over the allowable range. The proposed scheme shows the simulation result that the ABE is closest to the setting bandwidth for simulation compared to the existing algorithms.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.11 no.2
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• pp.1070-1088
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• 2017

Recently, the applications of Internet of Things (IoTs) are growing rapidly. Wireless Sensor Network (WSN) becomes an emerging technology to provide the low power wireless connectivity for IoTs. The IPv6 over low-power wireless personal area networks (6LoWPAN) has been proposed by IETF, which gives each WSN device an IPv6 address to connect with the Internet. The transmission congestion in IoTs could be a problem when a large numbers of sensors are deployed in the field. Therefore, it is important to consider whether the WSN devices have be completely integrated into the Internet with proper quality of service (QoS) requirements. The Software Defined Network (SDN) is a new architecture of network decoupling the data and control planes, and using the logical centralized control to manage the forwarding issues in large-scale networks. In this research, the SDN-based 6LoWPAN Border Router (6LBR) is proposed to integrate the transmission from WSNs to Internet. The proposed SDN-based 6LBR communicating between WSNs and the Internet will bring forward the requirements of end-to-end QoS with bandwidth guarantee. Based on our experimental results, we have observed that the selected 6LoWPAN traffic flows achieve lower packet loss rate in the Internet. Therefore, the 6LoWPAN traffic flows classified by SDN-based 6LBR can be reserved for the required bandwidth in the Internet to meet the QoS requirements.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.5 no.4
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• pp.720-740
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• 2011

Wi-Fi-enabled portable devices have recently been introduced into the consumer electronics market. These devices download or upload content, from or to a host machine, such as a personal computer, a laptop, a home gateway, or a media server. This paper investigates the fairness among multiple Wi-Fi-enabled portable devices in a home network when they are simultaneously communicated with the host machine. First, we present that, a simple IEEE 802.11-based home network suffers from unfairness, and the fairness is exaggerated by the wireless link errors. This unfairness is due to the asymmetric response of the TCP to data-packet loss and to acknowledgment-packet loss, and the wireless link errors that occur in the proximity of any node; the errors affect other wireless devices through the interaction at the interface queue of the home gateway. We propose a QoS-provisioning framework in order to achieve per-device fairness and service differentiation. For this purpose, we introduce the medium access price, which denotes an aggregate value of network-wide traffic load, per-device link usage, and per-device link error rate. We implemented the proposed framework in the ns-2 simulator, and carried out a simulation study to evaluate its performance with respect to fairness, service differentiation, loss and delay. The simulation results indicate that the proposed method enforces the per-device fairness, regardless of the number of devices present and regardless of the level of wireless link errors; furthermore it achieves high link utilization with only a small amount of frame losses.

• The KIPS Transactions:PartC
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• v.12C no.1 s.97
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• pp.121-128
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• 2005

Tho existing TCP(Transmission Control Protocol) is known to be unsuitable for a network with the characteristics of high RDP(Bandwidth-Delay Product) because of the fixed small or large buffer size at the TCP sender and receiver. Thus, some trial cases of adjusting the buffer sizes automatically with respect to network condition have been proposed to improve the end-to-end TCP throughput. ATBT(Automatic TCP fluffer Tuning) attempts to assure the buffer size of TCP sender according to its current congestion window size but the ATBT assumes that the buffer size of TCP receiver is maximum value that operating system defines. In DRS(Dynamic Right Sizing), by estimating the TCP arrival data of two times the amount TCP data received previously, the TCP receiver simply reserves the buffer size for the next arrival, accordingly. However, we do not need to reserve exactly two times of buffer size because of the possibility of TCP segment loss. We propose an efficient TCP buffer tuning technique(called TBT-PLR: TCP buffer tuning algorithm based on packet loss ratio) since we adopt the ATBT mechanism and the TBT-PLR mechanism for the TCP sender and the TCP receiver, respectively. For the purpose of testing the actual TCP performance, we implemented our TBT-PLR by modifying the linux kernel version 2.4.18 and evaluated the TCP performance by comparing TBT-PLR with the TCP schemes of the fixed buffer size. As a result, more balanced usage among TCP connections was obtained.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.2B
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• pp.44-52
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• 2005

This paper proposes a neural network prediction model to improve the valid packet transmission rate for the QoS(Quality of Service) of multimedia transmission. The Round Trip Time(RTT) and Packet Loss Rate(PLR) are predicted using a neural network and then the transmission rate is decided based on the predicted RTT and the PLR. The suggested method will improve the transmission rate since it uses the rate control factors corresponding to time of data is being transmitted, while the conventional one uses the transmission rate determined based on the past informations. An experimental set-up has been established using a Linux PC system, and the multimedia data are transmitted using UDP protocol in real time. The valid transmitted packets are about 5% higher than the one in the conventional TCP-Friendly congestion control method when the suggested algorithm was applied.

• The KIPS Transactions:PartC
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• v.8C no.6
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• pp.805-814
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• 2001

In this paper, we propose a new scheduling algorithm that guarantees the delay property of real-time traffic, not considered in previous DWRR(Dynamic Weighted Round Robin) algorithm and also transmits non-real-time traffic efficiently. The proposed scheduling algorithm is a variation of DWRR algorithm to guarantee the delay property of real-time traffic by adding cell transmission method based on delay priority. It also uses the threshold to prevent the cell loss of non-real-time traffic due to cell transmission method based on delay priority. Proposed scheduling algorithm may increase some complexity over conventional DWRR scheme because of cell transmission method based on delay priority. However, the consideration of delay priority can minimize cell delay and require less size of temporary buffer. Also, the results of our performance study shows that the proposed scheduling algorithm has better performance than conventional DWRR scheme due to reliable ABR service and congestion avoidance capacity.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.9A
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• pp.1372-1382
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• 1999

The major source of errors in high-speed networks such as Broadband ISDN(B-lSDN) is buffer overflow during congested conditions. These congestion errors are the dominant sources of errors in 1high-speed networks and result in cell losses. Conventional communication protocols use error detection and retransmission to deal with lost packets and transmission errors. However, these conventional ARQ(Automatic Repeat Request) methods are not suitable for the high-speed networks since the transmission delay due to retransmissions becomes significantly large. As an alternative, we have presented a method to recover consecutive cell losses using forward error correction(FEC) in ATM(Asynchronous Transfer Mode)networks to reduce the problem. The performance estimation based on the cell discard process model has showed our method can reduce the cell loss rate substantially. Also, the performance estimations in ATM networks by interleaving and IP multicast service are discussed.

• The Journal of the Korea institute of electronic communication sciences
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• v.15 no.6
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• pp.1045-1054
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• 2020

In this paper, we designed a TCP/IP based ring buffer system that can stably transfer bulk data streams in the unstable network environments. In the scheme we proposed, The observation data stream generated and output by each radio observatory's backend system as a UDP frame is stored as a UDP packet in a large capacity ring buffer via a socket buffer in the client system. Thereafter, for stable transmission to the remote destination, the packets are processed in TCP and transmitted to the socket buffer of server system in the correlation center, which packets are stored in a large capacity ring buffer if there is no problem with the packets. In case of errors such as loss, duplication, and out of order delivery, the packets are retransmitted through TCP flow control, and we guaranteed that the reliability of data arriving at the correlation center. When congestion avoidance occurs due to network performance instability, we also suggest that performance degradation can be minimized by applying parallel streams.