• The Transactions of the Korea Information Processing Society
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• v.2 no.6
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• pp.927-936
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• 1995

ATM Forum has defined a new service class for data applications called Available Bit Rate(ABR) Service, which has highly bursty traffic and unpredictable burst size. It is desirable that we reduce the probability of retrans mission of packets by minimizing the loss of cells because the traffic is much more sensitive to loss than delay. The Forum has also selected the Rate-Based Control for the ABR service and proposed EPRCA as the control mechanism for the service. This paper proposes the combination of EPRCA and the other feedb ack control mechanisms such as BECN and BP. The combined control mechanism control ABR traffic more efficiently and the simulation results show that the network performance can be improved by choosing the appropriate parameters.

• Journal of KIISE:Information Networking
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• v.31 no.1
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• pp.1-10
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• 2004

This study discusses Application-layer FEC using erasure codes. Because of the simple decoding process, erasure codes are used effectively in Application-layer FEC to deal with Packet-level errors. The large number of parity packets makes the loss rate to be small, but causes the network congestion to be worse. Thus, a redundancy control algorithm that can adjust the number of parity packets depending on network conditions is necessary. In addition, it is natural that high-priority frames such as I frames should produce more parity packets than low-priority frames such as P and B frames. In this paper, we propose a redundancy control algorithm that can adjust the amount of redundancy depending on the network conditions and depending on data priority, and test the performance in simple links and congestion links.

• The KIPS Transactions:PartC
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• v.9C no.4
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• pp.563-572
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• 2002

Wireless mobile networks tend to drop a large portion of packets due to propagation errors rather than congestion. To Improve reliability over noisy wireless channels, wireless networks can employ forward error correction (FEC) techniques. Static FEC algorithms, however, can degrade the performance by poorly matching their overhead to the degree of the underlying channel error, especially when the channel path loss rate fluctuates widely. This paper investigates the benefits of an adaptable FEC mechanism for wireless networks with severe packet loss by analytical analysis or measurements over a real wireless network called sensor network. We show that our adaptive FEC named FECA (FEC-level Adaptation) technique improves the performance by dynamically tuning FEC strength to the current amount of wireless channel loss. We quantify these benefits through a hybrid simulation integrating packet-level simulation with bit-level details and validate that FECA keeps selecting the appropriate FEC-level for a constantly changing wireless channel.

• 대한공업교육학회지
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• v.33 no.1
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• pp.233-248
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• 2008

Stream Control Transmission Protocol(SCTP) is a new transport protocol that is known to provide improved performance than Transmission Control Protocol(TCP) in multi-homing environment that is having two and more IP addresses. But currently single-homed computer is used primarily that is having one IP address. To identify whether mean transfer time for SCTP is faster that for TCP in single-homed environment considering packet loss, we make up real testbed regulating the bandwidth, delay time and packet loss rate on router and observe the transfer time. We write server and client applications to measure SCTP and TCP mean transfer time by C language. Analysis of these experimental results from the testbed implementation shows that mean transfer time of SCTP is not better than performance of TCP in single homed environment exceptional case. Main reasons of performance are that SCTP compared to TCP stops transmitting data by timeout and data transmission is often delayed when SACK congestion happens. The result of study shows that elaborate performance tuning is required in developing a new SCTP module or using a implemented SCTP module.

• Proceedings of the KIEE Conference
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• 2006.10c
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• pp.473-475
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• 2006

FieldBus communication systems in industrial wired and wireless network may cause the degradation of the TCP performance due to the racket loss. TCP is particularly targeted at the wired networks, a packet loss is assumed to be caused by the network congestion. As a result, the performance of TCP decreases significantly when used over networks that exhibit a high bit error rate. In order to solve this problem, this paper designs and implements the WFSnoop mechanism which offers a fast local retransmission. The proposed mechanism does not require any changes in customer premises. Base on the simulation in the wired and wireless network environment, we analyzed the performance of the WFSnoop mechanism.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.23 no.9A
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• pp.2249-2257
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• 1998

In this paper, we propose the enhanced priority control algorithm that adaptively controls the cell service ratio according to the relative cell occupancy ratio of buffer. The asynchronous transfer mode (ATM) provides the means to support various multimedia services in broadband networks. To support multimedia services, various data traffics of different priorities should be controlled effectively. And also it needs congestion control functions required in the netowrk to carry out the control operation. To accomplish this in a flexible and effective manner, priority classes for the different services ar ecommonly used. The proposed enhanced priority control mechanism have two service calsses of the delay sensitive class and the loss sensitive class. The simulation results show that te proposed control mechanism improves the QoS, the charateristics of cell loss probability and mean cell delay time, by selecting propeor relativ ecell occupancy ratio of buffer and the average arrival rate.

• The Journal of the Acoustical Society of Korea
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• v.21 no.4
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• pp.349-360
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• 2002

In ITU H.323 teleconference system, the RTP/RTCP protocol is offered to transfer real-time multimedia stream. Both sender and receiver hate experience in packet loss and jitter which result from network congestion over Internet. Audio quality over Internet depends on the number of lost packets and on jitter between successive packets. The goal of our study is to improve the speech quality over Internet by checking the packet loss characteristics of the network and adopting the but for control management mechanism at the receiver. We suggest a dynamic redundant audio transmission mechanism which examines the packet loss rate and uses the feedback information through RTCP.

• 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 the Korea Society of Computer and Information
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• v.6 no.1
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• pp.45-50
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• 2001

Congestion may take place in the ATM network because of high-speed cell transmission features, and cell delay and loss also can be caused by unexpected traffic variation. Thus. traffic control mechanisms are needed. One of them to decrease congestion is the cell shaping. This paper proposes a hybrid type cell shaper composed of a Leaky Bucket with token pool, EWMA with time window, and a spacing control buffer. The simulator BONeS with the ON/OFF traffic source model evaluates the performance of the proposed cell shaping method. Simulation results show that the cell shaping concerning the respective source traffics is adapted to and then controlled on the mean bit rate.

• Journal of the Korea Society of Computer and Information
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• v.7 no.3
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• pp.143-148
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• 2002

Congestion may take place in the ATM network because of high-speed cell transmission features, and cell delay and loss also can be caused by unexpected traffic variation. Thus, traffic control mechanisms are needed. One of them to decrease congestion is the Cell shaping. This paper proposes a hybrid type cell shaper composed of a Leaky Bucket with token pool, Tn with time window, and a spacing control buffer. The simulator BONeS with the ON/OFF traffic source model evaluates the performance of the proposed cell shaping method. Simulation results show that the cell shaping concerning the respective source traffics is adapted to and then controlled on the mean bit rate.