• Journal of KIISE
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• v.42 no.12
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• pp.1600-1610
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• 2015

While today's networks have been shown to exhibit HBDP (High Bandwidth Delay Product) characteristics, the legacy TCP increases the size of the congestion window slowly and decreases the size of the congestion window drastically such that it is not suitable for HBDP Networks. In order to solve this problem with the legacy TCP, many congestion control TCP mechanisms have been proposed. C-TCP (Compound-TCP) is a hybrid TCP which is a synergy of delay-based and loss-based approaches. C-TCP adapts the decreasing rate of the delay window without considering the congestion level, leading to degradation of performance. In this paper, we propose a new scheme to improve the performance of C-TCP. By controlling the increasing and decreasing rates according to the congestion level of the network, our proposed scheme can improve the bandwidth occupancy and fairness of C-TCP. Through performance evaluation, we show that our proposed scheme offers better performance in HBDP networks as compared to the legacy C-TCP.

• Journal of KIISE:Information Networking
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• v.27 no.1
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• pp.58-67
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• 2000

The paper presents a representative-based feedback mechanism and rate adaptation policy for congestion control in multicast traffic for multimedia applications. In multicast congestion control, feedback implosion occurs as many receivers send feedback to a sender. We propose to use representatives to avoid the feedback implosion. In our scheme, receivers feedback packet loss information periodically and a sender adapts the sending rate based on the information collected through a hierarchy of representatives. A representative is selected in each region and roles as a filter to decrease the number of feedbacks. The simulation results show that the proposed scheme solves the feedback implosion problem and well adapts in a congested situation.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.6 no.1
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• pp.241-269
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• 2012

Wireless networks composed of multimedia-enabled resource-constrained sensor nodes have enriched a large set of monitoring sensing applications. In such communication scenario, however, new challenges in data transmission and energy-efficiency have arisen due to the stringent requirements of those sensor networks. Generally, congested nodes may deplete the energy of the active congested paths toward the sink and incur in undesired communication delay and packet dropping, while bit errors during transmission may negatively impact the end-to-end quality of the received data. Many approaches have been proposed to face congestion and provide reliable communications in wireless sensor networks, usually employing some transport protocol that address one or both of these issues. Nevertheless, due to the unique characteristics of multimedia-based wireless sensor networks, notably minimum bandwidth demand, bounded delay and reduced energy consumption requirement, communication protocols from traditional scalar wireless sensor networks are not suitable for multimedia sensor networks. In the last decade, such requirements have fostered research in adapting existing protocols or proposing new protocols from scratch. We survey the state of the art of transport protocols for wireless multimedia sensor networks, addressing the recent developments and proposed strategies for congestion control and loss recovery. Future research directions are also discussed, outlining the remaining challenges and promising investigation areas.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.11 no.11
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• pp.5200-5222
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• 2017

The separation of control plane and data plane in Software Defined Network (SDN) makes it flexible to control the network behavior, while also causes some inconveniences to the link failure recovery due to the delay between fail point and the controller. To avoid delay and packet loss, pre-defined backup paths are used to reroute the disrupted flows when failure occurs. However, it may introduce large overhead to build and maintain these backup paths and is hard to dynamically construct backup paths according to the network status so as to avoid congestion during rerouting process. In order to realize congestion aware fast link failure recovery, this paper proposes a novel method which installs multi backup paths for every link via source routing and per-hop-tags and spread flows into different paths at fail point to avoid congestion. We carry out experiments and simulations to evaluate the performance of the method and the results demonstrate that our method can achieve congestion aware fast link failure recovery in SDN with a very low overhead.

• Journal of Internet Computing and Services
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• v.18 no.6
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• pp.65-73
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• 2017

Congestion occurring at wireless sensor networks(WSNs) causes packet delay and packet drop, which directly affects overall QoS(Quality of Service) parameters of network. Network congestion is critical when important data is to be transmitted through network. Thus, it is significantly important to effectively control the congestion. In this paper, new mechanism to guarantee reliable transmission for the important data is proposed by considering the importance of packet, configuring packet priority and utilizing the settings in routing process. Using this mechanism, network condition can be maintained without congestion in a way of making packet routed through various routes. Additionally, congestion control using packet service time, packet inter-arrival time and buffer utilization enables to reduce packet delay and prevent packet drop. Performance for the proposed mechanism was evaluated by simulation. The simulation results indicate that the proposed mechanism results to reduction of packet delay and produces positive influence in terms of packet loss rate and network lifetime. It implies that the proposed mechanism contributes to maintaining the network condition to be efficient.

• The KIPS Transactions:PartC
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• v.10C no.4
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• pp.479-484
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• 2003

In the current Internet, it is difficult for an Internet Phone to guarantee the QoS due to variable network conditions such as packet loss rate, delay and bandwidth. In addition, the QoS of an Internet Video Phone is more hard to guarantee because of video data. In this paper, we investigate application-level QoS control schemes that can adapt to variable network conditions, and describe an error control scheme and a congestion control scheme. Based on these QoS control schemes, we have designed and implemented an Internet Video Phone System that supports adaptive audio and video delivery. Through experiments, we found that the Internet Video Phone can reduce the packet loss rate considerably as well as adjust the transmission rate considering other TCP flows.

• The Journal of the Korea Contents Association
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• v.11 no.2
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• pp.69-78
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• 2011

Recently, the bandwidth available to an end user has been dramatically increasing with the advancing of network technologies. This high-speed network naturally requires faster and/or stable data transmission techniques. The UDT(UDP based Data Transfer protocol) is a UDP based transport protocol, and shows more efficient throughput than TCP in the long RTT environment, with benefit of rate control for a SYN time. With a NAK event, however, it is difficult to expect an optimum performance due to the increase of fixed sendInterval and the flow control based on the previous RTT. This paper proposes a rate control method on following a NAK, by adjusting the sendInterval according to some degree of RTT period which calculated from a set of experimental results. In addition, it suggests an improved flow control method based on the TCP vegas, in order to predict the network congestion afterward. An experimental results show that the revised flow control method improves UDT's throughput about 20Mbps. With combining the rate control and flow control proposed, the UDT throughput can be improved up to 26Mbps in average.

• Journal of KIISE:Information Networking
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• v.29 no.2
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• pp.181-187
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• 2002

Connection admission control(CAC), which decides whether or not to accept a new call request, is one of the most Important preventive congestion control techniques in asynchronous transfer mode(ATM) networks. To develop a practical CAC scheme, first we propose a "Modified Cell Loss Probability MP${\nu}"$, which is based on "Virtual Cell Loss Probability P${\nu}"$, taking into account mean burst duration of input traffic source and buffer size in ATM networks. MP${\nu}"$ computes more accurate cell loss probability than P${\nu}"$ without increasing computational complexity, since P${\nu}"$ is formulated simply form the maximum and the average cell rate of input traffic. P${\nu}"$ is overestimated as compared to the real cell loss probability when the mean burst duration is relatively small to the buffer capacity. Then, we Propose a CAC scheme, based on "Modified Virtual Bandwidth(MVB)" method, which may individualize the cell loss probabilities in heterogeneous traffic environments. For the proposed approach, we define the interference intensity to identify interferences between heterogeneous traffic sources and use it as well as MP${\nu}"$ to compute MVB. Our approach is well suitable for ATM networks since it provides high bandwidth utilization and guarantees simple and real time CAC computation for heterogeneous traffic environments.heterogeneous traffic environments.

• Journal of KIISE:Information Networking
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• v.32 no.3
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• pp.382-390
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• 2005

As today's networks evolve towards an If-based integrated network, the role of transmission control protocol(TCP) has been increasing as well. As a well-known issue, the performance of TCP is affected by its loss recovery mechanism that is comprised of two algorithms; fast retransmit and fast recovery. Although retransmission timeout(RTO) caused by multiple packet losses can be avoided by using selective acknowledgement(SACK) option, RTO cannot be avoided if a retransmitted packet is lost. Therefore, we propose a simple modification to make it possible for a TCP sender using SACK option to detect a lost retransmission. In order to evaluate the proposed algorithm, simulations have been performed for two scenarios where packet losses are random and correlated. Simulation results show that the proposed algorithm can improve TCP performance significantly.

• Proceedings of the IEEK Conference
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• 2006.06a
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• pp.749-750
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• 2006

TFRC is an equation-based rate control scheme originally developed for video transmission over wired networks. When applied to the wireless networks, it suffers from performance degradation. In this thesis, we propose an end-to-end loss discrimination algorithm to improve the performance of TFRC over wireless networks. The proposed WLD-TFRC scheme combines Spike and WLD(Wireless Loss Discount) algorithms to discriminate wireless loss from congestion loss, and to discount feedback loss event rate. Experimental results show that WLD-TFRC outperforms the original TFRC and effectively reduce the degradation of the video quality caused by the wireless link status.