• Proceedings of the Korean Information Science Society Conference
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• 2003.04d
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• pp.469-471
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• 2003

TCP는 패킷손실이 발생할 경우에 이를 혼잡에 의한 것이라고 판단하여 혼잡제어를 수행하는 프로토콜이다. 그러나 무선망은 그 특성상 에러가 많기 때문에 기존의 TCP프로토콜을 그대로 적용할 경우 불필요한 혼잡제어를 수행하게 되어 결국 전체 망의 성능을 떨어뜨리게 된다. 본 논문에서는 링크상태에 따라 송신률을 조절하는 TCP-ELSA 메커니즘 이용하여 링크상태 예측 방법을 개선하였다. TCP-ELSA는 무선링크의 링크상태를 예측하여 송신률을 조절하여 대역폭의 효율을 증가시키고 링크의 공평성을 보장하는 메커니즘이다. 본 논문에서 제안하는 개선된 TCP-ELSA는 링크상태 예측방법에 있어서 오류가 발생할 수 있는 경우를 고려함으로써 공평성을 유지하면서도 좀더 정확한 링크상태의 예측이 가능하도록 한다. 그리고 실시간 패킷을 비실시간 패킷과 구분하여 전송하도록 하여 Qos를 제공하였다.

• Journal of KIISE:Information Networking
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• v.30 no.5
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• pp.595-603
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• 2003

To provide mobile nodes with continuous communication services, it is important to reduce the packet losses during handoffs. The handoffs of mobile nodes cause packet losses and decrease of TCP throughput on account of a variety of factors. One of those is the congestion in the new cell. Due to the congestion, not only the node moving into the cell but also the already existing nodes that were successfully communicating in the cell suffer the performance degradation. In this paper we propose a new handoff mechanism called‘packet freeze control’, which avoids the congestion caused by handoffs by regulating the influx of traffic burst into the new cell. Packet freeze control is applicable to a wireless network domain in which FAs(Foreign Agents) are connected hierarchically and constitute a logical tree. It gradually increases the number of packets transferred to the new cell by buffering packets in the FAs on the packet delivery path over the wireless network domain. The simulation results show that the proposed mechanism not only reduces the packet losses but also enhances the TCP throughput of other mobile nodes in the cell.

• KIPS Transactions on Computer and Communication Systems
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• v.2 no.3
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• pp.117-124
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• 2013

In this paper we propose a TFRC(TCP Friendly Rate Control) which guarantees a minimum rate in order to improve the efficiency of the previous TFRC which cannot distinguish congestion losses and wireless losses and decreases throughput both in wired and wireless networks. This TFRC technique is able to guarantee a minimum rate for video by restricting a loss event rate with packet loss probability about existing TFRC and constraining a rate reduction from the feedback timeout. When we experimented both the existing TFRC and the new one with TCP in the same network, we found that the latter is better than the former. Consequently, it shows that the proposed TFRC can improve video streaming quality using a guaranteed minimum transmission rate.

• Journal of KIISE:Computing Practices and Letters
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• v.16 no.12
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• pp.1229-1233
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• 2010

In event-driven wireless sensor networks, network congestion occurs when event data, which have higher transmission rates than periodic sensing data, arc forwarded to bottleneck links. As the congestion continues, congestion collapse is triggered, so most of packets from source nodes are failed to transmit to a sink node. Rate control schemes can be a solution for preventing the congestion collapse problem. In this paper, a rate control scheme that each node controls child node's data rate based on congestion patterns is proposed. Experiments show that the proposed scheme effectively controls network congestion and successfully transmits more event data packets to a sink node than existing rate control schemes.

• Journal of the Korea Society of Computer and Information
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• v.7 no.4
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• pp.127-132
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• 2002

This paper aims to adjust the window size according to the network condition that the sender determines by making the receiver participating in the congestion levels. TCP-RLDM has the measurement-based transmission strategy based on the data-receiving rate complementing TCP with the property of Additive Increase / Multiplicative Decrease. The protocol can make an performance improvement by responding differently according to the property of errors-whether congestion losses or transient transmission errors - to confront dynamically in heterogeneous environments with wired or wireless networks and delay-sensitive or -tolerant applications. By collecting data-receiving rate and the cause of errors from the receiver and by enabling sender to use the congestion avoidance strategy before occuring congestion possibly, the protocol works well at variable network environments.

• 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.12C no.4 s.100
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• pp.571-580
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• 2005

In fast long-distance networks, TCP's congestion control algorithm has the problem of utilizing bandwidth effectively. Several window-based congestion control protocols for high-speed and large delay networks have been proposed to solve this problem. These protocols deliberate mainly three properties : scalability, TCP-friendliness, and RTT-fairness. These protocols, however, cannot satisfy above three properties at the same time because of the trade-off among them This paper presents a new window-based congestion control algorithm, called EM (Exponential Increase/ Multiplicative Decrease), that simultaneously supports all four properties including fast convergence, which is another important constraint for fast long-distance networks; it can support scalability by increasing congestion window exponentially proportional to the time elapsed since a packet loss; it can support RTT-fairness and TCP-friendliness by considering RTT in its response function; it can support last fair-share convergence by increasing congestion window inversely proportional to the congestion window just before packet loss. We evaluate the performance of EIMD and other algorithms by extensive computer simulations.

• Journal of KIISE
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• v.41 no.11
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• pp.974-981
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• 2014

These days, use of multimedia streaming service and demand of QoS (Quality of Service) improvement have been increased because of development of network. QoS of streaming service is influenced by a jitter, delay, throughput, and loss rate. For guaranteeing these factors which are influencing QoS, the role of transport layer is very important. But existing TCP which is a typical transport layer protocol increases the size of congestion window slowly and decreases the size of a congestion window drastically. These TCP characteristic have a problem which cannot guarantee the QoS of UHD multimedia streaming service. In this paper, we propose a router buffer based congestion control method for improving the QoS of UHD streaming services. Our proposed scheme applies congestion window growth rate differentially according to a degree of congestion for preventing an excess of available bandwidth and maintaining a high bandwidth occupied. Also, our proposed scheme can control the size of congestion window according to a change of delay. After comparing with other congestion control protocols in the jitter, throughput, and loss rate, we found that our proposed scheme can offer a service which is suitable for the UDH streaming service.

• 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.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.9B
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• pp.829-837
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• 2006

Due to the internet network congestion, packets may be dropped or delayed at routers. This phenomenon degrades the quality of streaming applications that require high QoS requirements. The proposed algorithm in this paper, called DBRC(Delay-Based Rate Control), tries to cause router queue occupancy to reach a steady state or equilibrium by throttling the transmission rate of the multimedia traffics when network delays tend to increase and also probing for more bandwidth when network delays tend to decrease. Simulation results show that the proposed algorithm provides smooth transmission rate, nearly constant delay and low packet loss rates, compared with TFRC(TCP Friendly Rate Control) that is one of dominant multimedia congestion control algorithms.