• 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.32 no.5
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• pp.583-592
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• 2005

An important factor influencing the robustness of the Internet is the end-to-end TCP congestion control. However, the congestion control scheme of TCP Reno, the most popular TCP version on the Internet, employs passive congestion indication. It makes the network congestion worse. Brakmo and Peterson proposed a congestion control algorithm, TCP Vegas, by modifying the congestion avoidance scheme of TCP Reno. Many studies indicate that Vegas is able to achieve better throughput and higher stability than Reno. But there are three unfairness problems in Vegas. These problems hinder the spread of Vegas in the current Internet. In this paper, in order to solve these unfairness problems, we propose a new congestion control algorithm called TCP NewVegas. The proposed NewVegas is able to solve these unfairness problems effectively by using the variation of the number of queued packets in a bottleneck router. To evaluate the proposed approach, we compare the performance among NewVegas, Reno and Vegas. Through the simulation, NewVegas is shown to be able to achieve throughput and better fairness than Vegas.

• Journal of Communications and Networks
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• v.4 no.2
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• pp.108-117
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• 2002

TCP, which was developed on the basis of wired links, supposes that packet losses are caused by network congestion. In a wireless network, however, packet losses due to data corruption occur frequently. Since TCP does not distinguish loss types, it applies its congestion control mechanism to non-congestion losses as well as congestion losses. As a result, the throughput of TCP is degraded. To solve this problem of TCP over wireless links, previous researches, such as split-connection and end-to-end schemes, tried to distinguish the loss types and applied the congestion control to only congestion losses; yet they do nothing for non-congestion losses. We propose a novel transport protocol for wireless networks. The protocol called VS-TCP (Variable Segment size Transmission Control Protocol) has a reaction mechanism for a non-congestion loss. VS-TCP varies a segment size according to a non-congestion loss rate, and therefore enhances the performance. If packet losses due to data corruption occur frequently, VS-TCP decreases a segment size in order to reduce both the retransmission overhead and packet corruption probability. If packets are rarely lost, it increases the size so as to lower the header overhead. Via simulations, we compared VS-TCP and other schemes. Our results show that the segment-size variation mechanism of VS-TCP achieves a substantial performance enhancement.

• Proceedings of the Korea Information Processing Society Conference
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• 2006.05a
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• pp.913-916
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• 2006

SNMP는 망 관리 시스템을 구현하는데 있어서 비교적 단순하고 안전한 일대일 통신 방안을 제공하고 있다. 원격 회의 또는 강의, VoIP, 네트워크 게임 등의 다양한 통신 컨텐츠에 대해 인터넷을 통한 이용이 급증하면서 UDP 기반의 SNMP 통신에 있어서도 Congestion Control을 적용하기 위한 방안이 필요하게 되었다. 본 논문에서는 UDP에서 Congestion Control을 이용하는 DCCP를 이용하여 SNMP를 기반의 망 관리 시스템을 구축할 수 있는 구조를 제안한다.

• Proceedings of the IEEK Conference
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• summer
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• pp.294-301
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• 2004

This paper presents a combination of optimization concept and congestion control for multicast communications to bring best benefit for the network. For different types of Internet services, there will be different utility functions and so there will be different ways to choose on how to control the congestion, especially for real time multicast traffic. Our proposed algorithm OMCC brings the first implementation experiment of utility-based Multicast Congestion Control. Simulation results show that OMCC brings better network performances in multicast session throughput while it still keeps a certain fairness of unicast and multicast sessions, and thus, provides better benefit for all network participants.

• 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.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.11A
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• pp.867-875
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• 2003

Current end-to-end congestion control depends only on the information of end points (using three duplicate ACK packets) and generally responds slowly to the network congestion. This mechanism can't avoid TCP global synchronization in which TCP congestion window size is fluctuated during congestion period. Furthermore, if RTT(Round Trip Time) is increased, three duplicate ACK packets are not correct congestion signals because congestion might already disappear and the host may send more packets until it receives three duplicate ACK packets. Recently there are increasing interests in solving end-to-end congestion control using AQM(Active Queue Management) to improve the performance of TCP protocols. AQM is a variation of RED-based congestion control. In this paper, we first evaluate the effectiveness of the current AQM schemes such as RED, CHOKe, ARED, FRED and SRED, over traffic with different rates and over traffic with mixed responsive and non-responsive flows, respectively. In particular, CHOKe mechanism shows greater unfairness, especially when more unresponsive flows exist in a shared link. We then propose a new AQM scheme using CHOKe mechanism, called DQC(Double Queue CHOKe), which uses two FIFO queues before applying CHOKe mechanism to adaptive congestion control. Simulation shows that it works well in protecting congestion-sensitive flows from congestion-causing flows and exhibits better performances than other AQM schemes. Also we use partial state information, proposed in LRURED, to improve our mechanism.

• International Journal of Computer Science & Network Security
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• v.22 no.10
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• pp.191-200
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• 2022

Constrained Application Protocol (CoAP) is a standardized protocol by the Internet Engineering Task Force (IETF) for the Internet of things (IoT). IoT devices have limited computation power, memory, and connectivity capabilities. One of the significant problems in IoT networks is congestion control. The CoAP standard has an exponential backoff congestion control mechanism, which may not be adequate for all IoT applications. Each IoT application would have different characteristics, requiring a novel algorithm to handle congestion in the IoT network. Unnecessary retransmissions, and packet collisions, caused due to lossy links and higher packet error rates, lead to congestion in the IoT network. This paper presents an adaptive congestion control protocol for CoAP, Adaptive Congestion Control with a Backoff algorithm (ACCB). AACB is an extension to our earlier protocol AdCoCoA. The proposed algorithm estimates RTT, RTTVAR, and RTO using dynamic factors instead of fixed values. Also, the backoff mechanism has dynamic factors to estimate the RTO value on retransmissions. This dynamic adaptation helps to improve CoAP performance and reduce retransmissions. The results show ACCB has significantly higher goodput (49.5%, 436.5%, 312.7%), packet delivery ratio (10.1%, 56%, 23.3%), and transmission rate (37.7%, 265%, 175.3%); compare to CoAP, CoCoA+ and AdCoCoA respectively in linear scenario. The results show ACCB has significantly higher goodput (60.5%, 482%,202.1%), packet delivery ratio (7.6%, 60.6%, 26%), and transmission rate (40.9%, 284%, 146.45%); compare to CoAP, CoCoA+ and AdCoCoA respectively in random walk scenario. ACCB has similar retransmission index compare to CoAp, CoCoA+ and AdCoCoA respectively in both the scenarios.

• International Journal of Control, Automation, and Systems
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• v.4 no.1
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• pp.53-62
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• 2006

In this paper, ATM network congestion control with explicit rate feedback is considered. In ATM networks, delays commonly appear in data transmission and have to be considered in congestion control design. In this paper, a bounded single round delay on the return path is considered. Our objective is to design an explicit rate feedback control that achieves a robust optimal $H_2$ performance regardless of the bounded time-varying delays. An optimization approach in terms of linear matrix inequalities (LMIs) is given. Saturation in source rate and queue buffer is also taken into consideration in the proposed design. Simulations for the cases of single source and multiple sources are presented to demonstrate the effectiveness of the design.

• 제어로봇시스템학회:학술대회논문집
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• 2004.08a
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• pp.474-479
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• 2004

This paper describes a feedback-based congestion control algorithm to improve TCP performance over wireless network. In this paper, we adjust the packet marking probability at the router for Max-Min fair sharing of the bandwidth and full utilization of the link. Using the successive ECN (Explicit Congestion Notification), the proposed algorithm regulates the window size to avoid the congestion and sees the packet loss only due to the wireless link error. Based on the asymptotic analysis, it is shown that the proposed algorithm guarantees the QoS of the wireless TCP. The effectiveness of the proposed algorithm is demonstrated by simulations.