• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.5B
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• pp.455-462
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• 2009

In this paper, we propose a buffer management scheme for decreasing the packet loss due to buffer overflow and improving the packet fairness between nodes in IEEE 802.11 based multi-hop mesh networks. In the proposed scheme, each mesh router that is an intermediate node receives fairly packet sent from neighboring mesh routers and mobile nodes, and it improves the reception ratio of multi-hop traffic of neighboring mesh routers. Therefore, the proposed scheme can reduce transmission delay and energy consumption. In order to improving the packet loss and the packet fairness, the proposed scheme uses the modified RTS/CTS under the IEEE 802.11 MAC protocol and reduces the packet loss by recognizing the packet size to send to the destination in advance. By using the simulation, we evaluated the proposed scheme in terms of the packet loss ratio and the number of received packet in each mesh router, and compare it to a traditional scheme.

• Proceedings of the Korea Information Processing Society Conference
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• 2002.11b
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• pp.1663-1666
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• 2002

PIM-SM은 특정 라우터를 RP(Rendezvous Point)로 설정하여 이 RP를 루트로 하고 한 그룹 내에서 공유하는 RPT(RP-rooted Shared Tree)를 사용한다. 초기에는 RPT를 통해 전송이 이루어지지만 마지막 홉 라우터에 트래픽 임계치의 값이 설정되어, 한 그룹에 대한 임계값이 초과되었을 경우 RPT에서 수신자가 소스를 루트로하는 SPT(Shotrest Path Tree)로 트리 전환을 시도한다. 그러나 RPT에서 SPT로 전환하는 동안 네트워크 내부의 전송상태 결여로 인해 패킷손실이 발생하고 소스의 데이터 패킷이 수신자에게 전달되지 못한 가능성이 발생한다. 본 연구에서는 유동적인 망의 전송되는 데이터의 양이 일시적으로 증가했을 때 임계치의 값에 따른 패킷 손실율을 조사하여 최적의 임계피를 찾아낸다.

• Proceedings of the Korea Information Processing Society Conference
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• 2007.05a
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• pp.1286-1287
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• 2007

무선 네트워크에서 TCP Vegas는 다양한 원인으로 발생하는 패킷 손실로 인해 전송률이 떨어진다. 다수의 패킷 손실이 발생하면 TCP Vegas의 혼잡제어 기법은 패킷 손실로 인해 혼잡윈도우가 증가되지 않는다. 본 논문은 TCP Vegas의 혼잡제어 기법을 수정하여 무선 네트워크에서 TCP Vegas의 성능을 향상시킨 wirelessVegas를 제안한다. wirelessVegas는 NS-2를 이용하여 시뮬레이션 한 결과 TCP Vegas 보다 1%의 무선 링크 에러율에서 약 20% 높은 성능을 보인다.

• Journal of Advanced Navigation Technology
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• v.15 no.2
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• pp.241-247
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• 2011

Most of network congestions are caused by TCP packet losses in the wired network ecosystems. On the contrary, high BER (Bit Error Rate), which is characteristic of the wireless networks, is a main factor in wireless network environments. Many wireless networks carry out the congestion control mechanisms frequently because they estimate that packet losses are not due to the wireless signal qualities but the congestion controls in their networks. To solve this problem, we propose the improved EWLN (Explicit Wireless Loss Notification) algorithm that more efficiently utilize the congestion window size to increase the wireless network throughput than legacy EWLN algorithm.

• Proceedings of the Korean Information Science Society Conference
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• 2004.10c
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• pp.175-177
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• 2004

무선망에서의 TCP 성능은 않은 연구가 되어 왔지만 손실율이 큰 경우의 TCP의 성능은 아직도 개선의 여지가 많다. 본 논문은 두 가지 점에서 새로운 TCP를 제안한다. 첫째, 혼잡제어(Congestion Control)와 손실제어(Loss Recovery)가 분리(Decoupled)되었다. 기존의 연구들은 무선손실과 혼잡손실을 구분하여야 한다는 점에는 많은 연구를 해왔지만 무선 손실에 대하여 어떻게 대처할 것 인가에 관한 문제는 상대적으로 간과하였다. 둘째, 기종의 TCP-Westwood와 TCP-Jersey에서 사용되는 ABE(Available Bandwidth Estimation)가 무선네트워크에 의해 발생하는 패킷 손실로 인해 부정확해지는 것을 지적하고 새로운 ABE방법을 제시한다. 시뮬레이션을 통하여 우수한 성능을 보여주었다.

• Journal of the Korea Society of Computer and Information
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• v.6 no.4
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• pp.154-162
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• 2001

In this paper, we investigate the scheduling algorithm of router system for Internet service based on the qualify-of-service (QoS) level of the input source traffics. We suggest an approprite scheduling algorithm in order to satisfy their QoS requirements for the loss-sensitive traffic and delay-sensitive traffic. For this purpose, we first study the service requirements of the multiplexer in Internet and the definition of QoS based on the ITU-T white recommendations. Second. we suggest a functional architecture of the multiplexer and the scheduling algorithm to satisfy various QoS requirements for Internet service. Finally. the performance measures of interest, namely steady-state packet loss probability and average delay, are discussed by simulation results.

• Journal of KIISE:Information Networking
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• v.27 no.2
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• pp.160-174
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• 2000

Nowadays, most widely used transport protocol, TCP is tuned to perform well in traditional networks where packet losses occur mostly because of congestion. TCP performs reliable end-to-end packet transmission under the assumption of low packet error rate. However, networks with wireless links suffer from significant losses due to high error rate and handoffs. TCP responds to all losses by invoking congestion control and avoidance algorithms, resulting in inefficient use of network bandwidth and degraded end-to-end performance in that system. To solve this problem, several methods have been proposed. In this paper, we analyse and compare these methods and propose appropriate model for improving TCP performance in the network with wireless links. This model uses TCP selective acknowledgement (SACK) option between TCP ends, and also uses caching method at the base station. Our simulation results show that using TCP SACK option with base station caching significantly reduces unnecessary duplicate retransmissions and recover packet losses effectively.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.5C
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• pp.490-496
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• 2006

H.264 is the most recent video coding standard, containing improved error resilience tools than previous video compression schemes. This paper shows an analysis of the dependency of error concealment (EC) performance on the expected number of correctly received neighboring macroblock(MB)s for a lost MB, applying error concealment schemes to the raster scan mode that is used in the previous video coding standard and the flexible macroblock ordering (FMO) which is one of error-resilience tools in H.264. We also present simulation results and performance evaluation with various packet loss rates. Simulation results show that the FMO mode provides better EC performances of $1{\sim}9dB$ PSNR improvements compared to the raster scan mode because of larger expected number of correctly received neighboring MBs. The PSNR improvement by FMO mode becomes higher as the intra-frame period is larger and the packet loss rate is higher.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.13 no.5
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• pp.967-973
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• 2009

Recently, as 3rd-generation mobile communication services using high-speed data rate system are widely employed, the demand for a variety of real-time data services such as VoIP service are also increased. Unlike typical data packets, VoIP packets have delay bound and low loss rate requirement. In this paper we propose a new scheduling algorithm that schedule two deferent kinds of packets efficiently, considering the characteristics of VoIP. Basically this algorithm considers both time delay and channel condition and gives priority depending on the time delay. Simulation results show that the proposed algorithm works more efficiently than conventional algorithms.

• 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.