• Journal of IKEEE
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• v.7 no.1 s.12
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• pp.97-106
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• 2003

In ARQ based error control, imperfect error detection leaves error remains on a packet. Aiming for a reduction of error remains in multicopy transmission ARQ system, we propose a rule of requesting a retransmission and deciding a correct copy, (identified as $(m, \;{\sigma})$rule). While the probability of error remains is reduced by the employment of the $(m, \;{\sigma})$ rule at multicopy transmission ARQ, delay and throughput performance may be degraded in comparison with those of conventional single copy transmission ARQ. Thus, we develop an analytical method to evaluate the performance trade-off in multicopy transmission ARQ following the $(m, \;{\sigma})$ rule. From the numerical results obtained by the analytical method, we investigate the effect of channel characteristics on the performance of error remains, packet loss, throughput, and packet delay, and confirm that the adaptability of the $(m, \;{\sigma})$ rule to conform to various QoS requirements with ease.

• Journal of KIISE:Information Networking
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• v.36 no.1
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• pp.34-42
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• 2009

In this paper, we propose a practical hybrid transport error control scheme to provide SVC video streaming service over error-prone IP networks. Many error control mechanisms for various video coding standards have been proposed in the literature. However, there is little research result which can be practically applicable to the multilayered coding structure of SVC(the scalable extension of H.264/AVC). We present a new hybrid transport error control scheme that efficiently combines layered Forward Error Correction(FEC) and Automatic Repeat Request(ARQ) for better packet-loss resilience. In the proposed hybrid error control, we adopt ACK-based ARQ instead of NACK-based ARQ to maximize throughput which is the amount of effective data packets delivered over a physical link per time unit. In order to prove the effectiveness of the proposed hybrid error control scheme, we adopt NIST-Net network emulator which is a general-purpose tool for emulating performance dynamics in IP networks. It is shown by simulations over the NIST-Net that the proposed hybrid error control scheme shows improved packet-loss resilience even with much less number of overhead packets compared to various conventional error control schemes.

• 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 Korean Institute of Communications and Information Sciences
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• v.21 no.4
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• pp.911-925
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• 1996

Emerging service classes need an error model which efficiently satisfies the requirements for each media and application. However limited flexibility in the error control has been provided. Therefore we propose a new error control model which applies error control scheme per media flexibly with respect to ETL(Error Tolerance Leverl) and delay. In order to select the error control scheme in this model, the attributes of media(i.e., ETL, isochronism), the attributes of application(i.e., delay) and environment parameters(i.e., packet loss ration, network types, connection modes of the application) are considered as effect factors.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.14 no.5
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• pp.1139-1145
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• 2010

Wireless mesh network is similar to ad-hoc network, so when transferred to the data packet in the wireless environment, interfered factor arise. When TCP(Transport Control Protocol) was created, however as it was design based on wired link, wireless link made more transmission error than wired link. It is existent problem that TCP unfairness and congestion collapse over wireless mesh network. But packet losses due to transmission errors are more frequent. The cause of transmission error in wireless ad-hoc network may be inexactly regarded as indications of network congestion. And then, Congestion Control Algorithm was running by this situation causes the TCP performance degradation. In this paper, proposed TCP can adaptively regulate the congestion window through moving node in the Wireless Mesh Network. And it enhanced the performance.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2002.11a
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• pp.605-609
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• 2002

Reliable multicast data transmission in a 1:N environment needs more sophisticated error control mechanism than that of in 1:1 environment due to ACK implosion and duplicated retransmission. Although there have been many related research on error control in reliable multicast, real implemented protocols are rare. As one of the reliable multicast transport protocols, ECTP is selected as an international standard reliable multicast protocol by ITU-T and ISO and implemented on RedHat 7.2 machine by us. In this paper, we evaluate the performance of the error control mechanism in the respect of throughput and generated control packet numbers with a real implementation code. From the results, it is concluded that the suitable values of error control parameters can be obtained from the local group size and network environments.

• Proceedings of the IEEK Conference
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• 2004.06a
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• pp.111-114
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• 2004

Sensor networks are deployed with a limited energy source. Thus, energy efficient design can be challenging. This paper has been studied optimal packet length with energy efficiency for sensor networks. And using Power Management can not improve energy efficiency. Power Management is turning off transceiver when transceiver is idle statue. We show that BCH code for error control can improve energy efficiency better than Convolutional code.

• 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.

• Journal of the Korean Institute of Telematics and Electronics A
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• v.32A no.2
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• pp.11-18
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• 1995

Error detection in ARQ(Automatic Repeat Request) protocols is very important in wireless data communication systems. The throughput efficiency of ARQ protocols can be improved by dynamically adapting the protocol packet length so that it approaches the optimum value for throuhput efficiency. In this paper, a simple and novel adaptive packet length allocation method is proposed which transmits the packets with variable length by dyanmically estimating the channel codition. The simulation results show that the average of throughput is improved by 315.4% in the stop-and-wait protocol, 41.4% in the go-back-N protocol and 155.9% in the selective repeat protocol respectively. And the throughput performances of adaptive packet length allocation method approximately approach the theoritically optimal throughput performances.

• IE interfaces
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• v.4 no.1
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• pp.63-69
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• 1991

In this paper, the Go-Back-N ARQ protocol with decoding in communication network is considered. The time delay and throughput are respectively analyzed as a function of window size and decoding time out. Packets arrive continuously at the decoder, and are stored in a buffer if the decoder is busy upon its arrival. The decoder devotes no more than a time-out period of predetermined length to the decoding of any single packet. If packet decoding is completed within that period, the packet leaves the system. Otherwise, it is retransimitted and its decoding starts anew. The time delay and throughput are obtained using recursive formula and difference equation. An appropriate time out and window size that satisfies the grade of service can be determined.