• Journal of KIISE:Information Networking
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• v.28 no.4
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• pp.517-526
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• 2001

In order to reduce the increasing packet loss rates caused by an exponential increase in network traffic, the IETF(Internet Engineering Task Force) is considering the deployment of active queue management techniques such as RED(Random Early Detection). While active queue management in routers and gateways can potentially reduce total packet loss rates in the Internet, this paper has demonstrated the inherent weakness of current techniques and shows that they are ineffective in preventing high loss rates. The inherent problem with these queue management algorithms is that they all use queue lengths as the indicator of the severity of congestion. In this paper, in order to solve this problem, a new active queue management algorithm called MRED(Modified Random Early Detection) is proposed. MRED computes the packet drop probability based on our heuristic method rather than the simple method used in RED. Using simulation, MRED is shown to perform better than existing queue management schemes. To analyze the performance, we also measure throughput of traffics under the FIFO control, and compared the performance with that of this MRED system.

• Journal of the Institute of Electronics and Information Engineers
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• v.52 no.3
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• pp.25-31
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• 2015

In this paper, we propose PEP based on cross-layer and network coding in digital video broadcasting-return channel via satellite (DVB-RCS) networks. In particular, we propose not only a scheme of cross-layer information exchange for interaction between TCP and the resource allocation (RA) scheme in the link layer but also a tuning algorithm for the TCP contention window (CWND) by using information on the RA in the link layer and the redundancy rate for network-coded packets. The simulation results show that TCP CWND can be adjusted by RA information in the proposed protocol. Furthermore, through the transmission of network-coded packets and the proposed CWND tuning algorithm, TCP throughput is enhanced in lossy environment due to user mobility without the unnecessary resource overhead.

• The Journal of the Korea Contents Association
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• v.10 no.4
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• pp.55-65
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• 2010

We present a method for increasing network lifetime without link failure due to lack of battery capacity of nodes in wireless ad-hoc networks with low mobility. In general, a node with larger remaining battery capacity represents the one with lesser traffic load. Thus, a modified AODV routing protocol is proposed to determine a possible route by considering a remaining battery capacity of a node. Besides, the total energy consumption of all nodes increase rapidly due to the huge amount of control packets which should be flooded into the network. To reduce such control packets efficiently, a source node can store information about alternative routes to the destination node into its routing table. When a link failure happens, the source node should retrieve the route first with the largest amount of the total remaining battery capacity from its table entries before initiating the route rediscovery process. To do so, the possibility of generating unnecessary AODV control packets should be reduced. The method proposed in this paper increases the network lifetime by 40% at most compared with the legacy AODV and MMBCR.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.2B
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• pp.44-52
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• 2005

This paper proposes a neural network prediction model to improve the valid packet transmission rate for the QoS(Quality of Service) of multimedia transmission. The Round Trip Time(RTT) and Packet Loss Rate(PLR) are predicted using a neural network and then the transmission rate is decided based on the predicted RTT and the PLR. The suggested method will improve the transmission rate since it uses the rate control factors corresponding to time of data is being transmitted, while the conventional one uses the transmission rate determined based on the past informations. An experimental set-up has been established using a Linux PC system, and the multimedia data are transmitted using UDP protocol in real time. The valid transmitted packets are about 5% higher than the one in the conventional TCP-Friendly congestion control method when the suggested algorithm was applied.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.21 no.6
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• pp.1127-1136
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• 2017

In wireless networks, the packet loss due to the bit error is misinterpreted as loss due to the congestion state, so TCP congestion control occurs frequently and performance degradation occurs. This degradation also occurs in MPTCP(Multipath TCP), which is an extension protocol of original TCP. In MPTCP, the overall performance of the multipath is degraded. In this paper, we propose a congestion control scheme which measures the bandwidth on each path of MPTCP and reduces the congestion window size by the measured bandwidth when packet loss occurs, in order to solve the MPTCP performance degradation in the wireless environment. We also implemented the proposed congestion control in the Linux kernel and compared it with the original MPTCP in the testbed and real wireless networks. Experimental results show that the proposed congestion control has better throughput performance than original MPTCP congestion control in the wireless environment.

• Journal of Korea Multimedia Society
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• v.10 no.8
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• pp.1039-1051
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• 2007

In this paper, a new lip synchronization control method is proposed to achieve very low delay in the video communication. The lip control is so much vital in video communication as delay reduction. In a general way, to control the lip synchronization, both the playtime and capture time calculated from RTP time stamp are used. RTP timestamp is created by stream sender and sent to the receiver along the stream. It is extracted from the received packet by stream receiver to calculate playtime and capture time. In this paper, we propose the method of searching most adjacent corresponding frame of the audio signal, which is assumed to be played with uniform speed. Encoding buffer of stream sender is removed to reduce the buffering delay. Besides, decoder buffer of receiver, which is used to correct the cracked packet, is resulted to process only 3 frames. These mechanisms enable us to achieve ultra low delay less than 100 ms, which is essential to video communication. Through simulations, the proposed method shows below the 100 ms delay and controlled the lip synchronization between audio and video.

• Journal of Korea Multimedia Society
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• v.7 no.5
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• pp.698-712
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• 2004

DiffServ network architecture does not define a call admission control procedure explicitly. In this paper, a new DiffServ QoS control mechanism is suggested which, after the call admission control, can execute packet process according to the class while, at the same time, executing on the flow based call admission control in the DiffServ network. Routers on the path from the source to the destination estimate the aggregated class traffic of the existing flows and the new incoming flow and then, perform a call admission control in accordance with the type of classes efficiently based on the required bandwidth per each class that can meet the user's QoS requirements. In order to facilitate the packet process according to the class after the flow based call admission control, a mechanism is suggested that can adjust the network resources to classes dynamically. The performance analysis on this mechanism is carried out through a simulation.

• Journal of Advanced Navigation Technology
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• v.14 no.1
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• pp.49-56
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• 2010

Severe energy constraints and the low power consumption require the significance of the energy efficient error control mechanisms in wireless sensor network (WSN). In this paper, an automatic repeat request (ARQ) methodology for the analysis of error control schemes in WSN is presented such that the effects of packet length, the modulation scheme and the interference effect of the wireless channel are investigated. Moreover, an analyis of ARQ error control is provided by considering two major architectures for wireless sensor network, i.e., Mica2 and MicaZ sensor nodes. And the throughput performance of WSN with asynchronous FSK signal and DSSS-OQPSK signal with selective repeat ARQ scheme are analyzed in multiple interference environment, and the probability of receiving a correct bit and packet from target node to sink node is evaluated as a function of the channel parameter, the number of wireless sensor node, and the spreading factor.

• Journal of Internet Computing and Services
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• v.6 no.5
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• pp.1-9
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• 2005

In a role based access control through attribute certificate, the use of role assignment certificates and role specification certificates can reduce management cost and the overhead incurred by changing roles, Highly distributed computing environments such as ubiquitous computing environments not having global or broad control. need another attribute certificate management technique, Actually just having role specification certificate separately reduce management cost, But for better performance we structure role specification, We group roles and make the role group relation tree, It results secure and efficient role renewing and distribution, For scalable role specification certificate distribution, the multicasting of packets is used, We take into account the packet lass and quantify performance enhancements of structuring role specification certificates.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.49 no.2
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• pp.1-8
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• 2012

We propose a new frame length control scheme for cognitive radio systems. In the proposed algorithm, the property that the probability of a primary packet arrival continually increases as time elapses during the primary user's idle period is utilized. When reducing the secondary frame length in accordance with the increasing primary packet arrival probability, spectrum sensing is conducted more frequently as time elapses and the primary packet arrival is detected more quickly. We obtain the optimal frame lengths by maximizing the secondary throughput while satisfying the interference constraint. Numerical results show a significant throughput enhancement via the frame length control.