• Proceedings of the KIEE Conference
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• 2006.07d
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• pp.1828-1829
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• 2006

In this paper, we propose sliding mode congestion controller for differentiated-services network. Two important issue in differentiated-services architecture are bandwidth guarantee and fair sharing of unsubscribed bandwidth among TCP flows with and without bandwidth reservation. We use tight upper and lower bounds for various settings of differentiated-services parameters using the loss-bounded model. The Sliding mode congestion controller scheme is designed using nonlinear control theory based on a nonlinear model of the network that is generated using fluid flow consideration. The methodology used is general and independent of technology, as for example TCP/IP or ATM. The sliding mode congestion controller methodology has been applied to an TCP network. We use NS-2 simulation to demonstrate that the proposed control methodology achieves the desired behavior of the network, and possesses important attributes. as e.g, stable and robust behavior, high utilization with bounded delay and loss, together with good steady-state and transient behavior.

• Journal of the Korea Society of Computer and Information
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• v.12 no.3
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• pp.177-183
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• 2007

Active queue management (AQM) refers to a family of packet dropping mechanisms for router queues that has been proposed to support end-to-end congestion control mechanisms in the Internet. The proposed AQM algorithm by the IETF is Random Early Detection (RED). The RED algorithm allows network operators simultaneously to achieve high throughput and low average delay. However. the resulting average queue length is quite sensitive to the level of congestion. In this paper, we propose the Refined Adaptive RED(RARED), as a solution for reducing the sensitivity to parameters that affect RED performance. Based on simulations, we observe that the RARED scheme improves overall performance of the network. In particular, the RARED scheme reduces packet drop rate and improves goodput.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.6A
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• pp.667-667
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• 2004

We must design the buffer algorithm that protects traffic congestion and decreasing throughput at satellite communication network. It is important that buffer algorithm is satisfied with the good performance of transmission packet, responsibility of many connecting traffic and the QOS for connecting character. Old buffer algorithms are not the suitable algorithms when we have the satellite communication network environment. RED buffer algorithm is proposed by Floyd. It has a better performance than old buffer algorithm. But this algorithm is not well adapted a number of connecting TCP packet and changing network, so this algorithm has a bad performance on satellite communication network that is many of connecting user at same time. This paper propose the TARED(Target Adaptive RED). It has a good performance, adaptation and stability on satellite communication network and has not overflow and underflow of the buffer level.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.6A
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• pp.666-676
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• 2004

We must design the buffer algorithm that protects traffic congestion and decreasing throughput at satellite communication network. It is important that buffer algorithm is satisfied with the good performance of transmission packet, responsibility of many connecting traffic and the 005 for connecting character. Old buffer algorithms are not the suitable algorithms when we have the satellite communication network environment. RED buffer algorithm is proposed by Floyd. It has a better performance than old buffer algorithm. But this algorithm is not well adapted a number of connecting TCP packet and changing network, so this algorithm has a bad Performance on satellite communication network that is many of connecting user at same time. This paper Propose the TARED(Target Adaptive RED). It has a good performance, adaptation and stability on satellite communication network and has not overflow and underflow of the buffer level.

• Journal of KIISE:Information Networking
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• v.29 no.3
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• pp.242-252
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• 2002

When a Mobile IP handoff occurs, the packets in flight can be lost because they are tunneled based on out-of-date location information. In this parer, we propose an enhanced handoff protocol that achieves no packet loss during Mobile IP handoff over a wireless LAN. Our handoff protocol predicts the next foreign agent that a mobile host is to visit by using: the information from the data link layer of wireless LAN. After that, when a Mobile IP handoff occult, the current foreign agent forwards the packets destined to a mobile host to the predicted foreign agent which buffers them. This eliminates packet loss and reduces the packet forwarding delay. Our handoff protocol is simulated using the Network Simulator-2 (ns-2) and shows the substantial performance enhancement of TCP with much less overhead up to 6.2 times compared to standard Mobile IP.

• Journal of Information Processing Systems
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• v.2 no.2
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• pp.76-81
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• 2006

A PDA is used mainly for downloading data from a stationary server such as a desktop PC in an infrastructure network based on wireless LAN. Thus, the overall performance depends heavily on the performance of such downloading with PDA. Unfortunately, for a PDA the time taken to receive data from a PC is longer than the time taken to send it by 53%. Thus, we measured and analyzed all possible factors that could cause the receiving time of a PDA to be delayed with a test bed system. There are crucial factors: the TCP window size, file access time of a PDA, and the inter-packet delay that affects the receiving time of a PDA. The window size of a PDA during the downstream is reduced dramatically to 686 bytes from 32,581 bytes. In addition, because flash memory is embedded into a PDA, writing data into the flash memory takes twice as long as reading the data from it. To alleviate these, we propose three distinct remedies: First, in order to keep the window size at a sender constant, both the size of a socket send buffer for a desktop PC and the size of a socket receive buffer for a PDA should be increased. Second, to shorten its internal file access time, the size of an application buffer implemented in an application should be doubled. Finally, the inter-packet delay of a PDA and a desktop PC at the application layer should be adjusted asymmetrically to lower the traffic bottleneck between these heterogeneous terminals.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.13 no.7
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• pp.3470-3493
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• 2019

IEEE 802.11s-based infrastructure Wireless Mesh Networks (iWMNs) are envisaged as a promising solution to provide ubiquitous wireless Internet access. The limited network capacity is a problem mainly caused by the medium contention between mesh users and the mesh access points (MAPs), which gets worst when the mesh clients employ the Transmission Control Protocol (TCP). To mitigate this problem, we use wireless network coding (WNC) in the MAPs. The aim of this proposal is to take advantage of the network topology around the MAPs, to alleviate the contention and maximize the use of the network capacity. We evaluate WNC when is used in MAPs. We model the formation of coding opportunities and, using computer simulations, we evaluate the formation of such coding opportunities. The results show that as the users density grows, the coding opportunities increase up to 70%; however, at the same time, the coding delay increments significantly. In order to reduce such delay, we propose to adaptively adjust the time that a packet can wait to catch a coding opportunity in an MAP. We assess the performance of moving-average estimation methods to forecast this adaptive sojourn time. We show that using moving-average estimation methods can significantly decrease the coding delay since they consider the traffic density conditions.

• Journal of information and communication convergence engineering
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• v.7 no.3
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• pp.309-313
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• 2009

In this paper, end-to-end VoIP(Voice over Internet Protocol) performance is evaluated by simulation with NS-2 simulation tool. There are many results studied and published for VoIP performance over TCP/IP networks. But, almost all of them were focused on wired or wireless Internet environments. About MANET (Mobile Ad Hoc Network), VoIP is currently studying several points of research. In this paper, analysis of VoIP performance is done with focusing on the mobility of MANETs. MOS(Mean Opinion Score), network delay, packet loss rates are considered as end-to-end QoS performance parameters.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.14 no.6
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• pp.75-84
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• 2014

In order to solve the problems such as overload, single point of failure, non-optimized data path, and network scalability in conventional central mobility management protocols, distributed mobility management schemes have been continually studied in and around the IETF. In this paper, a network-based partially-distributed mobility management mechanism, pDMMv6, is suggested and the performance comparison with traditional protocols such as PMIPv6 and MIPv6 is made through simulation under the various user traffic environment. The simulation results include UDP packet delivery ratio, end-to-end packet delay, binding delay for registration signaling, CPU utilization in each node, and response delays in several server-client TCP applications such as web browsing, e-mail, telnet remote login, FTP file up/down-load, and database access.

• Journal of the Institute of Electronics and Information Engineers
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• v.52 no.5
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• pp.3-9
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• 2015

The wireless communication system adopts an appropriate retransmission scheme on each system protocol layer to improve reliability of data transmission. In each system protocol layer, the retransmission scheme operates in independently other layers and operates based on the parameters without reference to end-to-end performance of wireless communication system. For this reason, it is difficult to design the optimal system parameters that satisfy the QoS requirements for each service class. Thus, the performance analysis of wireless communication system is needed to design the optimal system parameters according to the end-to-end QoS requirements for each service class. In this paper, we derive the mathematical model to formulate the end-to-end performance of wireless communication system. We also evaluate the performance at the MAC and transport layers in terms of average spectral efficiency and average transmission delay. Based on the results of performance evaluations, we design the optimal system parameters according to the QoS requirements of service classes. From the results, the HARQ combined with AMC is appropriate for the delay-sensitive service and the ARQ combined with AMC is appropriate for a service that is insensitive to transmission delay. Also, the TCP can be applied for the delay-insensitive service only.