• Journal of the Institute of Electronics Engineers of Korea CI
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• v.48 no.4
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• pp.35-42
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• 2011

Link quality assessment is a crucial part of sensor network formation to stably operate large-scale wireless sensor networks (WSNs). A stability of path consisting of several nodes strongly depends on all link quality between pair of consecutive nodes. Thus it is very important to assess the link quality on the stage of building a routing path. In this paper, we present a link quality assessment method, Hybrid Link Quality Metric (HQLM), which uses both of LQI and RSSI from RF chip of sensor nodes to minimize set-up time and energy consumption for network formation. The HQLM not only reduces the time and energy consumption, but also provides complementary cooperation of LQI and RSSI. In order to evaluate the validity and efficiency of the proposed method, we measure PDR (Packet Delivery Rate) by exchanging multiple messages and then, compare PDR to the result of HQLM for evaluation. From the research being carried out, we can conclude that the HQLM performs better than either LQI- or RSSI-based metric in terms of recall, precision, and matching on link quality.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.41 no.8
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• pp.974-985
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• 2016

Riverbed Modeler, which is a commercial packet-level discrete event simulator is used to model, design, and simulate complicated communication protocols and large-scale network. Riverbed Modeler got credit for its reliability in field of network simulation. In the MANET simulation environment using Riverbed Modeler, it is very complicated to add a new routing protocol into existing architecture of routing protocols because it is required lots of modifications of protocol recognition. In this paper, we propose Routing Adding Framework which can reduce errors or mistakes during modifying the existing routing support architecture. Routing Adding Framework is provided as a adapter API for protocol recognition. and it is only minimum modifications for protocol identifiers when a new routing protocol is added to the child process of manet_mgr process which manages routing protocols for IP layer. With Routing Adding Framework, we can reduce less than half modification. Then, we shows an example of implementation of a hybrid routing protocol AntHocNet using Routing Adding Framework, and we verify its design and application of the Routing Adding Framework by obtaining simulation result with similar result given by AntHocNet.

• The KIPS Transactions:PartC
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• v.8C no.1
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• pp.60-67
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• 2001

Multicast Protocol for multimedia service on the Internet can be classified into two types, e.g., source based tree and shared tree according to difference of tree construction method. Shared tree based multicast is known to show outstanding results in the aspect of scalability than source based tree. Generally, There have been lots of researches on the method to satisfy QoS constraints through proper Rendezvous Point (RP) in the shared tree. In addition, as the multicast group members join and leave dynamically in the service time, RP of the shared tree should b be reselected for guranteeing Qos to new member, But, RP reselection method has not been considered generally as the solution to satisfy QoS C constraints. In this paper, new initial RP selection and RP reselection method are proposed, which utilize RTCP (Real Time Control Protocol) report packet fields. Proposed initial RP selection and RP reselection method use RTCP protocol which underlying multimedia application service So, the proposed method does not need any special process for collecting network information to calculate RP. New initial RP selection method s shows better performance than random and topology based one by 40-50% in simulation. Also, RP reselection method improves delay p performance by 50% after initial RP selection according to the member’s dynamicity.

• Journal of KIISE:Information Networking
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• v.35 no.1
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• pp.46-55
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• 2008

As a mobility support for IP have studied, Internet Engineering Task Force(IETF) standardized the Mobile IPv6(MIPv6) protocol. When a mobile node moves between subnets, MIPv6 maintains connectivity to network and supports seamless communication, and these processes are called a Handover. Whenever the mobile node moves between subnets, the Handover is performed. The mobile node can not communicate during the Handover. This period is Galled Handover latency. To reduce this latency, mipshop working group standardizes Fast Handovers for Mobile IPv6(FMIPv6), but latency which the mobile node registers its new care-of address to a home agent and a correspondent node is still long. To solve this problem, we propose a scheme that the mobile node registers the new care-of address to the home agent and initiates Return Routability procedure in advance during layer 2 handover, based on FMIPv6 and IEEE 802.11. We analyze MIPv6, FMIPv6 and the proposed scheme in term of packet transmission cost during the Handover. Compared to MIPv6 the proposed scheme gains 79% improvement, while it gains 31% improvement compared to FMIPv6.

• Journal of KIISE:Software and Applications
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• v.35 no.3
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• pp.197-206
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• 2008

In the mobile Internet environment, frequent communications between a mobile device and a content server are required for searching or downloading learning materials. In this paper, we propose an efficient prefetching technique to reduce the network cost and to improve the communication efficiency in the mobile dictionary. Our prefetching-based approach can be explained as follows. First, we propose an overall framework for the prefetching-based mobile dictionary. Second, we present a systematic way of determining the amount of prefetching data for each of packet-based and flat-rate billing cases. Third, by focusing on the English-Korean mobile dictionary for middle or high school students, we propose an intuitive method of determining the words to be prefetched in advance. Fourth, based on these determination methods, we propose an efficient prefetching algorithm. Fifth, through experiments, we show the superiority of our prefetching-based method. From this approach, we can summarize major contributions as follows. First, to our best knowledge, this is the first attempt to exploit prefetching techniques in mobile applications. Second, we propose a systematic way of applying prefetching techniques to a mobile dictionary. Third, using prefetching techniques we improve the overall performance of a network-based mobile dictionary. Experimental results show that, compared with the traditional on-demand approach, our prefetching based approach improves the average performance by $9.8%{\sim}33.2%$. These results indicate that our framework can be widely used not only in the mobile dictionary but also in other mobile Internet applications that require the prefetching technique.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.43 no.10 s.352
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• pp.8-17
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• 2006

Recently, the high-speed Internet users increase rapidly and broadband networks have been widely deployed. However, the current TCP congestion control algorithm was designed for relatively narrowband network environments, and thus its performance is inefficient for traffic transport in broadband networks. To remedy this problem, the TCP having an enhanced congestion control algorithm is required for broadband networks. In this paper, we propose an improved TCP congestion control that can sufficiently utilize the large available bandwidth in broadband networks. The proposed algorithm predicts the available bandwidth by using ACK information and RTT variation, and prevents large packet losses by adjusting congestion window size appropriately. Also, it can rapidly utilize the large available bandwidth by enhancing the legacy TCP algorithm in congestion avoidance phase. In order to evaluate the performance of the proposed algorithm, we use the ns-2 simulator. The simulation results show that the proposed algorithm improves not only the utilization of the available bandwidth but also RTT fairness and the fairness between contending TCP flows better than the HSTCP in high bandwidth delay product network environment.

• Journal of IKEEE
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• v.5 no.2 s.9
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• pp.136-145
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• 2001

As with appearing new applications that requires QoS guarantee such as VoIP, VPN in Internet, problems of IP QoS has been one of most important issues in next-generation Internet. IETF has proposed integrated services model(Int-Serv) and differentiated service(Diff-Serv) to supply IP QoS in Internet. Int-Serv model uses the state information of each IP flow, so satisfies QoS according to traffic characteristics, but increases the amount of flow state information with increasing flow number. Diff-Serv model uses PHP(Per Hop Behavior), and there are well-defined classes to provide differentiated traffics with different services according to delay and loss sensitivity. Diff-Serv model can provide diverse services in Internet because of having no the state and signal information of each flow. As MPLS uses the packet forwarding technique based-on label, it implements the traffic engineering in the networks easily. The MPLS can set up the path with different traffic parameters, and assign each path to particular Class of Services. Therefore it is possible to support the Diff-Serv model with well-defined classes. In this paper we investigate the performance improvement of Diff-Serv function in the MPLS network to guarantee class of services in Internet.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.47 no.7
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• pp.118-127
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• 2010

Current TCP/IP-based Internet architecture has been used for over 30 years, however it will confront with fundamental problems due to new protocol extension limitation since communication environments will change drastically and various user requirements will be emerging in near future. To solve these problems, major countries have started Future Internet researches based on clean slate approach and they will deploy large-scale testbed to experiment and verify new functions. OpenFlow switch technology has been proposed as a new experimental technology for independent protocol that can utilized the legacy network devices and does not interfere with the production Internet traffic. Korea also started Future Internet testbed project called FIRST and OpenFlow switch with NetFPGA card will be used to deploy this testbed. To interconnect distributed testbed using OpenFlow switches, logical tunnel should be established by encapsulating MAC frame inside a unicast IP packet between OpenFlow switches because OpenFlow switches are not directly connected. In this paper, we have implemented a NetFPGA-based that performs MAC in IP tunneling between various OpenFlow switch sites implemented in domestic research network KOREN. The performance evaluation shows that the NetFPGA-based capsulator reveals better performance than the software-based tunneling and it can be utilized as a testbed for experimentation of Future Internet technologies.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.12 no.5
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• pp.267-273
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• 2012

The short-range communication module was developed in this paper which is suitable for the point-to-multipoint circumstances of 40 terminals communicating frequently in about 50 meters. The media access control layer of this communication module using asynchronous time-division multiplexing provides a fast and robust performance even in the worst case of simultaneous transmission events, and low packet error rate was measured a in LOS (Line-of-sight) circumstance by adding the function of acknowledge response to media access control layer. The difference test was carried out in order to measure the performance of point-to-multipoint communication. Two communication modules are respectively measured and graphed in 10 branches to 40 branches. The communication module developed in this paper showed a faster performance than the commercial Zigbee module in the specific case presented in this paper. Especially, in over 20 branches showed wide differences of the transmission speed. This results is caused by more network overhead of Zigbee whose wider applications needs the network layer and applicaiton layer besides media access control layer. Also, the asynchronous time-division multiplexing proposed in this paper are more suitable than CSMA-CA of Zigbee module when a lot of module ought to be frequently communicated in small area.

• Journal of KIISE:Information Networking
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• v.31 no.3
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• pp.269-279
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• 2004

The most 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 worse the network congestion. Recently, Brakmo and Peterson have proposed a new version of TCP, which is named TCP Vegas, with a fundamentally different congestion control scheme from that of the Reno. Many studies indicate that the Vegas is able to achieve better throughput and higher stability than the Reno. But there are two unfairness problems in Vegas. These problems hinder the spread of the Vegas in current Internet. In this paper, in order to solve these unfairness problems, we propose a new congestion control algorithm called TCP PowerVegas. The existing Vegas depends mainly only on the rtt(round trip time), but the proposed PowerVegas use the new congestion control scheme combined the Information on the rtt with the information on the packet loss. Therefore the PowerVegas performs the congestion control more competitively than the Vegas. Thus, the PowerVegas is able to solve effectively these unfairness problems which the Vegas has experienced. To evaluate the proposed approach, we compare the performance among PowerVegas, Reno and Vegas under same network environment. Using simulation, the PowerVegas is able to achieve better throughput and higher stability than the Reno and is shown to achieve much better fairness than the existing Vegas.