• The Journal of Korean Institute of Communications and Information Sciences
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• v.36 no.10B
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• pp.1182-1191
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• 2011

In this paper, we propose a network-based IP mobility management scheme, called Access Independent Mobile Service with IPv4/IPv6 Dual Stack (AIMS-DS), which can provide high-quality multimedia services to IPv4/IPv6 dual-stack mobile nodes (MNs) without any interruption over various wireless/wired access networks. The proposed scheme provides an MN with a fast and reliable mobility service among heterogeneous wireless access networks through the network-based control, the complete separation method of control and data plane, the cross-layer (layer2 and layer3) interworking method for handover control acceleration, etc, In addition, the proposed AIMS-DS can provide seamless mobility service to an MN under the environments of IPv4/IPv6 coexisting networks through the home address mobility support and transport network support. Through performance evaluation with computer simulations, we have shown the superiority of the proposed AIMS-DS in terms of handover latency, packet 1085 and packet delivery latency.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.37 no.1C
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• pp.63-75
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• 2012

This paper researched the operation methods of software video teleconferencing on the tactical communication networks under 1Mbps. The tactical communication networks have limited bandwidths, frequent data losses and transmission delays due to the unstable networks. In addition, the bandwidth for video teleconferencing has to be much smaller since the Army Tactical Command Information System(ATCIS) has priority of using the bandwidth. This paper analyzed such restrictions of tactical communication networks, presented some methods to improve the quality of the software video teleconferencing on the tactical communication networks and their actual experiments as well. It is applied in the first place to re-transmit the lost packets and to reduce the image size for the data traffic. Nothing is better for the video teleconferencing than to provide the bandwidth enough for every user. However, on the tactical communication networks with the limited bandwidth, video teleconferencing can be improved by optimizing the compression rate of image data, the number of image frames, the audio codec and the usage of audio compensation data.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.8B
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• pp.839-845
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• 2009

The Vehicular Ad-hoc Network (VANET) has two main problems to be overcome due to high mobility and frequently changing density: one is short link duration time and the other is high packet loss ratio. To solve the problems, there have been many studies to predict vehicular mobility. Most of the studies try to enhance link expire time and link reliability, however the distance between two relay nodes becomes too short to have high network throughput. In this paper, we propose a new routing algorithm that considers both link expire time and network throughput in the VANET. The proposed algorithm aims to find path with long link expire time and high throughput. Our simulation results show that the proposed algorithm outperforms the legacy greedy algorithm and its variants.

• Convergence Security Journal
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• v.10 no.1
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• pp.41-48
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• 2010

Greedy protocols show good performance in Vehicular Ad-hoc Networks (VANETs) environment in general. But they make longer routes causing by surroundings or turn out routing failures in some cases when there are many traffic signals which generate empty streets temporary, or there is no merge roads after a road divide into two roads. When a node selects the next node simply using the distance to the destination node, the longer route is made by traditional greedy protocols in some cases and sometimes the route ends up routing failure. Most of traditional greedy protocols just take into account the distance to the destination to select a next node. Each node needs to consider not only the distance to the destination node but also the direction to the destination while routing a packet because of geographical environment. The proposed routing scheme considers both of the distance and the direction for forwarding packets to make a stable route. And the protocol can configure as the surrounding environment. We evaluate the performance of the protocol using two mobility models and network simulations. Most of network performances are improved rather than in compared with traditional greedy protocols.

• Journal of Digital Contents Society
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• v.18 no.2
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• pp.365-371
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• 2017

IEEE 802.11 Wireless Local Area Network (WLAN) adopts Power Saving Mode(PSM) to save the power. Unlike existing PSM, this paper proposes a new scheme for the power saving of the Mobile Host(MH) when a MH performs the data transmission after the competition-based DCF(Distributed Coordination Function) and competition free-based PCF(Point Coordination Function). In this paper, The proposed scheme estimates the distance between the MH with the authority of data transmission and the Access Point(AP) and then adaptively controls the power of the MH considering the distance. Through the simulation result, we find that the proposed scheme consumes the smaller transmission power and has the similar success rate of packet transmission when it is compared to the existing scheme which uses the same power without the consideration of the distance.

• Journal of the Korean Institute of Telematics and Electronics S
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• v.36S no.7
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• pp.1-8
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• 1999

In this paper, we have proposed and evaluated the performance of the routiong protocols servicing the mobile and fixed subscribers in a large e scale circuit switched telecommunication network, connected by gateways. The large scale network consists of several subnetworks, and a subnetwork is composed of $M{\times}N$ nodes in grid topology. When a call for mobile subscriber occurs, the current routing protocols search the whole large scale network to find a mobile subscriber. Therefore, it causes many redundant packets and long call setup delay. So, we have proposed a new routing protocol, in which the destination subscriber is first searched at the subnetwork where the call is proposed protocol is better than that of the current protocol.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.46 no.5
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• pp.44-56
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• 2009

Multimedia streaming service is susceptible to loss and delay of data as it requires high bandwidth and real time processing. Therefore QoS cannot be guaranteed due to data loss caused by heavy network traffic and error of wireless channel. To solve these problems, studies about algorithms which improve the quality of multimedia by serving differently according to the priority of packets in multimedia stream. Two algorithms are proposed in this paper. The first algorithm proposed is WMS-1(Wireless Multimedia Scheduling-1) algorithm which acts like IWFQ when any wireless loss is occurred but assigns channels first in case of urgent situation like when the running time of multimedia runs out. The second algorithm proposed is WMS-2(Wireless Multimedia Scheduling-2) algerithm that assigns priority to multimedia flow and schedules flow that has higher priority according to type of frame first. The comparison with other existing scheduling algorithms shows that multimedia service quality of the proposed algorithm is improved and the larger the queue size of base station is, the better total quality of service and fairness were gained.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.46 no.5
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• pp.37-43
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• 2009

Streaming video service over wireless and mobile communication networks has received significant interests from both academia and industry recently. Specifically, mobile WiMAX (IEEE 802.16e) is capable of providing high data rate and flexible Quality of Service (QoS) mechanisms, supporting mobile streaming very attractive. However, we need to note that streaming videos can be partially deteriorated in their macroblocks and/or slices owing to errors on OFDMA subcarriers, as we consider that compressed video sequence is generally sensitive to the error-prone channel status of the wireless and mobile network. In this paper, we introduce an OFDMA subcarrier-adaptive mobile streaming server based on cross-layer design. This streaming server system is substantially efficient to reduce the deterioration of streaming video transferred on the subcarriers of low power strength without any modifications of the existing schedulers, packet ordering/reassembly, and subcarrier allocation strategies in the base station.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.45 no.5
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• pp.125-133
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• 2008

In this paper, we propose active queue management mechanism (Active-WRED) to guarantee quality of the high priority service class in multi-class traffic service environment. In congestion situation, this mechanism increases drop probability of low priority traffic and reduces the drop probability of the high priority traffic, therefore it can improve the quality of the high priority service. In order to analyze the performance of our mechanism we introduce the stochastic analysis of a discrete-time queueing systems for the performance evaluation of the Active Queue Management (AQM) based congestion control mechanism called Weighted Random Early Detection (WRED) using a two-state Markov-Modulated Bernoulli arrival process (MMBP-2) as the traffic source. A two-dimensional discrete-time Harkov chain is introduced to model the Active-WRED mechanism for two traffic classes (Guaranteed Service and Best Effort Service) where each dimension corresponds to a traffic class with its own parameters.

• Journal of the Korea Society of Computer and Information
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• v.12 no.1 s.45
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• pp.137-144
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• 2007

Advances in networking technology digital media, and codecs have made it possible for the Internet evolves into a Broadband convergence Network (BcN) and provides various services including Voice over Internet Protocol (VoIP) and IPTV over their high-speed IP networks. In order for the Internet to make a profit as traditional Public Switched Telephone Network (PSTN), it must provide high qualify VoIP services. Therefore, real time qualify measurement framework is the most important requisite to provide VoIP service. For this, IETF (Internet Engineering Task Force) defined RTCP-Extended Reports (RTCP-XR) that extend RTCP (Real-Time Transport Protocol Control Protocol). However, procedure and method tot actually VoIP qualify measurement did not recommended nothing but defined item to measure voice quality. Our objective in this paper is to describes a practical measuring framework for end-to-end QoS of switched voice packet in an IP environment. It includes concepts as well as step-by-step procedures for measuring packetized voice streams. It also proposes new formats that extend RTCP-XR's concept.