• Journal of the Institute of Electronics Engineers of Korea TC
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• v.47 no.12
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• pp.69-77
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• 2010

This paper presents a fast IP mobility management scheme in heterogeneous networks using the multiple wireless network interlaces. More specifically, in order to minimize the packet loss and handover latency due to handover, the E-HMIPv6, IETF HMIPv6 has been extended, is presented where the multiple tunnels between E-MAP and mobile node are dynamically constructed. E-HMIPv6 is composed of the extension of IETF HMIPv6 MAP, handover procedure, and simultaneous multiple tunnels. In order to demonstrate superior to the proposed method, the NS-2 simulation has done for performance evaluation of TCP and UDP-based application comparison with the existing mobility management method.

• The Journal of the Acoustical Society of Korea
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• v.21 no.2
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• pp.93-101
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• 2002

This paper presents the Internet audio broadcasting technology based on the streaming technology. In this paper, we choose the MPEG-2 AAC for multimedia data, and for the streaming of this data we use RTP/RTCP protocol. We use RTSP protocol for the control of streaming data and TCP/IP for the exchange of information between server and client. By using all of these protocols and MPEBG-2 AAC, we explain the implementation method for the unicast/multicast streaming server/client system. Our system was tested by ETRI intranet, which is connected by 2000 researchers. Experimental result show that our system can be process the packet loss and jitter by retransmission and variable length buffer. Multicast streaming server can be used for the audio broadcasting service inside the company, unicast streaming server can be used for the AOD (Audio On Demand) service.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.17 no.2
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• pp.719-723
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• 2016

Owing to the recent increased interest in wireless communication technology and rapid technology development, a range of applied technologies utilizing them are being released. In addition, at school, by adopting an attendance management system using wireless communication technology, attempts to solve problems caused by attendance books are being made. Representative attendance management systems include those using RFID, Bluetooth and clicker. Although these systems have solved the problem of wasting paper and time due to calling and writing attendance, they have other problems of generating additional expenses of purchasing or renting more equipment. To solve all of these problems, this paper suggests prototype system that can manage attendance by using NFC (Near Field Communication), which most smartphones provide. The attendance management system using NFC consists of two applications; one for professors and the other for students. The system solves problems, such as proxy attendance, loss of lesson time and additional cost by automatically managing attendance information using NFC and TCP/IP technologies.

• Journal of KIISE:Computer Systems and Theory
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• v.26 no.8
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• pp.975-983
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• 1999

최근 네트워크의 속도가 빨라지고 멀티미디어 데이터를 다루기 위한 기술들이 개발됨에 따라 많은 멀티미디어 응용 프로그램들이 인터넷에 등장하고 있다. 그러나 이들 응용프로그램들은 수신자에게 전송되는 영상.음성의 품질이 낮기 때문에 기대만큼 빠르게 확산되지 못하고 있다. 영상.음성의 품질이 낮은 이유는 현재 인터넷이 실시간 응용프로그램이 요구하는 만큼 빠르고 신뢰성 있게 데이터를 전송할 수 없기 때문이다. 현재 인터넷의 내부구조를 바꾸지 않고 품질을 높이기 위해 많은 연구들이 진행되고 있는데 그 중 하나는 동적으로 변화하는 인터넷의 상태에 맞게 멀티캐스트 트래픽의 전송율을 조절하는 종단간의 흐름제어이다. 본 논문은 기존의 흐름제어 기법인 IVS와 RLM의 성능을 개선시키기 위한 두 가지 흐름제어 기법을 소개한다. IVS는 송신자가 주기적으로 측정된 네트워크 상태에 따라 전송율을 일정하게 조절한다. 송신자가 하나의 데이타 스트림을 생성하는 IVS와는 달리 RLM에서는 송신자가 계층적 코딩에 의하여 생성된 여러개의 데이타 스트림을 전송하고 각 수신자는 자신의 네트워크 상태에 맞게 데이타 스트림을 선택하는 기법이다. 그러나 IVS는 송신자가 전송율을 일정하게 증가시키고, RLM은 각자의 네트워크 상태를 고려하지 않고 임의의 시간에 하나 이상의 데이타 스트림을 받기 때문에 성능을 저하시킬 수 있다. 본 논문에서는 TCP-like IVS와 Adaptive RLM이라는 두 가지 새로운 기법을 소개한다. TCP-like IVS는 송신자가 전송율을 동적으로 결정하고, Adaptive RLM은 하나 이상의 데이타 스트림을 받기 위해 적당한 시간을 선택할 수 있다. 본 논문에서는 시뮬레이션을 통해 여러 가지 네트워크 구조에서 두 가지 방식이 기존의 방식에 비하여 더욱 높은 대역폭 이용율과 10~20% 정도 적은 패킷손실율을 이룬다는 것을 보여준다.Abstract Nowadays, many multimedia applications for the Internet are introduced as the network gets faster and many techniques manipulating multimedia data are developed. These multimedia applications, however, do not spread widely and are not fast as expected at their introduction time due to the poor quality of image and voice delivered at receivers. The poor quality is mainly attributed to that the current Internet can not carry data as fast and reliably as the real-time applications require. To improve the quality without modifying the internal structure of the current Internet, many researches are conducted. One of them is an end-to-end flow control of multicast traffic adapting the sending rate to the dynamically varying Internet state. This paper proposes two flow-control techniques which can improve the performance of the two conventional techniques; IVS and RLM. IVS statically adjusts the sending rate based on the network state periodically estimated. Differently from IVS in which a sender produces one single data stream, in RLM a sender transmits several data streams generated by the layered coding scheme and each receiver selects some data streams based on its own network state. The more data streams a receiver receives, the better quality of image or voice the receiver can produce. The two techniques, however, can degrade the performance since IVS increases its sending rate statically and RLM accepts one more data stream at arbitrary time regardless of the network state respectively. We introduce two new techniques called TCP-like IVS and Adaptive RLM; TCP-like IVS can determine the sending rate dynamically and Adaptive RLM can select the right time to add one more data stream. Our simulation experiments show that two techniques can achieve better utilization and less packet loss by 10-20% over various network topologies.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.4C
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• pp.461-471
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• 2004

Recently, subscribers have strong desire to get QoS based personalized services in high-speed Internet access. Service providers have been rapidly replacing ADSL, cable broadband access networks with Metro-Ethernet based VDSL. But, it is difficult for Motto-Ethernet based broadband access networks to provide QoS based personalized services, because already deployed network elements cannot distinguish subscribers by specific traffic characteristics. In this paper, when the access network has tree topology, we show that it is possible to provide QoS for each downstream flow with only per flow traffic shaping at the edge node without QoS functions in access networks. In order to show that our suggested scheduling algorithm at the edge node can support the assured service in tree topology access networks, we evaluated its performance by simulation. The suggested scheduling algorithm can shape per-flow traffic based on the maximum bandwidth, and guarantees minimum bandwidth per flow by modifying the DRR scheduler. Simulation results show that congestion and loss in the access network elements are greatly reduced, TCP performance is highly enhanced and loss for assured CBR service flows is reduced by only shaping per-flow traffic at the edge node using our proposed scheduling algorithm.

• Journal of the Korea Society of Computer and Information
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• v.12 no.4
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• pp.1-10
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• 2007

This paper describes the development of a network-based passenger information system(PIS) which provides the convenience of the passenger of the train and heightens the effect of the subtitle service, the advertising and the shelter guidance broadcasting against the urgent event. The existing system uses VGA signal distributor in order to broadcast information with image and subtitle and voice guidance. In this paper we improve the existing system by applying the UDP and TCP/IP protocol and use a reference clock to solve a data loss and synchronization problem which occurs in this case. We also developed an XML-based subtitle authoring tool which can edit and play the subtitles with various 3D to improve the automatic guidance broadcasting and advertisement effect according to the operation schedule of the train. The system performance was evaluated through a simulation.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.9 no.10
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• pp.4015-4033
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• 2015

The future mobile networks consist of hyper-dense heterogeneous and small cell networks of same or different radio access technologies (RAT). Integrating mobile networks of different RATs to provide seamless and smooth mobility service will be the target of future mobile converged network. Generally, handover from high-speed networks to low-speed networks faces many challenges from application perspective, such as abrupt bandwidth variation, packet loss, round trip time variation, connection disruption, and transmission blackout. Existing inter-RAT handover solutions cannot solve all the problems at the same time. Based on the high-layer convergence sublayer design, a new receiver-aided soft inter-RAT handover is proposed. This soft handover scheme takes advantage of multihoming ability of multi-mode mobile station (MS) to smooth handover procedure. In addition, handover procedure is seamless and applicable to frequent handover scenarios. The simulation results conducted in UMTS-WiMAX converged network scenario show that: in case of TCP traffics for handover from WiMAX to UMTS, not only handover latency and packet loss are eliminated completely, but also abrupt bandwidth/wireless RTT variation is smoothed. These delightful features make this soft handover scheme be a reasonable candidate of mobility management for future mobile converged networks.

• Proceedings of the Korean Society of Precision Engineering Conference
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• 2001.04a
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• pp.279-283
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• 2001

We described an integrated management system, which is a design system of suction muffler(SMDS). SMDS constructs a virtual design system and possesses a Mutual Interfacing Module function using a Remote Analysis function and a GUI(Graphic User Interface). This system consists of a sever and clients. Client performs modeling and preprocessing, and server analyzes the results. The system uses Telnet and FTP based on TCP/IP protocol for connecting a client with a server.. It uses a PC at each work place as a basic platform for design and analysis of goods, and si able to manage a project as a unit. It is shown through an exsample that it is useful as a design tool in the fields.

• 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.

• IEMEK Journal of Embedded Systems and Applications
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• v.5 no.1
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• pp.7-17
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• 2010

ITS(Intelligent Transport System) as things are used for Broadcast service using TDMB/TPEG/NAVI rather than personal seamless service. It is attaching weight to Traffic information gathering, Charging, Settlement service. This research is applied to improve DCCP(Datagram Congestion Control Protocol) which has function as protecting data and preserving message boundary. The improving method is like that we solve data trust in UDP because Connection and Transmission overhead in UDP is less than in TCP. We fix the data loss which is generated from unordered delivery section of IP base wireless service by using DCCP protocol. We guarantee of connection with OBE(On-Board Equipment) and reliance about transmission of data by complement to mapping table and multi-hoping. Finally, We evaluate the performance about transmission of IP based data. We constructed a test-bed near research center for this test.