• KSII Transactions on Internet and Information Systems (TIIS)
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• v.5 no.5
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• pp.1013-1027
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• 2011

In this paper we propose ATCS, a practical TCP protocol coding scheme based on network coding for satellite IP networks. The proposal is specially designed to enhance TCP performance over satellite networks. In our scheme, the source introduces a degree of redundancy and transmits a random linear combination of TCP packets. Since the redundant packets are utilized to mask packet loss over satellite links, the degree of redundancy is determined by the link error rates. Through a simple and effective method, ATCS estimates link error rates in real time and then dynamically adjusts the redundant factor. Consequently, ATCS is adaptable to a wide range of link error rates by coding TCP segments with a flexible redundancy factor. Furthermore, the scheme is compatible with traditional TCP variants. Simulation results indicate that the proposal improves TCP performance considerably.

• The Journal of Information Technology
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• v.5 no.4
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• pp.155-164
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• 2002

MPLS is a integrated technology by using routing function and label swapping in the network layer. Based on the previous forwarding equivalence classes, it adds the fixed length label in ingress of the MPLS domain. For the routing, without the packet header information, it uses label for the forwarding decisions. In this paper, traffic engineering requirements in the MPLS internet will be setup. The traffic engineering function have to be performed previously with the network topology. In addition to, we presents the IP network topology and main function with MPLS signaling protocol.

• Journal of Navigation and Port Research
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• v.41 no.6
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• pp.395-400
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• 2017

As a serial communications protocol, Modbus has become a practically standard communication protocol and is now a commonly available means of connecting industrial electronic devices. Therefore, all devices can be connected using the Modbus protocol with the measurement and remote control on ships, buildings, trains, airplanes and more. The existing Modbus that has been used is based on serial communication. Modbus TCP uses Ethernet communication based on TCP / IP which is the most widely used Internet protocol today; so, it is faster than serial communication and can be connected to the Internet of Things. In this paper, we designed an algorithm to control LED lighting in a wireless Wi-Fi environment using the Modbus TCP communication protocol, and designed and implemented a LED controller circuit that can check external environmental factors and control remotely through the integrated management system of a ship. Temperature, humidity, current and illuminance values, which are external environmental factors, are received by the controller through the sensors, and these values are communicated to the ship's integrated management system via the Modbus protocol. The Modbus can be connected to Master devices via TCP communication to monitor temperature, humidity, current, illuminance status and LED output values, and also users can change the RGB value remotely in order to change to the desired color. In addition, in order to confirm the implementation of the controller, we developed a simulated ship management system to monitor the temperature, humidity, current and illumination conditions, and change the LED color of the controller by changing the RGB value remotely.

• Journal of Navigation and Port Research
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• v.27 no.3
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• pp.283-288
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• 2003

Recently, the growth of applications and services over high-speed Internet increases, ATM networks as wide area back-bone has been a major solution. As the conventional TCP/IP suite is still the standard protocol used to support upper application on current. Internet, the issues regarding whether TCP/IP will operate efficiently on top of an ATM infrastructure and how to control its QoS still remain for studies. TCP uses a window-based protocol for flow control in the transport layer. When TCP uses the UBR service in ATM layer, the control method is only buffer management. If a cell is discarded in ATM layer, one whole packet of TCP will be lost; this fact occur the most TCP performance degradation. Several dropping strategies, such as Tail Drop, EPD, PPD, SPD, FBA, have been proposed to improve the TCP performance over ATM. In this paper, to improve the TCP performance, we propose a packet dropping scheme that is based on comparison with EPD, SPD and FBA. Our proposed scheme is applied to schemes discussed in the previous technology. Our proposed scheme does not need to know each connection's mean packet size. When the buffer exceeds the given threshold, it is based on comparison between the number of dropped packet and the approved packet. Our results are reported and discussed for comparing these discarding schemes under similar conditions. Although the number of virtual channel (VC) is increased, the simulation results showed that the proposed scheme can allocate more fairly each VC than other scheme.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.40 no.10
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• pp.119-128
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• 2003

The rapid increase of Internet users, diversity of application services, high speed data transmission, and extension of application areas have resulted in exponential growth of Internet traffic. In order to satisfy the increasing demand of bandwidth, the MPλS protocol, which is based on MPLS for efficient integration of WDM and IP protocols, has been suggested by IETF. In this paper, we present an MPλS simulator which enables to simulate various MPλS function such as optical crossconnect (OXC), multi-wavelength links, routing and wavelength assignment(RWA), and MPλS signaling and control. The simulator is developed based on the ns-2, an widely used multi-protocol network simulator. The function of the simulator is validated by running many simulation based on various scenarios and performance measures such as throughput and blocking probability. The simulator could be widely utilized for validation of proposed protocols before developing real optical network systems.

• 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.

• The KIPS Transactions:PartD
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• v.16D no.6
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• pp.945-960
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• 2009

The purpose of this study is to define determinants of VoIP user acceptance and to verify significant causality among latent variables - performance expectancy, effort expectancy, cost expectancy, social influence, facilitating conditions, behavioral intend, use behavior - based on UTAUT model. We presented the expanded hypotheses including the new factor, cost expectancy and analyzed the moderating effect of user age, gender and usage experience variables. For a accuracy of predicted results, we focused on survey analysis with 641 real user samples. Compared to previous studies, it is meaningful that this research verified the conceptual difference between behavioral intention and usage behavior. As a result, all proposed hypotheses accepted and moderating effects are supported significantly in age and use experience moderating variables.

• Journal of Korea Multimedia Society
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• v.10 no.3
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• pp.335-347
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• 2007

VoIP (Voice over IP) is a technology to transport video and voice traffic over IP networks such as Internet. Today, the VoIP technology is viewed as the right choice for providing voice, video, and data communication among various terminals over the next generation network. This paper discusses a multipoint VoIP implementation with end-point mixing model which can support multipoint conference without a conference bridge. The multipoint VoIP is implemented with SIP (Session Initiation Protocol), and supports STUN (Simple Traversal of UDP Through NATs) since it works in an asymmetric NAT (Network Address Translator) environment. The characteristics of this paper are as follows. It is possible that all terminals in the hierarchical conference don't receive the duplicated media information because we use the end-point mixing model with the new media processing module. And, the paper solves the problem that the hierarchical conference session should be separated into several sessions when a mixing terminal terminates the hierarchical conference session.

• ETRI Journal
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• v.35 no.1
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• pp.7-17
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• 2013

The ongoing proliferation of new services, applications, and contents is leading the Internet to an architectural crisis owing to its inability to provide efficient solutions to new requirements. Clean-slate architectures for the future Internet offer a new approach to tackle current and future challenges. This proposal introduces a novel clean-slate architecture in which the TCP/IP protocol stack is decoupled in basic functionalities, that is, atomic services (ASs). A negotiation protocol, which enables context-aware service discovery for providing adapted communications, is also specified. Then, we present how ASs can be discovered and composed according to requesters' requirements. In addition, a media service provisioning use case shows the benefits of our framework. Finally, a proof-of-concept implementation of the framework is described and analyzed. This paper describes the first clean-slate architecture aligned with the work done within the ISO/IEC Future Network working group.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2011.07a
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• pp.482-483
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• 2011

본 논문에서는 스마트TV향 VoIP(Voice over Internet Protocol) 컨퍼런스 기능을 위한 잡음제거 알고리즘의 성능비교 하였다. 기존에 연구 되어져 있는 Improved Minima Controlled Recursive Averaging(IMCRA)방식과 Gaussian분포 기반의 잡음제거 알고리즘, IMCRA방식과 Gamma분포 기반의 잡음제거 알고리즘, IMCRA방식과 Mel-filter를 적용한 잡음제거 알고리즘, R&L 알고리즘들의 방식을 비교하였으며, 성능 비교를 위해 각 알고리즘을 통해 나온 다양한 잡음 환경에서의 잡음이 제거된 신호의 PESQ와 연산속도를 비교한다.