• IEMEK Journal of Embedded Systems and Applications
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• v.3 no.4
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• pp.198-206
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• 2008

Internet protocol television (IPTV), one of the most emerging services, offers multimedia streaming services with security, reliability, and relevant quality of service (QoS)/quality of experience (QoE). It provides added values to all the involving players and also brings technical and business challenges to them. For IPTV services, we expect to adopt the next generation network (NGN) environment for high quality and the Web technologies for personalization to meet the customer's necessity. Web can provide an open, flexible, and agile platform. Therefore, in this paper, we propose personalized IPTV services based on Web-based open platform in NGN environment and present functional architecture for Web-based personalized IPTV services. Technical issues for deploying the proposed IPTV services using Web are also provided. The objective of this paper is to analyze the critical architectural issues for developing viable and feasible networking platform models for personalized IPTV services.

• Convergence Security Journal
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• v.20 no.3
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• pp.3-11
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• 2020

The social networking service (SNS) is free, but the data usage fee paid to the telecommunications company and the member's information must be provided directly or indirectly. In addition, while SNS' specifications for transmitting and receiving devices such as smart-phones and PCs are increasing day by day, using universal transmission protocols in special environments such as contaminated areas or semiconductor manufacturing plants where work instructions are mainly made using short messages is not easy. It is not free and has a problem of weak security. This paper verified the practicality through the operation test by implementing IMCP, a low-power, low-cost message transmission protocol that aims to be wearable in special environments such as risk, pollution, and clean zone based on ATmega128.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2009.05a
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• pp.459-462
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• 2009

The Growth in wireless Internet has made Networking possible at anytime and anywhere so that application program development is vigorous for mobile service implementation. In particular, mobile service has a feature that can present flexible service at anytime, and anywhere. This study proposes a system which is able to make a remote control of computer by means of mobile as an example of the production of application for such mobile service. This paper researches a RDP(Remote Desktop Protocol), VNC(Virtual Network Computing) system that control screen of remote computer and input devices by using smart-phone. The advantage of this system is the function that turning on and off the computer, file transfer, file upload/download, and paperwork. Therefore users are expected to use more conveniently.

• 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.

• Journal of KIISE:Information Networking
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• v.29 no.2
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• pp.141-155
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• 2002

A medium access control protocol is proposed for integrated voice and data services in the packet CDMA network with a small coverage. Uplink channels are composed of time slots and multiple spreading codes for each slot. This protocol gives higher access priority to the delay-sensitive voice traffic than to the data traffic. During a talkspurt, voice terminals reserve a spreading code to transmit multiple voice packets. On the other hand, whenever generating a data packet, data terminals transmit a packet based on the status information of spreading codes in the current slot, which is received from base station. In this protocol, voice packet does not come into collision with data packet. Therefore, this protocol can increase the maximum number of voice terminals.

• Journal of KIISE:Information Networking
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• v.30 no.6
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• pp.729-742
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• 2003

Mobile Communication has a possibility of eavesdropping by nature of wireless channel. It is known that eavesdropping of CDMA system is impossible because the voice data spreads with the PN. First of all, we show that it is possible to eavesdrop the CDMA channel by analysis of the forward channel in case that we know the ESN and the MIN. We can monitor the forward traffic channel with easy since ESN and MIN are exposed during the call processing in CDMA service in Korea. In this paper, we will show security weakness and propose an enhanced mechanism for CDMA service. We consider the problem of security in the CDMA service. CDMA system has wireless channels to transmit voice or data. By this reason, CDMA communication has a possibility of being eavesdropped by someone. It is known that eavesdropping in CDMA system is impossible because the voice data spreads with the PN. However, we can eavesdrop the CDMA data in FCM protocol in case that we know the ESN and the MIN. In CDMA system, ESN and MIN are exposed to the wireless channel. In this paper, we analyze the flow of the voice and signal in the CDMA system and monitor the forward traffic channel by the FCM protocol. The FCM protocol is proposed to monitor the forward channel in CDMA system. We can show the possibility of monitoring in one-way channel of CDMA system by the FCM protocol. The test instrument based on the FCM protocol is proposed to monitor the CDMA forward channel. We will show the system architecture of the test instrument to monitor the forward channel in CDMA.

• ETRI Journal
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• v.38 no.5
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• pp.922-933
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• 2016

IP header compression schemes offer a valuable measure for bandwidth preservation. Such schemes have been practically implemented in infrastructure-based IP networks for point-to-point links. However, minimal research and practical implementation efforts have been conducted in the direction of an IP header compression strategy that can meet the peculiar requirements of multi-hop ad hoc wireless networks. In this paper, we present a practically implemented multi-hop IP header compression scheme using the Robust Header Compression (ROHC) protocol suite. The scheme runs on a novel identifier (ID) based networking architecture, known as an ID-based ad hoc network (IDHOCNET). IDHOCNET additionally solves a number of bottlenecks of pure IP-based ad hoc networks that have emerged owing to IP address auto-configuration service, distributed naming and name resolution, and the role of an IP address as an identifier at the application layer. The proposed scheme was tested on a multi-hop test bed. The results show that the implemented scheme has better gain and requires only O (1) ROHC contexts.

• Journal of KIISE:Information Networking
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• v.31 no.6
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• pp.566-572
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• 2004

In this paper, we define the secure authentication model to provide a mobile node with global roaming service and integrate the Fast Handoff scheme with our approach to minimize the service latency. By starting the AAA(Authentication, Authorization and Account) procedure with Fast Handoff simultaneously when a roaming occurs, authentication latency is reduced significantly and provision of fast and seamless service is possible. The previous works such as IPsec(Internet Protocol Security), RR (Return Routability) and AAA define the procedures performed after the completion of Layer2 Handoff which leads us to study a way of providing the real time and QoS guaranteed service during this period. The proposed scheme is for this goal and when appling it to roaming environment it shows the cost reduction up to 55% and 17% for the case of the MN receiving the FBACK and not respectively before L2 Handoff occurs.

• Journal of KIISE:Information Networking
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• v.34 no.2
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• pp.100-109
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• 2007

Streaming protocol with the Available Bandwidth measurement scheme has problems that are to measure a Available Bandwidth uncorrectly and slowly. In this basis, in order to overcome limitations of the previous streaming protocols, we propose the NABO that is a New Available Bandwidth measurement scheme used by OWD(One-Way Delay). Proposed NABO(New Available Bandwidth measurement based on OWD) measures the constant transmission delay occurred by bottleneck link capacity and the variable delay. Competing traffic contribute to the variable delay. Through the measurement of the constant transmission delay and the competing traffic, a NABO can measure the Available Bandwidth correctly and fast in network. The simulation result proves that the proposed NABO has a performance that satisfies both accuracy viewpoint and measurement speed viewpoint.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.5B
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• pp.430-441
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• 2006

As the various A/V devices and home networks are delivered to users, home networks are changing to an entertainment network. It is expected that the required network bandwidth and the amount of usage of media content in home entertainment networks will be increased. Although the access networks and home networks becoming a high speed network, there remains the problems for QoS-guaranteed media content transfer in home networks. Also, in the home network, there can be network traffic caused by applications like video conferencing, video telephone, and VoIP(voice over IP) as well as inner network traffic of home network. Since media content transfer requires the real-time delivery, it is very important and basic requirement that is to transfer media content to A/V device user wants while keeping the media quality. Even though there are many middleware protocol for home networking, they provide basic device discovery and control or simple functions for QoS-guaranteed media content transfer that are not enough to provide QoS-guaranteed media transfer service that user wants. Thus, in this paper, we propose the technique based on UPnP(universal plug and play) protocol for QoS-guaranteed media content transfer in the home network. The proposed technique is compatible with UPnP and can be used with UPnP as additional functions. In this paper, we utilize VideoLAN application to verify the proposed technique. We add the additional modules that support the proposed technique's function to VideoLAN and we verify the its functions through various test scenarios.