• Journal of the Korea Institute of Information and Communication Engineering
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• v.22 no.6
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• pp.929-934
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• 2018

The Internet of Things (IoT) is the network of devices embedded with electronics, software, sensors, actuators, and connectivity which enables these objects to connect and exchange data without any human intervention. The importance of time information is increased in order to impose order on scattered sensor data streams, resolving conflicts through time stamp information and so on. Time information and time synchronization are critical building blocks in the IoT. They allow devices to share a consistent notion of time and it is easier to build efficient and robust collaborative services. This paper proposes a standard time management scheme in the Internet of Things. Many IoT applications involve collection and forwarding of event data. It is useful to know when an event occurs for the purposes of triggering an action. In order to verify the feasibility of the proposed scheme, it is implemented and evaluated in the Arduino development environment.

• Journal of KIISE:Information Networking
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• v.33 no.2
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• pp.139-148
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• 2006

Recently with the advancement of local wireless communication technologies, digital trends of audio contents and devices are paving the way for new experiences to network-based audio streaming services. In this paper, we present the Wireless Audio Sharing (WASH) technology over Bluetooth and Wireless Local Area Network (WLAN). WASH system provides an audio sharing mechanism over multiple users and also provides stereo audio streaming between Bluetooth-enabled devices and Universal Plug and Play (UPnP) media devices connected to wired/wireless LANs. To achieve these, we extended the Bluetooth audio distribution profile and combined the extended profile with the UPnP AV architecture. With the implementation of this WASH technology, we show some experimental results of the stereo audio streaming in a real environment.

• The KIPS Transactions:PartC
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• v.14C no.1 s.111
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• pp.55-64
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• 2007

Behind the popularity of VoIP in these days, it may present significant security challenges in privacy and accounting. Authentication and message encryption are considered to be essential mechanisms in VoIP to be comparable to PSTN. SIP is responsible for setting up a secure call in VoIP. SIP employs TLS, DTLS or IPSec combined with TCP, UDP or SCTP as a security protocol in VoIP. These security mechanisms may introduce additional overheads into the SIP performance. However, this overhead has not been understood in detail by the community. In this paper we present the effect of the security protocol on the performance of SIP by comparing the call setup delays among security protocols. We implement a simulation of the various combinations of three security protocols and three transport layer protocols suggested for SIP. UDP with any combination of security protocols performs a lot better than the combination of TCP. TLS over SCTP may impose higher impact on the performance in average because TLS might have to open secure channels as the same number of streams in SCTP. The reasons for differences in the SIP performances are given.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2004.11a
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• pp.271-276
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• 2004

디지털방송이 시작되면서 고품질의 A/V(비디오/오디오)프로그램과 다양한 멀티미디어 컨텐츠를 제공하는 데이터방송을 처리하기 위한 수신 장치의 비중이 커지고 있다 이와 관련하여 데이터방송 환경에서 새로운 기술과 서비스가 등장할 때마다 이를 수용할 수 있는 소프트웨어를 탑재한 새로운 수신 장치가 필요하다. 일반적으로 한번 가정 내에 보급된 디지털 수신 장치의 소프트웨어 업그레이드가 용이하지 않기 때문에, 방송을 통해 이를 실현한다. 본 논문은 TV 셋탑박스 (STB) 내에 상주하는 middleware native application software 를 방송으로 다운받아 수정하는 기능을 지닌 STB 의 구현에 대해서 기술한다. 소프트웨어 업데이트 시스템은 소프트웨어를 포함하는 데이터 카루셀 스트림을 다운받아 파싱하는 다운로더, 추출된 소프트웨어를 설치하는 업데이트 로더, 그리고 예치상황이 발생하면 셋탑박스가 새로 부팅될 때, 로그 파일을 이용하여 소프트웨어를 옛날 상태의 회복시켜주는 리커버러 (recoverer)로 구성되어 있다. 다운로더는 지상파 디지털 방송 규격인 ATSC 규약에 맞게 구현하고, ATSC용 STB환경에서 테스트하고 있다.

• Journal of KIISE:Computing Practices and Letters
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• v.5 no.4
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• pp.440-448
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• 1999

본 논문은 객체지향형 분산처리 환경 하에서 오디오나 비디오 등과 같은 실시간(real-time) 스트림(stream) 데이타를 처리하는 데 필요한 소프트웨어 기반구조를 설계하고 구현한 내용을 기술한다. 본 논문에서 제시한 DAViS(Distributed Object Framework supporting Audio/Video Streaming)는, 오디오/비디오 데이타의 처리와 관련된 여러 소프트웨어 구성요소들을 분산객체로 추상화하고, 그 객체들간의 제어정보 교환경로와 오디오/비디오 데이타 전송경로를 서로 분리하여 처리한다. 분산응용프로그램 작성자는 DAViS에서 제공하는 서비스들을 이용하여, 기존의 분산프로그래밍 환경이 제공하는 것과 동일한 수준에서 오디오/비디오 데이타에 대한 처리를 표현할 수 있다. DAViS는, 새로운 형식의 오디오/비디오 데이타를 처리하는 부분을 손쉽게 통합하고, 하부 네트워크의 전송기술이나 컴퓨터시스템 관련 기술의 진보를 신속하고 자연스럽게 수용할 수 있도록 하는 유연한 구조를 가지고 있다. Abstract This paper describes the design and implementation of software framework which supports the processing of real-time stream data like audio and video in distributed object-oriented computing environment. DAViS(Distributed Object Framework supporting Audio/Video Streaming), proposed in this paper, abstracts software components concerning the processing of audio/video data as distributed objects and separates the transmission path of data between them from that of control information. Based on DAViS, distributed applications can be written in the same abstract level as is provided by the existing distributed environment in handling audio/video data. DAViS has a flexible internal structure enough to easily incorporate new types of audio/video data and to rapidly accommodate the progress of underlying network and computer system technology with very little modifications.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.15 no.1
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• pp.99-106
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• 2011

The emerging of ubiquitous computing and healthcare technologies provides us a strong platform to build sustainable healthcare applications especially those that require real-time information related to personal healthcare regardless of place. We realize that system stability, reliability and data protection are also important requirements for u-healthcare services. Therefore, in this paper, we designed a u-healthcare system which can be attached to the patient's body to measure vital signals, enhanced with USN secure sensor module. Our proposed u-healthcare system is using wireless sensor modules embedded with NLM-128 algorithm. In addition, PS-LFSR technique is applied to the NLM-128 algorithm to enable faster and more efficient computation. We included some performance statistical results in term of CPU cycles spent on NLM-128 algorithm with and without the PS-LFSR optimization for performance evaluation.

• Journal of Broadcast Engineering
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• v.15 no.4
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• pp.540-546
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• 2010

Since the stereoscopic 3-dimensional (3D) video that provides users with a realistic multimedia service requires twice as much data as 2-dimensional (2D) video, it is difficult to construct the fast system. In this paper, we propose a fast stereoscopic 3D broadcasting system based on the depth information. Before the transmission, we encode the input 2D+depth video using x264, an open source H.264/AVC fast encoder to reduce the size of the data. At the receiver, we decode the transmitted bitstream in real time using a compute unified device architecture (CUDA) video decoder API on NVIDIA graphics processing unit (GPU). Then, we apply a fast view synthesis method that generates the virtual view using GPU. The proposed system can display the output video in both 2DTV and 3DTV. From the experiment, we verified that the proposed system can service the stereoscopic 3D contents in 24 frames per second at most.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.21 no.6
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• pp.1490-1500
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• 1996

One of the important issues for multimedia communications over ATM netowrks is efficient use of net-work resources since transmitting video at low cost requires high utilization of channel bandwidth. In this paper, we propose an effective bandwidth estimation scheme for MPEG sources in ATM networks. It has been known difficult to allocate effective transmission bandwidth to MPEG sources due to its bursty charactersitics. The MPEG traffic stream can bemodeled using five traffic parameters:service rate, burst load ratio, interburst load ratio, burst period, and interburst period. Using these parameters, a transmission bandwidth for each source can be estimated, and the estimated bandwidth is applied synchronously to reduce the queue size in a buffer. For a simple network modelof a high speed link that multiplexes a number of virtual-circuit connections, simulations on adaptive bandwidth allocation were performed, and the results show that the queueing delay is significantly reduced, when compared to a fixed bandwidth allocation.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.23 no.6
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• pp.1526-1537
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• 1998

The split-screen function on the multipoint control unit(MCU) which is usually using processing method based on pixel domain has many problems for manipulating the video signal in real-time. Although the researches and the developements to cope ith such problems are processing, these have too complex architecture to implement and are limited to method for H.261 video signal. In this paper, we propose a new mechanism for split-screen that can actually apply to ISO/IEC MPEG video standard. The new method that is proposed in this paper do the processes in complete compression domain, thus it is suitable for the application of real-time multipoint multimedia communication service. By simple interpreting and manipulating the MPEG video element stream, the split-screen functional module can be implemented easily and the result of the procedures does not accompany image degradation at all. Finally, the complexity of implementation, the aspect for processing delay and the loss of image quality as compared to that resulting in the case of applying the previous split-screen method has been investigated. And it is confirmed that the proposed mechanism has a significant advantage as a split-screen method.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.8 no.6
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• pp.1076-1085
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• 2004

Recently, the demands for real-time service and multimedia data are rapidly increasing. There are significant redundancies between header fields both within the same packet header and in consecutive packets belonging to the same packet stream. And there are many overheads in using the current UDP/IP protocol. Header compression is considered to enhance the transmission efficiency for the payload of small size. By sending the static field information only once initially and by utilizing dependencies and predictability for other fields, the header size can be significantly reduced for most packets. This work describes an implementation for header compression of the headers of IP/UDP protocols to reduce the overhead on Ethernet network. Typical UDP/IP Header packets can be compressed down to 7 bytes and the header compression system is designed and implemented in Linux environment. Using the Header compression system designed between a server and clients provides have the advantage of effective data throughput in network. Since the minimum packet size in Ethernet is 64 bytes, the amount of reduction by header compression in practical chatting environment was 6.6 bytes.