• Journal of Internet Computing and Services
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• v.20 no.2
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• pp.21-27
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• 2019

The terrestrial ATSC 3.0 broadcasting system, which is capable of converging broadcast and communication services, uses IP based technology for data transmission between broadcasting equipment. In addition, data transmission between broadcasting equipment uses IP-based technology like existing wired communication network, which has advantageous in terms of equipment construction and maintenance In case IP based data transmission technology is used, however, it may inevitably cause an error that a packet is lost during transmission depending on the network environments. In order to cope with a broadcasting accident caused by such a transmission error or a malfunction of a broadcasting apparatus, a broadcasting system is generally configured as a duplication, which can transmit a normal packet when various types of error may occur. By this reason, correction method of error packets and intelligent switching technology are essential. Therefore, in this paper, we propose a design and implementation of intelligent IP switch for Ensuring Reliability of ATSC 3.0 Broadcast Streams. The proposed intelligent IP consists of IP Stream Analysis Module, ALP Stream Analysis Module, STL Stream Analysis Module and SMPTE 2022-1 based FEC Encoding/Decoding Module.

• Proceedings of the IEEK Conference
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• 2003.11c
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• pp.149-152
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• 2003

In this paper, we propose a scheme that applies to the VoIP application based on H.323 protocol to enhance the bandwidth efficiency. We multiplex the audio and video stream. In this scheme, audio frame is carried with video stream. And we applies not only multiplexing but also (in header compressing to the real audio/video stream to increase the bandwidth efficiency. With the multiplexing and RTP header compressing, we gain the bandwidth efficiency. In the finite network environment, We can assign bandwidth to other users who want to use other service. and other VoIP users. If we can apply the real time network situation to the our VoIP application, we can get more efficient performance.

• Journal of Institute of Control, Robotics and Systems
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• v.22 no.8
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• pp.671-677
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• 2016

A Controller Area Network (CAN) is a serial communication protocol that is highly reliable and efficient in many aspects, such as wiring cost and space, system flexibility, and network maintenance. Therefore, it is chosen for the communication protocol between a single chip controller based on Field Programmable Gate Array (FPGA) and peripheral devices. In this paper, the design and implementation of CAN IP, which is written in VHSIC Hardware Description Language (VHDL), is presented. The implemented CAN IP is based on the CAN 2.0A specification. The CAN IP consists of three processes: clock generator, bit timing, and bit streaming. The clock generator process generates a time quantum clock. The bit timing process does synchronization, receives bits from the Rx port, and transmits bits to the Tx port. The bit streaming process generates a bit stream, which is made from a message received from a micro controller subsystem, receives a bit stream from the bit timing process, and handles errors depending on the state of the CAN node and CAN message fields. The implemented CAN IP is synthesized and downloaded into SmartFusion FPGA. Simulations using ModelSim and chip test results show that the implemented CAN IP conforms to the CAN 2.0A specification.

• Journal of the Institute of Convergence Signal Processing
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• v.12 no.2
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• pp.102-106
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• 2011

The transfer capability of HFC (Hybrid Fiber Coax) network is superior to xDSL. HFC network, however, is not suitable for transferring high quality video due to cable model interfaces. For the services of high quality IPIV or VOD, the extra exclusive downstream transfer system is required without upgrading pre-equipped cable modem and service capability. This paper is aimed to develop the extra exclusive downstream transfer system not changing existing cable modem system but providing same quality of services. This system is composed of the extra exclusive downstream IP-cable sender and modem. This sender and modem have 30 Mbps transfer capability and HDTV stream can be served in the Cable TV network using 21 Mbps HDTV transport stream.

• Journal of the Institute of Electronics and Information Engineers
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• v.53 no.8
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• pp.19-27
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• 2016

Nowadays the adaptive buffering techniques for burst stream packet configuration and its operation algorithm to save energy in IP core network have been studied. This paper explains the selection method of packet buffering interval for energy saving when configuring burst stream packet at the ingress router in IP core network. Especially the adaptive buffering interval and its implementation scheme are required to improve the energy saving efficiency at the input part of the ingress router. In this paper, we propose the best adaptive buffering scheme that a current buffering interval is adaptively buffering scheme based on the input traffic of the past buffering interval, and analyze its characteristics of energy saving and end-to-end delay by computer simulation. We show the improvement of energy saving effect and reduction of mean delay variation when using an appropriate inverse-function selecting the buffering interval for the configuration of burst stream packet in this paper. We confirm this method have superior properties compared to other method. The proposed method shows that it is less sensitive to the various input traffic type of ingress router and a practical method.

• Proceedings of the IEEK Conference
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• 2002.06a
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• pp.311-314
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• 2002

This paper deals with an improved method of utilizing end-to-end bandwidth in the All-IP environment. The proposed method includes compression of UDP/RTP headers, and multiplexing of the RTP stream packets over the end-to-and media transfer. Although the conventional method of using TCRTP(Tunneling Multiplexed Compressed RTP) is an efficient mettled of maximizing tile network throughput, it is inadequate for the All-IP based end-to-end communication. The method is a link-layer independent solution that can be easily implemented in the NGN(Next Generation Network).

• The KIPS Transactions:PartC
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• v.10C no.7
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• pp.937-942
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• 2003

In this paper, we measured and examined RTT delays and packet losses according to the changes of stationary loads for two typical stream-type traffics, a DV and a MPGE2 on the R＆D Gigabit Network testbed, JGN. As the result of our actual measurements, we realized that the packet size of stationary load have no effects on a DV and a MPGE2 stream on the very high-speed network(50Mbps, IP over ATM). When its bandwidth and stationary load exceeds 95％ of network bandwidth, packet losses appeared and RTT delay increased rapidly. Also we realized that the number and size of Receive ＆ Transmit buffer on the end systems have no effects on packet losses and RTT delays.

• The KIPS Transactions:PartC
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• v.9C no.2
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• pp.257-266
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• 2002

In 3GPP2 wireless data communications, Mobile IP is used to support macro mobility and PDSN performs the function of foreign agent. The mobility supported when a mobile station moves from one PDSN to another is called a macro mobility. In this Paper, we first examine the possibilities of packet loss and change of packet sequences that can be occurred in macro mobility. Then, to resolve such Problems, we suggest a seamless handoff algorithm in PDSNs based on packet sequence control for each of down-stream and up-stream cases respectively.

• The Journal of the Korea Contents Association
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• v.10 no.8
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• pp.59-65
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• 2010

This paper proposes an implementation of a hardware module for transmitting MPEG-2 TS data over the internet protocol (IP)-based network. This implementation consists of two modules; one is an encapsulation module which bridges between n TS packets, where $1\;{\leq}\;n\;{\leq}\;7$, and an IP packets, the other is a packet conversion module which extracts an DSM-CC PS packet from consecutive TS packets and then reconstructing an IP packet. So, these IP packets are carried over 150 megabits per second. Although overall work flow of the proposed DeMUX is based on the reference design of ALTERA, the DeMUX is enhanced by modifying it and performs more functions by adding a packet conversion module. The DeMUX is described by Verilog-HDL (hardware description language) and shows the faithful functionality and throughput through the simulation.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.41 no.10
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• pp.1190-1196
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• 2016

In this paper, a cross-layer interworking scheme for different protocols is proposed to provide an efficient internet services in very small aperture terminal (VSAT) based satellite communication systems. In addition, we implement the proposed interworking model and prove the feasibility of the proposed system. VSAT based satellite communication systems commonly use digital video broadcasting (DVB)-S2 standard. Unfortunately, DVB-S2 has inefficient parts to support IP based internet services because it has originally been designed to support broadcasting services. Generic stream encapsulation (GSE) protocol, which is a layer 2 protocol, has been proposed to mitigate this inefficiency. We propose a cross-layer interworking scheme to cooperate efficiently between IP and GSE protocols and between GSE protocol and DVB-S2, respectively. In addtion, we implement the proposed interworking schemes via computer softwares and prove the feasibility using NI-USRP and commercial DVB receiver.