• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.6A
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• pp.690-696
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• 2004

Advanced Television Systems Committee (ATSC) -A/90, which is a standard for terrestrial data transmission in Korea, defines synchronized data that has a strong timing association with a separate Program Element. It is classified as synchronized streaming data that is carried in packetized elementary stream (PES) packets or a synchronized non-streaming data that shall be carried in digital storage media command and control (DSM-CC) section. In this paper, we study the design and verification of synchronized streaming data processing algorithm based on ATSC -A/90. We designed a parser and a player for the algorithm development. The received PES packet including synchronized streaming data is parsed in the parser. The parsed synchronized streaming data is synchronized and displayed by player. Finally, we ascertained that STB was working properly with MPEG-2 transport stream (TS) containing synchronized streaming data, as the proposed algorithm is implemented on a set-top box.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2014.05a
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• pp.571-573
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• 2014

In today's technology streaming service's QoS technologiey is an issue. On quality video streaming service there are some technical issues exist, such as buffering. This submission is "Adaptive dual-streaming system design" which is for the integrity of the data streaming that is sent to TCP and UDP for faster transmission of data to the stream. This system provides real-time incoming video encoding in bitrate of h.264-based H through a process based on the video footage of several server and client-to-TCP and UDP via Adaptive providing streaming services in a network environment. This is an unspecified number of buffers in a network environment and continued through the minimization of various streaming for playback of videos and multimedia will be utilized in the field.

• Proceedings of the IEEK Conference
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• 2003.11a
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• pp.417-420
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• 2003

Most Embedded Web Camera Server products currently deployed on the market adopt JPEG for compression of video data continuously acquired from the cameras. However, JPEG does not efficiently compress the continuous video stream, and is not appropriate for the Internet where the transmission bandwidth is not guaranteed. In our previous work, we presented the result of designing and implementing an embedded web camera streaming server using MPEG4 codec. But the server in our previous work did not show good performance since one CPU had to both compress and process the network transmission. In this paper, we present our efforts to improve our previous result by using dual CPUs, where DSP is employed for data compression and StrongARM is used for network processing. Better performance has been observed, but it is found that still more time is needed to optimize the performance.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.22 no.7
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• pp.1403-1410
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• 1997

Because streaming mode connection server suitable to wide area ATM networks performs transmission, reception and lookup with time restriction for the transmission time of a cell, it has demerits of large cell loss incase that burst traffic occurs. Therefore, in this paper to decrease cell loss we propose a high speed CAM (Content Addressable Memory) which is capable of processing data of streaming mode connections server at a high speed. the proposed CAM is applied to forwarding table VPC map which performs function to output connection numbers about input VPI(Virtual Path Identifier)/VCI(Virtual Channel Identifier). The designed high speed CAM consist of DBL(Dual Bit Line) CAM structure performed independently write operation and match operation and two-port SRAM structure. Also, its simulation verification and full-custom layout is performed by Hspice and Composs tools in 0.8 .$\mu$m design rule.

• International Journal of Contents
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• v.14 no.4
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• pp.30-38
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• 2018

Recently, HTTP Adaptive Streaming(HAS), a video streaming service over the HTTP based web platform has become common. The use of HAS service in mobile communication devices such as mobile phones and tablet PCs is rapidly expanding. This paper addresses ways to improve the quality of HAS service by enhancing the terms of viewer satisfaction. HAS systems have several internal operational processes, which can affect viewer satisfaction. Such processes include, the quality determination for the next video chunk, the TCP connections-setup procedure and the congestion control operation of the TCP. This paper proposes a transmission scheme to improve the HAS quality services over mobile web. The proposed scheme takes into consideration the past implicit communication state of the receiver's playback buffer occupancy. The results of these experiments indicate that the proposed scheme can improve the quality of HAS service from the mobile viewer's point of view.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2019.05a
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• pp.456-458
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• 2019

In this paper, transmission performance of streaming audio as a railway communication service based on LTE-R is analyzed. Performance analysis is perfomed with computer simulation based on NS(Network Simulation)-3, audio frame of MPEG(Moving Picture Experts Group)-4 is used as target application service for straming audio traffic. Results of this paper can be used to implement LTE-R networks and develope application services over LTE-R network.

• Journal of KIISE:Computer Systems and Theory
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• v.33 no.10
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• pp.722-733
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• 2006

In scalable streaming application, there are two important knobs to tune to effectively exploit the underlying network resource and to maximize the user perceivable quality of service(QoS): layer selection and packet scheduling. In this work, we propose Semantics Aware Packet Scheduling (SAPS) algorithm to address these issues. Using packet dependency graph, SAPS algorithm selects a layer to maximize QoS. We aim at minimizing distortion in selecting layers. In inter-frame coded video streaming, minimizing packet loss does not imply maximizing QoS. In determining the packet transmission schedule, we exploit the fact that significance of each packet loss is different dependent upon its frame type and the position within group of picture(GOP). In SAPS algorithm, each packet is assigned a weight called QoS Impact Factor Transmission schedule is derived based upon weighted smoothing. In simulation experiment, we observed that QOS actually improves when packet loss becomes worse. The simulation results show that the SAPS not only maximizes user perceivable QoS but also minimizes resource requirements.

• Journal of KIISE
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• v.44 no.12
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• pp.1296-1306
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• 2017

Recently, as the number of Internet of Things users has increased, IETF (Internet Engineering Task Force) has released the CoAP (Constrained Application Protocol). So Internet of Things have been researched actively. However, existing studies are difficult to adapt to streaming service due to low transmission rate that result from buffer underflow. In other words, one block is transmitted one block to client's one request according to the internet environment of limited resources. The proposed scheme adaptively adjusts the rate of CON(Confirmable) message among all messages for predicting the exact network condition. Based on this, the number of blocks is determined by using buffer occupancy rate and content download rate. Therefore it improves the quality of user experience by mitigating playback interruption. Experimental results show that the proposed scheme solves the buffer underflow problem in Internet of Things streaming environment by controlling transmission rate according to the network condition.

• Journal of Advanced Navigation Technology
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• v.14 no.3
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• pp.358-366
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• 2010

In the wireless network the congestion and delay occurs mainly when there are too many packets for the network to process or the sender transmits more packets than the receiver can accept. The congestion and delay is the reason of packet loss which degrades the performance of multimedia streaming. This paper proposes a novel transmission rate monitoring-based optimization mechanism to optimize packet loss and to improve QoS. The proposed scheme is based on the trade-off relationship between transmission rate monitoring and overhead monitoring. For this purpose this paper processes a source rate control-based optimization which optimizes congestion and delay. Performance evaluated RED, TFRC, and the proposed mechanism. The simulation results show that the proposed mechanism is more efficient than REC(Random Early Detection) mechanism and TFRC(TCP-friendly Rate Control) mechanism in packet loss rate, throughput rate, and average response rate.

• Journal of KIISE:Information Networking
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• v.30 no.2
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• pp.260-269
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• 2003

The traditional distributed object middleware platform is not suitable for the transmission of stream data, because RPC(Remote Procedure Call)-based message transmission have a great overhead. Therefore, the OMG(Object Management Group) proposes the AV(Audio and Video) stream reference model for streaming on the distributed object middleware platform. But, this reference model has not a detail of implementation. Particularly it also has not congestion control scheme for improvement of network efficiency on the real network environment. It is a very important and difficult technical issue to provide the stream transmission platform with advanced congestion control scheme. In this paper, we propose an architecture of a distributed stream transport platform and deal with the design and implementation concept of our proposed platform. Also, we present a mechanism to improve streaming utilization by SRTP(Smart RTP). SRTP is our proposed TCP-Friendly scheme.