• KIISE Transactions on Computing Practices
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• v.21 no.12
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• pp.780-785
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• 2015

Recently HTTP (Hypertext Transfer Protocol) adaptive streaming services have been the subject of much attention. The video quality control scheme of conventional HTTP adaptive streaming services estimates bandwidth using segment throughput and smooths out the sample of segment throughput. However, the conventional scheme has the problem of QoE (Quality of experience) degradation occurring with buffer underflow and frequent quality change due to the fixed number of samples. In order to solve this problem, we propose a video quality control scheme based on segment throughput and buffer occupancy. The proposed scheme determines the number of samples according to the variation of segment throughput. The proposed scheme also controls video quality based on the threshold of bitrate to keep stable buffer occupancy. The simulation results show that proposed scheme improves QoE by preventing buffer underflow and decreasing quality change when compared with the conventional scheme.

• 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 the Korea Institute of Information and Communication Engineering
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• v.17 no.8
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• pp.1851-1856
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• 2013

An XML is widely used for web services, such as SOAP(Simple Object Access Protocol) and REST (Representational State Transfer), and also de facto standard for representing data. Since the XML parser using DOM(Document Object Model) requires a preprocessing task creating a DOM-tree, and then storing it into memory, embedded systems with limited resources typically employ a streaming XML parser without preprocessing. In this paper, we propose a new architecture for the streaming XML parser using an APSearch(Adaptive Parallel Search) on FPGA(Field Programmable Gate Array). Compared to other approaches, the proposed APSearch parser dramatically reduces overhead on the software side and achieves about 2.55 and 2.96 times improvement in the time needed for an XML parsing. Therefore, our APSearch parser is suitable for systems to speed up XML parsing.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2009.01a
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• pp.668-672
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• 2009

In wireless ad-hoc network, knowing the available bandwidth of the time varying channel is imperative for live video streaming applications. This is because the available bandwidth is varying all the time and strictly limited against the large data size of video streaming. Additionally, adapting the encoding rate to the suitable bit-rate for the network, where an overlarge encoding rate induces congestion loss and playback delay, decreases the loss and delay. While some effective rate controlling methods have been proposed and simulated well like VTP (Video Transport Protocol) [1], implementing to cooperate with the encoder and tuning the parameters are still challenging works. In this paper, we show our result of the implementation experiment of VTP based encoding rate controlling method and then introduce some techniques of our parameter tuning for a video streaming application over wireless environment.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2009.01a
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• pp.49-54
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• 2009

Synchronization between media is an important aspect in the design of multimedia streaming system. This paper proposes a precise media synchronization mechanism for digital video and audio transport over IP networks. To support synchronization between video and audio bitstreams transported over IP networks, RTP/RTCP protocol suite is usually employed. To provide a precise mechanism for media synchronization between video and audio, we suggest an efficient media synchronization algorithm based on NPT (Normal Play Time) which can be derivable from the timestamp information in the header part of RTP packet generated for the transport of video and audio streams. With the proposed method, we do not need to send and process any RTCP SR (sender report) packet which is required for conventional media synchronization scheme, and accordingly could reduce the number of required UDP ports and the amount of control traffic injected into the network.

• 대한공업교육학회지
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• v.32 no.2
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• pp.199-216
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• 2007

Stream Control Transmission Protocol(SCTP) is a transport layer protocol to support the data transmission. SCTP is similar to Transmission Control Protocol(TCP) in a variety of aspects. However, several features of SCTP including multi-homing and multi-streaming incur the performance difference from TCP. This paper highlights the data transfer during the initial slow start phase in SCTP congestion control composed of slow start phase and congestion avoidance phase. In order to compare the mean transfer time between SCTP and TCP, we experiment with different performance parameters including bandwidth, round trip time, and data length. By varying data length, we also measure the corresponding initial window size, which is one of factors affecting the mean transfer time. For the experiment, we have written server and client applications by C language using SCTP socket API and have measured the transfer time by ethereal program. We transferred data between client and server using round-robin method. Analysis of these experimental results from the testbed implementation shows that larger initial window size of SCTP than that of TCP brings the reduction in the mean transfer time of SCTP compared with TCP by 15 % on average during the initial slow start phase.

• Electronics and Telecommunications Trends
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• v.18 no.3 s.81
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• pp.11-20
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• 2003

최근 SCTP(Stream Control Transmission Protocol)는 TCP/UDP 이후의 차세대 수송계층 프로토콜로서 주목 받고 있다. SCTP는 기존 TCP 및 UDP의 문제점을 극복하도록 설계되었으며 특히 multi-streaming 및 multi-homing 특성을 제공한다. 본 고에서는 SCTP 프로토콜의 기본 특징에 대하여 알아보고, 현재 논의중인 확장작업의 주요 골자를 살펴본다.

• Proceedings of the Korea Multimedia Society Conference
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• 2003.11a
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• pp.457-460
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• 2003

RTS(Real Time Streaming Protocol) 프로토콜은 어플리케이션 레벨에서 실시간 데이터 전송의 세션과 제어를 표현한다. 하지만, RTSP는 오든 스트리밍 솔루션에 대한 내용을 포함하고 있어 그 내용이 포괄적이고. 회사별 자신의 응용과 환경에 맞게 RTSP 표준을 구현하고 있어 상호호환이 어렵다. 본 논문에서는 주문형 미디어 시스템에 필요한 기본 기능을 RTSP 표준을 이용하여 어떻게 구현해야 하는지에 대한 방법을 제시함으로써, RTSP 지원 업체간의 호환성을 유도한다.

• Proceedings of the Korea Information Processing Society Conference
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• 2002.11b
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• pp.1357-1360
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• 2002

무선 데이터 통신이 하루가 다르게 증가하면서 무선망에서 핸드폰을 이용해 멀티미디어 데이터를 스트리밍하는 기술들이 연구되며 소개되어지고 있다. 이러한 기술을 연구하는데 있어서는 첫째로 데이터 전송 실험을 통한 무선 망에 대한 이해 및 특성 파악이 필요하고, 둘째로 이런 망 특성에 따른 적합한 스트리밍 기법 연구가 필요하다. 본 논문에서는 이러한 방법에 따라 망의 특성 파악에 관한 데이터를 기반으로 실험을 통해 EVRC 오디오 코텍을 이용한 핸드폰에서의 오디오 스트리밍에 있어 좀 더 나은 에러 대처 방법과 트랜스포트 프로토콜의 사용 기법을 제안하고 있다.

• Proceedings of the Korea Multimedia Society Conference
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• 2003.05b
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• pp.318-321
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• 2003

최근 ITU와 FS-VDSL 등의 국제 표준 기구에서 ATM과 IP 기반에서 VDSL 기술을 이용한 VOD 서비스 실현을 위해 서비스, 망 구조 및 프로토콜 등에 대해 연구가 진행 중에 있다 본 논문에서는 ATM/VDSL 기반의 VOD 서비스 제공을 위해 종단간 ATM 모델의 망 구성과 프로토콜 스택을 제시하고 분석한다. 그리고, 이 모델에 대해 IGMP (Internet Group Management Protocol)HTSP(Real Time Streaming Protocol)와 DSM-CC (Digital Storage Media Command and Control) 을 이용하는 서버와 셋탑박스 간의 VOD 서비스 신호 절차를 제안하고 분석한다.