• Proceedings of the Korea Multimedia Society Conference
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• 2003.11b
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• pp.503-507
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• 2003

사용자의 고화질 요구와 사업자의 차별화된 서비스 제공 노력의 결과로 기존의 MPEG-4 기반이 아닌 고화질 전용의 MPEG-2 화질을 사용하는 VOD 서비스가 새로운 대안으로 제시되고 있다. MPEG-2 비디오는 높은 네트워크 대역폭을 요구하는 단점이 있는 반면, 사용자에게 양질의 화질을 제공할 수 있으며 표준의 사용으로 컨텐츠 유지. 보수에 유리하다. 본 논문에서는 상용 스트리밍 서버인 nCUBE 서버와 연동하여 MPEG-2 TS 데이터를 스트리밍 하는 VOD 시스템에 대하여 설명한다. VOD 제어 프로토콜로 RTSP(Real Time Streaming Protocol)를 사용하였으며, 스트림 전송 프로토콜로 UDP/IP 방식을 사용하였다. 지원하는 VCR 기능으로는 FF, RW, STOP. Pause가 있다.

• Proceedings of the Korea Information Processing Society Conference
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• 2003.05c
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• pp.2229-2232
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• 2003

인터넷이 활성화되면서 다양한 서비스가 제공되어지고 있다. 그 가운데 하나가 MOD(Multimedia on Demand) 서비스이다. MOD는 다양한 프로그램을 사용자가 원하는 시점에 볼 수 있도록 한다. MOD 서비스를 유료화 할 경우 시청한 양만큼 요금을 지불하는 pay-per-view 방법을 생각할 수 있다. 본 논문에서는 RTP(Realtime Tansport Protocol)/RTSP(Real Time Streaming Protocol)를 이용한 스트리밍 서버와 클라이언트를 구현하고 여기에 PPV(Pay-Per-View)를 적용해 본다.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.8 no.11
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• pp.4068-4086
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• 2014

Due to the prevalence of powerful mobile terminals and the rapid advancements in wireless communication technologies, the wireless video streaming service has become increasingly more popular. Recent studies show that video streaming services via Transmission Control Protocol (TCP) are becoming more practical. TCP has more advantages than User Diagram Protocol (UDP), including firewall traversal, bandwidth fairness, and reliability. However, each video service shares an equal portion of the limited bandwidth because of the fair sharing characteristics inherent in TCP and this bandwidth fair sharing cannot always guarantee the video quality for each user. To solve this challenging problem, an Adaptive Multiple TCP (AM-TCP) scheme is proposed in this paper to guarantee the video quality for mobile devices in wireless networks. AM-TCP adaptively controls the number of TCP connections according to the video Rate Distortion (RD) characteristics of each stream and network status. The proposed scheme can minimize the total distortion of all participating video streams and maximize the service quality by guaranteeing the quality of each video streaming session. The simulation results show that the proposed scheme can significantly improve the quality of video streaming in wireless networks.

• Proceedings of the Korean Information Science Society Conference
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• 2005.07a
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• pp.799-801
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• 2005

There have been many streaming servers that provide a large number of contents for a user's preference. General purpose streaming sewer makes use of a RTSP protocol for streaming controls such as message passing with client players. To date, there has been minimal research regarding streaming server's performance test tools. For measuring streaming server's performance, performance evaluation technique is needed and also achieved by RTSP based controls, a server's performance result and its miscellaneous test tools such the PseudoPlayer for pumping data to a specified port and the PseudoMonitor for gathering information. In this paper, We implement a test toolkit for evaluating a streaming server's performance and show the case of its application

• Journal of Broadcast Engineering
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• v.10 no.3
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• pp.297-312
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• 2005

The multimedia traffic of continuous video and audio data via streaming service accounts for a significant and expanding portion of the Internet traffic. This streaming data delivery is mostly based on RTP with UDP. However, UDP does not support congestion control. For this reason, UDP causes the starvation of congestion controlled TCP traffic which reduces its bandwidth share during overload situation. In this paper, we propose a new TCP-friendly rate control scheme called 'TF-RTP(TCP-Friendly RTP)'. In the congested network state, the TF-RTP exactly estimates the competing TCP's throughput by using the modified parameters. Then, it controls the sending rate of the video streams. Therefore, the TF-RTP adjusts its sending rate to TCP-friendly and fair share with competing TCP traffics. Through the simulation, we prove that the TF-RTP correctly estimates the TCP's throughput and improves the TCP-friendliness and fairness.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2003.10a
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• pp.837-840
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• 2003

Recently the multimedia streaming traffic such as digital audio and video in the Internet has increased tremendously. Unlike TCP, the UDP protocol, which has been used to transmit streaming traffic through the Internet, does not apply any congestion control mechanism to regulate the data flow through the shared network. If this trend is let go unchecked, these traffic will effect the performance of TCP, which is used to transport data traffic, and may lead to congestion collapse of the Internet. To avoid any adverse effort on the current Internet functionality, A study on a new protocol of modification or addition of some functionality to existing transport protocol for transmitting streaming traffic in the Internet is needed. TCP-frienly congestion control mechanism is classified with window-based congestion control scheme and rate-based congestion control scheme. In this paper, we propose an algorithm for improving the transmitting rate on a hybrid TCP-friendly congestion control scheme combined with widow-based and rate-based congestion control for multimedia streaming in the internet.

• Journal of the Korea Institute of Military Science and Technology
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• v.23 no.4
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• pp.371-380
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• 2020

Video is principal information that facilitates commander's immediate command decision. Due to fading characteristics of radio link, however, it is difficult to stably transmit video in a multi-hop wireless environment. In this paper, we propose a MANET structure composed of a link adaptive routing protocol and a TDMA MAC protocol to stably transmit video traffic in a ultra-narrowband video streaming network. The routing protocol can adapt to link state change and select a stable route. The TDMA protocol enables collision-free video transmission to a destination using multi-hop dynamic resource allocation. As a result of simulation, the proposed MANET structure shows better video transmission performance than proposed MANET structure without link quality adaption, AODV with CSMA/CA, and OLSR with CSMA/CA structures.

• Journal of KIISE
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• v.41 no.11
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• pp.974-981
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• 2014

These days, use of multimedia streaming service and demand of QoS (Quality of Service) improvement have been increased because of development of network. QoS of streaming service is influenced by a jitter, delay, throughput, and loss rate. For guaranteeing these factors which are influencing QoS, the role of transport layer is very important. But existing TCP which is a typical transport layer protocol increases the size of congestion window slowly and decreases the size of a congestion window drastically. These TCP characteristic have a problem which cannot guarantee the QoS of UHD multimedia streaming service. In this paper, we propose a router buffer based congestion control method for improving the QoS of UHD streaming services. Our proposed scheme applies congestion window growth rate differentially according to a degree of congestion for preventing an excess of available bandwidth and maintaining a high bandwidth occupied. Also, our proposed scheme can control the size of congestion window according to a change of delay. After comparing with other congestion control protocols in the jitter, throughput, and loss rate, we found that our proposed scheme can offer a service which is suitable for the UDH streaming service.

• The Journal of the Acoustical Society of Korea
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• v.21 no.2
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• pp.93-101
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• 2002

This paper presents the Internet audio broadcasting technology based on the streaming technology. In this paper, we choose the MPEG-2 AAC for multimedia data, and for the streaming of this data we use RTP/RTCP protocol. We use RTSP protocol for the control of streaming data and TCP/IP for the exchange of information between server and client. By using all of these protocols and MPEBG-2 AAC, we explain the implementation method for the unicast/multicast streaming server/client system. Our system was tested by ETRI intranet, which is connected by 2000 researchers. Experimental result show that our system can be process the packet loss and jitter by retransmission and variable length buffer. Multicast streaming server can be used for the audio broadcasting service inside the company, unicast streaming server can be used for the AOD (Audio On Demand) service.

• 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.