• Journal of the Korea Institute of Information and Communication Engineering
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• v.8 no.3
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• pp.637-644
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• 2004

This paper proposed and implemented the adaptive transmission middleware for video streaming, which is able to support the adaptive transmission of video data to the fluctuating changes of network environment in the packet-based network and the properties of transmitted video data. The adaptive transmission middleware is made up SR-RTP-based transfer module and TFRC(TCP Friendly Rate Control)-based transfer-rate control module. The SR-RTP-based transfer module supports RTP-based real-time transfer of video data and packet retransmission scheme retransmitting the high-priority packets selectively in the damaged video data to reduce the error induced by the packet loss. Sharing the transmission bandwidth of network with the TCP-based data transfer, the TFRC-based transfer-rate control module controls the transfer rate of video data according to the most allowable transmission bandwidth in the network, so that the transfer rate is controlled adaptively to the fluctuating changes of transmission bandwidth. This paper, for the experiment, applied the adaptive transmission middleware to video streaming in the external Internet environment, and analyzed the effective frame transfer rate and the degree of the streaming jitter to evaluate the performance of packet-loss recovery and adaptive transfer rate control. In the external Internet environment where the packet-loss rate is high a bit, the relatively high streaming performance was showed compared with the case that didn't apply the adaptive transmission middleware.

• Journal of Korea Multimedia Society
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• v.12 no.5
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• pp.653-663
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• 2009

Synchronization between media is an important aspect in the design of multimedia communication-system. This paper proposes a precise media synchronization mechanism for 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. In 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.

• Proceedings of the Korea Contents Association Conference
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• 2009.05a
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• pp.491-496
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• 2009

In this paper, we propose USN Protocol of Room Control with high efficiency and new routing techniques. I establish a network of cluster hierarchy structure under bases standard for IEEE 802.15.4 and another networks to manage each room use a control packet transmitted and The proposed protocol does not perform BroadCast over a network and transmits a packet to same layer to the destination and then passes the packet after doing searches. If a destination to be transmitted cannot be looked up, the protocol transmits AODV routing packet so that not only traffic load of a wireless network is decreased, but also efficient routing path can be obtained.

• Journal of Broadcast Engineering
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• v.16 no.3
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• pp.530-541
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• 2011

In this paper, we propose an effective network-adaptive ARQ based error control scheme to provide video streaming services through IP networks where packet error usually occurs. If time delay and feedback channel are allowed, client can request server to retransmit lost packets through IP networks. However, if retransmission is unconditionally requested without considering network condition and number of simultaneous feedback messages, retransmitted packets may not arrive in a timely manner so that decoding may not occur. In the proposed ARQ, a client conditionally requests retransmission based on assumed network condition, and it further determines valid retransmission time so that effective ARQ can be applied. In order to verify the performance of the proposed adaptive ARQ based error control, NIST-Net is used to emulate packet-loss network environment. It is shown by simulations that the proposed scheme provides noticeable error resilience with significantly reduced traffics required for ARQ.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.26 no.12B
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• pp.1665-1676
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• 2001

The effect of spreading gain control on the throughput of a CDMA slotted ALOHA system is considered. Mobile users transmit packets over a shared channel, and the packets transmitted in the same time slot over the shared channel act as simultaneous access interference (SAI). When using spread-spectrum signal, a CDMA slotted ALOHA channel achieves high probability of capture due to the property of high title resolution, and the bit rate of user information is determined by spreading gain. When the SAI level gets larger, the high value of spreading gain enhances the packet throughput by increasing the probability of a successful packet transmission, while it degrades the of the effective throughput by reducing the user information bits carried within a packer. To solve the problem, we investigated the effect of the capture probability and the SAI level on these system throughputs, and evaluated the throughput performance of the system for each spreading gain control scheme. The results showed that the maximum effective throughput could be achieved with an unified method despite the variation of the SAI level by deriving an optimal value of the spreading gain according to 171e system states.

• The KIPS Transactions:PartC
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• v.13C no.3 s.106
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• pp.345-352
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• 2006

In the integrated wired/wireless network which consists of the Internet and moving networks, multiple wireless links are used to connect a fixed host(FH) in the Internet to a mobile host(MH) in the moving network. For use in such an environment, we propose a scheme to overcome the TCP performance degradation due to the packet losses over the wireless links without losing the end-to-end TCP semantics. The proposed scheme in each mobile router(MR) allows to obtain the information regarding packet losses over the upstream wireless links based on the received packet sequence number and the ACK number. This information is delivered to the upstream router, which enables the upstream access router(AR) or MR to quickly retransmit the lost packets. The proposed scheme has the feature to quickly recover the packet losses incurred over the upstream wireless links and the performance of the proposed scheme is evaluated through simulation. It is shown that the significant performance gain can be obtained using the proposed scheme compared with the snoop mechanism which maintains end-to-end TCP semantics and does not require any additional features at the source and/or destination nodes.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.25 no.7
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• pp.923-929
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• 2021

TCP communication and packet communication require transmission control technology to ensure high quality and high reliability. However, in the case of real-time data transmission, an inefficient transmission problem occurs. In order to overcome this problem and transmit the packet reliability, in general, early congestion control using the buffer level as an index was used. Control of the congestion control point and the cancellation point is delayed because the point at which congestion is controlled is based on the buffer level. Therefore, in this paper, not only the buffer level indicator, but also the ideal buffer level, which determines the packet discard probability, is classified so that the transmission rate and buffer level that measure network congestion are close to the level above the optimal setting. As a result, it was shown that the average buffer level can be directly controlled by maintaining the average buffer level by the ideal buffer level set in the experiment to prove the proposed method.

• Proceedings of the Korean Information Science Society Conference
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• 2004.04a
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• pp.499-501
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• 2004

현재 IPv6에서 이동 통신에 끊김 없는 서비스를 제공하기 위해 Mobile IPv6에 관한 많은 연구가 이루어지고 있다. FMIPv6는 핸드오버 절차를 간소화시켜서 빠른 속도의 핸드오버와 데이터 전송이 가능하며 SIP나 VoIP 및 무선 인터넷 동영상 서비스와 같은 경우에 많이 사용될 것으로 예상되지만 패킷 손실의 문제점을 지니고 있다. 본 논문은 FMIPv6에서 핸드오버시 발생하는 패킷 손실을 줄이기 위하여 라우터에서 사용하는 여러 패킷 관리 스케줄링 기법 중 WFQ기법을 사용한 트래픽 관리 메커니즘을 설계하고 평가하였다.

• Proceedings of the Korea Multimedia Society Conference
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• 2003.11b
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• pp.786-789
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• 2003

현재 IPv6에서 이동 통신에 끊김 없는 서비스를 제공하기 위해 Mobile IPv6에 관한 많은 연구가 이루어지고 있다. FMIPv6는 핸드오버 절차를 간소화시켜서 빠른 속도의 핸드오버와 데이터전송이 가능하며 SIP나 VoIP 및 무선 인터넷 동영상 서비스와 같은 경우에 많이 사용될 것으로 예상되지만 패킷 손실의 문제점을 지니고 있다. 본 논문은 FMIPv6에서 핸드오버시 발생하는 패킷 손실을 줄이기 위하여 라우터에서 사용하는 여러 패킷 관리 스케줄링 기법 중 WFQ기법을 사용한 트래픽 관리 메커니즘을 설계하였다.

• Proceedings of the Korean Information Science Society Conference
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• 2000.10c
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• pp.600-602
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• 2000

본 논문에서는 VOD시스템, 혹은 영상회의 시스템에서의 오디오 패킷의 QOS를 보장해 주기 위해 클라이언트의 버퍼 상태에 따른 전송속도 제어와 부가 전송 기법을 동적으로 사용하여 재전송 함으로써 오디오 패킷의 손실문제를 해결하기 위한 방법과 함께 클라이언트의 버퍼 상황을 파악하여 전송속도를 조절하여 클라이언트의 버퍼에서의 오버프로우와 언더플로우를 방지하여 VOD와 영상회의 시스템에서 오디오 데이터의 전송시 안정적인 서비스를 보장할 수 있다.