• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.6C
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• pp.614-622
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• 2006

In this paper, we propose an efficient packetization method to reduce the packetization overhead. For the purpose, we firstly verify the relationship between packet length and packet loss rate. The empirical results show that as the packet length is larger than the path MTU, the packet loss rate is drastically increased, producing poor visual quality at the receiver side. However, as the length of the packet is reduced, we should transmit more packets per frame and the packetization overhead will be increased. This increase in the packetization overhead reduces the number of bits allocated to the video data, resulting in the low visual quality. Therefore, each packet should be packetized to have the packet length close to the path MTU. In this paper, we show that the this process of the packetization with the constraint on the packet length is very similar to the dynamic storage allocation in the operating system. We had thoroughly surveyed the dynamic storage allocation methods used in the recent operating systems and propose to use the allocation methods for the video packetization. We empirically show that the proposed method can reduce the packetization overhead upto 28.3%, compared with the conventional sequential packetization method which have been widely used in Internet video transmission.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.36 no.9A
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• pp.749-757
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• 2011

In this paper, we propose a robust And-Or tree-based Luby transform encoded symbol packetization algorithm to minimize the quality degradation of video streaming service caused by packet losses over wireless network. To achieve this goal, the relationship among Luby transform encoded symbols are analyzed by using And-Or tree, and the proposed packetization algorithm is designed to minimize packet loss effects by reducing the dependency among packets conveying Luby transform encoded symbols. Finally, experimental results are provided to show the performance of the proposed algorithm.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.8C
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• pp.810-823
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• 2005

The transmission of duplicate packets provides a great loss-resilience without undue time-delay in the video transmission over packet loss networks. But this method generally deteriorates the problem of traffic congestion because of the increased bit-rate required for duplicate transmission. In this paper, we propose an efficient packetization and duplicate transmission of video packets. The proposed method transmits only the video signal with high priority for each video macroblock that is quite small in volume but very important for the reconstruction of the video. The proposed method significantly reduces the required bit-rate for duplicate transmission. An efficient packetization method is also proposed to reduce additional packet overhead which is required for transmitting the duplicate data. The duplicated high priority data of the Previous video slice is transmitted as a Piggyback to the data Packet of the current video slice. It is shown by simulations that the proposed method remarkably improves the packet loss-resilience for video transmission only with small increase of redundant duplicated data for each slice.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.11C
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• pp.1068-1075
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• 2005

In this paper we propose a video coding method and associated packetization and decoding methods for error resilient transmission over the Internet. The proposed method re-organizes the input image into several mutually similar subimages. For this case, if the one of the subimage is lost in the network, the lost one is recovered by the proposed error concealment method which uses the correctly received other subimages. The performance of the proposed method is confirmed by the empirical results. The proposed method is not limited to the Internet communications but is applicable to the other packet-based networks.

• Proceedings of the Korea Information Processing Society Conference
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• 2001.04b
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• pp.623-626
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• 2001

본 논문에서는 별도 장비에 의하여 음성 서비스와 ADSL 인터넷 접속 서비스를 각각 제공하는 현재의 통신망 환경에서 음성 서비스와 ADSL 서비스를 한 시스템에서 모두 제공하는 통합 액세스 장비를 이용한 액세스 망의 패킷화 방법과, VoATM/IP 게이트웨이를 이용한 중계망을 패킷화 방법, 그리고 통합 액세스 장비, VoATM/IP Gateway 그리고 SoftSwitch를 결합하여 통신망 전체를 패킷화 하는 차세대 통신망 진화 전략을 비교한다. 또한 경제성을 최대로 활용한 통신망 진화 전략과 패킷화 도입에 따른 고려사항을 방법을 도출하고자 한다.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2017.05a
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• pp.621-624
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• 2017

In this paper, we propose a protocol synchronizing to packet trains in a underwater acoustic networks with limitations such as long propagation delay, low data rate, high packet loss. The proposed protocol employs an one-way communication scheme to calculate packet arrival time differences of consecutively transferred packets and minimizes uncertain variation effect of long propagation delay. Simulation results shows the superiority of our protocol, compared with a previously proposed one under the given network conditions.

• 전자공학회논문지 IE
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• v.48 no.3
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• pp.16-19
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• 2011

Differentiated Service is an IP QoS ensuring method based on packet marking that allows packets to be prioritized according to user requirements. During the time of congestion, more low priority packets are dropped than high priority packets. Different policy models are used to determine how to mark the packet. This paper investigated the performance of Differentiated Service using time sliding window with 3 color marking (TSW3CM). Simulation results using NS-2 showed that Differentiated Service can provide the quality of service requirements.

• The KIPS Transactions:PartB
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• v.10B no.5
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• pp.553-560
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• 2003

Since Wavelet Transform decomposes a video frame into subbands with various frequencies and resolutions, the reconstructed video qualify at a receiver fluctuates according to the location of transmission errors within frames. This deteriorates the whole visual duality of the video. Specifically, for a wavelet based video which exploits the motion estimation prediction scheme, the transmission errors of a subband not only have a bad effect on other subbands within a same frame but also propagates to the subsequent frames. In this paper, we propose BDP(Block Based Dispersive Packetization) scheme, for a wavelet based video stream, which maintains constant video quality despite packet location that a transmission error occurs. To evaluate the performance of the proposed scheme, we use MRME(Multi-Resolution Motion Estimation) scheme to compress a video in Inter coding mode and Gilbert´s error model to generate the error patterns in wireless network environment. The simulation results show that BDP is more efficient than BP (Block based Packetization) or DP (Dispersive Packetization) in both PSNR and visual quality.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2008.11a
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• pp.239-242
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• 2008

지상파 DMB는 이동 환경에서 QVGA 급의 영상과 스테레오 오디오를 제공하는 방송 서비스로서 2005년 12월부터 본격적으로 서비스되고 있는데, 최근에는 DMB 환경에서 고품질의 영상과 오디오를 제공하려는 기술에 대한 연구가 이루어지고 있다. 지상파 DMB 환경에서 고품질의 영상 또는 오디오를 제공하기 위해서는 기존의 DMB 서비스에 추가적인 데이터들을 전송하는 것이 필요한데, 하나의 지상파 DMB 방송 채널에 할당되는 전송 비트율이 높지 않다는 점을 감안하면, 이러한 추가적인 데이터들을 효율적으로 전송하는 것이 서비스의 상용화 입장에서는 중요한 요소가 될 수 있다. 본 논문에서는 지상파 DMB 환경에서 멀티채널 오디오 서비스를 제공하고자 할 때, 추가적으로 전송되어야 하는 부가정보 스트림의 효율적인 전송을 위한 패킷화 방법을 제안한다. 지상파 DMB 환경에서 멀티채널 오디오 서비스를 제공하기 위한 부가정보 스트림은 일반 오디오 스트림과 마찬가지로 프레임 단위로 생성이 되며, 약 12kbps의 비트율을 가진다. 그러나, 부가정보 스트림을 지상파 DMB 환경에서 전송하기 위하여, MPEG-2 TS로 패킷화하여 전송하게 되면, 부가정보 스트림의 비트율보다 훨씬 높은 약 32kbps의 전송율을 가지게 된다. 본 연구에서는 이와 같은 문제점을 해결하기 위하여, 멀티채널 오디오 서비스를 위해 필요한 부가정보 스트림의 비트율을 분석하고, 이를 바탕으로 하나의 TS 패킷에 하나 이상의 부가정보 프레임을 포함하여 전송하는 방법을 제안한다. 제안한 방법의 성능 검증을 위해 제안한 방법에 따라 하나 이상의 부가정보 프레임을 하나의 TS 패킷에 포함하여 패킷화하는 것을 시뮬레이션하고, 그 결과를 제시하였다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.27 no.1C
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• pp.48-58
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• 2002

As internet traffic increases, the problem that it should be efficiently accepted in to the existing voice network in the pending problem importantly to the existing communication corporations. The feature of next generation exchange network is made up of the form of integration network that connect data traffic(internet service. Etc) with the existing voice network and it can be showed with very diverse aspects according to the constitution time of network or the characteristics of business. The progress strategy that develop the existing circuit based communication network into packet-based communication network can be divided into two in a large scale according to the application position These are VoDSL application method(Technology that packetize access network) and softswitch technology application method(after packetizing relay network, packetizing that into the access network). In this paper, we will deduce the desirable technology that can construct packet-based next generation exchange networks in the structure of the existing communication network environment. We will perform the research on a device to offer the necessary core technique VoSDL service with realistic resolutions primarily.