• Journal of the Korea Institute of Information and Communication Engineering
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• v.9 no.8
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• pp.1696-1702
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• 2005

Real Time Transport Protocol (RTP) is the Internet standard protocol for transport of real time data audio/video IP Telephony, Multimedia Seivece. In case of 8kbps voice codec, the size of packet per data is 20bytes and become more large to minimal 40bytes with adding each layer's header in RTP/UDP/IP. To solve this problem, various header compression skill were suggested on point-to-point networks. But it compress even IP header and cannot be suitable to apply to end-to-end network Thus, We will renew header compression protocol to apply wired router-based network.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• v.9 no.2
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• pp.146-149
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• 2005

Confusion of network traffic is increased by increasing of internet user and large of network, Specially olded one and one communication caused loss of bandwidth because of redundant packet by increasing video conference and internet broadcasting. Thereupon multicast technique, method reducing loss of bandwith, for multimedia data transmission was proposed. This paper proposes method to solve overhead problem in the middle router through group management and capsuling with the Xcast technique added Disignated Router(DR). To solve the middle router not supporting IPv6, Xcast using tunneling technique in the IPv6 design and analyze the performance through a simulated examination.

• The Transactions of the Korea Information Processing Society
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• v.6 no.2
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• pp.459-466
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• 1999

I broadband ISDN every packet will show different result if it would be processed according to its usage by the server. That is, normal data won't show big differences if they would be processed at normal speed. But it will improve the quality of service to process some kinds of data - for example real time video or voice type data or some data for a bid to by something through the internet - more fast than the normal type data. solution for this problem was suggested - priority packets. But the analyses of them are under way. Son in this paper a switching system for an output queuing model in a single server was assumed and some packets were given priorities and analysed. And correlation, simulating real life situation, was given too. These packets were analysed through three cases, first packets having no correlation, second packets having only correlation and finally packets having priority three cases, first packets having no correlation, second packets having only correlation and finally packets having priority and correlation. The result showed that correlation doesn't affect the mean delay time and the high priority packets have improved mean delay time regardless of the arrival rate. Those packets were assumed to be fixed-sized like ATM fixed-sized cell and the contention strategy was assumed to be short bus contention method for the output queue, and the mean delay length and the maximum 버퍼 length not to lose any packets were analysed.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.39 no.6
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• pp.623-629
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• 2002

In transmitting compressed video bit-stream over Internet, packet losses cause error propagations in both spatial and temporal domains, which in turn leads to severe degradation I image quality. In this paper, a new error concealment algorithm, called EBMA(Edge Detection based Boundary Matching Algorithm), is proposed to repair damaged portions of the video frames in the receiver. Conventional BMA(Boundary Matching Algorithm) assumes that the pixels on the boundary of the missing block and its neighboring blocks are very similar, but has no consideration of edges across the boundary. In our approach, the edges are detected across the boundary of the lost or erroneous block. Once the orientation of each edge is found, only the pixel difference along the expected edges across the boundary is measured instead of the calculation of difference along the expected edges across the boundary is measured instead of the calculation of differences between all adjacent pixels on the boundary Therefore, the proposed approach needs very few computations and the experiment shows and improvement of the performance over the conventional BMA in terms of both subjective and objective quality of video sequences.

• Journal of KIISE:Information Networking
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• v.33 no.4
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• pp.333-342
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• 2006

The cdma2000 lxEV - DO mobile communication system provides broadcast and multicast services (BCMCS) to meet an increasing demand from multimedia data services. The servicing of video streams over a BCMCS network must, however, face a challenge from the unreliable and error-prone nature of the radio channel. The BCMCS network uses Reed-Solomon coding integrated with the MAC protocol for error recovery. We analyze this coding technique and show that it is not effective in the case of slowly moving mobiles. To improve the playback quality of an MPEG-4 FGS video stream, we propose the Hybrid error recovery scheme, which combines Reed-Solomon with ARQ, using slots which are saved by reducing the Reed-Solomon coding overhead. The target packets to be retransmitted are prioritized by a utility function to reduce the packet error rate in the application layer within a fixed retransmission budget. This is achieved by considering of the map of the error control block at each mobile node. The proposed Hybrid error recovery scheme also uses the characteristics of MPEG-4 FGS (fine granularity scalability) to improve the video quality even when conditions are adverse: slow-moving nodes and a high error rate in the physical channel.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.39 no.6
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• pp.53-67
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• 2002

The multimedia synchronization is one of the key issues to be resolved in order to provide a good quality of multimedia related services, such as Video on Demands(VoD), Lecture on Demands(LoD), and tele-conferences. In this paper, we introduce an adaptive multimedia synchronization algorithm using the level of buffers and load of servers, which are modeled and analyzed by ExSpect, a Petri net based simulation tool. In the proposed algorithm, the audio and video buffers are divided to 5 different levels, and the pre-defined play-out speed controller tries to make the buffer level to be normal in different temporal relations between multimedia streams using buffer levels and server loads. Because each multimedia packet is played by the pre-defined play-out speed, the media data can be reproduced within the permissible limit of errors while preserving the level of buffers to be normal. The proposed algorithm is able to handle and support various communication restrictions between providers and users, and offers little jitter play-out to many users in networks with the limited transmission capability. The performance of the developed algorithm is analyzed in various network conditions using a Petri net simulation tool.

• The KIPS Transactions:PartC
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• v.13C no.4 s.107
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• pp.471-480
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• 2006

Recently, multimedia communication services, such as video conferencing and voice over IP, have been rapidly spread. H.323 is an international standard that specifies the components, protocols and procedures that provide multimedia communication services of real-time audio, video, and data communications over packet networks, including IP based networks. H.323 is applied to many commercial services because it supports various network environments and has a good performance. But communication services based on H.323 may have some problem because of current network trouble or mis-implementation of H.323. The understanding of this problem is a critical issue because it improves the quality of service and is easy to service maintenance. In this paper, we implement the analysis system for H.323 protocol wihch includes H.245, H.225.0, RTP, RTCP, and so on. Tills system is able to capture, parse, and present the H.323 protocol in real-time. Through the operation test and performance evaluation, we prove that our system is a useful to analyze and understand the problems for communication services based on H.323.

• The KIPS Transactions:PartB
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• v.9B no.5
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• pp.571-578
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• 2002

In this paper, we propose a video sequence coding scheme called AMV (Auxiliary Motion Vector) to minimize error propagation caused by transmission errors over the Internet. Unlike the conventional coding schemes the AMY coder, for a macroblock in a frame, selects two best matching blocks among several preceding frames. The best matching block, called a primary block, is used for motion compensation of the destination macroblock. The other block, called an auxiliary block, replaces the primary block in case of its loss at the decoder. When a primary block is corrupted or lost during transmission, the decoder can efficiently and simply suppress error propagation to the subsequent frames by replacing the block with an auxiliary block. This scheme has an advantage of reducing both the number and the impact of error propagations. We implemented the proposed coder by modifying H.263 standard coding and evaluated the performance of our proposed scheme in the simulation. The simulation results show that AMV coder is more efficient than the H.263 baseline coder at the high packet loss rate.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.47 no.1
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• pp.171-184
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• 2010

In this work, we develop a novel channel power allocation method for the real-time multimedia over the wireless network environment. Since each frame has different effect on the user perceivable QoS, improving packet loss does not necessarily coincide with perceivable improvements in QoS. A new channel power control scheme is suggested based on the quantified importance of each frame in terms of user perceivable QoS. Dynamic programming formulation is used to obtain optimal transmit power which minimizes power consumption and maximizes user perceivable QoS simultaneously. The experiment is performed by using publicly available video clips. The performance is evaluated using network simulator version 2 (NS 2) and decoding engine is embedded at the client node, and calculated PSNR over the every frame transmitted. Through the semantics aware power allocation (SAPA) scheme, significant improvement on the QoS has been verified, which is the result of unequal protection to more important packets. SAPA scheme reduced the loss of I frame by upto 27% and reduced power consumption by upto 19% without degradation on the user perceivable QoS.

• Journal of KIISE:Information Networking
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• v.29 no.4
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• pp.375-385
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• 2002

The IETF has defined the Intserv model and the RSVP signaling protocol to improve QoS capability for a set of newly emerging services including voice and video streams that require high transmission bandwidth and low delay. However, since the current Intserv model requires each router to maintain the states of each service flow, the complexity and the overhead for processing packets in each rioter drastically increase as the size of the network increases, giving rise to the scalability problem. This motivates our work; namely, we investigate and devise new control schemes to enhance the scalability of the Intesev model. To do this, we basically resort to the SCORE network model, extend it to fairly well adapt to the three services presented in the Intserv model, and devise schemes of the QoS scheduling, the admission control, and the edge and core node architectures. We also carry out the computer simulation by using ns-2 simulator to examine the performance of the proposed scheme in respects of the bandwidth allocation capability, the packet delay, and the packet delay variation. The results show that the proposed scheme meets the QoS requirements of the respective three services of Intserv model, thus we conclude that the proposed scheme enhances the scalability, while keeping the efficiency of the current Intserv model.