• The Transactions of the Korea Information Processing Society
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• v.7 no.6
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• pp.1949-1955
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• 2000

This paper proposes a multistage interconnection ATM switching network without internal blocking. The first is recirculating shuffle-exchange network improved on hardware complexity. The next is connected to Rank network with tree structure. In this network, after the packets transferred to the same output ports are given each priority, only a packet with highest priority is sent to the next, an the others are recirculated to the first. Rearrangeability through decomposition and composition algorithm is applied for the transferred packets in hanyan network and all they arrive at a final destinations. To analyze throughput, waiting time and packet loss ratio according tothe size of buffer, the probabilities are modeled by a binomial distribution of packet arrival.

• Journal of Information Technology Applications and Management
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• v.23 no.2
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• pp.61-80
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• 2016

The wireless mesh network (WMN) is a next-generation technology for data networking that has the advantage in cost and the flexibility in its construction because of not requiring the infra-structure such as the ethernet. This paper focuses on the optimal link scheduling problem under the wireless mesh network to effectuate real-time streaming by using the constraint programming. In particular, Under the limitation of half-duplex transmission in wireless nodes, this paper proposes a solution method to minimize the makespan in scheduling packet transmission from wireless nodes to the gateway in a WMN with no packet transmission conflicts due to the half-duplex transmission. It discusses the conflicts in packet transmission and deduces the condition of feasible schedules, which defines the model for the constraint programming. Finally it comparatively shows and discusses the results using two constraint programming solvers, Gecode and the IBM ILOG CP solver.

• Proceedings of the KIEE Conference
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• 2006.04a
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• pp.105-107
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• 2006

As internet streaming data increase, transport protocol such as TCP, TGP-Friendly is important to study control transmission rate and share of Internet bandwidth. In this paper, we propose a TCP-Friendly protocol using Neural Network for media delivery over wired Internet which has various traffic size(PTFRC). PTFRC can effectively send streaming data when occur congestion and predict one-step ahead round trip time and packet loss rate. A multi-layer perceptron structure is used as the prediction model, and the Levenberg-Marquardt algorithm is used as a traning algorithm. The performance of the PTFRC was evaluated by the share of Bandwidth and packet loss rate with various protocols.

• Journal of the Korea Society of Computer and Information
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• v.8 no.3
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• pp.47-52
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• 2003

In this paper. a new wavelet packet image coder is proposed for images that include many high-frequency components using the relation between subbands. The new algorithm presents parent-child relationship for reducing image reconstruction error using relations between individual frequency subbands. By parent-child relationship, every coefficient is partitioned and encoded for the zerotree structure. It demonstrates higher PSNR under the same bit rate. These results show that the encoding process of the proposed coder is more accurate than the conventional ones for images that include many high-frequency elements.

• Journal of information and communication convergence engineering
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• v.3 no.2
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• pp.90-92
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• 2005

As the result of the study shows, Round Robin Algorithm allows servers to ensure definite traffic distribution, unless incoming data loads differ much. Although error levels are high in some cases, they were eventually alleviated by repeated tests for a long period of time. The study uses two software packages (Packet Capture and Round Robin Test Package) to check packet volume from Virtual Network Structure (data generator, virtual server, Server 1, 2, 3), and finds out traffic distribution toward Server 1, 2, and 3. The functions of implemented Round Robin Load Balancing Monitoring System include round robin testing, system monitoring, and graphical indication of data transmission and packet volume.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.20 no.5
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• pp.471-479
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• 2009

In this paper, we proposed an improved ray tracing method with an amelioration of visible tree structure, a visible face determination method, and non-uniform random test point method. In a proposed visible tree structure, it reduces tree nodes by means of merging similar nodes. In a visible face determination method, it shows that a ray hit test with a packet ray method can reduce a test time. A ray tracing method involving with a packet ray hit test method can improve a tree construction time up to 3.3 times than a ray tracing method with a single ray hit test method. Furthermore, by seeding a non-uniform and random test point on a face, tree construction time is improved up to 1.11 times. Received powers from the proposed ray tracing results and measured results have good agreement with 1.9 dB RMS error.

• Journal of Broadcast Engineering
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• v.1 no.1
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• pp.67-73
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• 1996

In general, the high quality audio coding(HQAC) has the structure of the convertional data compression techniques combined with moodels of human perception. The primary auditory characteristic applied to HQAC is the masking effect in the spectral domain. Therefore spectral techniques such as the subband coding or the transform coding are widely used[1][2]. However no effort has yet been made to apply the temporal masking effect and temporal redundancy removing method in HQAC. The audio data compression method proposed in this paper eliminates statistical and perceptual redundancies in both temporal and spectral domain. Transformed audio signal is divided into packets, which consist of 6 frames. A packet contains 1536 samples($256{\times}6$) :nd redundancies in packet reside in both temporal and spectral domain. Both redundancies are elminated at the same time in each packet. The psychoacoustic model has been improved to give more delicate results by taking into account temporal masking as well as fine spectral masking. For quantization, each packet is divided into subblocks designed to have an analogy with the nonlinear critical bands and to reflect the temporal auditory characteristics. Consequently, high quality of reconstructed audio is conserved at low bit-rates.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.19 no.1
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• pp.47-55
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• 1994

This paper investigates the performance analysis of the packet DS/SS receiver with a PJED(phase-jump error detector) using the block signal processing(BSP) methods. The conventional packet DS/SS block receiver has a high probability of mistaking the phase-jump detection, which causes the frequency estimation error. The conventional receiver uses a Matched-Pulse Timing Extractor which has a complicated structure. The proposed packet DS/SS block receiver with the PJED which uses libearity of the phase has little probability of mistaking the phase-jump detection. The proposed Matched Pulse Timing Extractor gas the more simple structure but obtains the same performance on the exact matched-pluse timing as the conventional one does. The simulation results show that the proposed receiver gives about 2dB improvement in the BER compared with the conventional receiver.

• The KIPS Transactions:PartC
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• v.10C no.5
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• pp.641-646
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• 2003

Packet processing speed of routers as well as transmission speed of physical links gives a great effect on IP packet transfer rate in Internet. The router forwards a packet after determining the next hop to the packet's destination. IP address lookup is a main design issue for high performance routers. In this paper, we propose a partitioned compressed-trie for speeding-up IP address lookup algorithms based on tie data structure by exploiting path compression. In the ,proposed scheme, IP prefixes are divided into several compressed-tries and lookup is performed on only one partitioned compressed-trie. Memory access time for IP address lookup is lessen due to compression technique and memory required for maintaining partition does not increased.

• Journal of information and communication convergence engineering
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• v.14 no.4
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• pp.240-245
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• 2016

Scalable video streaming over wireless networks has many challenges. The most significant challenge is related to packet loss. To overcome this problem, in this paper, we propose an unequal loss protection (ULP) method using a new forward error correction (FEC) mechanism for robust scalable video streaming over wireless networks. For an efficient FEC assignment considering video quality, we first introduce a simple and efficient performance metric, the layer-based recovery rate (LRR), for quantifying the unequal error propagation effects of the temporal and quality layers on the basis of packet losses. LRR is based on the unequal importance in both the temporal and the quality layers of a hierarchical scalable video coding structure. Then, the proposed ULP-LRR method assigns an appropriate number of FEC packets on the basis of the LRR to protect the video layers against packet lossy network environments. Compared with conventional ULP algorithms, the proposed ULP-LRR algorithm demonstrates a higher performance for various error-prone wireless channel statuses.