• Journal of KIISE:Information Networking
• /
• v.29 no.6
• /
• pp.696-705
• /
• 2002

This paper presents a new copy network for multicast ATM switches. Many studies have been carried out up to date since the proposition of Lee's copy network. However, the overflows and cell conflicts within the switch have still been raised a problem in argument. In order to reduce those problems, we proposed a 3-dimensional multicast switching architecture which has shared buffers in this paper. The proposed architecture can reduce the overflows and cell conflicts through multiple paths and output ports even in the high load environments. Also, we proposed a cell splitting algorithm which handles the cell in the case of large fan-out, and a copy network to increase throughput by expanding the Lee's Broadcast Banyan Network(BBN). Cell copy uses the Boolean interval splitting algorithm and the multicast pattern of the cells according to the self-routing characteristics of the network. In the proposed copy network, we improve the problems such as overflow, cell splitting of large fanout, cell conflicts, etc., which were still existed in the Lee's network. The results of performance evaluation by computer simulation show that the proposed scheme has better throughput, cell loss rate and cell delay than the conventional method.

• Journal of KIISE:Computing Practices and Letters
• /
• v.9 no.6
• /
• pp.623-634
• /
• 2003

In this paper, we propose a collaborative multimedia authoring system. Our authoring system represents a multimedia presentation in a 3D coordinate system. One axis represents the traditional timeline information (T-zone), and the other two axes represent spatial coordinates (XY-zone). Our system represents a visual media objects as a 3D parallelepipeds and audio media objects as cylinders. This interface allows for simultaneous authoring and manipulation of both the temporal and the spatial aspects of a presentation. Using our system, users can design multimedia presentations collaboratively in the unified spatio-temporal space while freely traversing the spatial domain and the temporal domain without changing the context of authoring. In addition, we suggest an efficient mechanism of concurrency control for shared objects generated by our collaborative writing system. The mechanism is mainly based on the user awareness, the multiple versions, and the access permission of shared objects. Our concurrency control mechanism is designed to keep data consistency by minimizing the collision due to the delay or the failure of network communication and to allow maximum responsiveness for users using optimistic concurrency control. Also, the mechanism maximize the responsiveness by refining the locking granularity and applying different concurrency control mechanisms to each.

• Journal of KIISE:Information Networking
• /
• v.32 no.3
• /
• pp.391-403
• /
• 2005

Several nv6 mobility support protocols for mobile Internet services are proposed in IETP : Mobile Ipv6, Hierarchical Mobile Ipv6, and Fast Handovers over Mobile Ipv6. Recently, IEEE 802.11 network has also been widely deployed in public areas for mobile Internet services. In the near future, IPv6 mobility support over IEEE 802.11 network is expected to be a key function to actualize the All If-based mobile various services. For appropriate application of these protocols, the IPv6 mobility support protocols should be analyzed according to their characteristics in terms of signaling, handover latency, lost packets, and required buffer sire as well as the impact of lower layer such as IEEE 802.11 network. In this paper, we analyze the performance of the protocols over IEEE 802.11 network. We define a packet-level traffic model, a network system model, and a mobility model. From these models, we construct a framework for the performance analysis. We also make cost functions to formalize each protocol's performance. Lastly, we analyze the effect of varying parameters used to show diverse numerical results, and compare with each other. From the analysis results, it is concluded that each Protocol has contrary or contrastive advantages with other Protocols, so there is no protocol that holds a dominant position.

• KIPS Transactions on Computer and Communication Systems
• /
• v.6 no.5
• /
• pp.219-230
• /
• 2017

General-Purpose Graphics Processing Units (GPGPUs) build massively parallel architecture and apply multithreading technology to explore parallelism. By using programming models like CUDA, and OpenCL, GPGPUs are becoming the best in exploiting plentiful thread-level parallelism caused by parallel applications. Unfortunately, modern GPGPU cannot efficiently utilize its available hardware resources for numerous general-purpose applications. One of the primary reasons is the inefficiency of existing warp/thread block schedulers in hiding long latency instructions, resulting in lost opportunity to improve the performance. This paper studies the effects of hardware thread scheduling policy on GPGPU performance. We propose a novel warp scheduling policy that can alleviate the drawbacks of the traditional round-robin policy. The proposed warp scheduler first classifies the warps of a thread block into two groups, warps with long latency and warps with short latency and then schedules the warps with long latency before the warps with short latency. Furthermore, to support the proposed warp scheduler, we also propose a supplemental technique that can dynamically reduce the number of streaming multiprocessors to which will be assigned thread blocks when encountering a high contention degree at the memory and interconnection network. Based on our experiments on a 15-streaming multiprocessor GPGPU platform, the proposed warp scheduling policy provides an average IPC improvement of 7.5% over the baseline round-robin warp scheduling policy. This paper also shows that the GPGPU performance can be improved by approximately 8.9% on average when the two proposed techniques are combined.

• Journal of the Institute of Convergence Signal Processing
• /
• v.9 no.1
• /
• pp.24-30
• /
• 2008

In this paper we analyze the performance of an adaptive sector cell system, which is adopted to maintain the traffic balance between sectors and to utilize the cell resources effectively, using the data collected from real channel environments. In the data measurements, we transmitted the QPSK modulated signal with carrier frequency of 1.95GHz and received the signals using the 8x4 array antenna equipped on the top of buildings in the urban area. We analyzed the angular distribution and the delay spread of a user signal and analyzed angular distribution of mobile users in a cell using the collected data. Also, we propose the vector channel modeling using the estimated pdf(probability distribution function) of the analyzing results. Through the proposed channel modeling the improvement of the call blocking rate was analyzed when using the adaptive sector cell system, and computer simulations show that the call blocking rate of the adaptive sector cell system was much lower than that of the fixed sector cell system. Additionally, it shows that the call blocking rate increases severely in the fixed sector cell system while the difference of the call blocking rate was smaller in the adaptive sector cell system, as the user density of the spatial distribution increases.

• KIPS Transactions on Computer and Communication Systems
• /
• v.9 no.6
• /
• pp.121-130
• /
• 2020

In order to provide customized services according to a specific user or environment, various service scenarios should be developed based on context-aware workflow model. As the context-aware workflow model is constructed using limited context information and service information in the service domain, overlapping elements can occur in many service scenarios. The repetitive work process that results from these overlapping elements delays the development process of the service scenario. Therefore, the elements of the context-aware workflow model must be reused to solve the unnecessary work processes of service scenario development. In this paper, we propose a reuse method XML Object in context-aware workflow model to improve the process of service scenario development. The proposed method documents and manages the independent XML Object of the context-aware workflow model and reuses it by invoking the unit document in the service scenario development process. It can also be applied to new service scenarios by changing the attribute values of reusable elements. Experiments show example that the development process of the service scenario is simplified by reusing the elements of the context-aware workflow model.

• 전자공학회논문지 IE
• /
• v.44 no.1
• /
• pp.27-32
• /
• 2007

This study was to suggest the predictive model of propagation, considering the effect by the multipath waves produced by the sending and receiving vehicles' left/right reflectors and the adjacent vehicles when the communication between the vehicles on the one-way two-lanes road in the urban city with a lot of traffic jams. Then, the radius of curved road was 600[m], the length of curved roads $52.4\sim471.2[m]$, and the bridge's pier of road was $5o\sim45o$. Also, it was simulated by changing the receiving vehicle located on the curved road's gap from minimum 3.3[m] to maximum 29.5[m], corresponding to the change of distance of the bridge's pier of road and curved road. As a result of this research above, in case of $5o\sim15o$ bridge's pier of road, it was within l[dB] regardless of the receiving vehicle's position on the curved road in case of propagation path loss. In case of $15o\sim45o$, it was approximately $1\sim8[dB]$ as the bridge's pier of road is changed. And, in case of propagation path, it found out that it was changed to $0.4\sim120[m]$ according to the change of bridge's pier of road. Then, the delay time of propagation was 400[nsec] as it produced 120[m] in the difference of propagation path.

• 한국벤처창업학회:학술대회논문집
• /
• 2010.08a
• /
• pp.71-89
• /
• 2010

It is strongly believed that multicast will become one of the most promising services on internet for the next generation. Multicast service can be deployed either on network-layer or application-layer. IP multicast (network-layer multicast) is implemented by network nodes (i.e., routers) and avoids multiple copies of the same datagram on the same link. Despite the conceptual simplicity of IP multicast and its obvious benefits, it has not been widely deployed since there remain many unresolved issues. As an alternative to IP multicast, overlay multicast (application-layer multicast) implements the multicast functionality at end hosts rather than routers. This may require more overall bandwidth than IP multicast because duplicate packets travel the same physical links multiple times, but it provides an inexpensive, deployable method of providing point-to-multipoint group communication. In this paper we develop an efficient method applied greedy algorithm for solving two models of overlay multicast tree building problem that is aimed to construct MDST (Minimum Diameter Spanning Tree : minimum cost path from a source node to all its receivers) and MST (Minimum Spanning Tree : minimum total cost spanning all the members). We also simulate and analyze MDST and MST.

• Journal of the Institute of Electronics Engineers of Korea TC
• /
• v.43 no.7 s.349
• /
• pp.143-149
• /
• 2006

Turbo code, a kind of error correction coding technique, has been used in the field of digital mobile communication system. As the number of iterations increases, it can achieves remarkable BER performance over AWGN channel environment. However, if the number of iterations is increased in the several channel environments, any further iteration results in very little improvement, and requires much delay and computation in proportion to the number of iterations. To solve this problems, it is necessary to device an efficient criterion to stop the iteration process and prevent unnecessary delay and computation. In this paper, it proposes an efficient and simple criterion for stopping the iteration process in turbo decoding. By using sign changed ratio of extrinsic information values in turbo decoder, the proposed algorithm can largely reduce the average number of iterations without BER performance degradation. As a result of simulations, the average number of iterations is reduced by about $12.48%{\sim}22.22%$ compared to CE algorithm and about $20.43%{\sim}54.02%$ compared to SDR algorithm.

• Journal of the Institute of Electronics Engineers of Korea SP
• /
• v.38 no.3
• /
• pp.304-315
• /
• 2001

In this paper, we peformed the real-time implementation of AMR(Adaptive Multi-Rate) speech coding algorithm which is adopted for IMT-2000 service using TMS320C6201, i.e., a Texas Instrument´s fixed-point DSP. With the ANSI C source code released from ETSI, optimization is performed to make it run in real-time with memory as small as possible using the C compiler and assembly language. Implemented AMR speech codec has the size of 32.06 kWords program memory, 9.75 kWords data RAM memory, and 19.89 kWords data ROM memory. And, The time required for processing one frame of 20 ms length speech data is about 4.38 ms, and it is short enough for real-time operation. It is verified that the decoded result of the implemented speech codec on the DSP is identical with the PC simulation result using ANSI C code for test sequences. Also, actual sound input/output test using microphone and speaker demonstrates its proper real-time operation without distortions or delays.