• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2011.05a
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• pp.336-339
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• 2011

In the RTLS(Real Time Location Based System), in case of existing a number of moving target, extremely complecated data flow is can be occurred. In the network where single gateway exists, various data which was collected from sensor node is transmitted along the simple route as time goes by. In case of multi-gateway configuration, the collected data is transmitted through diverse routes rather than simple route. This kind of data causes jams on nodes and this brings down the performance of the network. Different from existing studies, in this thesis, MAC (Media Access Control) protocol which minimizes data collision between nodes and guarantees QoS(Quality of Service) is suggested, in order to communicate efficiently in multi-gateway underwater sensor network environment. In the suggested protocol, source node which wants to transmit data makes a channel reservation to a number of destination node using a RTS packet. Source node reserves a channel without collision, by scheduling CTS response time using expected delay information from neighbor nodes. Once the reservation is made, source node transmit data packet without collision. This protocol analyzes/estimates the performance compared to a method provided from existing studies via simulation. As a results of the analysis, it was comfirmed that the suggested method has better performance, such as efficiency and delay.

• The Journal of The Korea Institute of Intelligent Transport Systems
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• v.11 no.3
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• pp.48-54
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• 2012

The quality of real-time voice communication over Internet Protocol networks based on telematics is affected by network impairments such as delays, jitters, and packet loss. To resolve this issue, this paper proposes a receiver-based enhancing method of VoIP speech quality. The proposed method enables users to deliver high-quality voice using playout control and signal reconstruction, which consists of concealment of lost packets, adaptive playout-buffer scheduling using active jitter estimation, and smooth interpolation between two signals in a transition region. The proposed algorithm achieves higher Perceptual Evaluation of Speech Quality (PESQ) values and low buffering delay than the reference algorithm.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2006.05a
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• pp.538-541
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• 2006

Recently researches of P2P media streaming have been actively published, but the research what kind of P2P overlay networks are better to delivering media data has not been attended. This paper designs a P2P streaming media system model. In this system, we constructs a Half-Structure P2P overlay protocol based on node's capability as the infrastructure of streaming application, which utilizes the heterogeneity of the nodes to maintain topology, presents a active contents diffusing algorithm and Two-stage search algorithm, make it possible for nodes in P2P system to collect information according their capacity and reduces the number of forwarding packet compared with flooding. Also, we give an optimal scheme to scheduling media data.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.45 no.9
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• pp.39-48
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• 2008

Recently, many multimedia services have emerged in Internet such as real-time and non- real time services. However, in this Internet environment, we have some limitations to satisfy each service feature. To guarantee the service features in Measurement-Based Admission Control(MBAC) based system on the flow-aware network, there is the method applying Dynamic Priority Scheduling(DPS) algorithm that gives a higher priority to an earlier incoming flow in all of the link bandwidth. This paper classifies all flows under several groups according to flow characteristics on per-flow MBAC algorithm based system. In each flow group, DPS algorithm is applied. This paper proposes two methods that are a DPS based bandwidth borrowing method and a bandwidth dynamic allocation method. The former is that if low priority class has available bandwidths, the flow of high priority class borrows the bandwidth of low priority class when high priority flow has insufficient bandwidth to connect a flow call. The later is that the each group has a minimum bandwidth and is allocated the bandwidth dynamically according to the excess rate for available bandwidth. We compare and analyze the characteristics of the two proposed methods through the simulation experiments. As the results of the experiment, the proposed methods are more effective than existing DPS based method on the packet loss and delay characteristics. Consequently the proposed two methods are very useful in various multimedia network environments.

• Journal of the Institute of Electronics Engineers of Korea SD
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• v.42 no.2 s.332
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• pp.89-96
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• 2005

In this paper, we propose a customer premise platform for processing multimedia data service on the ATM network. The proposed platform has a specific AAL2 processor that includes AAL2 protocol and scheduler algorithm so as to off-load large potion of burden from host processor and make it easy to process multimedia data from the ATM network in real time compared with conventional platform in which AAL/ATM tasks are processed by software. The ATS scheduler that is implemented based on 2-level time slot ring provides a simple and efficient method for scheduling data of VBR-rt, UBR and CBR traffics. TMS320C5402 DSP is used to process voice-related tasks such as voice compression and voice packet manupulation and AAL2 processor is implemented on $0.35\;{\mu}m$ process line. We implemented the customer premise equipment for VoDSL service and tested the proposed platform on a test bed network. The experimental results show that the proposed equipment has the call success rate of $97\%$ at least and provides voice service of toll-qualify.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.48 no.6
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• pp.140-148
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• 2011

Multimedia streaming service is susceptible to loss and delay of data as it requires high bandwidth and real time processing. Therefore QoS cannot be guaranteed due to data loss caused by heavy network traffic. To deal with these problems, a study on algorithm that can provide high quality multimedia service by considering both network congestion and characteristics of multimedia data is required. In this paper, we propose APQM algorithm which probabilistically removes packet by the congestion level of the queue in wireless station. The comparison with other existing scheduling algorithms shows tht congestion in the network is reduced and multimedia service quality of the proposed algorithm is improved.

• Journal of the Korea Society for Simulation
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• v.20 no.1
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• pp.107-116
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• 2011

Recently, femtocell has been concerned as one of effective solutions to relieve shadow region and provide high quality services to users in indoor environments. Even though femtocell offers various benefits to cellular operators and users, many technical issues, such as interference coordination, network synchronization, self-configuration, self-optimization, and so on, should be solved to deploy the femtocell in current network. In this paper, we develop a simulator for evaluating performance of long term evolution (LTE) femtocell systems under various interference scenarios. The simulator consists of a main-module and five sub-modules. The main-module connects and manages five sub-modules which have the functionality managing user mobility, packet scheduling, call admission control, traffic generation, and modulation and coding scheme (MCS). To provide user convenience, the simulator adopts graphical user interface (GUI) which can observes simulation results in real time. We expect that this simulator can contribute to developing effective femtocell systems by supporting a tool for analyzing the effect of interference between macrocell and femtocell.

• Journal of Broadcast Engineering
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• v.15 no.2
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• pp.205-216
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• 2010

In this paper, a method of MPEG-2 Transport Stream (TS) multiplexing for Ultra HDTV (UHDTV) and its design and implementation as a SW tool is described. In practice, UHD video may be divided into several HD videos and each video is encoded in parallel. Therefore, it is necessary to synchronize and multiplex multiple bitstreams encoding each HD video for transmitting and storing UHD video. In this paper, it is assumed that 4 HD videos partitioning a UHD spatially are encoded as H.264/AVC and two 5.0 channel audios are encoded by AC-3. Therefore, 4 H.264/AVC elementary streams (ESs) and 2 AC-3 ESs is mainly considered in the TS multiplexing of UHD. For the carriage of H.264/AVC and AC-3 over MPEG-2 TS, PES packetization and TS multiplexing are designed and implemented based on the extended specification of the MPEG-2 Systems and ATSC (Digital audio compressed standard), respectively. The implemented UHD TS multiplexing tool emulates real time HW operation in the time unit corresponding to the duration of one TS packet transmission in a given TS rate. In particular, in order to satisfy the timing model, the buffers defined in the TS System Target Decoder (T-STD) are monitored and their statuses are considered in the scheduling of TS multiplexing. For UHD multiplexing, two kinds of multiplexing structures, which are UHD re-multiplexing and UHD program multiplexing, are implemented and their strength and weakness are investigated. The developed UHD TS multiplexing tool is tested and verified in terms of the syntax and semantics conformance and functionalities by using a commercial analyzer and real-time presentation tools.