• Journal of the Institute of Electronics Engineers of Korea TC
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• v.42 no.12
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• pp.25-30
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• 2005

We propose and design a fully conversed Ethernet and TDM transport system to guarantee qualify of service for Ethernet packet. Developed convergence system can support not only L2 VPN service and premium multimedia service based on MPLS protocol but also TDM leased line service, simultaneously. Developed convergence system has the advantage of providing high reliability that realizes high-quality and real time communications due to assign QoS profile and guarantee bandwidth per channel consist of VCGs without affect adjacent channels. Evaluation for proposed system was successfully performed within the ling network.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.21 no.4
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• pp.911-925
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• 1996

Emerging service classes need an error model which efficiently satisfies the requirements for each media and application. However limited flexibility in the error control has been provided. Therefore we propose a new error control model which applies error control scheme per media flexibly with respect to ETL(Error Tolerance Leverl) and delay. In order to select the error control scheme in this model, the attributes of media(i.e., ETL, isochronism), the attributes of application(i.e., delay) and environment parameters(i.e., packet loss ration, network types, connection modes of the application) are considered as effect factors.

• The Transactions of the Korea Information Processing Society
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• v.5 no.11
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• pp.2954-2972
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• 1998

It is expected that the transport of compressed video will become a significant part of total network traffic because of the widespread introduction of multimedial services such as VOD(video on demand). Accordingly, VBR(variable bit-rate) encoded video will be widely used, due to its advantages in statistical multiplexing gain and consistent vido quality. Since the transport of video traffic requires larger bandwidth than that of voice and data, the characterization of video source and traffic modeling is very important for the design of proper resource allocation scheme in ATM networks. Suitable statistical source models are also required to analyze performance metrics such as packet loss, delay and jitter. In this paper, we analyzed and described on the characterization and traffic modeling of MPEG video sources. The models are broadly classified into two categories; i.e., statistical models and deterministic models. In statistical models, the models are categorized into five groups: AR(autoregressive), Markov, composite Marko and AR, TES, and selfsimilar models. In deterministic models, the models are categorized into $({\sigma},\;{\rho}$, parameterized model, D-BIND, and Empirical Envelopes models. Each model was analyzed for its characteristics along with corresponding advantages and shortcomings, and we made comparisons on the complexity of each model.

• Journal of the Korea Convergence Society
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• v.7 no.4
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• pp.25-32
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• 2016

When using reliable multicast protocol over air links, the multicast packets lost in the air link cause the initiation of retransmission request packets and the implosion of retransmission packets, which deteriorate multicast session performance. This paper proposes on the efficient reliable multicast mechanism in wireless networks utilizing the Agents. In this paper we show the design of a retransmission agent which improves the performance of reliable multicast sessions in wireless network. The main idea is to cache reliable multicast packets at the base station and perform local retransmissions across the wireless link. MATLAB has been used to simulate and to get performance results for signaling overhead and processing delay through the comparison of the proposed agent model to the Multicast File Transfer Protocol. It has been proven from the simulation results that the proxy module make pass trials shorter in Multicast File Transfer Protocol.

• The Journal of The Korea Institute of Intelligent Transport Systems
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• v.10 no.4
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• pp.71-77
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• 2011

In this paper, we propose a conversational quality measurement (CQM) system for providing the objective QoS of high quality mobile VoIP voice telecommunication. For measuring the conversational quality, the VoIP telecommunication system is implemented in two smart phones connected with VoIP. The VoIP telecommunication system consists of echo cancellation, noise reduction, speech encoding/decoding, packet generation with RTP (Real-Time Protocol), jitter buffer control and POS (Play-out Schedule) with LC (loss Concealment). The CQM system is connected to a microphone and a speaker of each smart phone. The voice signal of each speaker is recorded and used to measure CE (Conversational Efficiency), CS (Conversational Symmetry), PESQ (Perceptual Evaluation of Speech Quality) and CE-CS-PESQ correlation. We prove the CQM system by measuring CE, CS and PESQ under various SNR, delay and loss due to IP network environment.

• The Journal of The Korea Institute of Intelligent Transport Systems
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• v.10 no.4
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• pp.78-85
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• 2011

In vehicle ad-hoc network (VANET) environments, traffic related information such as accident information, emergency information and real time traffic condition have to be delivered to on-board-unit (OBU) or/and road-side-equipment (RSE) for preventing traffic accidents in advance. In this paper, we introduce a Multi-Channel MAC (MCM) since the existing single channel operation may cause packet transmission delay and unexpected communication failure. To offer a seamless safety message transmission during the various services, it is necessary to manage the MAC scheduler in wireless access in vehicular environments (WAVE) systems. The MCM consists of MAC softwares and MAC hardwares where the former and the later ones are implemented with real time operation system based C language and FPGA module with VHDL language, respectively. The performance and QoS are verified by practical measurements and compared with the scheme using single channel operation.

• Journal of Broadcast Engineering
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• v.16 no.3
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• pp.520-529
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• 2011

In this paper, we propose the extension of the MPEG-2 Transport Stream (TS) header for efficient adaptation of multi-layer coded video such as scalable video coding (SVC) and multiview video coding (MVC) in the HTTP streaming. First of all, the limit of the existing TS in terms of flexible adaptation of multi-layer video is investigated, and the signaling by extending TS header is proposed to provide efficient adaptation in a TS level. The proposed extension utilizes the private data field in the adaptation field of TS header to signal scalability and/or view information, which enable us to support diverse adaptation that suits underlying constraints of client capabilities, network conditions and user preferences. In short, the extension enables adaptation of scalable video with full scalability as well as view selection of multiview video in a TS level while keeping backward compatibility with the existing TS syntax/semantics. The performance of the proposed extension is compared with the existing adaptation using PID (packet ID) in terms of efficiency and complexity of adaptation. Furthermore, the increase of TS overhead caused by proposed extension is analyzed and an extension scheme to minimized the overhead is proposed.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.9B
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• pp.610-618
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• 2005

As developed Internet and Computer Capability, Many Users take the many information through the network. So requirement of User that use to network was rapidly increased and become various. But it spend much time to accept user requirement on current network, so studied such as Active network for solved it. This Active node on Active network have the capability that stored and processed execution code aside from capability of forwarding packet on current network. So required execution code for executed packet arrived in active node, if execution code should not be in active node, have to take by request previous Action node and Code Server to it. But if this execution code take from previous active node and Code Server, bring to time delay by transport execution code and increased traffic of network and execution time. So, As used execution code stored in cache on active node, it need to increase execution time and decreased number of request. So, our paper suggest ANC caching technique that able to decrease number of execution code request and time of execution code by efficiently store execution code to active node. ANC caching technique may decrease the network traffic and execution time of code, to decrease request of execution code from previous active node.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.12 no.2
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• pp.860-872
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• 2018

In this Paper, we propose 8K UHD (Ultra High Definition) encapsulation method for efficient transmission and reception based on MMT (MPEG Media Transport). Broadcasting services for 8K UHD allow users to feel the maximized reality. However, present technology is difficult to provide 8K UHD in broadcasting networks, because the 8K UHD bitrate is too high to be transmitted in the current broadcasting networks. Research for transmitting 8K UHD is underway. In some researches, a receiver is implemented with four 4K UHD display instead of a 8K UHD display. In order to transmit 8K UHD within the limited transmission bitrate of broadcasting network, 8K UHD contents encoded by SHVC (Scalable High Efficiency Video Coding) and then transmitted over heterogeneous network. For using the broadcasting and communication networks, MMT standard is used. MMT is IP based transmission protocol as the next generation transmission protocol. According to the MMT standard, video stream encapsulated and transmitted in MMTP (MMT Protocol) packet. IP-based broadcasting and communication networks can be used to transmit simultaneously, and the receiver can synchronize and play it. We propose an encapsulation method that can efficiently transmit and receive 8K UHD. The proposed method increases a payload rate and decreases an initial delay at the receiver. We show that the efficiency of the proposed method is verified by experimental tests.

• The Journal of The Korea Institute of Intelligent Transport Systems
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• v.8 no.4
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• pp.53-64
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• 2009

Inter-vehicular communication that propagates information without infrastructures has drawn a lot of interest. However, it is difficult to apply conventional ad-hoc routing protocols directly in inter-vehicular communication due to frequent changes in the network topology caused by high mobility of the vehicles. MMFP(Multi-hop MAC Forwarding) is a unicast forwarding protocol that transport packets based on the reachability information instead of path selection or position information. However, delivering public safety messages informing road conditions such as collision, obstacles and fog through inter-vehicular communication requires broadcast rather than unicast since these messages contain information valuable to most drivers within a close proximity. Flooding is one of the simplest methods for multi-hop broadcast, but it suffers from reduced packet delivery-ratio and high transmission delay due to an excessive number of duplicated packets. This paper presents two multi-hop broadcast protocols for inter-vehicular communication that extend the MMFP. UMHB(Unreliable Multi-Hop Broadcast) mitigates the duplicated packets of MMFP by limiting the number of nodes to rebroadcast packets. UMHB, however, still suffers from low delivery ratio. RMHB(Reliable Multi-Hop Broadcast) uses acknowledgement and retransmission in order to improve the reliability of UMHB at the cost of increase in transmission delay, which we show through simulation is within an acceptable range for collision avoidance application.