• Journal of KIISE:Computing Practices and Letters
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• v.6 no.5
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• pp.505-512
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• 2000

In this paper, an adaptive multimedia synchronization scheme for VOD (Video On Demand) services in internet environments is proposed. This scheme considers the characteristics of MPEG (Moving Picture Expert Group) system stream. Consequently, the intra-synchronization is handled at the pack layer, and the inter-synchronization is handled at the packet layer. The proposed scheme can cope adaptively with variation of packet loss, jitter and client's playback capacity. If there are variations of the packet loss or client's playback capacity, the server will change the transmission rate by selective picture skip. The client can then adjust and control the playback time according to the variation of the network jitter. Our experimental results show that the proposed scheme can quickly adapt to the network condition, and can guarantee a better quality of service than the other existing schemes.

• Journal of Korea Society of Industrial Information Systems
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• v.22 no.1
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• pp.33-40
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• 2017

In this Paper, Analyse the Steady State Behavior of TCP and TFRC with Packet Error when both TCP and TFRC Flows Co-exist in the Network. First, Model the Network with TCP and TFRC Connections as a Discrete Time System. Second, Calculate Average Round Trip Time of the Packet Between Source and Destination on Packet Loss Environment. Then Derive the Steady State Performance i.e. Throughput of TCP and TFRC, and Average Buffer Size of RED Router Based on the Analytic Network Model. The Throughput of TCP and TFRC Connection Decrease Rapidly with the Growth of Sending Window Size and Their Transmission Rate but Their Declines become Smoothly when the Number of Sending Window Arrives on Threshold Value. The Average Queue Length of RED Router Increases Slowly on Low Transmission Rate but Increases Rapidly on High Transmission Rate.

• Journal of the Korean Society for Railway
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• v.10 no.6
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• pp.758-765
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• 2007

In this paper, we propose an efficient seamless handoff scheme to minimize packet loss and unnecessary packets on the Internet using the peculiar mobility characteristics of public vehicles such as trains. MR (Mobile Router) in a train visits each AR (Access Router) in the fixed order. As the MR detects reachability to the NAR (Next Access Router) on the new link, the PAR (Previous Access Router) can directly deliver packets from MR's HA (Home Agent) to the NAR according to the HML (Handoff Mobile router List). Then. the NAR buffers them until the MR finishes L3 (Layer 3) handoff procedure with the NAR. Therefore, our scheme can support a seamless handoff without the packet loss and unnecessary packets on the Internet. The result of our performance evaluation has shown that the proposed scheme could provide excellent performance, compared with the NEMO basic support protocol and the Bi-casting protocol.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.1A
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• pp.30-37
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• 2006

In this paper, we propose a QoS-aware efficient bandwidth re-provisioning scheme in a next generation wireless packet transport network. At the transport network layer, it classifies the traffic of the radio network layer into a real time class and a non-real time class. Using an auto-regressive time-series model and a given packet loss probability, our scheme predicts the needed bandwidth of the non-real time class at every re-provisioning interval. Our scheme increases the system capacity by releasing the unutilized bandwidth of the non-real time traffic class for the real-time traffic class while insuring a controllable upper bound on the packet loss probability of a non-real time traffic class. Through empirical evaluations using the real Internet traffic traces, our scheme is validated that it can increase the bandwidth efficiency while guaranteeing the quality of service requirements of the non-real time traffic class.

• Journal of Korea Multimedia Society
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• v.9 no.3
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• pp.323-332
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• 2006

Fair packet scheduling algorithms supporting quality-of-services of real-time multimedia applications can be classified into the following two schemes in terms of the reference time used in calculating the timestamp of arriving packet; the Finish-Time (FT) and Start-Time (ST) schemes. The FT scheme, used in most schedulers, that has the property of an inversely rate-proportional latency is suitable to support various delay bounds because it can adjust the latency of a flow with raising the flow's reserved rate. However, the scheme may incur some bandwidth loss due to excess rate reservation. Meanwhile, although the ST scheme does not suffer from the bandwidth loss, it is hard to support multiple delay bounds because of its latency property relying on the number of flows. This paper is devoted to propose a ST scheme based scheduler to effectively support multiple delay bounds and analyze its performance comparing to the FT scheme based scheduler. The comparison results show that the proposed scheduler gives better utilization by up to 50%.

• Journal of Korea Multimedia Society
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• v.7 no.1
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• pp.54-63
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• 2004

This paper describes a hybrid QoS guarantee scheme for high quality audio streaming services on the Internet. The continuous playback of the audio data requires the isochronous transmission of the audio data packet through the Internet. In order to retain the QoS at the ultimate destination (client) as the same as servers provide, the transmission protocols should consider the error conditions such as packet loss, and out of order delivery. Generally, the protocols supporting the transmission of continuous media data do not try to recover the errors. The protocols are working somehow for the toll quality multimedia streaming services, but rot for the high quality streaming services, such as the DVD sound/music payback. The hybrid QoS guarantee scheme includes the three mechanisms to overcome the problem. The selective retransmission for the lost packet, the adaptive buffering at client-side, and the adaptive transmission rate at server-side are totally adopted to recover the packet loss with the minimal overhead, to prevent from the buffer starvation during the retransmission, and to maintain the isochronous transmission even after the retransmission. The experiments have shown good results for the high Quality audio streaming services on the Internet.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.1B
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• pp.1-8
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• 2006

Currently, there are many efforts underway to provide Internet service on integrated wireless and wired networks. Supporting IP mobility is one of the major issues to construct IP based wireless network. Mobile IP has been proposed to solve the IP Mobility problem. But, in processing frequent handoffs in cellular based wireless access network, Micro mobility protocols have been proposed to solve these problems. Micro mobility protocols proposed the Cellular IP, HAWII, and Hierarchical Mobile IP. Cellular IP attracts special attention for it's seamless mobility support in limited geographical areas. New BS must be known to occur begging of handoff in Cellular IP indirect handoff. Therefore during perceiving of hanoff, packet loss or packet duplication still can occur in Cellular IP indirect handoff, which results in the degradation of UDP and TCP performance. In this paper, we propose a enhanced indirect handoff scheme for Cellular IP. Proposed handoff scheme is using a crossover node to minimize the signalling procedure and using a buffering to minimize the packet loss or packet duplication.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.6B
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• pp.396-405
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• 2005

Snoop protocol is one of the efficient schemes to compensate TCP packet loss and enhance TCP throughput in wired-cum-wireless networks. However, Snoop protocol has a problem; it cannot perform local retransmission efficiently under the bursty-error prone wireless link. In this paper, we propose Enhanced Snoop(E-Snoop) protocol to solve this problem of Snoop protocol. With E-Snoop protocol, packet losses can be noticed by receiving new ACK packets as well as by receiving duplicate ACK packets or local retransmission timeout. Therefore, TCP throughput can be enhanced by fast recognition of bursty packet losses and fast local retransmissions. From the simulation results, E-Snoop protocol can improve TCP throughput more efficiently than Snoop protocol and can yield more TCP improvement especially in the channel with high packet loss rates.

• The KIPS Transactions:PartC
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• v.18C no.6
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• pp.423-432
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• 2011

Recently, OpenFlow has been paid attention to as a fundamental technology which provides a function of virtualization and programmability in network. In Korea, deployment of OpenFlow networks in campuses and the interconnection between them through tunneling in layer 3 has been performed. However, the performance of the interconnected networks is decreased due to delay in IP layer. In this paper, we design and deploy nation-wide, not local, OpenFlow networks in a pure layer 2 environment over KREONET. After that, we do end-to-end Round-trip Time measurements and TCP/UDP performance tests in OpenFlow and normal networks, and do comparison and analysis on the test results. The results show that the nation-wide OpenFlow networks provide equal performance to normal networks except for the initial packet loss for UDP streaming. In regards to the performance decrease due to early UDP packet loss, we can mitigate it by implementing exceptional procedures in a controller which deal with the same continuous "Packet_in" events.

• The Journal of the Acoustical Society of Korea
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• v.34 no.5
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• pp.403-409
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• 2015

In this paper, we propose Packet Loss Concealment (PLC) method using adaptive signal scale estimation for performance improvement of G.711 PLC. The conventional method controls a gain using 20 % attenuation factor when continuous loss occurs. However, this method lead to deterioration because that don't consider the change of signal. So, we propose gain control by adaptive signal scale estimation through before and after frame information using Least Mean Square (LMS) predictor. Performance evaluation of proposed algorithm is presented through Perceptual Evaluation of Speech Quality (PESQ) evaulation.