• 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.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.8 no.3
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• pp.637-644
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• 2004

This paper proposed and implemented the adaptive transmission middleware for video streaming, which is able to support the adaptive transmission of video data to the fluctuating changes of network environment in the packet-based network and the properties of transmitted video data. The adaptive transmission middleware is made up SR-RTP-based transfer module and TFRC(TCP Friendly Rate Control)-based transfer-rate control module. The SR-RTP-based transfer module supports RTP-based real-time transfer of video data and packet retransmission scheme retransmitting the high-priority packets selectively in the damaged video data to reduce the error induced by the packet loss. Sharing the transmission bandwidth of network with the TCP-based data transfer, the TFRC-based transfer-rate control module controls the transfer rate of video data according to the most allowable transmission bandwidth in the network, so that the transfer rate is controlled adaptively to the fluctuating changes of transmission bandwidth. This paper, for the experiment, applied the adaptive transmission middleware to video streaming in the external Internet environment, and analyzed the effective frame transfer rate and the degree of the streaming jitter to evaluate the performance of packet-loss recovery and adaptive transfer rate control. In the external Internet environment where the packet-loss rate is high a bit, the relatively high streaming performance was showed compared with the case that didn't apply the adaptive transmission middleware.

• Journal of KIISE:Information Networking
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• v.30 no.2
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• pp.233-243
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• 2003

With the original Transmission Control Protocol(TCP) design, which is particularly targeted at the wired networks, a packet loss is assumed to be caused by the network congestion. In the wireless environment where the chances to lose packets due to transmission bit errors are not negligible, though, this assumption may result in unnecessary TCP performance degradation. In this paper, we propose three schemes that improve the ability to conceal the packet losses in the wireless network while limiting the degree of violating TCP end-to-end semantics to a temporary incidents. If there happens a packet loss at the wireless link and there is a chance that the loss is noticed by the sending TCP, the proposed schemes send an indirect acknowledgement. Each of the proposed schemes uses different criteria to decide whether there is a chance that the packet loss occurred in the wireless part is noticed by the sender. In order to limit the buffer overhead in the base, the indirect acknowledgements are issued only when the length of buffer is less than a certain threshold. We use simulation to compare the overhead and the performance of the proposed schemes, and to show that the proposed schemes improve the TCP performance compared to Snoop with a limited amount of buffer at the base station.

• The KIPS Transactions:PartC
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• v.15C no.4
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• pp.281-288
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• 2008

Recently, the advance of multimedia hardware has fostered the development of wireless multimedia sensor network which is able to ubiquitously obtain multimedia content such as image or audio from the environment. The multimedia data which has several characteristics such as large size and correlation between the data requires reliability in transmission. However, the existing solution which take the focus on the efficiency of network mainly, is not appropriate to transmit the multimedia data. In the paper, we proposes a reliable asynchronous image transfer protocol, RAIT. RAIT applies double sliding window method in node-to-node image tansfer to prevent the packet loss caused by network congestion. The double sliding window consists of one sliding window for the receiving queue, which is used for prevention of packet loss caused by communication failure between nodes and the other sliding window for the sending queue which prevents the packet loss caused by network congestion. the routing node prevents the packet loss and guarantees the fairness between the nodes by scheduling the packets based on the image non-preemptively. The RAIT implements the double sliding window method by cross layer design between RAIT layer, routing layer, and queue layer. The experiment shows that RAIT guarantees the reliability of image transmission compared with the existing protocol.

• Journal of the Korean Institute of Telematics and Electronics T
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• v.35T no.2
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• pp.35-38
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• 1998

We need the packet transmission of voice, data and video according to the development of ATM technology and packet network. In ATM, packet is parted to the constant length data, we called cell, the method is needed to minimize data loss. Because over ATM, we discard error cell without correction in transmission. So, about video, there is layered coding method. We simulated the coding using wavelet transform, which is structurally layed coded, so need not the additional coder.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.6 no.6
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• pp.513-520
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• 2005

High-speed Portable Internet system provides 1-3 Mbps data transmission speed to terminals moving up to 60 km/sec. Since High-speed Portable Internet system supports services requiring different QoS, it needs an efficient scheduling method based on those different QoSs. This paper shows the performance comparisons of several different packet scheduling schemes for minimizing the mean delay over the downlink of High-speed Portable Internet system to support the packet data service. Simulation results show that proposed scheme superior to other schemes at side throughput and data loss rate.