• Journal of the Korea Institute of Information and Communication Engineering
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• v.5 no.5
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• pp.923-930
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• 2001

In this paper, in order to improve the performance of TCP short traffic services in wireless Internet environments, the Split-ACKs(SPACK) scheme is proposed. In wireless networks, unlike wired networks, packet losses will occur more often due to high bit error rates. Therefore, each packet loss over wireless links results in congestion control procedure of TCP being invoked at the source. This causes severe end-to-end performance degradation of TCP. In this paper, to alleviate the TCP performance, the SPACK method, split acknowledgement packets in the base station, is proposed. Using computer simulation, the performance of TCP using SPACK is analyzed and shows better performance than traditional TCP protocol.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.36 no.9A
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• pp.749-757
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• 2011

In this paper, we propose a robust And-Or tree-based Luby transform encoded symbol packetization algorithm to minimize the quality degradation of video streaming service caused by packet losses over wireless network. To achieve this goal, the relationship among Luby transform encoded symbols are analyzed by using And-Or tree, and the proposed packetization algorithm is designed to minimize packet loss effects by reducing the dependency among packets conveying Luby transform encoded symbols. Finally, experimental results are provided to show the performance of the proposed algorithm.

• Proceedings of the IEEK Conference
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• 2002.07c
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• pp.2075-2078
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• 2002

Internet became the most popular network in spite of its weakness in realtime multimedia service. Many experts believe that the Internet has the potential to become the main multimedia distribution network of the near future. Currently, it does not provide any (BoS guarantees and, even when it does, guaranteed quality delivery of video may turn out to be too expensive. Unavoidable packet losses and delay jitter caused by congestion in a best effort delivery environment require use of intelligent transport techniques for effective video delivery. According to market needs of better quality of service (QoS) fur realtime multimedia services over Internet, they have been standardizing RSVP, IntServ, and DiffServ This paper combines the benefits of QoS mechanisms such as RSVP/IntServ with scalable video encoding. We propose that more important bit stream is given more priority such that limited network resources are guaranteed far the stream. Various prioritizing approaches are proposed and compared to normal approach by using Network Simulator. The calculated QoS parameters such as packet loss rate are used to calculate degree of degradation in video quality. In this Paper, proposed methods can be implemented adaptively to Von protocol, such as H.323, SIP.

• Proceedings of the IEEK Conference
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• 2007.07a
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• pp.41-42
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• 2007

DMAC is an energy efficiency and low latency protocol designed for data gathering tree structures. However, it causes the funneling effect which is many-to-one traffic patterns in tree structures, consequently, results in packet collisions, losses, and energy consumptions in low depth nodes. In this paper, we present an energy efficient MAC protocol with fairness-based scheduling to avoid the funneling effect of DMAC protocol. By using traffic information from children nodes, our protocol dynamically adjusts duty cycles of last-depth nodes to mitigate overloaded packets in the vicinity of the sink node. Therefore, we expect our protocol to save more energy and achieve better packet delivery ratio, compared to DMAC protocol.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.14 no.5
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• pp.1139-1145
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• 2010

Wireless mesh network is similar to ad-hoc network, so when transferred to the data packet in the wireless environment, interfered factor arise. When TCP(Transport Control Protocol) was created, however as it was design based on wired link, wireless link made more transmission error than wired link. It is existent problem that TCP unfairness and congestion collapse over wireless mesh network. But packet losses due to transmission errors are more frequent. The cause of transmission error in wireless ad-hoc network may be inexactly regarded as indications of network congestion. And then, Congestion Control Algorithm was running by this situation causes the TCP performance degradation. In this paper, proposed TCP can adaptively regulate the congestion window through moving node in the Wireless Mesh Network. And it enhanced the performance.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.10 no.10
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• pp.4977-4996
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• 2016

The existence of excessively large and too filled network buffers, known as bufferbloat, has recently gained attention as a major performance problem for delay-sensitive applications. Researchers have made three types of suggestions to solve the bufferbloat problem. One is End to End (E2E) congestion control, second is deployment of Active Queue Management (AQM) techniques and third is the combination of above two. However, these solutions either seem impractical or could not obtain good bandwidth utilization. In this paper, we propose a Transmission Control Protocol（TCP）delayed window update mechanism which uses a congestion detection approach to predict the congestion level of networks. When detecting the network congestion is coming, a delayed window update control strategy is adopted to maintain good protocol performance. If the network is non-congested, the mechanism stops work and congestion window is updated based on the original protocol. The simulation experiments are conducted on both high bandwidth and long delay scenario and low bandwidth and short delay scenario. Experiment results show that TCP delayed window update mechanism can effectively improve the performance of the original protocol, decreasing packet losses and queuing delay while guaranteeing transmission efficiency of the whole network. In addition, it can perform good fairness and TCP friendliness.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.3B
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• pp.130-137
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• 2005

Wireless communication environment is changing rapidly as we use new wireless communication technology such as WiBro to access high speed Internet. As a result, reliable data transmission using TCP is also expected to increase. Since TCP assumes that it is used in wired network, TCP suffers significant performance degradation over wireless network where packet losses are related to non-congestion loss. Especially RTO imposes a great performance degradation of TCP. In this paper, we analyze the loss recovery probabilities based on previous researches, and use simulation results of our algorithm to show that it prevents performance degradation by quickly detecting and recovery losses without RTO during fast recovery.

• Journal of Information Processing Systems
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• v.7 no.4
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• pp.627-634
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• 2011

This paper addresses the Proxy Mobile IPv6 (PMIP) handover using bicasting in mobile/wireless networks. The bicasting scheme can be used to support the PMIP handover, which tends to waste the network resources of wireless links and incurs data losses during handover. We propose an enhanced scheme of PMIP handover, called the partial bicasting with buffering for PMIP (PBB-PMIP). In the PBB-PMIP handover, the bicasting is performed in the "partial" region between the Local Mobility Anchor (LMA) and the Mobile Access Gateway (MAG), when a mobile node is in the handover area. The data packets are buffered at the new MAG during handover to reduce data losses and are then forwarded to mobile nodes after handover. By ns-2 simulations, the proposed PBB-PMIP scheme is compared with the existing schemes of PMIP and PMIP with bicasting. The proposed scheme can benefit from the reduction of handover delay and packet loss, and the effective use of the network resources of wireless links, as compared to the existing handover schemes.

• KIPS Transactions on Computer and Communication Systems
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• v.2 no.3
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• pp.117-124
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• 2013

In this paper we propose a TFRC(TCP Friendly Rate Control) which guarantees a minimum rate in order to improve the efficiency of the previous TFRC which cannot distinguish congestion losses and wireless losses and decreases throughput both in wired and wireless networks. This TFRC technique is able to guarantee a minimum rate for video by restricting a loss event rate with packet loss probability about existing TFRC and constraining a rate reduction from the feedback timeout. When we experimented both the existing TFRC and the new one with TCP in the same network, we found that the latter is better than the former. Consequently, it shows that the proposed TFRC can improve video streaming quality using a guaranteed minimum transmission rate.

• Journal of Broadcast Engineering
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• v.14 no.5
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• pp.573-588
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• 2009

In this paper, we propose a novel video quality degradation monitoring scheme titled VR-VQMS(Visual Rhythm based Video Quality Monitoring Scheme) over an IPTV service prone to packet losses during network transmission. Proposed scheme quantifies the amount of quality degradation due to packet losses, and can be classified into a RR(reduced-reference) based quality measurement scheme exploiting visual rhythm data of H.264-encoded video frames at a media server and reconstructed ones at an Set-top Box as feature information. Two scenarios, On-line and Off-line VR-VQMS, are proposed as the practical solutions. We define the NPSNR(Networked Peak-to-peak Signal-to-Noise Ratio) modified by the well-known PSNR as a new objective quality metric, and several additional objective and subjective metrics based on it to obtain the statistics on timing, duration, occurrence, and amount of quality degradation. Simulation results show that the proposed method closely approximates the results from 2D video frames and gives good estimation of subjective quality(i.e.,MOS(mean opinion score)) performed by 10 test observers. We expect that the proposed scheme can play a role as a practical solution to monitor the video quality experienced by individual customers in a commercial IPTV service, and be implemented as a small and light agent program running on a resource-limited set-top box.