• Journal of Korea Multimedia Society
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• v.17 no.9
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• pp.1084-1091
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• 2014

When demand for transmitting multimedia data increases, network congestion is more likely to occur and users will suffer high loss rate as well as high delay. In order to enhance quality-of-service (QoS) of video multicasting, we need to raise transmission reliability and reduce end-to-end delay. This paper proposes a routing mechanism for a OpenFlow-based small-scale network in order to multicast video reliably with low delay. In our method, multipath routing will be applied to Multiple Description (MD) Coded video to exploit its multi-description property. Through performance evaluation, our method shows improvement on loss rate, delay and video distortion.

• Journal of Digital Convergence
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• v.10 no.1
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• pp.49-57
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• 2012

The Internet succeeded because of the end-to-end principle which allowed anyone to add functionality to the network. However, as the internet is increasingly becoming the platform for smart IT applications such as VoIP, IPTV, Cloud Computing and Smart Phone, networks are now under increasing strain of traffic congestion and the absence of quality of service insurances. To date, the debate over internet rules has focused on network neutrality rather than network efficiency. This article emphasizes the well-functioning role of market mechanism for the efficient use and further development of the network. To maximize the value of the network, this article proposes a differential treatment to packets based on customer types, and a two-part tariff pricing rule to secure funding to expand and upgrade networks.

• Journal of Internet Computing and Services
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• v.3 no.3
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• pp.67-74
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• 2002

Asynchronous transfer mode (ATM) can be efficiently used to transport packet data services. The switching system will support voice and packet data services simultaneously from end to end applications. To guarantee quality of service (QoS) of the offered services, source rate to send packet data is needed to control the network overload condition. Most existing control algorithms are shown to provide the threshold-based feedback control technique. However, real-time voice calls can be dynamically connected and released during data services in the network. If the feedback control information delays, quality of the serviced voice can be degraded due to a time delay between source and destination in the high speed link, An adaptive algorithm based on the optimal least mean square error technique is presented for the predictive feedback control technique. The algorithm attempts to predict a future buffer size from weight (slope) adaptation of unknown functions, which are used for feedback control. Simulation results are presented, which show the effectiveness of the algorithm.

• Journal of KIISE:Information Networking
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• v.32 no.2
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• pp.156-166
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• 2005

Ethernet passive optical network is an economical and efficient access network that has received significant research attention in recent years. A MAC(Media Access Control) protocol of PON, the next generation access network, is based on TDMA(Time Division Multiple Access) basically. In this thesis, we addressed the problem of dynamic bandwidth allocation in QoS based Ethernet PONs. We augmented the bandwidth allocation to support QoS in a differentiated services framework. Our differentiated bandwidth guarantee allocation(DBGA) allocates effectively and fairly bandwidths among end users. Moreover, we showed that DBGA that perform weighted bandwidth allocation for high priority packets result in better performance in terms of average and maximum packet delay, as well as network throughput compared with existing dynamic bandwidth allocations. We used simulation experiments to study the performance and validate the effectiveness of the proposed bandwidth allocations.

• The Journal of the Korea Contents Association
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• v.10 no.9
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• pp.36-44
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• 2010

In this paper, we propose video quality control system(VQCS) which can control video service quality through the monitoring of end-to-end available bandwidth for video streaming service like IPTV in NGN convergence network. Various multimedia services such as video, voice and gaming service can be provided by IPTV, and these services require large amounts of bandwidth. At this time, video quality degradation like video jerkiness, block distortion and blurring is caused when network available bandwidth is insufficient. Available bandwidth monitoring method is need to stably control video streaming quality. So, we periodically calculate the amount of the packets in link and measure available bandwidth by using total length field in IP header at terminal. Scalability extractor in network selects suitable video streaming data rate based on available bandwidth and transports video streaming with adaptive data rate to prevent video quality deterioration.

• ETRI Journal
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• v.37 no.2
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• pp.380-386
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• 2015

Guaranteeing quality of service over a multihop wireless network is difficult because end-to-end (ETE) delay is accumulated at each hop in a multihop flow. Recently, research has been conducted on network coding (NC) schemes as an alternative mechanism to significantly increase the utilization of valuable resources in multihop wireless networks. This paper proposes a new section-based joint NC and scheduling scheme that can reduce ETE delay and enhance resource efficiency in a multihop wireless network. Next, this paper derives the average ETE delay of the proposed scheme and simulates a TDMA network where the proposed scheme is deployed. Finally, this paper compares the performance of the proposed scheme with that of the conventional sequential scheduling scheme. From the performance analysis and simulation results, the proposed scheme gives more delay-and energy-efficient slot assignments even if the NC operation is applied, resulting in a use of fewer network resources and a reduction in ETE delay.

• Journal of KIISE:Information Networking
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• v.29 no.4
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• pp.428-436
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• 2002

Providing a mobile host with its required QoS is highly influenced by its mobility. The resource ReSerVation Protocol(RSVP) establishes and maintains a reservation state to ensure a given QoS level along the path from the sender to the receiver. However, RSVP is designed for use in fixed networks and thus it is inadequate in the mobile networking environment where a host changes its point of attachment. In this paper, we propose a new resource reservation protocol, RSVP-RA(RSVP by RSVP Agent) for mobile hosts. Our protocol assumes IETF Mobile IP as a mobility support mechanism. The proposed protocol introduce a new protocol entity - RSVP agent - to manage reservations in a mobile host's current visiting network. RSVP Agent is located in a local network and makes resource reservations in neighboring cells that the mobile host is expected to visit in the future. Thus, the proposed Protocol can provide a seamless QoS to the mobile host and significantly improve the scalability problem of RSVP by reducing the end-to-end signalling messages acrossing the backbone networks. The proposed protocols reduce packet delay, bandwidth overhead and the number of RSVP messages to maintain reservation states. We compared the performance of our proposed protocol with other proposed protocols in terms of signalling overhead, packet delay by simulation.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.6 no.3
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• pp.386-392
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• 2002

Internet telephony is an application that transmits voice data for conversation. Therefore it must provide high sound quality. However while audio packets are transferred through the network, they are affected by delay variations and jitters, which could result in poor sound quality of the receiving end does not have an appropriate jitter buffer to overcome network factors. This thesis introduces a buffer management algorithm that could be used to provide better sound quality for Internet phone terminals. This algorithm actively responds to both the compression algorithms that are used by the terminals, as well as to the received data to provide an improvement in sound quality. In order to verify the effectiveness of the proposed algorithm, we experimented in variance network settings. The results show that the proposed algorithm improves on the performance of the conventional buffer management algorithm.

• Proceedings of the Korean Information Science Society Conference
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• 2005.07a
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• pp.454-456
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• 2005

본 논문은 Wireless Mesh Network를 위한 새로운 MAC 프로토콜의 설계에 관해 기술한다. Mesh Network는 Mesh Router에 의해 무선 인프라를 구성한다. WMN(Wireless Mesh Network)는 고정된 상태의 Mesh Router에 의한 인프라를 구성하고 인터넷과 같은 기존의 인프라와 연결되는 구조적 특징을 가진다. 또한 다양한 응용분야의 지원을 위한 QoS(Quality of Service)가 요구되어지기 때문에 end-to-end 전송 지연, 공정성, 그리고 데이터 손실 등 많은 사항을 고려한 프로토콜의 설계가 요구된다. 본 논문에서는 WMN의 이러한 특징을 지원하기 위한 토폴로지의 제안과 TDMA를 활용한 MAC 프로토콜을 설계하였으며, NS-2 시뮬레이션을 통해 성능을 측정하였다.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.15 no.4
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• pp.839-844
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• 2011

IPTV contents are based on MPEG2-TS(Moving Picture Expert Group-2 Transport Stream). The Created content is delivered to each home's Set-Top box through the Internet service network, which will be visible on television through the decoding process. The using and spending of content expanding were quite limited in the other existing lines' devices. For this end, provided methods are to convert MPEG2-TS to other format and then transmit it. These methods are causing not only an overload to the system but also its increasing the hardware resource requirements. In this paper, the MPEG2-TS of IPTV's content using RTP(Real time Transport Protocol) provided by the applicable variety of devices and environmental system, was designed and implemented. Through this paper, the existing MPEG2-TS is providing other devices for distribution and consumption that can give better content to the consumers which means that the QoS(Quality of Service) was enhanced. And, the increased consumption and use of content will help the IPTV business to get more development and activation.