• 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 the Korea Computer Industry Society
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• v.9 no.2
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• pp.53-62
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• 2008

Intensive studies have been made in the area of IPTV VOD server. The basic goal of the study is to find an efficient mechanism to allow maximum number of users under the limited resources such as Buffer utilization, disk performance and network bandwidth. The overlay multicast that has been recently presented as an alternative for the IP multicast has been getting much persuasion by the system resource and the network bandwidth and the advancement of the network cost. we propose a efficient overlay multi_casting network policy for multimedia services with multi media partition storage. Simulation results show that the rate of service number and service time of proposed scheme are about 23% performance improved than that of traditional methods. This implies that our method can allow much more users for given resources.

• Journal of KIISE:Information Networking
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• v.35 no.5
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• pp.402-408
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• 2008

This paper considers the problem of minimizing network-wide broadcast latency in multi-rate wireless mesh networks where a node can dynamically adjust its link layer transmission rates to its neighbors. We propose a broadcast algorithm that complements existing broadcast construct algorithm which chooses a multicast node randomly when each candidate node has same metric. We consider the currently accumulated broadcast latency from source node to the each candidate node so far to choose the next broadcast node. The proposed broadcast algorithm for minimizing latency in a multi-rate mesh networks which exploit wireless advantage and the multi-rate nature of the network. Simulation based on current 802.11 parameters shows that proposed MinLink_WCDS algorithm improves overall latency than the previous existing broadcast algorithm.

• Journal of Satellite, Information and Communications
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• v.1 no.1
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• pp.40-47
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• 2006

The next-generation mobile communication system are required to provide high-data-rate and high quality transmission over hostile radio channels. Orthogonal Frequency Division Multiplexing(OFDM) technique regarded as a most promising solution in mobile environment. This paper considers the issue of efficient subcarrier allocation for OFDM system in multi-service application. Proposed algorithm can consider channel gain of each service and divide subchannel. Proposed method considers priority of service and divide more subchannels in high priority service. When proposed algorithm compares with existent algorithm, the convergence rate improves and the calculation amount is decreased.

• Journal of Communications and Networks
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• v.18 no.3
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• pp.375-386
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• 2016

Most existing studies on broadcast services in orthogonal frequency division multiplexing (OFDM) systems have focused on how to allocate the transmission power to the subcarriers. However, because a broadcasting system must guarantee quality of service to all users, the performance of the broadcast service dominantly depends on the channel state of the user who has the lowest received signal-to-noise ratio among users. To reduce the effect of the worst user on the system performance, we propose a joint delivery scheme of unicast and broadcast transmissions in OFDM systems with broadcast and unicast best-effort users. In the proposed joint delivery scheme, the BS delivers the broadcast information using both the broadcast and unicast subcarriers at the same time in order to improve the performance of the broadcast service. The object of the proposed scheme is to minimize the outage probability of the broadcast service while maximizing the sum-rate of best-effort users. For the proposed joint delivery scheme, we develop an adaptive power and subcarrier allocation algorithm under the constraint of total transmission power. This paper shows that the optimal power allocation over each subcarrier in the proposed scheme has a multi-level water filling form. Because the power allocation and the subcarrier assignment problems should be jointly solved, we develop an iterative algorithm to find the optimal solution. Numerical results show that the proposed joint delivery scheme with adaptive power and subcarrier allocation outperforms the conventional scheme in terms of the outage probability of the broadcast service and the sum-rate of best-effort users.

• Journal of Korea Multimedia Society
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• v.14 no.1
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• pp.33-43
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• 2011

Routing technologies considering QoS-based hypermedia services have been seen as a crucial network property in next generation optical Internet (NGOI) networks based on IP/dense-wavelength division multiplexing (DWDM). The huge potential capacity of one single fiber. which is in Tb/s range, can be exploited by applying DWDM technology which transfers multiple data streams (classified and aggregated IP traffics) on multiple wavelengths (classified with QoS-based) simultaneously. So, DWDM-based optical networks have been a favorable approach for the next generation optical backbone networks. Finding a qualified path meeting the multiple constraints is a multi-constraint optimization problem, which has been proven to be NP-complete and cannot be solved by a simple algorithm. The majority of previous works in DWDM networks has viewed heuristic QoS routing algorithms (as an extension of the current Internet routing paradigm) which are very complex and cause the operational and implementation overheads. This aspect will be more pronounced when the network is unstable or when the size of network is large. In this paper, we propose a flooding-based unicast and multicast QoS routing methodologies(YS-QUR and YS-QMR) which incur much lower message overhead yet yields a good connection establishment success rate. The simulation results demonstrate that the YS-QUR and YS-QMR algorithms are superior to the previous routing algorithms.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.1B
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• pp.26-38
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• 2006

To construct seamless collaborative environments, what all participants intent should be delivered, and visual elements such gesture, facial expression, and ambiance should be shared with all participants. In this paper, we propose high-quality video service to support DV(digital video) and HDV(high-definition DV) based on Access Grid(AG) which is a prevalent collaborative system. The proposed service is designed for employing versatile media tools and codecs with SDP(session description protocol) and SAP(session announcement protocol). We also design network-adaptive video transmission module to mitigate the impact of network fluctuation. This periodically monitors multicast performance and controls frame rate on sender side considering network condition. The experimental results over the test bed show that proposed service enhances quality of AG video service and provides seamless high-quality video transport by mitigating the impact of network fluctuation.

• The Transactions of the Korea Information Processing Society
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• v.5 no.12
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• pp.3047-3062
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• 1998

This paper deseribes the desihn and the implementation of the Realtime Transport Protocol(RTP)/ Rdaltime Control Protocol(RTCP) (RFC 1889,1890) that is used to transmit the audio/video data to any destination and to feedback the Quality of Service (QoS) information of the received media data to the sender, in the teleconferencing systems proposed by ITU-T. These protocols are implemented with multi thead technique and run on top of UDP/IP-Multicast through the socket interface as the underlying protocol. The upper layer is impelmented such that in can be accessed by the H245 comference control protocol. The RTP packetizes the digitized audio/video data from the encoder info a fixed format, and multieast to the participants. The RTCP monitors RTP packets and extracts the QoS values from it such as round-trip delay, jiter and packet loss to form RTCP packets and non periokically sends them to the sender site. In this Paper, we also descritx the study of measurement and analysis for QoS factors that observed on performing teleconferencing system over Internet. The results from this experiment is indicate that RTT and Jitter value are acceptable even entwork load is high. However, it appears that packet loss rate is high in daytime and most losses periods have length one or two.