• Journal of the Korea Society of Computer and Information
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• v.15 no.11
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• pp.135-142
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• 2010

QoS in the proxy system are under heavy influence from interferences such as congestion, latency, and retransmission. Also, multi-level streaming services affects from temporal synchronization, which lead to degrade the service quality. This paper proposes a new segment-based buffer management mechanism which reduces performance degradation of streaming services and enhances throughput of streaming due to drawbacks of the proxy system. The proposed paper optimizes streaming services by: 1) Use of segment-based buffer management mechanism, 2) Minimization of overhead due to congestion and interference, and 3) Minimization of retransmission due to disconnection and delay. This paper utilizes fuzzy value $\mu$ and cost weight $\omega$ to process the result. The simulation result shows that the proposed mechanism has better performance in buffer cache control rate, average packet loss rate, and delay saving rate with stream relevance metric than the other existing methods of fixed segmentation method, pyramid segmentation method, and skyscraper segmentation method.

• Journal of the Korea Society for Simulation
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• v.23 no.4
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• pp.65-74
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• 2014

IEEE 802.11n standard has introduced the new schemes in MAC and PHY layers to improve network throughput. Frame aggregation and Block ACK are mainly defined to increase the efficiency of the MAC layer. There exists still problem in IEEE 802.11n. When block ACK request and/or response frames are missing or received in error, the sender does not know the status (success/failure) of each frame in the aggregated large frame and retransmits all the frames. This can cause a lower network performance. To solve this problem, we propose a new effective scheme, called reduced retransmission of MPDUs (RRM) scheme. In the proposed scheme, when a sender does not receive a block ACK response frame, it just transmits a next data frame and requests a block ACK. Therefore, it can retransmits the erroneous frames. Performance of the proposed scheme is investigated by simulation. Our results show that the proposed scheme is very effective and improves the performance under a wide range of channel error conditions.

• Journal of the Institute of Electronics and Information Engineers
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• v.52 no.8
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• pp.10-17
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• 2015

A satellite communication system is one of viable solutions for Internet applications running in wide areas. However, the performance of TCP can be seriously degraded in the satellite networks due to long round-trip time (RTT) and high bit error rate (BER) over satellite links. Therefore, a performance enhancing proxy(PEP) based TCP splitting connection scheme is used in the satellite link to improve the TCP performance. In this paper, we implement PEP testbed and conduct experiment to evaluate the performance of TCP splitting connection by comparing with high-speed TCP solutions in various environments. In our experimental environment, we consider multiple connections, high packet loss, and limited bandwidth. The experiment results show that PEP improves the TCP throughput than high-speed TCP variants in various environments. However, there is no improvement of the TCP throughput with the limited bandwidth because there is packet loss caused by both the congestion and the channel error.

• The Journal of the Korea Contents Association
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• v.10 no.3
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• pp.80-91
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• 2010

Due to the convergence of broadcasting and communications, IPTV services are spotlighted as the that next-generation multimedia services. IPTV services should have functionality such as unlimited channel capacity, extension of media, QoS awareness and are required increasing traffic and quality control technology to adapt the attributes of IPTV service. Consequently, flow based quality control techniques are needed. Therefore, many studies for providing Internet QoS are performed at IETF (Internet Engineering Task Force). As the buffer management mechanism among IP QoS methods, active queue management method such as RED(Random Early Detection) and modified RED algorithms have proposed. However, these algorithms have difficulties to satisfy the requirements of various Internet user QoS. Therefore, in this paper we propose the Flow based AQM(Active Queue Management) algorithm for the multimedia services that request various QoS requirements. The proposed algorithm can converge the packet loss ratio to the target packet loss ratio of required QoS requirements. And we present a performance evaluation by the simulations using the ns-2.

• 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.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.19 no.8
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• pp.1901-1907
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• 2015

Today's smartphones and user devices are equipped with multiple radio interfaces increasingly. Aggregating theses multiple radio interfaces and using them concurrently will increase a user's communication speed immediately, but at the expense of increased power consumption. In this paper, we develop a mathematical performance model of an adaptive radio activation scheme by which a radio interface is activated only when needed for performance increase and deactivated otherwise. The developed model shows that the adaptive scheme reduces delay significantly and almost halves power consumption below a certain level of traffic input.

• Journal of Broadcast Engineering
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• v.13 no.4
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• pp.440-451
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• 2008

With the advent of IP-based multimedia service based on IP network, there is a rapidly increasing demand for IPTV. Unlike the previous coaxial cable based TV, IPTV provides a variety of convergence services based on IP newark. However, since the IPTV service quality is a lot affected by the network degradation such as packet loss and jitter, it may not be guaranteed. In this paper, we propose an objective measure for various degradations of IPTV-based videos considering subjective assessment. To this end, we first determine QoE(Quality of Experience) indicators, which can affect human visual perception. Then we develop the video quality metric for each QoE indicator. Subjective assessment based on MOS is conducted and used to construct mapping relationship between each measure and perceived visual quality. Experiments are performed on various videos to confirm the efficiency and robustness of the proposed method and show high correlation with subjective assessment.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.17 no.7
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• pp.1618-1626
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• 2013

Virtualization service processes all operation including the data creation, storing, and disposal in a server and transmits processed data as the streaming media form. Therefore, client can use the same environment as the traditional desktop environment without considering the type of device. Virtualization service should consider not only the video quality but also the delay bounds and continuity of video playback for improving the user perceived Quality of Service(QoS) of streaming service. In this paper, we propose a network-adaptive transmission architecture that focuses on guaranteeing QoS requirements for virtualization service. In order to provide those, the proposed architecture have the transmission rate adaptation function based on available bandwidth and the content bit-rate control function based on sender buffer state. Through each function, proposed architecture guarantee the delay bounds and continuity of virtualization contents playback. The simulation results show that proposed network-adaptive transmission architecture provides a improve performance of throughput and transmission delay.

• The KIPS Transactions:PartC
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• v.9C no.4
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• pp.543-550
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• 2002

This paper proposes the dynamic slot allocation method for real-time VBR (rt-VBR) services in wireless ATM networks. The proposed method is characterized by a contention-based mechanism of the reservation request, a contention-free polling scheme for transferring the dynamic parameters. The base station scheduler allocates a dynamic parameter minislot to the wireless terminal for transferring the residual lifetime and the number of requesting slots as the dynamic parameters. The scheduling algorithm uses a priority scheme based on the maximum cell transfer delay parameter. Based on the received dynamic parameters, the scheduler allocates the uplink slots to the wireless terminal with the most stringent delay requirement. The simulation results show that the proposed method guarantee the delay constraint of rt-VBR services along with its cell loss rate significantly reduced.

• Journal of KIISE:Computer Systems and Theory
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• v.32 no.11_12
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• pp.654-662
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• 2005

Mobile portable devices for wireless network solely depend on a limited battery power. Therefore, we need to design for wireless communication protocols with an energy efficiency. TCP-Westwood is one of the most important approaches on TCP performance improvement in wireless environments that estimates the available bandwidth by using the sampling mechanism. The advantage is that data can be transmitted efficiently using the estimation of available bandwidth. However, when the sender with TCP-Westwood is in a wireless environment, it does not consider of the sampling mechanism operation. In this thesis, a new energy saving transport protocol, called E2TP(Energy Efficient Transport Protocol), is proposed to solve problems which occur when the sender with TCP-Westwood is in a wireless environment. Also, when there are packet loss while doing frequent link error in a wireless environment, E2TP provides the instantaneous segment size adjustment for a more efficient data retransmission. The simulation result proves that the proposed E2TP has better performance in energy efficiency and throughput than both TCP and TCP-Westwood.