• The Transactions of the Korea Information Processing Society
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• v.4 no.4
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• pp.1059-1075
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• 1997

The transprot layer protocol for distrubuted multimedia applications in high speed network should gurantee the Quality of Serivce(QoS)requested by uwer.,The QoS Parmenters can be divied into two calssifications:those depend on the speed of a network such as bandiwidth,end-to-end transmission delay and throughput;and net- woek independent parameters such as various typed of group communications, retransmission method based on multimeda characteristic,acceptable packet wrror rate and transmission priority.In this proposed protocol,we divided user's QoS into performance redated parmeters and non-performance related parameters.The perform- ance-related paramenters are mapped into ATM traddic paramenters by the Distributed QoS Manger(DQM),the QoS manager,and the non-performance redlted paramenters are supported by the Distributed Multimedia Trans-port protocol(DMTP),a high-performance group transport protocol.Especially,becaues the DMTP is designed with considering the IP protocol,it can be efficiently used as an underlying protocol not onlu in ATM,but also in Ethernet,Token ring and FDDI LAN.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.12 no.6
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• pp.229-240
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• 2012

The development of multimedia applications followed by development of high-speed networks has improved the performance of mobile devices with high transfer speed broadband. Mobile internet access has made possible seamless indoor and outdoor mobile multicast services. Multicasting services are used to support efficient group communications. However, mobile multicasting services have two constraints: tunnel convergence and handover latency. Many protocols and handover methods have been proposed to address these problems. The inter-LMA optimized handover model for multicasting services has previously been proposed for PMIPv6-based networks. The proposed model removes the tunnel convergence issue and reduces router processing costs. It also makes possible the performance of fast handover operations with adaptive transmission mechanisms. In addition, the proposed scheme exhibits low packet delivery costs and handover latency in comparison with existing schemes, and ensures fast handover when moving the inter-LMA domain

• Journal of the Korean Society for Aeronautical & Space Sciences
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• v.36 no.6
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• pp.600-607
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• 2008

Reliable wireless transmission of data generated by the flight critical subsystems or mission equipments of the unmanned aerial system is critical for mission success. As the UAS system becomes more sophisticated, its dependency on a reliable high rate radio communication system also increases. This requirement is applied not only during the operation phase but also in the early development test phase. This paper introduces a practical cost-effective communication system for a UAS. The downlink module combines analog NTSC video signal with onboard data, and send them using 2.4 GHz carrier wave. The uplink system has less a severe requirement on the bandwidth, and thus uses 430 MHz signal. This paper also presents a sample packet structure which can be adopted for many UAS of similar class.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.7A
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• pp.770-777
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• 2007

Smaller cell size in the micro-cell or pico-cell structure brings about more frequent hand-overs between cells, and higher speed movement of the mobile terminal makes short the permissible time interval for executing these hand-over procedures. In this situation the hand-over failure may occur or some packets may be lost during the hand-over. Moreover a rapid degradation of throughput is triggered by packet re-transmission for compensating such errors. The QoS (Quality of Services) of mobile multimedia applications with higher bits rate requirements and higher speed mobility are severely affected by even shot service interruption. This paper proposes a new hand-over scheme to provide seamless services in the next generation mobile communication systems. Simulation is done to evaluate the performance of the proposed scheme based on its hand-over failure rate and packet loss rate.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.16 no.1
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• pp.121-127
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• 2016

Satellite communication is a communication method using a satellite as a repeater communication on the air and have a communication range is wide and is possible to transmit high-speed data, so Satellite communication has been highlighted as the next generation communication. However, because the distance between the ground and the satellite is far away, there is a disadvantage that propagation delay time is very long and packet is damaged. Therefore, this study was applied to Link Relay H-ARQ system in order to complement the damage to the packet. Furthermore, we propose the Modified Link Relay H-ARQ scheme is a new scheme to supplementation the delay time. By applying the Markov channel for implementing a channel of satellite communications, the results of analyzing the performance, when applying the Link Relay HARQ scheme, BER and PER performance was improved, when applying the Modified Link Relay H-ARQ scheme, Throughput performance was confirmed improvement.

• The Transactions of the Korea Information Processing Society
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• v.7 no.4
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• pp.1211-1216
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• 2000

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 rateot 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 fro feedback control. Simulation results are presented, which show the effectiveness of the algorithm.

• 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 Internet Computing and Services
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• v.7 no.5
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• pp.43-57
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• 2006

Next-generation mobile communication systems evolve in the form of high speed data transmission along with integration of wired-wireless network. Therefore, it needs researches on integrating service to heterogeneous networks to offer high speed data transmission and various services while supporting user mobility. In existing studies, heterogeneous networks are linked to single core network separatively, Since vertical handoff between heterogeneous networks leads to some delay, packets may be lost during vertical handoff. To solve this problem, this paper proposes an hierarchically integrated network architecture considering the characteristics of CDMA2000, WiBro, and WLAN. The hierarchically integrated networks are overlaid according to coverage of each network. Therefore, the proposed architecture can minimize handoff delay and packet loss. In addition, this paper proposes an integrated framework for next generation mobile communication networks.

• KIPS Transactions on Computer and Communication Systems
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• v.2 no.9
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• pp.381-392
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• 2013

During the last decade, the multitude of advances attained in terminal computers, along with the introduction of mobile hand-held devices, and the deployment of high speed networks have led to a recent surge of interest in Quality of Service (QoS) for video applications. The main difficulty is that mobile devices experience disparate channel conditions, which results in different rates and patterns of packet loss. One way of making more efficient use of network resources in video services over wireless channels with heterogeneous characteristics to heterogeneous types of mobile device is to use a scalable video coding (SVC). An SVC divides a video stream into a base layer and a single or multiple enhancement layers. We have to ensure that the base layer of the video stream is successfully received and decoded by the subscribers, because it provides the basis for the subsequent decoding of the enhancement layer(s). At the same time, a system should be designed so that the enhancement layer(s) can be successfully decoded by as many users as possible, so that the average QoS is as high as possible. To accommodate these characteristics, we propose an efficient transmission scheme which incorporates SVC-aware dual-channel repetition to improve the perceived quality of services. We repeat the base-layer data over two channels, with different characteristics, to exploit transmission diversity. On the other hand, those channels are utilized to increase the data rate of enhancement layer data. This arrangement reduces service disruption under poor channel conditions by protecting the data that is more important to video decoding. Simulations show that our scheme safeguards the important packets and improves perceived video quality at a mobile device.

• 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.