• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.6B
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• pp.595-604
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• 2009

The HCCA reserves the channel resources based on the mean data rate in IEEE 802.11e. It may cause either the waste of channel resource or the increase of transmission delay at MAC layer if the frame size is rapidly varied when a compressed mode video codec such as MPEG video is used. To solve these problems, it is developed that the packet scheduler allocates the wireless resource adaptation by according to the packet size. However, it is difficult to perform the admission control because of the difficulty with calculating the available resources. In this paper, we propose a CAC mechanism to solve the problem that may not satisfy the QoS by increasing traffic load in case of using EDCA. Especially, the proposed CAC mechanism calculates the EB of TSs using the traffic information transmitted by the application layer and the number of average transmission according to the wireless channel environment, and then determines the admission of the TS based on the EB. According to the simulation results of the proposed CAC mechanism, it admitted the TSs under the loads which are satisfied within the delay bound. Therefore, the proposed mechanism guarantees QoS of streaming services effectively.

• Journal of Advanced Navigation Technology
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• v.14 no.4
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• pp.530-536
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• 2010

This paper proposes a QoS/QoE guaranteeing scheme by reassembling packets that have already been delivered to the end systems under the condition that there is no other way to fulfill user's QoS/QoE with the current network conditions. The proposed scheme is a QoS/QoE provisioning scheme that can be applied when IPTV streaming packets have not been consecutively received by the receiver. The receiving end system modifies the continuous counter field in the MPEG-2 Transport Stream and reassembles packets based on this information. In the simulation, broadcasting streaming environment has been implemented in order to verity the QoS/QoE performance of the proposed scheme under the good and poor network conditions.

• Journal of the Korea Society of Computer and Information
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• v.10 no.5 s.37
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• pp.187-198
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• 2005

In this paper, we propose the enhanced header compression scheme for the efficient data traffic transmission in Ipv6 networks. The bandwidth of wireless links and IP networks will probably be limited due to properties of the physical medium and regulatory limits on the use of frequencies for radio communications. That is major cause of user throughput reduction. Therefore, We discuss the IPHC(RFC2507) and ROHC(RFC3095) scheme. IPHC is simple header compression scheme and ROHC is enhanced header compression that have fast optimal recovery scheme. We have studied the enhanced header compression scheme in ROHC. We will show that indication of compression context values preventing from packet losses can provide the fast recovery of compression state. Computer simulations show that the proposed scheme has better performance than the previous one.

• Journal of the Korea Society of Computer and Information
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• v.13 no.7
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• pp.91-98
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• 2008

In digital broadcasting, it is possible to multiplex maximum one hundred audio or music programs into MPEG-2 transport stream, which is suitable for transmitting through one channel. In order to check if multiplex music programs are transmitted well, we need a multiplex audio monitoring system that monitors the programs in real-time. In analog broadcasting, we have used hardware-based audio monitoring system for a small number music programs. However, the effectiveness of hardware-based audio monitoring system from the cost and function viewpoint is so low that a new system is needed for digital broadcasting. In this paper, we have designed and implemented a software-based audio monitoring system to satisfy these requirements. In this implementation, only one PC is used without other hardware facilities, and the system monitors digital broadcasting music programs effectively. Transmitted digital broadcasting streams are demultiplexed into many music programs and the realtime value of audio level and packet error information for these programs are displayed in the screen. Thus, the system detects and shows the abnormal transmitting programs automatically. Simulation results show that effective realtime multiplex audio monitoring is possible for digital broadcasting music programs.