• 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.

• Journal of Institute of Control, Robotics and Systems
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• v.18 no.8
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• pp.750-757
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• 2012

This paper focuses on human adaptive HVAC system that can regulate the thermal comfort of the resident in intelligent buildings. The thermal comfort is represented in this paper by PMV (Predicted Mean Vote) as defined by ISO 7730. This PMV value indicates how hot or cold a resident feels by considering temperature, humidity, resident's metabolic rate, relative air velocity, and clothing insulation. In order to develop such a system, a location detection system based on ZigBee module was used along with temperature sensors, other environment sensors. The human adaptive HVAC system was evaluated experimentally on a test bed emulating a room.

• Journal of KIISE:Information Networking
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• v.37 no.6
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• pp.483-488
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• 2010

Due to the prevalence of various mobile devices and wireless broadband networks, there has been a significant increase in interest and demand for multimedia streaming services. Moreover, the user can service the participatory video broadcasting service in the mobile device and it can be used to deliver the real-time news and more variety information in the user side. Live multimedia service of user participation should consider not only the video quality but also the delay bounds and continuity of video playback for improving the user perceived QoS (Quality of Service) of streaming service. In this paper, we propose an adaptive rate control scheme, called DeBuG (Delay Bounds Guaranteed), to guarantee the delay bounds and continuity of video playback for the real-time streaming in mobile devices. In order to provide those, the proposed scheme has a quality adaptation function based on the transmission buffer status and network status awareness. It also has a selective frame dropper, which is based on the media priority, before the transmission video frames. The simulation results demonstrate the effectiveness of our proposed scheme.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.8 no.11
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• pp.4068-4086
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• 2014

Due to the prevalence of powerful mobile terminals and the rapid advancements in wireless communication technologies, the wireless video streaming service has become increasingly more popular. Recent studies show that video streaming services via Transmission Control Protocol (TCP) are becoming more practical. TCP has more advantages than User Diagram Protocol (UDP), including firewall traversal, bandwidth fairness, and reliability. However, each video service shares an equal portion of the limited bandwidth because of the fair sharing characteristics inherent in TCP and this bandwidth fair sharing cannot always guarantee the video quality for each user. To solve this challenging problem, an Adaptive Multiple TCP (AM-TCP) scheme is proposed in this paper to guarantee the video quality for mobile devices in wireless networks. AM-TCP adaptively controls the number of TCP connections according to the video Rate Distortion (RD) characteristics of each stream and network status. The proposed scheme can minimize the total distortion of all participating video streams and maximize the service quality by guaranteeing the quality of each video streaming session. The simulation results show that the proposed scheme can significantly improve the quality of video streaming in wireless networks.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.6B
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• pp.588-603
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• 2003

In this paper, we propose an Aggregate Three Color Marker without per flow management which is required for an Edge router to improve End-to-End QoS of Assured Service in DiffServ. Proposed Aggregate Three Color Marker is used with the Adaptive RIO-DC scheme to achieve the minimum rate guarantee without per flow management. Assuming that the admission control for Assured Service has been performed, proposed Aggregate Three Color Marker measures incoming In-profile traffic rate at the output link of an edge router using a token-bucket with a token rate equal to the sum of contracted rates of admitted flows passing the edge router. If there are token losses from the token bucket, out-of-profile packets are promoted to Yellow packets within the aggregate traffic profile. And yellow packets are demoted to out-of-profile packets at the input link to an Edge router fer the purpose of fairness maintenance. In-profile packets and Yellow packets are processed identically at the RIO-DC buffer management scheme in our proposed method. Simulation results show that through using proposed Aggregate Three Color Marker with the Adaptive RIO-DC scheme, the minimum rate guarantee for Assured Service can be achieved without per flow management at multiple DiffServ domains.

• The KIPS Transactions:PartC
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• v.13C no.2 s.105
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• pp.241-248
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• 2006

Mobile computing has such characteristics as portability, wireless network, mobility, etc. These characteristics cause various problems to mobile terminals like frequent disconnection, high error rate, and varying network status. These problems motivate us to develop an adaptive error control mechanism for supporting multimedia service in mobile computing environment. In this paper, we propose the Adaptive Error Control(AEC) scheme using client's buffer size and current error rate. After categorizing the status into four groups according to client's buffer size and current error rate, this scheme applies an appropriate error control scheme to each status. In this scheme, thresholds of buffer size and error rate are determined by the data transmission time, play rate and average VOP size, and by the probability of error for a sequence of packets. The performance of proposed scheme is evaluated by flaying MPEG-4 files on an experimental client/server environment, respectively. The results show that error correcting rate is similar to other schemes while the time for correcting error reduce a little. In addition, the size of data for correcting error is decreased by 23% compared with FEC and Hybrid FEC, respectively. Theses results demonstrate that the proposed scheme is more suitable in mobile computing environment with small bandwidth and varying environment than existing schemes.

• Proceedings of the IEEK Conference
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• summer
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• pp.147-151
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• 2004

QoS is a subject of high interest for successful deployment of various services in a home gateway and the gateway is possible to support QoS by installing existing queuing disciplines, which control the outgoing traffic to guarantee only QoS of the traffic. But m the home gateway it is also important to guarantee QoS of the incoming traffic. This paper proposes an adaptive control of the traffic to guarantee QoS of incoming traffic into the home gateway. In the proposed method, the upper limit of the available bandwidth of sending rate varies with receiving rate. And the proposed method makes the gap between the allocated rate and the actual service rate of the traffic narrow. Some experiments on a test bed show that the proposed method is valid.

• International Journal of Contents
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• v.2 no.2
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• pp.10-16
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• 2006

In this paper, we propose a predictive resource metric region (RMR) based radio resource metric estimation (RRME) mechanism, which utilizes a resource metric mapping function (RMMF), both of which permit efficient inter-working between the physical layer and higher layers for envisaging multimedia service applications over a CDMA communication system platform. The RMR can provide the acceptable resource region where QoS and acceptable link quality can be guaranteed with an achievable resource margin to be utilized in terms of capacity margin, the degree of confidence (DCL) of user, second-order statistics of Eb/Io. With predicted capacity margin and variance, DCL can deliver decision parameters with which an adaptive QoS based admission control can perform well taking capacity and resource availability into account in a dynamic and predictive manner. Combined with advanced techniques such as adaptive modulation or rate control and power control, the proposed mechanism can adjust the conventional stringent link quality information efficiently, and deliver accurate information of the resource availability. Thus, these can guarantee the maximization of resource utilization of multimedia service applications.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.10A
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• pp.1532-1538
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• 1999

In general, because error rate of wireless link is higher than that of wired link, DLC layer protocol for wired network with low error rate is not proper for wireless environments. In addition, the conventional DLC layer protocol for wireless network is optimized for the low-speed data service, so it is difficult to use conventional DLC protocol in the current mobile communication environments handing high-speed and multimedia services. Therefore, a DLC layer protocol that is suitable to current wireless communication environments is required. In this paper, we propose a novel error control scheme that supports a variety of traffic attribute and is applicable to high-speed and multimedia data service in WATM. The proposed scheme provides enhanced throughput performance for real-time traffic by using modified ASR ARQ without ACK and reduces loss rate by using FEC in the case of high error condition. Also, for non real-time traffic, the use of ASR ARQ without ACK enhances throughput performance and delay time is decreased by using FEC in the case of high error rate channel. As a result of simulation, the proposed scheme has better performance than conventional ASR ARQ protocol in view of delay and throughput.

• Journal of Communications and Networks
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• v.11 no.4
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• pp.375-383
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• 2009

The WiMedia alliance has specified a distributed medium access control (WiMedia MAC) protocol based on ultra wideband (UWB) for high data rate WPANs (HR-WPANs). The merits of WiMedia MAC such as distributed nature and high data rate make it a favorite candidate in HR-WPAN. Although QoS parameters such as the range of service rates are provided to a traffic stream, the WiMedia MAC is not able to use the QoS parameters and to determine or adjust a service rate using the QoS parameters for the traffic stream. In this paper, we propose a fair and adaptive resource allocation method that allocates time slots to isochronous streams according to QoS parameters and the current traffic load condition in a fully distributed manner. Although the traffic load condition changes, each device independently recognizes the changes and calculates fair and maximum allowable service rates for traffic streams. From the numerical and simulation results, it is proved that the proposed method achieves high capacity of traffic streams and fair QoS provisioning under various traffic load condition.