• Journal of KIISE:Computer Systems and Theory
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• v.32 no.11_12
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• pp.617-628
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• 2005

The services for ubiquitous computing environments have to provide automatically user-specific adaptive services in dynamically changed environments with many informations provided by both a user and his/her environment. Workflows used in business and distributed computing environments support service automation by connecting many tasks with rules or orderings of tasks. Therefore we must specify context information on transition condition to support context-aware services by adapting a workflow to ubiquitous computing environments. In this paper, we present a structural context model to specify context information on transition constraints of the workflow. And we propose an uWDL Ubiquitous Workflow Description Language) based on web services, which provides web service interfaces which are standardized and independent on heterogeneous and various platforms, protocols, and languages. The uWDL tan be interpreted and executed by a workflow engine, and provide users autonomic services based on context-awareness.

• Journal of the Korean Institute of Intelligent Systems
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• v.19 no.4
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• pp.511-518
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• 2009

A lot of methods are proposed to provide services for object informations in distributed domain to satisfy the recent increase of user-centered services. This paper proposed a method called fuzzy drop manager for the service of reliable distribution application domain objects. The proposed system accesses the domain using replica parameter ci,j and access matrix Z, and evaluates the reference relatedness inside the domain using the relatedness, given by the mapping of intra-domain fuzzy relevance, between fuzzy sets. Objects in the domain generated an $\alpha$-level set according to the reference relatedness obtained by applying $\alpha$-level to extend queries. Simulation results showed that the proposed method has better performance than the other methods.

• 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.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.8 no.11
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• pp.3916-3936
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• 2014

Smartphone applications like games, image processing, e-commerce and social networking are gaining exponential growth, with the ubiquity of cellular services. This demands increased computational power and storage from mobile devices with a sufficiently high bandwidth for mobile internet service. But mobile nodes are highly constrained in the processing and storage, along with the battery power, which further restrains their dependability. Adopting the unlimited storage and computing power offered by cloud servers, it is possible to overcome and turn these issues into a favorable opportunity for the growth of mobile cloud computing. As the mobile internet data traffic is predicted to grow at the rate of around 65 percent yearly, even advanced services like 3G and 4G for mobile communication will fail to accommodate such exponential growth of data. On the other hand, developers extend popular applications with high end graphics leading to smart phones, manufactured with multicore processors and graphics processing units making them unaffordable. Therefore, to address the need of resource constrained mobile nodes and bandwidth constrained cellular networks, the computations can be migrated to resourceful servers connected to cloud. The server now acts as a bridge that should enable the participating mobile nodes to offload their computations through Wi-Fi directly to the virtualized server. Our proposed model enables an on-demand service offloading with a decision support system that identifies the capabilities of the client's hardware and software resources in judging the requirements for offloading. Further, the node's location, context and security capabilities are estimated to facilitate adaptive migration.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.8 no.6
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• pp.1881-1900
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• 2014

Since 802.11 wireless LANs are so widely used, it has become common for numerous access points (APs) to overlap in a region, where most of those APs are managed individually without any coordinated control. This pattern of wireless LAN usage is called the private OBSS (Overlapping Basic Service Set) environment in this paper. Due to frame collisions across BSSs, each BSS in the private OBSS environment suffers severe performance degradation. This study approaches the problem from the perspective of congestion control rather than noise or collision resolution. The retry limit, one of the 802.11 attributes, could be used for traffic control in conjunction with TCP. Reducing the retry limit causes early discard of a frame, and it has a similar effect of random early drops at a router, well known in the research area of congestion control. It makes the shared link less crowded with frames, and then the benefit of fewer collisions surpasses the penalty of less strict error recovery. As a result, the network-wide performance improves and so does the performance of each BSS eventually. Reducing the retry limit also has positive effects of merging TCP ACKs and reducing HOL-like blocking time at the AP. Extensive experiments have validated the idea that in the OBSS environment, reducing the retry limit provides better performance, which is contrary to the common wisdom. Since our strategy is basically to sacrifice error recovery for congestion control, it could yield side-effects in an environment where the cost of error recovery is high. Therefore, to be useful in general network and traffic environments, adaptability is required. To prove the feasibility of the adaptive scheme, a simple method to dynamically adjust the value of the retry limit has been proposed. Experiments have shown that this approach could provide comparable performance in unfriendly environments.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.7A
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• pp.987-995
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• 1999

The handoff is a distinctive characteristic of the mobile networks. In the CDMA systems, if base stations support multiple Frequency Assignment (FA), they provide both soft handoff and hard handoff. Under the CDMA environments, the soft handoff guarantees the favorable service quality and the continuity of call connection without interruption, and increases the service capacity of the base stations. This paper proposes call connection control schemes with handoff queue for supporting efficient handoff processing. The proposed schemes are divided into two categories: single handoff queue scheme and multiple handoff queue scheme. We analyze the performance of the proposed call connection control schemes using numerical analysis. From the analysis results, we can say that it is more desirable to avoid hard handoff as long as handoff queues are used. When a single handoff queue used, adaptive scheme that properly mixes avoidable and avoidable hard handoff method under the given traffic condition is more desirable. In case that multiple handoff queues are used, the suitable trade-off needs to be developed between handoff blocking probability and hard handoff probability to guarantee a given blocking probability threshold.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.46 no.4
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• pp.67-75
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• 2009

Multi-layered video coding that provides scalability across the visual content has emerged for easily adaptive service over current heterogeneous network. However, the network is still error prone environment so that video service may suffer packet loss or erroneous decoding of the video. Especially distortion caused by the burst error may propagate to several pictures until intra refreshing, which will raise a terrific degradation of picture quality. To overcome the problem at terminal independently, we propose a new error concealment algorithm for the multi-layered video coding. The proposed method uses the similarity of between layers in the multi-layered video coding and POCS (Projections Onto Convex Sets) which is a powerful error concealment tool, but heavily dependent on initial values. To find adequate initial value which can reduce iteration times as well as achieve high performance, we took consideration into both features of layered approach coding and the correlation in neighbor blocks. The simulation results show that the proposed concealment method works well.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.8B
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• pp.797-804
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• 2009

Recently, SIP(Session Initiation Protocol) is used to set up and manage sessions for multimedia applications such as VoIP(Voice over IP) and IMS(IP Multimedia Subsystem). However, because SIP operates over the Internet, it is exposed to pre-existed internet security threats such as service degradation or service disruptions. Multimedia applications which are delay sensitive even suffers more from the threats mentioned above. The proposed methods so far to detect SIP INVITE flooding are CUSUM(Cumulative Sum), Hellinger distance and adaptive threshold, but among methods only take normal state into consideration. So, it is not capable of adapting the condition of the network congestion which are dynamically changing. In this paper, SIP INVITE flooding detection algorithm considering network congestion which enables efficient detections of such attacks is proposed. The proposed algorithm is expected to detect other types of attacks such as BYE and CANCEL more precisely compared to other methods.

• Journal of the Korea Society of Computer and Information
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• v.10 no.4 s.36
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• pp.29-36
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• 2005

As computing environments are more rapidly developed, an adaptive and intelligent services using post PC such as PDA. Laptop, and Tablet PC in case of rounding and examining patients are highly demanded. The objective of this study is to design and implement a context-awareness support system based on voice service for medical environments. To achieve it, we propose a context middleware which plays an important role in recognizing a client with PDA by using a Bluetooth wireless communication technology as well as in executing an appropriate execution module, like delivery for diagnosis information of patients, according to the staff's context acquired from a context server. In addition, the context server functions as a manager that efficiently stores context information such as client's current status, physical environment, and device resources into a database server. Finally, for verifying the usefulness of the proposed system, we develop an application system which provides voice playing services for notification of other physicians through our context middleware.

• The KIPS Transactions:PartC
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• v.9C no.1
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• pp.65-72
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• 2002

In this paper, we implement a high-speed swatch tilth the function for label integration to enhance the expansion of networks using the label space of routers efficiently on MPLS over ATM networks. We propose an appropriate hardware structure to support the VC-merge function and differentiated services simultaneously. In this paper, we use the adaptive congestion control method such as EPD algorithm in carte that there is a possibility of network congestion in output buffers of each core LSR. In addition, we justify the validity of the proposed VC-merge method through simulation and comparison to conventional Non VC-merge methods. The proposed VC-merge capable switch is modeled in VHDL. synthesized, and fabricated using the SAMSUNG 0.5um SOG process.