• KSII Transactions on Internet and Information Systems (TIIS)
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• v.10 no.4
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• pp.1481-1500
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• 2016

Device-to-Device (D2D) communication underlaying macro-small cell networks, as one of the promising technologies in the era of 5G, is able to improve spectral efficiency and increase system capacity. In this paper, we model the cross- and co-tier D2D communications in two-tier macro-small cell networks. To avoid the complicated interference for cross-tier D2D, we propose a mode selection scheme with a dedicated resource sharing strategy. For co-tier D2D, we formulate a joint optimization problem of power control and resource reuse with the aim of maximizing the overall outage capacity. To solve this non-convex optimization problem, we devise a heuristic algorithm to obtain a suboptimal solution and reduce the computational complexity. System-level simulations demonstrate the effectiveness of the proposed method, which can provide enhanced system performance and guarantee the quality-of-service (QoS) of all devices in two-tier macro-small cell networks. In addition, our study reveals the high potential of introducing cross- and co-tier D2D in small cell networks: i) cross-tier D2D obtains better performance at low and medium small cell densities than co-tier D2D, and ii) co-tier D2D achieves a steady performance improvement with the increase of small cell density.

• Journal of Internet Computing and Services
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• v.12 no.2
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• pp.75-83
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• 2011

Wireless services and devices that use frequency increase more and more because of advancement of the industry. Therefore, the available spectrum band becomes increasingly insufficient. Cognitive Radio, which adaptively utilizes the vacant licensed spectrum band, is considered as an effective way to utilize the spectrum resource shortage. CR user should move from current allocated channel to vacant channel to avoid the interference to the primary user when the primary user appears in the current channel. In this case, CR system undergoes the break off time until handshake is completed. So, in order to guarantee the Quality of Service(QoS) of CR system, fast handshake method is required. In this paper, we propose the embedded control signal transmission technique to reduce the break off time. This method can transmit the control signals in data transmission period as well as the broadcasting period. Proposed method can improve the data throughput and decrease the break off time. Computer based simulation proves that our proposed scheme outperforms conventional one.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.45 no.5
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• pp.125-133
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• 2008

In this paper, we propose active queue management mechanism (Active-WRED) to guarantee quality of the high priority service class in multi-class traffic service environment. In congestion situation, this mechanism increases drop probability of low priority traffic and reduces the drop probability of the high priority traffic, therefore it can improve the quality of the high priority service. In order to analyze the performance of our mechanism we introduce the stochastic analysis of a discrete-time queueing systems for the performance evaluation of the Active Queue Management (AQM) based congestion control mechanism called Weighted Random Early Detection (WRED) using a two-state Markov-Modulated Bernoulli arrival process (MMBP-2) as the traffic source. A two-dimensional discrete-time Harkov chain is introduced to model the Active-WRED mechanism for two traffic classes (Guaranteed Service and Best Effort Service) where each dimension corresponds to a traffic class with its own parameters.

• 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.

• The KIPS Transactions:PartC
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• v.15C no.4
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• pp.253-262
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• 2008

Providing Quality-of-Service (QoS) guarantee requires the reservation of resource needed to accommodate the traffic flow of service for which resource reservation protocol (RSVP) was introduced. However, if any RSVP-capable router on the path fails to accommodate the traffic flow due to lack of resource, the flow is not allowed no matter how much surplus resources other routers on the path have. In order to solve this problem, this paper proposes a resource reservation method with available resource migration between RSVP-capable routers in which the routers use the recently developed latency optimized fair queuing scheduler. The results of the simulation applying the proposed method to an evaluation network show that it may yield the gain of up to 165% compared to that in the original one in terms of the number of admitted flows.

• The KIPS Transactions:PartA
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• v.8A no.4
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• pp.287-292
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• 2001

Multimedia service systems should have efficient capacity to serve the growing clients and new data. In the general streaming services, users can endure the small amount of time delay at the beginning of service. But they want to have good quality of service. A streaming server tries to transfer video files to clients from a repository of files in real time. The server must guarantee concurrent and uninterrupted delivery of each video stream requested from clients. To achieve its purpose, many stream servers adopt multi-processors, sufficient memory, and RAID or SAN in their systems. In this paper, we propose a Linux-based parallel media streaming server. It is superior to the other systems in the storing structure, fault-tolerance, and service capacity. Since this system supports the web interlace, users can operate easily through the www. This system uses unique striping policy to distribute multimedia files into the parallel storage nodes. If a service request occurs, each storage node transmits striped files concurrently to the client. Its performance is better than the single media streaming service because of the parallel architecture.

• The KIPS Transactions:PartC
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• v.8C no.6
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• pp.805-814
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• 2001

In this paper, we propose a new scheduling algorithm that guarantees the delay property of real-time traffic, not considered in previous DWRR(Dynamic Weighted Round Robin) algorithm and also transmits non-real-time traffic efficiently. The proposed scheduling algorithm is a variation of DWRR algorithm to guarantee the delay property of real-time traffic by adding cell transmission method based on delay priority. It also uses the threshold to prevent the cell loss of non-real-time traffic due to cell transmission method based on delay priority. Proposed scheduling algorithm may increase some complexity over conventional DWRR scheme because of cell transmission method based on delay priority. However, the consideration of delay priority can minimize cell delay and require less size of temporary buffer. Also, the results of our performance study shows that the proposed scheduling algorithm has better performance than conventional DWRR scheme due to reliable ABR service and congestion avoidance capacity.

• Journal of Internet Computing and Services
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• v.5 no.6
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• pp.31-43
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• 2004

Voice over Internet Protocol (VoIP) is currently a popular research topic as a real time voice packet transmission method. But current Internet environment do not guarantee the quality of voice when we take a side view of delay, jitter and loss. Up to now, many voice based evaluation algorithms have been used to measure speech quality of VoIP systems. However, these algorithms have the defects that their results are different according to voice samples and some algorithms can not take network environment for speech transmission path. The E-model can be used to solve the problems of these algorithms. In this paper. we introduce VoIP planning guidelines through the various analysis of E-model which can model impairments of network quality as well as VoIP equipment quality systematically, We, also, show the evaluation method and results of speech transmission quality.

• Journal of Korea Multimedia Society
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• v.11 no.7
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• pp.966-976
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• 2008

Providing Quality-of-Service(QoS) guarantee requires the reservation of resource needed to accommodate the traffic flow of service for which resource reservation protocol(RSVP) was introduced. However, if any RSVP-capable router on the path fails to accommodate the traffic flow due to lack of resource, the flow is not allowed no matter how much surplus resources other routers on the path have. In order to solve this problem, this paper proposes a resource reservation method that can adjust adaptively the delay budget of traffic flow allocated to each RSVP-capable router in which the router uses the recently developed general-time fair queuing scheduler. The results of the simulation applying the proposed method to an evaluation network show that it may yield the gain of up to 90% compared to that in the original one in terms of the number of admitted flows.

• The KIPS Transactions:PartC
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• v.18C no.6
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• pp.433-440
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• 2011

In this paper, we propose a new scheduling method that can guarantee reliable jitter by minimizing the queue length variation in the streaming service provisioning such as IPTV and VoD. The amount of traffic to be delivered within a certain time is very fluid because MPEG-4 and H.264 encoders use VBR(Variable Bit Rate) for delivering video streaming traffic. This VBR characteristic increases the end-to-end propagation delay variation when existing scheduling methods are used for delivering video frames. Therefore, we propose the new scheduling method that can minimize change rate of queue length by adaptively controling service rate taking into account the size of bulk incoming packets and arrival rate for bulk streaming traffic. Video frames can be more reliably transmitted through the minimization of the queue length variation using the proposed method. We use the queueing model and also carry out OPNET simulation to validate the proposed method.