• Journal of Communications and Networks
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• v.18 no.5
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• pp.784-795
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• 2016

In this paper, we investigate the resource allocation problem in time-varying heterogeneous wireless networks (HetNet) with multi-homing user equipments (UE). The stochastic optimization model is employed to maximize the network utility, which is defined as the difference between the HetNet's throughput and the total energy consumption cost. In harmony with the hierarchical architecture of HetNet, the problem of stochastic optimization of resource allocation is decomposed into two subproblems by the Lyapunov optimization theory, associated with the flow control in transport layer and the power allocation in physical (PHY) layer, respectively. For avoiding the signaling overhead, outdated dynamic information, and scalability issues, the distributed resource allocation method is developed for solving the two subproblems based on the primal-dual decomposition theory. After that, the adaptive resource allocation algorithm is developed to accommodate the timevarying wireless network only according to the current network state information, i.e. the queue state information (QSI) at radio access networks (RAN) and the channel state information (CSI) of RANs-UE links. The tradeoff between network utility and delay is derived, where the increase of delay is approximately linear in V and the increase of network utility is at the speed of 1/V with a control parameter V. Extensive simulations are presented to show the effectiveness of our proposed scheme.

• The Journal of The Korea Institute of Intelligent Transport Systems
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• v.13 no.3
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• pp.66-77
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• 2014

UTIS(Urban Traffic Information System) directly collects link travel time in urban area by using probe vehicles. Therefore it can estimate more accurate link travel speed compared to other traffic detection systems. However, UTIS includes some missing data caused by the lack of probe vehicles and RSEs on road network, system failures, and other factors. In this study, we suggest a new model, based on k-NN algorithm, for imputing missing data to provide more accurate travel time information. New imputation model is an adaptive k-NN which can flexibly adjust the number of nearest neighbors(NN) depending on the distribution of candidate objects. The evaluation result indicates that the new model successfully imputed missing speed data and significantly reduced the imputation error as compared with other models(ARIMA and etc). We have a plan to use the new imputation model improving traffic information service by applying UTIS Central Traffic Information Center.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.12 no.8
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• pp.3856-3872
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• 2018

HTTP-based adaptive streaming (HAS) has recently been widely deployed on the Internet. In the HAS system, a video content is encoded at multiple bitrates and the encoded video content is segmented into small parts of fixed durations. The HAS client requests a video segment and stores it in the playout buffer. The rate adaptation algorithm employed in HAS clients dynamically determines the video bitrate depending on the time-varying bandwidth. Many studies have shown that an efficient rate adaptation algorithm is critical to ensuring quality-of-experience in HAS systems. However, existing algorithms have problems estimating the network bandwidth because bandwidth estimation is performed on the client-side application stack. Without the help of transport layer protocols, it is difficult to achieve accurate bandwidth estimation due to the inherent segment-based transmission of the HAS. In this paper, we propose an alternative approach that utilizes the playout buffer occupancy rather than using bandwidth estimates obtained from the application layer. We start with a queueing analysis of the playout buffer. Then, we present a buffer-aware rate adaptation algorithm that is solely based on the mean buffer occupancy. Our simulation results show that compared to conventional algorithms, the proposed algorithm achieves very smooth video quality while delivering a similar average video bitrate.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.9 no.8
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• pp.3121-3135
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• 2015

Owing to the increase in consumption of multimedia content and the improvement of device capacity, user demand for high-quality content has increased. However, it is difficult to transport such large amounts of content over the existing broadcasting network with limited bandwidth. To provide high-definition broadcasting, some studies suggest methods of transporting multimedia over heterogeneous networks after encoding content hierarchically. MPEG Media Transport (MMT), standardized by Moving Picture Experts Group (MPEG), is a solution that enables large-volume media transport over heterogeneous networks such as digital broadcasting networks and packet-switched networks. In the case of delivering a scalable encoded video over different networks, synchronization of each stream is an important issue. MMT defines a synchronization scheme, but does not contain sufficient functions to implement it. In this paper, we propose a synchronization scheme for media streams that are encoded hierarchically, divided into layers, and transported over heterogeneous networks. We implement our scheme using MMT and HTTP, and experimented using three encapsulated video streams with different durations. As a result, we show that the proposed scheme can reduce the waiting time to display high-quality video, relative to Dynamic Adaptive Streaming over HTTP-Scalable Video Coding (DASH-SVC) by requesting segments of enhanced layers after calculating the transmission time. Additionally, we find out that the selection of durations have a relation to the characteristics of the video.

• Journal of Broadcast Engineering
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• v.16 no.5
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• pp.722-737
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• 2011

In this paper, we propose an estimation of network jitter that occurs when transmitting TCP packets containing MPEG-2 TS in progressive streaming service over wired or wireless Internet networks. Based on the estimated network jitter size, we can calculate required de-jitter buffering time to absorb the network jitter at the receiver side. For this purpose, by exploiting the PCR timestamp existing in the TS packet header, we create a new timestamp information that is marked in the optional field of TCP packet header to estimate the network jitter. By using the proposed de-jitter buffering scheme, it is possible to employ the conventional T-STD buffer model without any modification in the progressive streaming service over IP networks. The proposed method can be applicable to the recently developed international standard, MPEG DASH (dynamic adaptive streaming over HTTP) technology.

• The Journal of The Korea Institute of Intelligent Transport Systems
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• v.2 no.2 s.3
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• pp.43-54
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• 2003

As the demand for ubiquitous mobile wireless Internet grows, vehicles are receiving a lot of attention as new networking platforms. The demand for 4G all-IP networks encourages vehicle networks to be connected using IPv6. By means of network mobility (NEMO) support, we can connect sensors, controllers, local ,servers as well as passengers' devices of a vehicle to the Internet through a mobile router. The mobile router provides the connectivity to the Internet and mobility transparency for the rest of the mobile nodes of an in-vehicle nv6 network. So, it is .important for the mobile router to assure reliable connection and a sufficient data rate for the group of nodes behind it. To provide reliability, this paper proposes an adaptive multihoming architecture of multiple mobile routers. Proposed architecture makes use of different mobility characteristics of different vehicles. Simulation results with different configurations show that the proposed architecture increases session preservation thus increases reliability and reduces packet loss. We also show that the proposed architecture is adaptive to heterogeneous access environment which provide different access coverage areas and data rates. The result shows that our architecture achieves sufficient data rates as well as session preservation.

• Journal of Broadcast Engineering
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• v.24 no.2
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• pp.251-263
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• 2019

Recently the huge demand for Internet media streaming has dramatically increased the cost of the CDN (Content Delivery Network) and the need for a means to reduce it is increasing day by day. In this situation, a P2P-assisted DASH technology has recently emerged, which uses P2P (Peer-to-Peer) communications based on WebRTC (Web Real-Time Communication) standards to reduce the CDN cost. This paper proposes an algorithm that can significantly improve CDN cost savings in this technology by selecting peers based on the transport history. Also we implemented this algorithm in an experimental system and, after setting experimental conditions that emulate the actual mobile network environment, we measured the performance of the experimental system. As a result, we demonstrated that the proposed algorithm can achieve higher CDN cost savings compared to the conventional algorithm where peers are selected at random.

• Journal of Satellite, Information and Communications
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• v.12 no.4
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• pp.131-140
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• 2017

Multipath TCP (MPTCP) is a transport layer protocol that simultaneously transmits data using multiple interfaces. MPTCP is superior to existing TCP in network environment with homogeneous subflows, but it shows worse performance compared to existing TCP in network environment with bufferbloat. If a bufferbloat occurs in one of the MPTCP multipaths, the packet will not arrive at the MPTCP receive buffer due to a sudden increase in delay time, resulting in a HoL blocking phenomenon. It makes the receive window of the other path to be zero. In this paper, we apply Adaptive Random Early Detection (ARED), Controlled Delay (CoDel) and Proportional Integral Controller Enhanced (PIE) among the proposed Active Queue Management (AQM) to limit the delay of bufferbloat path. Experiments were conducted to improve the performance of MPTCP in heterogeneous networks. In order to carry out the experiment, we constructed a Linux-based testbed and compared the MPTCP performance with that of the existing droptail.

• Journal of Internet Computing and Services
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• v.17 no.6
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• pp.9-15
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• 2016

In this paper, we propose a smart HLS(HTTP Live Stream) platform with dynamic single buffering for the best transmission of adaptive video bit-rates. This smart HLS can optimizes the channel transition zapping-time with the monitoring of bandwidth between HLS server and OTT(Over The Top) client. This platform is designed through the control of video stream due to proper multi-bitrates and bandwidths. This proposed OTT can decode the live and VOD(Video On Demand) videos with the buffering of optimumal bitrate. And, the HLS can be cooperated with a smart OTT, and segmented for the m3u8 files of H.265 MPEG-2 TS(Transport Stream) videos. As a resullt, this single buffer based smart OTT can transmit optimal videos with the maximum data buffering according to the adaptive bit-rate depending on the network bandwidth efficiency and the decoded VOD video, too.

• Journal of Satellite, Information and Communications
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• v.8 no.1
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• pp.1-7
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• 2013

Currently satellite communication systems usually use the ACM(Adaptive Coding and Modulation) to extend the link availability and to increase the bandwidth efficiency. However, when ACM system is used for satellite communications, we should carefully consider TCP congestion control to avoid network congestions. Because MODCODs in ACM are changed to make a packet more robust according to satellite wireless link conditions, bandwidth of satellite forward link is also changed. Whereas TCP has a severe problem to control the congestion window for the changed bandwidth, then packet overflow can be experienced at MAC or PHY interface buffers. This is a reason that TCP in transport layer does not recognize a change of bandwidth capability form MAC or PHY layer. To overcome this problem, we propose the adaptive congestion control scheme of TCP for supporting ACM in Satellite PEP (Performance Enhancing Proxy) systems. Simulation results by using ns-2 show that our proposed scheme can be efficiently adapted to the changed bandwidth and TCP congestion window size, and can be useful to improve TCP performance.