• KSII Transactions on Internet and Information Systems (TIIS)
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• v.11 no.1
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• pp.414-435
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• 2017

As network adaptive streaming technology becomes increasingly common, transport protocol also becomes important in guaranteeing the quality of multimedia streaming. At the same time, because of the appearance of high-quality video such as Ultra High Definition (UHD), preventing buffering as well as preserving high quality while deploying a streaming service becomes important. The Internet Engineering Task Force recently published Multipath TCP (MPTCP). MPTCP improves the maximum transmission rate by simultaneously transmitting data over different paths with multiple TCP subflows. However, MPTCP cannot preserve high quality, because the MPTCP subflows slowly increase the transmission rate, and upon detecting a packet loss, drastically halve the transmission rate. In this paper, we propose a new multipath congestion control scheme for high-quality multimedia streaming. The proposed scheme preserves high quality of video by adaptively adjusting the increasing parameter of subflows according to the network status. The proposed scheme also increases network efficiency by providing load balancing and stability, and by supporting fairness with single-flow congestion control schemes.

• Journal of the Korean Institute of Telematics and Electronics A
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• v.33A no.3
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• pp.47-54
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• 1996

In this paper we propose a new timing recovery method by means of utilizing service data filling rate instead of timing information of transmitter. A proposed algorithm controls the phase locked loop in the opposite direction ot data filling rate of FIFO in receiver, and it is based on the fact that average of cell jitters is zero. The proposed method is simple compared with timing information method of transmitter. It can be utilized for timing recovery in synchronous digital hierarchy as well as in plesiochronous digial hierarchy without common reference clocks in end-to-end erminals. We implement the interactive video communication system and test the proposed algorithm. As a result, we hav econfirmed that it yields good perfomrnces in terms of jitters characteristics and hardware complexity.

• Convergence Security Journal
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• v.7 no.2
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• pp.17-25
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• 2007

In this paper, an algorithm that provisions proportional differentiation of packet delays is proposed with an objective for enhancing quality of service (QoS) in future packet networks. It features an adaptive scheme that adjusts the target delay every time slot to compensate the deviation from the target delay which is caused by the prediction error on the traffic to be arrived in the next time slot. It predicts the traffic to be arrived at the beginning of a time slot and measures the actual arrived traffic at the end of the time slot. The difference between them is utilized to the delay control operation for the next time slot to offset it. As it compensates the prediction error continuously, it shows superior adaptability to the bursty traffic as well as the exponential rate traffic. It is demonstrated through simulations that the algorithm meets the quantitative delay bounds and shows superiority to the traffic fluctuation in comparison with the conventional non-adaptive mechanism. The algorithm is implemented with VHDL on a Xilinx Spartan XC3S1500 FPGA and the performance is verified under the test board based on the XPC860P CPU.

• Journal of KIISE
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• v.42 no.8
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• pp.1039-1048
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• 2015

Recently, the web-based adaptive streaming service, DASH (Dynamic Adaptive Streaming over HTTP), is receiving more attention. However, existing network-based and buffer-based video quality control schemes in DASH environments make oscillation of segment throughput, causing degradation of the quality of experience (QoE) with frequent quality changes and playback interruptions because these schemes do not consider the content characteristics. In this paper, we propose a C-DASH (Content Characteristics based Dynamic Adaptive Streaming over HTTP) scheme in order to improve the QoE in DASH environments. The C-DASH scheme performs seamless and smooth quality control based on the segment throughput, buffer status, and segment size of the content. Based on simulation results, it is confirmed that the C-DASH scheme can improve the QoE, when compared with the existing quality control schemes.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.7 no.4
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• pp.632-650
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• 2013

The paper considers a spectrum sharing cognitive radio (CR) network coexisting with a primary network under the average interference power constraint. In particular, the secondary user (SU) is assumed to carry delay-sensitive services and thus shall satisfy a given delay quality-of-service (QoS) constraint. The secondary receiver is also assumed to be equipped with multiple antennas to perform maximal ratio combining (MRC) to enhance SU performance. We investigate the effective capacity of the SU with MRC diversity under aforementioned constraints in Nakagami fading environments. Particularly, we derive the optimal power allocation to achieve the maximum effective capacity of the SU, and further derive the effective capacity in closed-form. In addition, we further obtain the closed-form expressions for the effective capacities under three widely used power and rate adaptive transmission schemes, namely, optimal simultaneous power and rate adaptation (opra), truncated channel inversion with fixed rate (tifr) and channel inversion with fixed rate without truncation (cifr). Numerical results supported by simulations are presented to consolidate our studies. The impacts on the effective capacity of various system parameters such as the number of antennas, the average interference power constraint and the delay QoS constraint are investigated in detail. It is shown that MRC diversity can significantly improve the effective capacity of the SU especially for cifr transmission scheme.

• The Transactions of the Korea Information Processing Society
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• v.7 no.12
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• pp.3923-3932
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• 2000

In mobile network, wireless links may be overloaded due to a high concentratioJl of mobile hosts in a given cell. As a result, the required QOS(Quality of Service) cannot be provided to all flows in that cell. So, there exists adaptive applications which can operate with various bandwidth levels. When a wireless link is overloaded, it can adapt by reducing the bandwidths of flows, is referred to as rate adaptation. In this paper, we analyze the performance of existing rate adaptation methods by simulations. The result of simulation experiments shows that average-fair adaptation scheme has very low network overhead but high bandwidth variance of flows, which causes the worse quality of service. We propose a rate adaptation method refined by combining fair adaptation and average-fair adaptation, and shows the refined method can provide lower bandwidth variance of flows by simulations.

• Journal of KIISE:Computer Systems and Theory
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• v.34 no.10
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• pp.529-538
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• 2007

Fair Queueing algorithms based on Generalized Processor Sharing (GPS) not only guarantee sessions with service rate and delay, but also provide sessions with instantaneous fair sharing. This fair sharing distributes server capacity to currently backlogged sessions in proportion to their weights without regard to the amount of service that the sessions received in the past. From a long-term perspective, the instantaneous fair sharing leads to a different quality of service in terms of delay and bandwidth to sessions with the same weight depending on their traffic pattern. To minimize such long-term unfairness, we propose a delay-bandwidth normalization model that defines the concept of value of service (VoS) from the aspect of both delay and bandwidth. A model and a packet-by-packet scheduling algorithm are proposed to realize the VoS concept. Performance comparisons between the proposed algorithm and algorithms based on fair queueing and service curve show that the proposed algorithm provides better long-term fairness among sessions and that is more adaptive to dynamic traffic characteristics without compromising its service rate and delay guarantees.

• Proceedings of the IEEK Conference
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• 2003.07d
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• pp.1661-1664
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• 2003

The transcoding is a method which converts a multimedia stream to an adaptive format, such as frame rate and resolution, that can be played in the mobile devices. Due to the heavy consumption of CPU power for transcoding, the server in charge cannot supply all streams requested from clients if it is already occupied by several streams. The load balancing among proxies is a common approach for solving such problem. In addition to collaborative works for load balancing, the mobility must be taken into account in the mobile environments. Thus, the wired-network based algorithm is not suitable for the streams service in the mobile environments. In this paper, we propose an alternative scheme for cooperative proxy architecture that considers the case in which mobile devices are moving over other zones.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.12B
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• pp.735-742
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• 2007

We propose an effective packet marking algorithm for video streaming in DiffServ network. Because legacy packet markers such as srTCM(single rate three color marker) cannot distinguish the importance of packet, these markers can cause quality degradation of streaming during the network congestion period. Recently proposed TMS (Two Marker System) [4] shows effectiveness in such scenario that video streaming service is struggling with other types of service traffic. However, if many video streaming services co-exist in DiffServ network and result in competition among themselves, then both legacy packet markers and even TMS cannot prevent drastic streaming quality degradation. To cope with this, we suggest A-TCPM (Adaptive time sliding window Three Color Marker) algorithm. In this algorithm, an A-TCPM module decides the color of a racket based upon the probability which is lead by current channel status and frame importance ratio. Simulation results show that proposed A-TCPM algorithm can enhance streaming service quality especially when overbooked video streaming sessions struggle with themselves.

• Journal of KIISE:Information Networking
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• v.32 no.3
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• pp.404-414
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• 2005

In the packet data network systems beyond 3G, the service quality of current users is affected by traffic load due to the packet burstiness. There also exists a hot cell problem, a well-known problem of cellular systems, caused by traffic centraliBation. Hot cell problem is one of the major reasons of degrading system performance because hot cell increases the call drop rate without fully utilization of system resource. Therefore, it is very important to distribute the traffic on the several neighboring cells so that system uses its resource effectively and maintains the quality of service. In this paper, we propose the adaptive handoff algorithms for distributing traffic in the packet data network systems. In addition, we propose a new load estimation method with MAC state diagram suitable for packet data network systems. Through the simulation results, we could find that proposed algorithm is able to improve efficiency of system resource and to assure the service quality of users through traffic distribution.