• Journal of the Institute of Electronics Engineers of Korea TC
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• v.38 no.10
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• pp.24-33
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• 2001

Delay, delay variation and packet loss rate are principal QoS(Quality of Service) elements of packet communication. This paper proposes a new buffer allocation mechanism to improve the packet loss performance in such a situation that multiple logical buffers share a single physical memory buffer. In the proposed buffer allocation mechanism, the movement of dynamic threshold follows a curved track instead of a straight line which is used in the DT(dynamic threshold) mechanism. In order evaluate the effectiveness of the proposed mechanism, it is compared with the existing previously proposed mechanisms in several aspects including NC(no control), ST(Static Threshold) and DT mechanisms.

• ETRI Journal
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• v.26 no.6
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• pp.653-656
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• 2004

This paper presents an effective algorithm for inserting an MPEG-2 transport stream (TS) into a Digital Audio Broadcasting (DAB) ensemble without any bandwidth waste in a Digital Multimedia Broadcasting (DMB) transmission system. The key technologies of this algorithm include packet rate control and program clock reference correction, which are important for TS processing. The proposed algorithms are applied to the various DMB transmission systems based on Eureka-147, and the performance of the proposed algorithm is confirmed through the experimental DMB broadcasting.

• Journal of Internet of Things and Convergence
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• v.1 no.1
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• pp.1-4
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• 2015

This paper aims to present the mathematical model to find the mean web object transfer latency in the slow-start phase of TCP congestion control mechanism, which is one of the main control techniques of Internet. Mean latency is an important service quality measure of end-user in the network. The application area of the proposed latency model is the narrowband environment including multi-hop wireless network and Internet of Things(IoT), where packet loss occurs in the slow-start phase only due to small window. The model finds the latency considering initial window size and the packet loss rate. Our model shows that for a given packet loss rate, round trip time and initial window size mainly affect the mean web object transfer latency. The proposed model can be applied to estimate the mean response time that end user requires in the IoT service applications.

• Journal of Multimedia Information System
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• v.3 no.4
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• pp.141-148
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• 2016

This paper proposes a medium access control (MAC) protocol for multiple-input multiple-output (MIMO) ad-hoc networks. Multiple-packet receptions in MIMO systems have attracted as a key technique to achieve a high transmission rate. In the conventional protocols for multiple-packet receptions, timing offsets among multiple-frame transmissions cause frame collisions induced by hidden nodes, which degrades network performance. In the proposed protocol, transmission synchronization among hidden nodes can be achieved by applying pulse/tone exchanges. By applying the pulse/tone exchanges, multiple-packet receptions among hidden nodes can be achieved, which enhances network throughputs compared with the conventional protocol. Simulation results show effectiveness of the proposed protocol.

• Journal of Advanced Navigation Technology
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• v.18 no.1
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• pp.67-73
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• 2014

Due to the development of the communication technology and the high-performance mobile device, the number of video streaming service user has been increasing. Video streaming services in the restricted network environments need error control to support users for the Ultra High Definition(UHD) video delivery. Packet-level FEC method was used to control the error for delay-sensitive video service, but it has a defect in huge occupation of network resources by redundant packets. In this paper, we proposed a packet-level FEC algorithm which maintains the video quality by previous FEC methods and also reduces the occupation of network resource in the restricted network environments. The proposed algorithm controls the redundant rate with using the delivery characteristic of the video coding technology, and improves the efficiency of network resources by reducing the traffic by around 33% in comparison with the previous FEC methods.

• Journal of KIISE:Computing Practices and Letters
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• v.15 no.2
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• pp.114-124
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• 2009

In this paper, we propose a practical TCP friendly rate control scheme that considers the minimum channel bandwidth of the network when transporting SVC (scalable video coding) video over IP netowrks such as Internet. RTP and RTCP is mainly designed for use with UDP (User Datagram Protocol) for real-time video transport over the Internet. TCP-friendly rate control was proposed to satisfy the demands of multimedia applications while being reasonably fair when competing for bandwidth with conventional TCP applications. However the rate control model of the conventional TCP-friendly rate control scheme does not consider the minimum channel bandwidth of the network. Thus the estimated channel bandwidth by the conventional rate control model might be quite different from the real channel bandwidth when the packet loss ratio of the network is very large. In this paper, we propose a modified TCP-friendly rate control scheme that considers the minimum channel bandwidth of the network. Based on the modified TCP-friendly rate control, we assign the minimum channel bandwidth to the base layer bitstream of SVC video, and remaining available bandwidth is allocated to the enhancement layer of SVC video for the TCP friendly scalable video transmission. It is shown by simulations that the modified TCP-friendly rate control scheme can be effectively used for a wider range of controlled bit rates depending on the packet loss ratio than the conventional TCP-friendly control scheme. Furthermore, the effectiveness of the proposed scheme in terms of objective video quality is proved by comparing PSNR performance with the conventional scheme.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.39 no.2
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• pp.59-68
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• 2002

In this paper, we compared and analyzed two new models of cyclic packet dropping algorithm, Adaptive Cyclic Packet Dropping algorithm (ACPD), and Non-adaptive Cyclic Packet Dropping algorithm (NCPD) with RIO. The ACPD algorithm drops adaptively packets for the congestion control, as predicting traffic pattern between each cycle. Therefore the ACPD algorithm makes up for the drawback of RIO algorithm and minimizes the wastes of the bandwidth being capable of predicting in the NCPD algorithm. We modelled two cyclic packet drop algorithms and executed a simulation and analyzed the throughput and packet drop rate based on Sending Priority changing dynamically depending on network traffic. In this algorithm, applying the strict drop precedence policy, we get better performance on priority levels. The results show that two new algorithms may provide more efficient and stricter drop precedence policy as compared to RIO independent of traffic load. The ACPD algorithm can provide better performance on priority levels and keep stricter drop policy than other algorithms.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.5 no.9
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• pp.1492-1512
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• 2011

Due to multiple hops, mobility and time-varying channel, supporting delay sensitive real-time traffic in wireless local area network-based (WLAN) mesh networks is a challenging task. In particular for real-time traffic subject to medium access control (MAC) layer control overhead, such as preamble, carrier sense waiting time and the random backoff period, the performance of real-time flows will be degraded greatly. In order to support real-time traffic, an efficient adaptive packet scheduling (APS) scheme is proposed, which aims to improve the system performance by guaranteeing inter-class, intra-class service differentiation and adaptively adjusting the packet length. APS classifies incoming packets by the IEEE 802.11e access class and then queued into a suitable buffer queue. APS employs strict priority service discipline for resource allocation among different service classes to achieve inter-class fairness. By estimating the received signal to interference plus noise ratio (SINR) per bit and current link condition, APS is able to calculate the optimized packet length with bi-dimensional markov MAC model to improve system performance. To achieve the fairness of intra-class, APS also takes maximum tolerable packet delay, transmission requests, and average allocation transmission into consideration to allocate transmission opportunity to the corresponding traffic. Detailed simulation results and comparison with IEEE 802.11e enhanced distributed channel access (EDCA) scheme show that the proposed APS scheme is able to effectively provide inter-class and intra-class differentiate services and improve QoS for real-time traffic in terms of throughput, end-to-end delay, packet loss rate and fairness.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.38 no.11
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• pp.1-8
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• 2001

Adaptive and Strict Packet Dropping mechanism we proposed in this paper drops adaptively packets for the congestion control, as predicting traffic pattern between each cycle. Therefore the proposed mechanism makes up for the drawback of RIO mechanism and minimizes the wastes of the bandwidth being capable of predicting in Dynamic and Strict Packet Dropping (DSPD) mechanism. And we executed a simulation and analyzed the throughput and packet drop rate based on the Sending Drop Precedence changing dynamically depending on the network traffic and compared RIO and the DSPD. The results show that the proposed mechanism provides better performance on drop precedence levels and stricter drop precedence policy for AF class than RIO and the DSPD mechanism.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.13 no.7
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• pp.3447-3469
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• 2019

It is well-known that the cooperative communication and error control technology can improve the network performance, but most existing cooperative MAC protocols have not focused on how to cope with the contention process caused by cooperation and how to reduce the bad influence of channel packet error rate on the system performance. Inspired by this, this paper first modifies and improves the basic rules of the IEEE 802.11 Medium Access Control (MAC) protocol to optimize the contention among the multi-relay in a cooperative ARQ scheme. Secondly, a hybrid ARQ protocol with soft combining is adopted to make full use of the effective information in the error data packet and hence improve the ability of the receiver to decode the data packet correctly. The closed expressions of network performance including throughput and average packet transmission delay in a saturated network are then analyzed and derived by establishing a dedicated two-dimensional Markov model and solving its steady-state distribution. Finally, the performance evaluation and superiority of the proposed protocol are validated in different representative study cases through MATLAB simulations.