• The Journal of Korean Institute of Communications and Information Sciences
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• v.38B no.7
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• pp.553-563
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• 2013

When the IEEE 802.11 wireless LAN-based broadcasting services, we use broadcast packets to broadcast multimedia contents to a large number of users using limited wireless resources. However, broadcast transmission is difficult to recover the loss packets compared with unicast transmission. Therefore, analysis of packet loss characteristics is required to perform efficient packet recovery. The packet loss in wireless transmissions is often bursty with high loss data rate. Even if loss patterns have the same average packet loss, they are different in the recovery rate of random loss and burst loss depending on the nature. Therefore, the analysis and research of the nature of the loss are needed to recover loss packets considering bursty characteristics. In this paper, we experimented Wi-Fi broadcast transmission according to transmission rate and analyzed bursty characteristics of loss patterns using 4-state markov model.

• Journal of Korea Multimedia Society
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• v.20 no.7
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• pp.1030-1037
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• 2017

Traffic flows of real-time multimedia services such as Internet phone and IPTV are bounded on the end-to-end delay. Packets violating their delay limits will be dropped at a router because of not useful anymore. Service providers promise the quality of their providing services in terms of SLA(Service Level Agreement), and they, especially, have to guarantee the packet loss rates listed in the SLA. This paper is about a method to guarantee the required packet loss rate of each traffic flow keeping the high network resource utilization as well. In details, it assures the required loss rate by adjusting adaptively the timestamps of packets of the flow according to the difference between the required and measured loss rates in the lossy Weighted Fair Queuing(WFQ) scheduler. The proposed method is expected to be highly applicable because of assuring the packet loss rates regardless of the fluctuations of offered traffic load in terms of quality of services and statistical characteristics.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.13 no.12
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• pp.5948-5971
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• 2019

Estimating loss rates with the packet traces captured from some point in the middle of the network has received much attention within the research community. Meanwhile, existing intermediate-point methods like [1] require the capturing system to capture all the TCP traffic that crosses the border of an access network (typically Gigabit network) destined to or coming from the Internet. However, limited to the performance of current hardware and software, capturing network traffic in a Gigabit environment is still a challenging task. The uncaptured packets will affect the total number of captured packets and the estimated number of packet losses, which eventually affects the accuracy of the estimated loss rate. Therefore, to obtain more accurate loss rate, a method of strengthening packet loss measurement from the network intermediate point is proposed in this paper. Through constructing a series of heuristic rules and leveraging the binomial distribution principle, the proposed method realizes the compensation for the estimated loss rate. Also, experiment results show that although there is no increase in the proportion of accurate estimates, the compensation makes the majority of estimates closer to the accurate ones.

• Journal of Internet Computing and Services
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• v.9 no.4
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• pp.1-10
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• 2008

In this paper, a packet scheduling algorithm, called the modified PLFS, is proposed for real-time traffic in IEEE 802.22 WRAN systems. The modified PLFS(Packet Loss Fair Scheduling) algorithm utilizes not only the delay of the Head of Line(HOL) packets in buffer of each user but also the amount of expected loss packets in the next-next frame when a service will not be given in the next frame. The performances of the modified PLFS are compared with those of PLFS and M-LWDF in terms of the average packet loss rate and throughput. The simulation results show that the proposed scheduling algorithm performs much better than the PLFS and M-LWDF algorithms.

• Journal of Advanced Navigation Technology
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• v.14 no.3
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• pp.358-366
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• 2010

In the wireless network the congestion and delay occurs mainly when there are too many packets for the network to process or the sender transmits more packets than the receiver can accept. The congestion and delay is the reason of packet loss which degrades the performance of multimedia streaming. This paper proposes a novel transmission rate monitoring-based optimization mechanism to optimize packet loss and to improve QoS. The proposed scheme is based on the trade-off relationship between transmission rate monitoring and overhead monitoring. For this purpose this paper processes a source rate control-based optimization which optimizes congestion and delay. Performance evaluated RED, TFRC, and the proposed mechanism. The simulation results show that the proposed mechanism is more efficient than REC(Random Early Detection) mechanism and TFRC(TCP-friendly Rate Control) mechanism in packet loss rate, throughput rate, and average response rate.

• The KIPS Transactions:PartC
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• v.11C no.7 s.96
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• pp.931-936
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• 2004

We need to have an adaptive TCP protocol that can be tolerable on wireless network environement. TCP Westwood for use in the environe-ment that have a very high loss rate like a sattelite was proposed by modifying the existing bulk retransmission protocol. Bulk retransmission mechanism shows a highly enhanced performance on networks that have a very high loss rate but are prone to bursty loss networks. Also, it can exprience less performance on low late transmission environement. This paper proposes Adaptive Bulk Retransmission Mechanism that adjusts the number of bulk retransmitted packets based on the network conditions. The proposed mechanism was evaluated by using NS-2.

• Proceedings of the IEEK Conference
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• 2000.11a
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• pp.37-40
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• 2000

Multi Protocol Label Switching (MPLS) is a high performance method for forwarding packets (frames) through a network. It enables routers at the edge of a network to apply simple labels to packets (frames). we use MPLS in the core network for internet. MPLS provide high speed switching and traffic engineering in MPLS domain but at the Label Edge Router(LER) there is frequently cell discarding via congestion and buffer management method. It is one of the most important reasons retransmission and congestion. In this paper we propose advanced LER scheme that provide less cell loss rate also efficient network infrastructure.

• The Transactions of The Korean Institute of Electrical Engineers
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• v.57 no.11
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• pp.2124-2126
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• 2008

In the wireless communication systems, the demand of multimedia services is also increased. Unlike typical data packets, realtime service such as VoIP packets have delay bound and low loss rate requirement. In this paper we propose a new scheduling algorithm that be able to allocate resources to the different kinds of services such as VoIP and data packet. The proposed algorithm considers both time delay and channel condition toe determine the priority. Simulation results show that the proposed algorithm works more efficiently than the conventional algorithms.

• Journal of IKEEE
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• v.22 no.4
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• pp.1147-1151
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• 2018

As personal users create and share lots of contents at any time and any places, new networking architecture such as content centric networking (CCN) has emerged. CCN utilizes content name as a packet identifier, not address. However, current CCN has a limitation for content source mobility management. The movement of content sources cause long delivery latency and long service disruption. To solve that, a hierarchical mobility management was was proposed. However, the hierarchical mobility management scheme has still the loss of interest packets and long handoff latency. So, this paper presents the hybrid hierarchical mobility management in mobile CCN environements to reduce both the loss rate of interest packets and the handoff latency. It is shown from the performance evaluations shows that the proposed scheme provides low loss rate of control message.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.27 no.7A
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• pp.711-723
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• 2002

A finer granular scalable (FGS) version of ISO/IEC MPEG-4 video streaming is investigated in this work with the prioritized stream delivery over loss-rate differentiated networks. Our proposed system is focused on the seamless integration of rate adaptation, prioritized packetization, and simplified differentiation for the MPEG-4 FGS video streaming. The proposed system consists of three key components: 1) rate adaptation with scalable source encoding, 2) content-aware prioritized packetization, and 3) loss-based differential forwarding. More specifically, a constant-quality rate adaptation is first achieved by optimally truncating the over-coded FGS stream based on the embedding rate-distortion (R-D) information (obtained from a piecewise linear R-D model). The rate-controlled video stream is then packetized and prioritized according to the loss impact of each packet. Prioritized packets are transmitted over the underlying network, where packets are subject to differentiated dropping and forwarding. By focusing on the end-to-end quality, we establish an effective working conditions for the proposed video streaming and the superior performance is verified by simulated MPEG-4 FGS video streaming.