• The Journal of the Korea institute of electronic communication sciences
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• v.8 no.1
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• pp.71-76
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• 2013

In this paper, end-to-end performances for some VoIP traffics are measured and analyzed over MANET(Mobile Ad-hoc Networks) with packet aggregation transmission under blackhole attacks. A VoIP simulator based on NS-2 is used for performance measure. In the simulation, MOS, call connection ratio, network delay and packet loss rate are measured for end-to-end transmission performance. With analysis of measured data, characteristics of each traffics for performance parameters of packet aggregation transmission is showed on MANET under blackhole attack, As a results of this paper, some considerations be required for packet aggregation transmission are suggested to implement VoIP services on MANET under blackhole attacks.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.14 no.2
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• pp.473-494
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• 2020

To provide a guaranteed Quality of Service (QoS) to real-time traffic in High-Speed Downlink Packet Access (HSDPA) core network, we proposed an enhanced mechanism. For an enhanced QoS, a Class-Based Low Latency Fair Queueing (CBLLFQ) packet scheduling algorithm is introduced in this work. Packet classification, metering, queuing, and scheduling using differentiated services (DiffServ) environment was the points in focus. To classify different types of real-time voice and multimedia traffic, the QoS provisioning mechanisms use different DiffServ code points (DSCP).The proposed algorithm is based on traffic classes which efficiently require the guarantee of services and specified level of fairness. In CBLLFQ, a mapping criterion and an efficient queuing mechanism for voice, video and other traffic in separate queues are used. It is proved, that the algorithm enhances the throughput and fairness along with a reduction in the delay and packet loss factors for smooth and worst traffic conditions. The results calculated through simulation show that the proposed calculations meet the QoS prerequisites efficiently.

• ETRI Journal
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• v.27 no.6
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• pp.777-787
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• 2005

In this paper, we propose an urgency- and efficiency-based wireless packet scheduling (UEPS) algorithm that is able to schedule real-time (RT) and non-real-time (NRT) traffics at the same time while supporting multiple users simultaneously at any given scheduling time instant. The UEPS algorithm is designed to support wireless downlink packet scheduling in an orthogonal frequency division multiple access (OFDMA) system, which is a strong candidate as a wireless access method for the next generation of wireless communications. The UEPS algorithm uses the time-utility function as a scheduling urgency factor and the relative status of the current channel to the average channel status as an efficiency indicator of radio resource usage. The design goal of the UEPS algorithm is to maximize throughput of NRT traffics while satisfying quality-of-service (QoS) requirements of RT traffics. The simulation study shows that the UEPS algorithm is able to give better throughput performance than existing wireless packet scheduling algorithms such as proportional fair (PF) and modified-largest weighted delay first (M-LWDF), while satisfying the QoS requirements of RT traffics such as average delay and packet loss rate under various traffic loads.

• The Journal of the Korea institute of electronic communication sciences
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• v.10 no.2
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• pp.289-296
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• 2015

Wireless multimedia sensor network senses and transfers mass multimedia data. Also, it is sensitive to latency. This thesis proposes a routing technique based on traffic priority in order to improve the network efficiency by minimizing latency. In addition, it proposes a congestion control mechanism that uses packet service time, packet inter-arrival time, buffer usage, etc. In this thesis, we verified the reduction of packet latency in accordance with the quality level of packet as a result of the performance analysis through the simulation method. Also, we verified that the proposed mechanism maintained a reliable network state by preventing packet loss due to network overload.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.6 no.12
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• pp.3257-3274
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• 2012

Broadcast routing has become an important research field for vehicular ad-hoc networks (VANETs) recently. However, the packet delivery rate is generally low in existing VANET broadcast routing protocols. Therefore, the design of an appropriate broadcast protocol based on the features of VANET has become a crucial part of the development of VANET. This paper analyzes the disadvantage of existing broadcast routing protocols in VANETs, and proposes an improved algorithm (namely ODAM-C) based on the ODAM (Optimized Dissemination of Alarm Messages) protocol. The ODAM-C algorithm improves the packet delivery rate by two mechanisms based on the forwarding features of ODAM. The first distance-based mechanism reduces the possibility of packet loss by considering the angles between source nodes, forwarding nodes and receiving nodes. The second mechanism increases the redundancy of forwarding nodes to guarantee the packet success delivery ratio. We show by analysis and simulations that the proposed algorithm can improve packet delivery rate for vehicular networks compared against two widely-used existing protocols.

• 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 Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.7B
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• pp.667-674
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• 2004

The performance of transmission control protocol (TCP) is largely dependent upon its loss recovery. Therefore, it is a very important issue whether the packet losses may be recovered without retransmission timeout (RTO) or not. Although TCP SACK can recover multiple packet losses in a window, it cannot avoid RTO if a retransmitted packet is lost again. In order to alleviate this problem, we propose a simple change to TCP SACK, which is called TCP SACK+ in simple. We use a stochastic model to evaluate the performance of TCP SACK+, and compare it with TCP SACK. Numerical results evaluated by simulations show that SACK+ can improve the loss recovery of TCP SACK significantly in presence of random losses.

• Journal of KIISE:Information Networking
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• v.29 no.4
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• pp.437-446
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• 2002

In order to reduce the increasing packet loss rates caused by an exponential increase in network traffic, the IETF (Internet Engineering Task Force) is considering the deployment of active queue management techniques such as RED (Random Early Detection). While active queue management in routers and gateways can potentially reduce packet loss rates in the Internet, this paper has demonstrated the inherent weakness of current techniques and shows that they are ineffective in preventing high loss rates. The inherent problem with these queue management algorithms is that they all use static parameter setting. So, in case where these parameters do not match the requirement of the network load, the performance of these algorithms can approach that of a traditional Drop-tail. In this paper, in order to solve this problem, a new active queue management algorithm called ARED (Active RED) is proposed. ARED computes the parameter based on our heuristic method. This algorithm can effectively reduce packet loss while maintaining high link utilizations.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.14 no.4
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• pp.247-256
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• 2021

In this paper, we propose a packet loss concealment (PLC) algorithm using pitch harmonic motion prediction and adaptive signal amplitude prediction and. The spectral motion prediction method divides the spectral motion of the previous usable frame into predetermined sub-bands to predict and restore the motion of the lost signal. In the proposed algorithm, the speech signal is classified into voiced and unvoiced sounds. In the case of voiced sounds, it is further divided into pitch harmonics using the pitch frequency to predict and restore the pitch harmonic motion of the lost frame, and for the unvoiced sound, the lost frame is restored using the spectral motion prediction method. When the continuous loss of speech frames occurs, a method of adjusting the gain using the least mean square (LMS) predictor is proposed. The performance of the proposed algorithm was evaluated through the objective evaluation method, PESQ (Perceptual Evaluation of Speech Quality) and was showed MOS 0.1 improvement over the conventional method.

• The KIPS Transactions:PartC
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• v.11C no.3
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• pp.361-370
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• 2004

All active nudes which have no physically direct connection with each other in If network must be able to compose and manage network topology Informations. Besides one active program can be performed by the active nodes when every active packet for this program is transmitted without any loss of packets. Also the active packets should be transmitted effectively to minimize the transmission delay and securely from threatens. In this thesis, the discovery scheme of active nodes is adapted for active nodes in IP networks to compose and manage the topology information. The scheme for the efficient, reliable and secure transmission of active packets is also proposed. The sequence number is assigned to every active packet. If a receiver detects the loss of active packet checking the sequence number, the receiver requests the retransmission of the lost packet to the previous active node. kiter receiving an active packet and adapting security and reliability schemes, intermediate active nodes not only copy and send the Packet Instantly but also apply some suity mechanisms to it. And the active packet transmission engine is proposed to provide these transmission schemes The simulation of the adapted active node discovery scheme and the proposed active packet transmission engine is performed. The simulation results show that the adapted active node discovery scheme is efficient and the proposed active engine has the low latency and the high performance.