• The KIPS Transactions:PartC
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• v.9C no.3
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• pp.375-384
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• 2002

In the current Internet, the QoS of interactive applications is hardly guaranteed because of variable bandwidth, packet loss and delay. Moreover, VoIP which is becoming an important part of the information infra-structure in these days, is susceptible to network packet loss and end-to-end delay. Therefore, it needs error control mechanisms in network level or application level. The FEC-based error control mechanisms are used for interactive audio application such as VoIP. The FEC sends a main information along with redundant information to recover the lost packets and adjusts redundant information depending on network conditions to reduce the bandwidth overhead. However, because most of the error control mechanisms do not consider end-to-end delay but packet loss rate, their performances are poor. In this paper, we propose a new error control algorithm, SCCRP, considering packet loss rate as well as end-to-end delay. Through experiments, we confirm that the SCCRP has a lower packet loss rate and a lower end-to-end delay after reconstruction.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.39B no.5
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• pp.296-303
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• 2014

Reference Broadcast Synchronization (RBS) is one of the most prominent synchronization protocols in wireless sensor nework. Given that the broadcasting medium is available, RBS can give very high accuracy of synchronization. However, RBS uses instantaneous synchronization and results in time discontinuity, which might cause serious faults in the distributed system. Also, RBS lacks packet loss tolerance, which brings about degraded performance in severe conditions of wireless channel. In this paper, the problem of time discontinuity in RBS is pointed out and the effect of packet loss on the performance of RBS is examined. Then, a continuous synchronization and a packet loss tolerance mechanism for RBS are proposed, and the result is verified through simulations.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.8 no.9
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• pp.3159-3176
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• 2014

This work presents an analysis of the buffer features of an access router, especially the size, the impact on delay and the packet loss rate. In particular, we study how these features can affect the Quality of Service (QoS) of multimedia applications when generating traffic bursts in local networks. First, we show how in a typical SME (Small and Medium Enterprise) network in which several multimedia flows (VoIP, videoconferencing and video surveillance) share access, the upgrade of the bandwidth of the internal network may cause the appearance of a significant amount of packet loss caused by buffer overflow. Secondly, the study shows that the bursty nature of the traffic in some applications traffic (video surveillance) may impair their QoS and that of other services (VoIP and videoconferencing), especially when a certain number of bursts overlap. Various tests have been developed with the aim of characterizing the problems that may appear when network capacity is increased in these scenarios. In some cases, especially when applications generating bursty traffic are present, increasing the network speed may lead to a deterioration in the quality. It has been found that the cause of this quality degradation is buffer overflow, which depends on the bandwidth relationship between the access and the internal networks. Besides, it has been necessary to describe the packet loss distribution by means of a histogram since, although most of the communications present good QoS results, a few of them have worse outcomes. Finally, in order to complete the study we present the MOS results for VoIP calculated from the delay and packet loss rate.

• Proceedings of the Korea Society for Simulation Conference
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• 1999.10a
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• pp.127-132
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• 1999

A mobile IP is proposed to support host mobility over the current Internet. One of the most important issues on the host mobility is location and routing schemes that allow mobile hosts to move effectively from one site to another. In a Mobile IP environment, frequent handoffs are likely to degrade the performance by minimizing the loss of datagrams during handoffs. The handoff scheme is using routing optimization and binding extension to improve the performance by minimizing the average transfer delay of messages and packet loss. Simulation details show the improvement of transport delays and packet loss rate.

• Journal of the Korean Institute of Telematics and Electronics S
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• v.34S no.8
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• pp.45-55
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• 1997

To design a ATM Switch which ahs advantages in high sped packet switching, it is essential to set multiple paths between input ports and output ports and to design a new packet transfer technique on that paths for decreasing Packet Loss by conflicts in internal Switch Plane. We propose new packet transfer method, Alternate Path Transfer Mechanism by Dynamic Bypass Transfer Method which can solve conflict problem in Banyan network easily. Proposed ATM Switch consists of Banyan networks, Input/Ouput Port, Bypass Link, and Bypass Link Controller. Packets caused conflicts in SEs have another chances of packet transfer over alternate switching planes by using this mechanism.

• Journal of KIISE:Information Networking
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• v.34 no.2
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• pp.110-119
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• 2007

Mobile IPv6 (MIPv6) enables a mobile node (MN) to maintain its connectivity with a correspondent node (CN) while changing its point of attachment. In MIPv6, packets sent from a CN to a MN during handover are lost. Several mechanisms including FMIPv6 and HMIPv6 have been proposed in order to minimize packet loss. However, such mechanisms still suffer from performance degradation due to not only packet loss but also out-of-sequence packets. In this paper, we propose I-FHMIPv6 to resolve packet loss as well as the out-of-sequence packet problem. In I-FHMIPv6, the flush message is newly defined in order to notify a home agent (HA) or CN of the fact that the binding cache entry of a MN is about to be updated. A MN receiving the flush message can know that there is no more packets transmitted via the previous route, which resolve the out-of-sequence packet problem. Moreover, with the proposed mechanism, we can minimize packet loss by integrating FMIPv6 and HMIPv6 efficiently. I-FHMIPv6 is evaluated by performing simulations, and the simulation results show that I-FHMIPv6 outperforms FMIPv6 and HMIPv6.

• Proceedings of the Korea Multimedia Society Conference
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• 2012.05a
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• pp.166-167
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• 2012

High packet loss rate and impatience of messages passing through optical switches are essential characteristics in Optical Burst Switching system equipped with Optic Delay Lines buffer, which have not been solved efficiently yet by current existing models. In order to capture both effects, this paper introduces an analytical model from the viewpoint of classical queuing theory with impatient customers. We then apply it to evaluate and compare two wavelength-sharing cases, (1) all delay lines share a common wavelength resource and (2) each wavelength is associated with a number of delay lines. Our numerical results suggest to implement the first case because of lower packet loss rate for a fairly broad range of traffic load.

• Proceedings of the KIEE Conference
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• 2006.04a
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• pp.105-107
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• 2006

As internet streaming data increase, transport protocol such as TCP, TGP-Friendly is important to study control transmission rate and share of Internet bandwidth. In this paper, we propose a TCP-Friendly protocol using Neural Network for media delivery over wired Internet which has various traffic size(PTFRC). PTFRC can effectively send streaming data when occur congestion and predict one-step ahead round trip time and packet loss rate. A multi-layer perceptron structure is used as the prediction model, and the Levenberg-Marquardt algorithm is used as a traning algorithm. The performance of the PTFRC was evaluated by the share of Bandwidth and packet loss rate with various protocols.

• Proceedings of the Korea Contents Association Conference
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• 2005.05a
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• pp.487-490
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• 2005

In this paper, proposed VoIP call quality Assessment model for Ubiquitous environment that apply the recency Effect and bust packet loss model. It is model that improved E-Model's problems for elaborate and reliable assessment. A new model makes the accurate VoIP call quality assessment possible by applying the burst packet loss and recency effect. Advanced E-model apply bust packet loss model potentialized elaborate and reliable assessment.

• Proceedings of the IEEK Conference
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• 2006.06a
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• pp.59-60
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• 2006

TCP applied to the wireless-wired integrated network is the one that was applied to the existing wired network. In the wireless-wired integrated network, both wireless and congestion loss can occur. When wireless packet losses occur, the congestion control of TCP causes performance degradation by reducing its transmission rate. In this paper, we propose the algorithm to distinguish the wireless packet loss from congestion packet loss using MIB of the 802.11 MAC which has been generally used recently in wireless links.