• Journal of Korea Multimedia Society
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• v.7 no.1
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• pp.54-63
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• 2004

This paper describes a hybrid QoS guarantee scheme for high quality audio streaming services on the Internet. The continuous playback of the audio data requires the isochronous transmission of the audio data packet through the Internet. In order to retain the QoS at the ultimate destination (client) as the same as servers provide, the transmission protocols should consider the error conditions such as packet loss, and out of order delivery. Generally, the protocols supporting the transmission of continuous media data do not try to recover the errors. The protocols are working somehow for the toll quality multimedia streaming services, but rot for the high quality streaming services, such as the DVD sound/music payback. The hybrid QoS guarantee scheme includes the three mechanisms to overcome the problem. The selective retransmission for the lost packet, the adaptive buffering at client-side, and the adaptive transmission rate at server-side are totally adopted to recover the packet loss with the minimal overhead, to prevent from the buffer starvation during the retransmission, and to maintain the isochronous transmission even after the retransmission. The experiments have shown good results for the high Quality audio streaming services on the Internet.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.6B
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• pp.396-405
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• 2005

Snoop protocol is one of the efficient schemes to compensate TCP packet loss and enhance TCP throughput in wired-cum-wireless networks. However, Snoop protocol has a problem; it cannot perform local retransmission efficiently under the bursty-error prone wireless link. In this paper, we propose Enhanced Snoop(E-Snoop) protocol to solve this problem of Snoop protocol. With E-Snoop protocol, packet losses can be noticed by receiving new ACK packets as well as by receiving duplicate ACK packets or local retransmission timeout. Therefore, TCP throughput can be enhanced by fast recognition of bursty packet losses and fast local retransmissions. From the simulation results, E-Snoop protocol can improve TCP throughput more efficiently than Snoop protocol and can yield more TCP improvement especially in the channel with high packet loss rates.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.15 no.10
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• pp.2257-2264
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• 2011

Sensor nodes in Wireless sensor network have limited resources and consume almost all energy to the communication. For its traffic feature as a burst traffic type toward a sink node, it has high probability to network congestion. Network congestion causes packet drops and retransmission of dropped packets draws energy consumption. In particular, the loss of packet that is from the sensor node far away from a sink node requires additional energy consumption by frequent retransmission. This paper presents a traffic control mechanism that determines packet transfer by considering priority of packet and congestion level as well as hop count. Analysis of proposed mechanism by simulation demonstrated that it improved energy efficiency.

• Journal of Communications and Networks
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• v.4 no.2
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• pp.148-157
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• 2002

This paper investigates schemes to improve TCP performance in multipath forwarding networks. In multipath routing, packets to the same destination are sent to multiple next-hops in either packet-level or flow-level forwarding mode. Effective bandwidth is increased since we can utilize unused capacity of multiple paths to the destination. In packet-level multipath forwarding networks, TCP performance may not be enhanced due to frequent out-of-order segment arrivals at the receiver because of different delays among paths. To overcome this problem, we propose simple TCP modifications. At the sender, the fast retransmission threshold is adjusted taking the number of paths into consideration. At the receiver, the delayed acknowledgment scheme is modified such that an acknowledgment for an out-of-order segment arrival is delayed in the same way for the in-order one. The number of unnecessary retransmissions and congestion window reductions is diminished, which is verified by extensive simulations. In flow-level multipath forwarding networks, hashing is used at routers to select outgoing link of a packet. Here, we show by simulations that TCP performance is increased in proportion to the number of paths regardless of delay differences.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.22 no.8
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• pp.768-775
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• 2011

Since error compensation schemes guaranteeing high reliability are required in wireless communication systems for the transmission of the large amount of data, as an efficient error compensation scheme the turbo HARQ scheme combining automatic repeat request and turbo coding has been studied in many places in the literature. In the case of conventional turbo HARQ schemes, the transmitter recognizing NAK signals repetitively sends the whole unit packet to the receiver although the received packet can be partially correctable. Through two successive processes, selection of uncorrectable error data and retransmission of only the relevant parts of the information data, transmission efficiency can be improved. In this paper, we present an error data selection criterion for retransmission using the sign transitions of A Posteriori values and propose a tubo HARQ scheme based on the partial retransmission technique. Through a computer simulation, we show and analyze the performance of the proposed scheme with transmission efficiency.

• Journal of Communications and Networks
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• v.12 no.6
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• pp.577-581
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• 2010

We introduce a novel packet retransmission technique which improves the efficiency of automatic retransmission query (ARQ) protocols in the context of satellite broadcast/multicast systems. The proposed coded ARQ technique, similarly to fountain coding, performs transmission of redundant packets, which are made by linear combinations of the packets composing the source block. Differently from fountain codes, the packets for the linear combinations are selected on the basis of the retransmission requests coming from the user terminals. The selection is performed in a way that, at the terminals, the source packets can be recovered iteratively by means of simple back-substitutions. This work aims at providing a simple and efficient alternative to reliable multicast protocols based on erasure correction coding techniques.

• Proceedings of the IEEK Conference
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• 1999.11a
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• pp.121-124
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• 1999

In a packet-switched wireless cellular network, a packet destined to a mobile station is queued at a base station and then broadcast over the base station's cell. When an active mobile station leaves a cell, there re-main packets which are destined to the mobile and not yet delivered to it at the cell's base station. For application which are sensitive to packet losses, such back- logged packets must be forwarded to the new base station. Otherwise, an end-to-end retransmission may be required. However, an increase in packet delay is incurred by employing the packet forward scheme, since a packet may be forwarded many times before it is delivered to the destined mobile station. For an enhanced quality-of-service level, it is preferred to reduce tile packet delay time. In this paper, we develop an analytical approximation method for deriving mean packet delay times. Using the approximation and simulation methods, we investigate the effect of network parameters on the packet delay time.

• Journal of the Korea Society for Simulation
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• v.17 no.4
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• pp.191-198
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• 2008

6LoWPAN (IPv6 Low-power Wireless Personal Area Network) is IPv6 packets transmission technology at Sensor network over the IEEE 802.15.4 Standard MAC and Physical layer. Adaptation layer between IP layer and MAC layer performs fragmentation and reassembly of packet for transmit IPv6 packets. RFC4944, IETF 6LoWPAN WG standard document define packet fragmentation and reassembly. In this paper, we propose the 6PASim (6LoWPAN Packet Simulator) to perform IPv6 packet fragmentation and reassembly for performance evaluation. The 6PASim consist of two parts. One is Packet_Transmit_module that makes IEEE 802.15.4 frames the IPv6 packet from upper layer, and transmit its. and the another is Packet_Receive_module that reassembles transferred frames and completes original IPv6 packets. we can evaluate frame transmit rate and amount of control message through 6PASim. The result of simulation shows the SRM (Selective Retransmission Method) scheme provider better performance than IRM (Immediate Retransmission Method) scheme.

• ETRI Journal
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• v.27 no.1
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• pp.1-12
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• 2005

We propose a heuristic buffer management scheme that uses both positive and negative acknowledgments to provide scalability and reliability. Under our scheme, most receiver nodes only send negative acknowledgments to their repair nodes to request packet retransmissions while some representative nodes also send positive acknowledgments to indicate which packets can be discarded from the repair node's buffer. Our scheme provides scalability because it significantly reduces the number of feedbacks sent by the receiver nodes. In addition, it provides fast recovery of transmission errors since the packets requested from the receiver nodes are almost always available in their buffers. Our scheme also reduces the number of additional retransmissions from the original sender node or upstream repair nodes. These features satisfy the original goal of treebased protocols since most packet retransmissions are performed within a local group.

• Journal of Korea Multimedia Society
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• v.8 no.6
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• pp.798-806
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• 2005

In this paper, we suggest a game system that can support network modules for multi-platform based video games, and built a system that can convert from a single-user game to multi-user game. In this system, we bring in an initial delay buffering scheme on clients to handle any periods of latency occurring from the load fluctuation in a network, when a real-time game is played, and shows that stable play for a game is achieved as the result of the scheme. This paper also presents a retransmission algorithm based on the history of game commands to handle drawbacks of UDP mechanism. And, we evaluate the network delay and packet loss using the simulation tool NS2, and shows the case of 0.3 second buffer delay is the most suitable for recovery.