• Proceedings of the IEEK Conference
• /
• 1999.11a
• /
• pp.121-124
• /
• 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 Korean Institute of Industrial Engineers
• /
• v.28 no.2
• /
• pp.139-146
• /
• 2002

The third generation mobile communication needs to provide multimedia service with increased data rates. Thus an efficient allocation of radio and network resources is very important. This paper models the 'burst switching' as an efficient radio resource allocation scheme and the performance is compared to the circuit and packet switching. In burst switching, radio resource is allocated to a call for the duration of data bursts rather than an entire session or a single packet as in the case of circuit and packet switching. After a stream of data burst, if a packet does not arrive during timer2 value ($\tau_{2}$), the channel of physical layer is released and the call stays in suspended state. Again if a packet does not arrive for timerl value ($\tau_{1}$) in the suspended state, the upper layer is also released. Thus the two timer values to minimize the sum of access delay and queuing delay need to be determined. In this paper, we focus on the decision of $\tau_{2}$ which minimizes the access and queueing delay with the assumption that traffic arrivals follow Poison process. The simulation, however, is performed with Pareto distribution which well describes the bursty traffic. The computational results show that the delay and the packet loss probability by the burst switching is dramatically reduced compared to the packet switching.

• Journal of Communications and Networks
• /
• v.16 no.5
• /
• pp.548-558
• /
• 2014

Load distribution is vital to the performance of multipath transport. The task becomes more challenging in real-time multimedia applications (RTMA), which impose stringent delay requirements. Two key issues to be addressed are: 1) How to minimize end-to-end delay and 2) how to alleviate packet reordering that incurs additional recovery time at the receiver. In this paper, we propose sub-packet based multipath load distribution (SPMLD), a new model that splits traffic at the granularity of sub-packet. Our SPMLD model aims to minimize total packet delay by effectively aggregating multiple parallel paths as a single virtual path. First, we formulate the packet splitting over multiple paths as a constrained optimization problem and derive its solution based on progressive approximation method. Second, in the solution, we analyze queuing delay by introducing D/M/1 model and obtain the expression of dynamic packet splitting ratio for each path. Third, in order to describe SPMLD's scheduling policy, we propose two distributed algorithms respectively implemented in the source and destination nodes. We evaluate the performance of SPMLD through extensive simulations in QualNet using real-time H.264 video streaming. Experimental results demonstrate that: SPMLD outperforms previous flow and packet based load distribution models in terms of video peak signal-to-noise ratio, total packet delay, end-to-end delay, and risk of packet reordering. Besides, SPMLD's extra overhead is tiny compared to the input video streaming.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.13 no.2
• /
• pp.171-181
• /
• 1988

In this study, a packet voice/data terminal(PVDT) that services both voice and data in the packet-switched network is implemented. The software structure of the PVDT is designed according to the OSI 7 layer architecture. The discrimination of voice and data is made in the link layer. Voice packets have priority over data packets in order to minimize the transmission delay, and are serviced by a simple protocol so that the overhead arising form the retransmission of packets may be minimized. The hardware structure of the PVDT is divided into five modules; a master control module, a speech proessing module, a speech activity detection module, a telephone interface module, and an input/output interface module. In addition to the hardware implementation, the optimal reconstruction delay of voice packets to reduce the influence of delay variance is analyzed.

• Journal of Internet Computing and Services
• /
• v.18 no.6
• /
• pp.65-73
• /
• 2017

Congestion occurring at wireless sensor networks(WSNs) causes packet delay and packet drop, which directly affects overall QoS(Quality of Service) parameters of network. Network congestion is critical when important data is to be transmitted through network. Thus, it is significantly important to effectively control the congestion. In this paper, new mechanism to guarantee reliable transmission for the important data is proposed by considering the importance of packet, configuring packet priority and utilizing the settings in routing process. Using this mechanism, network condition can be maintained without congestion in a way of making packet routed through various routes. Additionally, congestion control using packet service time, packet inter-arrival time and buffer utilization enables to reduce packet delay and prevent packet drop. Performance for the proposed mechanism was evaluated by simulation. The simulation results indicate that the proposed mechanism results to reduction of packet delay and produces positive influence in terms of packet loss rate and network lifetime. It implies that the proposed mechanism contributes to maintaining the network condition to be efficient.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.15 no.2
• /
• pp.98-111
• /
• 1990

In this paper a queneing model for the D channel access protocal recommeded by CCITT is developed, and delays of the signalling and packet messages are analyzed using the model, Behaviors of packet and signalling messages in the D-channel access system are also investigated. The analytical results have been verified by simulation.

• KSII Transactions on Internet and Information Systems (TIIS)
• /
• v.9 no.6
• /
• pp.2180-2200
• /
• 2015

This paper considers mobility control to improve packet delivery in delay-tolerant networks (DTNs) under group mobility. Based on the group structure in group mobility, we propose two mobility control techniques; group formation enforcement and group purposeful movement. Both techniques can be used to increase the contact opportunities between groups by extending the group's reachability. In addition, they can be easily integrated into some existing DTN routing schemes under group mobility to effectively expedite the packet delivery. This paper is divided into 2 parts. First, we study how our proposed mobility control schemes reduce the packet delivery delay in DTNs by integrating them into one simple routing scheme called group-epidemic routing (G-ER). For each scheme, we analytically derive the cumulative density function of the packet delivery delay to show how it can effectively reduce the packet delivery delay. Then, based on our second proposed technique, the group purposeful movement, we design a new DTN routing scheme, called purposeful movement assisted routing (PMAR), to further reduce the packet delay. Extensive simulations in NS2 have been conducted to show the significant improvement of PMAR over G-ER under different practical network conditions.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.30 no.10A
• /
• pp.896-905
• /
• 2005

In this paper, we propose a SPD(Selective Packet Delay) scheme to improve the performance of High-Speed Downlink Shared Channel(HS-DSCH) employing N-Channel Stop and Wait retransmission scheme in High Speed Downlink Packet Access(HSDPA) system. The proposed SPD coordinates packet transmissions according to the channel condition. When the channel condition is bad, packet transmission is forcedly delayed, and the designated time slot is set over to other users in good channel condition. Hence, the SPD is able to reduce the average transmission delay for packet transmission under the burst error environments. In addition, we propose two packet scheduling schemes called SPD-LDPF(Long Delayed Packet First) and SPD-DCRR(Deficit Compensated Round Robin) that are effectively combined with the SPD scheme. Simulation results show that the proposed scheme has better performance in terms of delay, throughput and fairness.

• Journal of Industrial Technology
• /
• v.27 no.B
• /
• pp.235-243
• /
• 2007

Sensor network is used to obtain sensing data in various area. The interval to sense the events depends on the type of target application and the amounts of data generated by sensor nodes are not constant. Many applications exploit long sensing interval to enhance the life time of network but there are specific applications that requires very short interval to obtain fine-grained, high-precision sensing data. If the number of nodes in the network is increased and the interval to sense data is shortened, the amounts of generated data are greatly increased and this leads to increased amount of packets to transfer to the network. To transfer large amount of packets fast, it is necessary that the delay between successive packet transmissions should be minimized as possible. In Sensor network, since the Operating Systems are worked on the event driven, the Timer Event is used to transfer packets successively. However, since the transferring time of packet completely is varies very much, it is very hard to set appropriate interval. The longer the interval, the higher the delay and the shorter the delay, the larger the fail of transfer request. In this paper, we propose ESTEO which reduces the delay between successive packet transmissions by using SendDone Event which informs that a packet transmission has been completed.In ESTEO, the delay between successive packet transmissions is shortened very much since the transmission of next packet starts at the time when the transmission of previous packet has completed, irrespective of the transmission timee. Therefore ESTEO could provide high packet transmission rate given large amount of packets.

• The KIPS Transactions:PartC
• /
• v.12C no.3 s.99
• /
• pp.437-442
• /
• 2005

The measurement of RTT (Round Trip Time) can be used for the analysis of Internet congestion. However, simple measuring of RTT which measures only hun around time of a packet can not infer a packet forward/backward delay variation. In this thesis, we present a new algorithm which can be used for the estimation of forward/backward delay variation of packets. These delay variations are implication of network congestion state. In this algorithm, the reference forward/backward delay can be determined based on the minimum RTT value. The delay variation of each packet can be calculated by comparing reference delay with the packet delay. We verified our proposed algorithm by NS-2 simulation and delay measuring in a real network.