• Journal of Communications and Networks
• /
• v.5 no.1
• /
• pp.25-32
• /
• 2003

4G mobile communication system requires the throughput of 10-100Mbps. Adaptive modulated OFDM system is promising technique for increasing the throughput. In the pilot symbol assisted high-speed packet transmission system, the data symbol duration is generally considered to be small compared to the coherence time. However, OFDM symbol duration is longer than the symbol duration of a single carrier system, so that the packet duration of the pilot symbol assisted high speed packet transmission system is long. In this case, the change of channel conditions is too fast to be accurately estimated by channel estimator at the receiver in high Doppler frequency, so that many errors occur during demodulation, especially with the data symbols at the end of each packet. In this paper, we consider the BER at various instantaneous $E_b/N_o$ that includes the demodulation errors in high Doppler frequency. When the coherence time is ten times longer than the duration of a single packet, the channel can be closely approximated as an AWGN channel. Otherwise, the approximation breaks down and the above-mentioned errors that occur during demodulation must be taken into consideration. In this paper, we propose the pilot symbol assisted high speed packet transmission system based on adaptive OFDM using a novel lookup table to consider the demodulated errors and evaluate the throughput performance.

• KSII Transactions on Internet and Information Systems (TIIS)
• /
• v.13 no.3
• /
• pp.1298-1310
• /
• 2019

Network coding (NC) is a promising technology that can improve available bandwidth and packet throughput in wireless sensor networks (WSNs). Sliding window is an improved technology of NC, which is a supplement of TCP/IP technology and can improve data throughput and network lifetime on WSNs. This paper proposes a network coding-based maximum lifetime algorithm for sliding window in WSNs (NC-MLSW) which improves the throughput and network lifetime in WSN. The packets on the source node are sent on the WSNs. The intermediate node encodes the received original packet and forwards the newly encoded packet to the next node. Finally, the destination node decodes the received encoded data packet and recovers the original packet. The performance of the NC-MLSW algorithm is studied using NS2 simulation software and the network packet throughput, network lifetime and data packet loss rate were evaluated. The simulations experiment results show that the NC-MLSW algorithm can obviously improve the network packet throughput and network lifetime.

• Journal of Korean Society of Industrial and Systems Engineering
• /
• v.46 no.2
• /
• pp.57-71
• /
• 2023

In this study, we analyze a finite-buffer M/G/1 queueing model with randomized pushout space priority and nonpreemptive time priority. Space and time priority queueing models have been extensively studied to analyze the performance of communication systems serving different types of traffic simultaneously: one type is sensitive to packet delay, and the other is sensitive to packet loss. However, these models have limitations. Some models assume that packet transmission times follow exponential distributions, which is not always realistic. Other models use general distributions for packet transmission times, but their space priority rules are too rigid, making it difficult to fine-tune service performance for different types of traffic. Our proposed model addresses these limitations and is more suitable for analyzing communication systems that handle different types of traffic with general packet length distributions. For the proposed queueing model, we first derive the distribution of the number of packets in the system when the transmission of each packet is completed, and we then obtain packet loss probabilities and the expected number of packets for each type of traffic. We also present a numerical example to explore the effect of a system parameter, the pushout probability, on system performance for different packet transmission time distributions.

• Journal of Korea Game Society
• /
• v.15 no.6
• /
• pp.97-106
• /
• 2015

In an Unity engine for multi-player online games, the main thread processing game logic must be separated from the network thread that is responsible for network packet communication. Packet communication between the network threads needs to drop packets that overlap in order to improve the rendering speed. In this paper, the packet discard policy of network thread is proposed for an Unity engine for multi-player online games. The proposed method is the hybrid method of both Partial Packet Discard and Periodic Packet Discard methods to improve the rendering speed by periodically discarding overlapped network packets managed by the queue. The rendering speed of the proposed method is analyzed and its effectiveness is verified by various packet generating simulations of the Unity engine for multi-player online games.

• Journal of KIISE:Computer Systems and Theory
• /
• v.33 no.10
• /
• pp.722-733
• /
• 2006

In scalable streaming application, there are two important knobs to tune to effectively exploit the underlying network resource and to maximize the user perceivable quality of service(QoS): layer selection and packet scheduling. In this work, we propose Semantics Aware Packet Scheduling (SAPS) algorithm to address these issues. Using packet dependency graph, SAPS algorithm selects a layer to maximize QoS. We aim at minimizing distortion in selecting layers. In inter-frame coded video streaming, minimizing packet loss does not imply maximizing QoS. In determining the packet transmission schedule, we exploit the fact that significance of each packet loss is different dependent upon its frame type and the position within group of picture(GOP). In SAPS algorithm, each packet is assigned a weight called QoS Impact Factor Transmission schedule is derived based upon weighted smoothing. In simulation experiment, we observed that QOS actually improves when packet loss becomes worse. The simulation results show that the SAPS not only maximizes user perceivable QoS but also minimizes resource requirements.

• The Journal of the Institute of Internet, Broadcasting and Communication
• /
• v.12 no.3
• /
• pp.1-7
• /
• 2012

Cognitive radio, a new paradigm for wireless communication, is being recently expected to support various types of multimedia traffics. To guarantee Quality of Service (QoS) from SUs, a static packet priority policy can be considered. However, this approach can easily satisfy Quality of Service of high priority application while that of lower priority applications is being degraded. In the paper, we propose a fuzzy-based dynamic packet scheduling algorithm to support multimedia traffics in which the dynamic packet scheduler modifies priorities of packets according to Fuzzy-rules with the information of priority and delay deadline of each packet, and determines which packet would be transmitted through the channel of the primary user in the next time slot in order to reduce packet loss rate. Our simulation result shows that packet loss rate can be improved through the proposed scheme when overall traffic load is not heavy.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.32 no.10C
• /
• pp.950-958
• /
• 2007

This paper presents an efficient packet loss resilient scheme for real-time video transmission over the Internet. By analyzing the temporal and spatial dependencies in inter- and intra-frames, we assign forward error correction codes (FEC) across video packets for minimizing the effect of error concealment and error propagation from packet loss. To achieve optimal allocation of FEC codes, we formulate the effect of packet loss on video quality degradation as packet distortion model. Then we propose an unequal FEC assignment scheme with low complexity based on packet correction rate, which uses the packet distortion model and includes channel status information. Simulation results show that the proposed FEC assignment scheme gives substantial improvement for the received video quality in packet lossy networks. Furthermore the proposed scheme achieves relatively smaller degradation of video quality with higher packet loss rates.

• 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.

• Proceedings of the IEEK Conference
• /
• 2004.06a
• /
• pp.317-320
• /
• 2004

In this paper, it is analyzed a performance improvement of ACL Packet using turbo code scheme in bluetooth system. In order for segment to improve the transfer capability, the transmission of messages have been simulated using a fragmentation that begins with the total package and incremental fragmentation for each layer using the TCP to define the resultant packet size and the level of fragmentation for each proceeding layer. This turbo code scheme decreases transmission time of L2CAP baseband packets by sending packets. From the results, we were able to obtain packet transmission time, optimal TCP packet size, ACL(DM packet) in AWGN.

• 한국정보통신설비학회:학술대회논문집
• /
• 2007.08a
• /
• pp.136-141
• /
• 2007

The frequency accuracy and phase alignment is necessary for ensuring the quality of service (QoS) for applications such as voice, real-time video, wireless hand-off, and data over a converged access medium at the telecom network. As telecom networks evolve from circuit to packet switching, proper synchronization algorithm should be meditated for IP networks to achieve performance quality comparable to that of legacy circuit-switched networks. The Time of Packet (ToP) specified in IEEE 1588 is able to synchronize distributed clocks with an accuracy of less than one microsecond in packet networks. But, The ToP can be affected by impairments of a network such as packet delay variation. This paper proposes the efficient method to minimize the expectable delay variation when ToP synchronizes the distributed clocks. The simulation results are presented to demonstrate the improved performance case when the efficient ToP transmit algorithm is applied.