• The Journal of the Acoustical Society of Korea
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• v.18 no.6
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• pp.71-76
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• 1999

In digital cellular communication systems, tandem coding occurs in communications between mobile phones with different speech codecs, resulting in poor voice quality, high computational load, and long transmission delay. In this paper, voice packet conversion technique is proposed to solve the tandem coding problems, and packet conversion algorithm from 13kbps QCELP to 8kbps QCELP is developed. Simulations using various speech data show that the proposed packet conversion method produces voice quality which is equivalent to that by the conventional tandem coding method with shorter transmission delay using about 33% computational load.

• Journal of Ubiquitous Convergence Technology
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• v.1 no.1
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• pp.28-34
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• 2007

Tree-based reliable multicast protocols provide scalability by distributing error-recovery tasks among several repair nodes. These repair nodes perform local error recovery for their receiver nodes using the data stored in their buffers. We propose a packet loss patterns adaptive feedback scheduling scheme to manage these buffers in an efficient manner. Under our scheme, receiver nodes send NAKs to repair nodes to request packet retransmissions only when the packet losses are independent events from other nodes. At dynamic and infrequent intervals, they also send ACKs to indicate which packets can be safely discarded from the repair node's buffer. Our scheme reduces delay in error recovery because the requested packets are almost always available in the repair node's buffers. It also reduces the repair node's workload because (a) each receiver node sends infrequent ACKs with non-fixed intervals and (b) their sending times are fairly distributed among all the receiver nodes.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2017.05a
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• pp.621-624
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• 2017

In this paper, we propose a protocol synchronizing to packet trains in a underwater acoustic networks with limitations such as long propagation delay, low data rate, high packet loss. The proposed protocol employs an one-way communication scheme to calculate packet arrival time differences of consecutively transferred packets and minimizes uncertain variation effect of long propagation delay. Simulation results shows the superiority of our protocol, compared with a previously proposed one under the given network conditions.

• Speech Sciences
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• v.11 no.4
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• pp.67-73
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• 2004

Packet loss, jitter, and delay in the Internet are caused mainly by the shortage of network bandwidth. It is due to queuing and routing process in the intermediate nodes of the packet network. In the Internet whose bandwidth is changing very rapidly in time depending on the number of users and data traffic, controlling the peak transmission bit-rate of a VoIP. system depending on the channel condition could be very helpful for making use of the available network bandwidth. Adapting packet size to the channel condition can reduce packet loss to improve the speech quality. It has been shown in [1] that a VoIP system with an AMR speech codec provides better speech quality than VoIP systems with fixed rate speech codecs. With the adaptive codec mode assignment. algorithm proposed in [1], in this paper, we performed the voice transmission experiments using the wireless LAN through the real Internet environment. Experimental results are analyzed and discussed with our findings.

• Proceedings of the Korea Information Processing Society Conference
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• 2008.11a
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• pp.1012-1015
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• 2008

The packet capturing becomes a bottleneck in the network intrusion detection and monitoring system as the network performance developing. Many approaches, zero copy, interrupt coalescing and NAPI which attempt to improve packet capturing performance of Linux, are inefficient. PF_RING is a new type of network socket that dramatically improves the packet capture speed, but not perfect. This paper proposes some solutions which can improve the memory utilization and save some data copy time based on the commodity network adapters rather than on the commercial network adapters.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.25 no.7
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• pp.940-945
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• 2021

Future network architectures such as Named Data Networking (NDN) were born to change the way data can be transmitted from current host-centric network technologies to information-centric network technologies. Recently, many studies are being conducted to graft Vehicular NDN to the communication network technology of smart vehicles including connected vehicles. Explosion of data traffic due to Interest/Data packet broadcasting in Vehicular NDN environment is a very important problem to be solved in order to realize VNDN-based data communication. In this paper, the generation of data packet copies according to the increase in network size, vehicle speed, and frequency of interest packets in VNDN network is simulated and evaluated using ndnSIM, in order to show how severe the data broadcast storm phenomenon. The CDP(Copies of Data Packets) increased proportionally in the increase of network size or Interest frequency.

• The KIPS Transactions:PartA
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• v.9A no.3
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• pp.317-326
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• 2002

The mobile telecommunication system has been growing exponentially after 1990s due to the high population in a city and the growth of mobile user. In this time, the current mobile system mainly concentrates on the voice communication. However, in the next generation, mobile users want to get very diverse services via mobile terminal such as the Internet access, web access, multimedia communication, and etc. For this reason, the next generation system, such as the UMTS system, has to support the packet data service and it will play the major role in the system. In different from the voice service, the packet data service should store packet data in a buffer at base station before it forwards. Consequently, the performance closely related to the buffer management and its operation. However, until now, there are rare study on the buffer behavior and management. In this paper, we will observe the buffer behavior in the UMTS systems by using simulation and analyze the results. For this research, we generate packet data traffics and model the UMTS system for a simulation. Then we analyze the buffer state by a simulation and calculate the buffer overflow probability by mathematical methods.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.26 no.5
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• pp.725-730
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• 2022

Named Data Networking (NDN) is a representative technology of ICN that realizes the future Internet architecture. NDN searches for data by its content and not by its host IP address. The consumer generates an interest packet and sends it to the NDN network. The NDN network uses three tables such as CS, FIB, and PIT and forwards the received interest packet to the next hop. The producer transmits the data packet to the consumer through a name-based forwarding scheme. In this paper, we design and implement an ndnSIM-based NDN network and perform performance evaluation. We analyze the ndnSIM structure and develop a 6-node congested NDN network and a 9-node grid NDN network using ndnSIM. In the simulation, the performance of latency and throughput of the interest packet rate are measured. We analyze the effect of congestion on the latency and throughput of the NDN network. This approach will help researchers in the future.

• Journal of the Institute of Electronics Engineers of Korea SC
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• v.43 no.1 s.307
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• pp.7-12
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• 2006

In this paper we proposed a real-time remote monitoring system for interoperability between local area and wide area for wireless data communication. In local area, we used a miniaturized low-power wireless module and in wide area used CDMA Cellular System's Packet Data Service. The measurement results can be spread via Internet access in real-time

• Journal of Korea Multimedia Society
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• v.20 no.2
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• pp.254-260
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• 2017

Data packets from sensor nodes scattered over measuring fields are generally forwarding to the sink node, which may be connected to the wired networks, in a wireless sensor network. So many data packets are gathered near the sink node, resulting in significant data packet collisions and severe transmission latency. In an event detection application such as object tracking and military, bursty data is generated when an event occurs. So many data packet should be transmitted in a limited time to the sink node. In this paper, we present a delay efficient and bursty traffic friendly MAC protocol called DEBF-MAC protocol for wireless sensor networks. The DEBF-MAC uses a slot-reserved mechanism and sleep period control method to send multiple data packets efficiently in an operational cycle time. Our simulation results show that DEBF-MAC outperforms DW-MAC and SR-MAC in terms of energy consumption and transmission delay.