• The KIPS Transactions:PartC
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• 제9C권4호
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• pp.563-572
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• 2002

Wireless mobile networks tend to drop a large portion of packets due to propagation errors rather than congestion. To Improve reliability over noisy wireless channels, wireless networks can employ forward error correction (FEC) techniques. Static FEC algorithms, however, can degrade the performance by poorly matching their overhead to the degree of the underlying channel error, especially when the channel path loss rate fluctuates widely. This paper investigates the benefits of an adaptable FEC mechanism for wireless networks with severe packet loss by analytical analysis or measurements over a real wireless network called sensor network. We show that our adaptive FEC named FECA (FEC-level Adaptation) technique improves the performance by dynamically tuning FEC strength to the current amount of wireless channel loss. We quantify these benefits through a hybrid simulation integrating packet-level simulation with bit-level details and validate that FECA keeps selecting the appropriate FEC-level for a constantly changing wireless channel.

• KSII Transactions on Internet and Information Systems (TIIS)
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• 제13권9호
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• pp.4572-4586
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• 2019

Mobile Edge Computing (MEC) and Information-Centric Networking (ICN) are essential network architectures for the future Internet. The advantages of MEC and ICN such as computation and storage capabilities at the edge of the network, in-network caching and named-data communication paradigm can greatly improve the quality of video streaming applications. However, the packet loss in wireless network environments still affects the video streaming performance and the existing loss recovery approaches in ICN does not exploit the capabilities of MEC. This paper proposes a Deep Learning based Loss Recovery Mechanism (DL-LRM) for video streaming over MEC based ICN. Different with existing approaches, the Forward Error Correction (FEC) packets are generated at the edge of the network, which dramatically reduces the workload of core network and backhaul. By monitoring network states, our proposed DL-LRM controls the FEC request rate by deep reinforcement learning algorithm. Considering the characteristics of video streaming and MEC, in this paper we develop content caching detection and fast retransmission algorithm to effectively utilize resources of MEC. Experimental results demonstrate that the DL-LRM is able to adaptively adjust and control the FEC request rate and achieve better video quality than the existing approaches.

• Journal of KIISE:Information Networking
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• 제31권1호
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• pp.20-26
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• 2004

A congestion control algorithm to prevent buffer overflow in MSR(Mobility Support Router) for I-TCP is proposed. Due to high bit error rate and frequent hand-offs over wireless environment, the current congestion control scheme in TCP Reno over mixed(wired and wireless) network exhibits lower throughput than the throughput achieved over wired only network. I-TCP has been proposed to address this by splitting a TCP connection into two TCP connections over wired section and wireless section, respectively. However, buffer overflow in MSR may occur whenever there are excessive bit errors or frequent hand-offs. This may lead to the loss of packets acked by MSR(resident in buffer) to the sender, but not received by the receiver, breaking TCP end-to-end semantics. In this Paper, a new scheme is proposed to prevent the MSR buffer from overflow by introducing “flow control” between the sender and the MSR. Advertised window for the TCP connection between the sender and the MSR is tied to the remaining MSR buffer space, controlling the flow of packets to the MSR buffer before overflow occurs.

• Journal of the Korean Society of Safety
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• 제25권6호
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• pp.203-209
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• 2010

A compact ANSI/EIA 709.1 protocol analyzer system suited for maintenance of LonWorks network for safety management of underground facilities was developed and tested. The hardware is based on the TMS320LF2406A embedded system, and the software was designed using Visual C++6.0 under Windows XP environment. Connected to the LonWorks network the developed protocol analyzer decodes the raw packets and pass them to the master PC through USB port. Then on the PC the packets are processed and analyzed in various aspects and the key features that are essential to the maintenance of LonWorks network installed at underground facilities are displayed in a user-friendly format. Performance of the developed protocol analyzer was evaluated through a series of experiments, by measuring the speed of packet analysis and the error rate. The protocol analyzer proved to work reliably even under the increased bandwidth. However, more comprehensive tests under various underground environmental conditions are desired.

• Journal of the Korean Society for Railway
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• 제15권5호
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• pp.454-458
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• 2012

The CBTC(Communication based Train Control) and WPAN(Wireless Personal Area Network) systems employed on 2.4GHz ISM band wireless networks provide complementary service within the same environment. Coexistence between the networks will be impaired if the mutual packets are uncertainty associated the timing or gaussian distance. This paper analyzes the impact of the mutual interference and proposed method to minimize the effect of the WPAN system signals on CBTC system. The performance analysis is illustrated by examining the symbol error rate versus signal to noise interference ratio in terms of carrier frequency offset. The method contributes to reduce the symbol error rate drastically. However it can't be a fundamental solution for the impact on the interference problem. It may be needed for railway only frequency after checking the problems of the current railway frequency.

• Progress in Medical Physics
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• 제14권2호
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• pp.124-130
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• 2003

Currently, as a consequence of PACS (Picture Archiving Communication System) implementation many hospitals are replacing conventional film-type interpretations of diagnostic medical images with new digital-format interpretations that can also be saved, and retrieve However, the big limitation in PACS is considered to be the lack of mobility. The purpose of this study is to determine the optimal communication packet size. This was done by considering the terms occurred in the wireless communication. After encoding medical image using JPGE2000 image compression method, This method embodied auto-error correction technique preventing the loss of packets occurred during wireless communication. A PC class server, with capabilities to load, collect data, save images, and connect with other network, was installed. Image data were compressed using JPEG2000 algorithm which supports the capability of high energy density and compression ratio, to communicate through a wireless network. Image data were also transmitted in block units coeded by JPEG2000 to prevent the loss of the packets in a wireless network. When JPGE2000 image data were decoded in a PUA (Personal Digital Assistant), it was instantaneous for a MR (Magnetic Resonance) head image of 256${\times}$256 pixels, while it took approximately 5 seconds to decode a CR (Computed Radiography) chest image of 800${\times}$790 pixels. In the transmission of the image data using a CDMA 1X module (Code-Division Multiple Access 1st Generation), 256 byte/sec was considered a stable transmission rate, but packets were lost in the intervals at the transmission rate of 1Kbyte/sec. However, even with a transmission rate above 1 Kbyte/sec, packets were not lost in wireless LAN. Current PACS are not compatible with wireless networks. because it does not have an interface between wired and wireless. Thus, the mobile JPEG2000 image viewing system was developed in order to complement mobility-a limitation in PACS. Moreover, the weak-connections of the wireless network was enhanced by re-transmitting image data within a limitations The results of this study are expected to play an interface role between the current wired-networks PACS and the mobile devices.

• Journal of Korea Society of Digital Industry and Information Management
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• 제5권2호
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• pp.141-156
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• 2009

As the TCP is the protocol designed for the wired network that packet loss probability is very low, because TCP transmitter takes it for granted that the packet loss by the wireless network characteristics is occurred by the network congestion and lowers the transmitter's transmission rate, the performance is degraded. The Snoop Protocol was designed for the wired network by putting the Snoop agent module on the BS(Base Station) that connect the wire network to the wireless network to complement the TCP problem. The Snoop agent cash the packets being transferred to the wireless terminal and recover the loss by resending locally for the error occurred in the wireless link. The Snoop agent blocks the unnecessary congestion control by preventing the dupack (duplicate acknowledgement)for the retransmitted packet from sending to the sender and hiding the loss in the wireless link from the sender. We evaluated the performance in the wired/wireless network and in various TCP versions using the TCP designed for the wired network and the Snoop designed for the wireless network and evaluated the performance of the wired/wireless hybrid network in the wireless link environment that the continuous packet loss occur.

• Journal of the Korea Institute of Information and Communication Engineering
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• 제15권10호
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• pp.2129-2135
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• 2011

The WLAN and WPAN systems employed on 2.4GHz ISM band wireless networks provide complementary services within the same environments. Coexistence between the networks will be impaired, if the mutual packets are uncertainty associated the timing or gaussian distance. This paper analyzes the impact of the mutual interference on the services performance and in order to minimize the effect of WPAN system signals on the WLAN system, proposed a method of suppressing the interference on the WLAN system. The analysis is illustrated by examining the symbol error rate versus signal to noise interference ratio in terms of the carrier frequency offset.

• Journal of Korean Society of Industrial and Systems Engineering
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• 제29권4호
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• pp.100-106
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• 2006

We analyze a voice/data integrated traffic model of the cellular mobile communication system with hybrid guard channels for voice and handoff calls. In a multi-service integrated wireless environment, quality of service guarantee is crucial for smooth transportation of real time information. Real time voice traffic requires a guaranteed upper bounded on both delay and packet error rate, whereas data traffic does not. Voice traffic has high transmission priority over data packets. Thus one of the important problems is the design of admission control schemes which can efficiently accommodate the differential quality of service requirements. In this paper, a hybrid guard channel scheme is considered in which arriving calls are assigned channels as long as the number of busy channels in the cell is below a predetermined first threshold. When the number of busy channels reaches the first threshold, new originating data calls are queued in the infinite data buffer. Then reaches second threshold, only handoff calls are assigned the remaining channels and new originating voice calls are blocked. We evaluate the system by a two-dimensional Markov chain approach and generating function method and obtain performance measures included blocking probability and forced termination probability.

• The Journal of Korean Institute of Communications and Information Sciences
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• 제26권12A호
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• pp.2027-2035
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• 2001

In this paper, we present an audio stream delivery using the AMR (Adaptive Multi-Rate) coder that was adopted by ETSI and 3GPP as a standard vocoder for next generation IMT-2000 service in which includes combined sender (FEC) and receiver reconstruction technique in the Internet. By use of the media-specific FEC scheme, the possibility to recover lost packets can be much increased due to the addition of repair data to a main data stream, by which the contents of lost packets can be recovered. The AMR codec is based on the code-excited linear predictive (CELP) coding model. So we use a frame erasure concealment for CELP-based coders. The proposed scheme is evaluated with ITU-T G.729 (CS-ACELP) coder and AMR - 12.2 kbit/s through the SNR (Signal to Noise Ratio) and the MOS (Mean Opinion Score) test. The proposed scheme provides 1.1 higher in Mean Opinion Score value and 5.61 dB higher than AMR - 12.2 kbit/s in terms of SNR in 10% packet loss, and maintains the communicab1e quality speech at frame erasure rates lop to 20%.