• Proceedings of the Korea Information Processing Society Conference
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• 2013.11a
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• pp.400-401
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• 2013

Mobility in a network causes link disconnections and link recovery is vital for reliability of a network. A link failure affects all the preceding nodes on a damaged routing path; creates communication delay, throughput degradation, and congestion. This paper proposes link recovery mechanisms in CCN based networks. Packet overhearing is used to update neighboring nodes information. The recovery is done by forwarding node resulting in low control overhead, and better efficiency. The proposed mechanisms increase overall performance of a typical CCN and simulation results show that our proposed scheme works very well in densely populated networks with high mobility.

• The KIPS Transactions:PartC
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• v.17C no.6
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• pp.465-470
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• 2010

In this paper we propose a method which can be used to enhance the reliability of E-ODMRP (Enhanced On-Demand Multicast Routing Protocol). E-ODMRP has low overhead compared to its predecessors since it performs periodic refresh at a rate dynamically adapted to the nodes' mobility and adopts the local recovery. Upon detecting a broken route, a node performs a local search to graft to the forwarding mesh proactively. However in E-ODMRP there is no packet recovery mechanism. A receiver may lose some packets when it is detached from the multicast tree. We propose a simple packet recovery mechanism that can be incorporated into E-ODMRP for enhanced reliability. We show via simulation that our mechanism effectively enhances the reliability of E-ODMRP.

• Journal of KIISE:Information Networking
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• v.31 no.2
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• pp.233-241
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• 2004

In this paper, we propose a packet-lossless fast rerouting scheme at a link/node fault in MPLS (Multiprotocol Label Switching) network with minimized accumulated buffer delay problem at ingress node. The proposed scheme uses a predefined, alternative LSP (Label Switched Path) In order to restore user traffic. We propose two restoration approaches. In the first approach, an alternative LSP is initially allocated with more bandwidth than the protected working LSP during the failure recovery phase. After the failure recovery, the excessively allocated bandwidth of the alternative LSP is readjusted to the bandwidth of the working LSP. In the second approach, we reduce the length of protected working LSP by using segment-based restoration. The proposed approaches have merits of (ⅰ) no buffer delay problem after failure recovery at ingress node, and (ⅱ) the smaller required buffer size at the ingress node than the previous approach.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.4 no.3
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• pp.341-357
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• 2010

Forward Error Correction (FEC) techniques have been adopted to overcome packet losses and to improve the quality of video delivery. The efficiency of the FEC has been significantly compromised, however, due to the characteristics of the wireless channel such as burst packet loss, channel fluctuation and lack of Quality of Service (QoS) support. We propose herein an Adaptive Cross-layer FEC mechanism (ACFEC) to enhance the quality of video streaming over 802.11 WLANs. Under the conventional approaches, FEC functions are implemented on the application layer, and required feedback information to calculate redundancy rates. Our proposed ACFEC mechanism, however, leverages the functionalities of different network layers. The Automatic Repeat reQuest (ARQ) function on the Media Access Control (MAC) layer can detect packet losses. Through cooperation with the User Datagram Protocol (UDP), the redundancy rates are adaptively controlled based on the packet loss information. The experiment results demonstrate that the ACFEC mechanism is able to adaptively adjust and control the redundancy rates and, thereby, to overcome both of temporary and persistent channel fluctuations. Consequently, the proposed mechanism, under various network conditions, performs better in recovery than the conventional methods, while generating a much less volume of redundant traffic.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.19 no.8
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• pp.1536-1547
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• 1994

This paper presented the implementation and design of narrowband wireless MODEM for packet transmission. The MODEM consists of transmitter, receiver, and the control unit. The BPSK modulation with narrowband filtering is used. The receiver consists of functional modules such as carrier recovery, bit synchronization, lock detector, etc. We evaluated the performance of packet transmission with three MODEM sets implemented in distributed packet radio network. We confirmed the transmission of packetized data through RS232C port of PC. Also, we presented results of experimental data by using measuring instruments. The implemented MODEM in this paper is expected to be useful for the design of wireless LAN system.

• The Journal of the Acoustical Society of Korea
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• v.21 no.4
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• pp.349-360
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• 2002

In ITU H.323 teleconference system, the RTP/RTCP protocol is offered to transfer real-time multimedia stream. Both sender and receiver hate experience in packet loss and jitter which result from network congestion over Internet. Audio quality over Internet depends on the number of lost packets and on jitter between successive packets. The goal of our study is to improve the speech quality over Internet by checking the packet loss characteristics of the network and adopting the but for control management mechanism at the receiver. We suggest a dynamic redundant audio transmission mechanism which examines the packet loss rate and uses the feedback information through RTCP.

• Proceedings of the Korean Institute of Navigation and Port Research Conference
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• 2003.05a
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• pp.311-315
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• 2003

Recently, there is asymmetrical transmission in Internet data stream. The asymmetrical transmission has much more downstream than upstream. Owing to this point, it needs to reduce the acknowledgement as element of the obsrtuction in downstream. In this paper, according to simulation's result, we know that Sack has good performance than New Reno in bottleneck environment Comparing two protocols in packet loss rate, NewReno is lower than Sack. And also comparing two protocols in throughput of ack packet, not only NewReno processes ack packet more quickly than Sack, but also NewReno processes more ack packet than Sack protocol during ten seconds in simulation. As a result, NewReno is batter than Sack in throughput of asymmetrical link.

• Proceedings of the Korean Institute of Communication Sciences Conference
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• 2004.07a
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• pp.212-212
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• 2004

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.3B
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• pp.130-137
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• 2005

Wireless communication environment is changing rapidly as we use new wireless communication technology such as WiBro to access high speed Internet. As a result, reliable data transmission using TCP is also expected to increase. Since TCP assumes that it is used in wired network, TCP suffers significant performance degradation over wireless network where packet losses are related to non-congestion loss. Especially RTO imposes a great performance degradation of TCP. In this paper, we analyze the loss recovery probabilities based on previous researches, and use simulation results of our algorithm to show that it prevents performance degradation by quickly detecting and recovery losses without RTO during fast recovery.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.23 no.6
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• pp.1496-1502
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• 1998

In this paper, a real-time audio broadcasting service system which is robust to loaded traffic on the Internet is developed. For implementing reliable real-time data transfer, the transfer characteristics of TCP/IP and UDP/IP was compared and analyzed. For lost packet recovery, redundant audio data algorithm was used and interleaving technique was applied for scattering consecutive packet loss. Test results showed, when using TCP/IP, pause occurred during playback, and when using UDP/IP, a stable receive rate was noticeable but the quality of the sound was lower than that of uisng TCP/IP. The recovery rate using redundant audio data and interleaving technique is shown in Fig. 9 and the delay is shown in Fig 4.