• 한국HCI학회:학술대회논문집
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• 2008.02a
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• pp.218-223
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• 2008

In this paper, we implement several methods to improve an audio collaboration system over IP networks, and then evaluate the performance of the implemented methods. In general, speech and audio quality degrades depending on the characteristics of IP networks such as jitter and packet loss. In order to reduce this quality degradation, we propose a lower bit rate audio delivery scheme using the MPEG-2 AAC (Advanced Audio Coding) audio codec in a viewpoint that a packet loss rate could be reduced by a smaller packet size. In addition, iLBC (Internet Low-Bitrate Codec) and the G.711 packet loss concealment algorithm defined by IEFT and ITU-T, respectively, are applied to a audio collaboration system. RAT (Robust-Audio Tool)[7] is used as a baseline platform for the implementation of the proposed methods. It is shown from the implementation that the implemented MPEG-2 AAC audio codec with a bitrate of 256 kbit/s performs as similar as the uncompressed audio quality with a bitrate of 512 kbit/s, and that iLBC and the G.711 packet loss concealment algorithm can improve speech quality when a packet loss rate is 2~10%.

• The Journal of the Acoustical Society of Korea
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• v.32 no.1
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• pp.86-94
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• 2013

In this paper, we propose a network coding scheme that is one of the most promising techniques for overcoming transmission errors in underwater acoustic communications. It is assumed that the proposed scheme operates in a Code Division Multiple Access (CDMA) network where multiple sensor nodes share the underwater acoustic channel in both the frequency and the time domains by means of orthogonal codes. The network topology deploys multi-hop transmission with relaying between multiple source nodes and one destination node via multiple relay nodes. The proposed scheme is evaluated in terms of the successful packet delivery ratio of end-to-end transactions under varying packet loss rates. A computer simulation shows that the successful delivery ratio is maintained at over 95% even when the packet loss rate reaches 50%.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.9 no.10
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• pp.4223-4249
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• 2015

A novel jamming detection technique to detect the presence of jamming in the downstream direction for cluster based wireless sensor networks is proposed in this paper. The proposed technique is deployed in base station and in cluster heads. The proposed technique is novel in two aspects: Firstly, whenever a cluster head receives a packet it verifies whether the source node is legitimate node or new node. Secondly if a source node is declared as new node in the first step, then this technique observes the behavior of the new node to find whether the new node is legitimate node or jammed node. In order to monitor the behavior of the existing node and new node, the second step uses two metrics namely packet delivery ratio (PDR) and received signal strength indicator (RSSI). The rationality of using PDR and RSSI is presented by performing statistical test. PDR and RSSI of every member in the cluster is measured and assessed by the cluster head. And finally the cluster head determines whether the members of the cluster are jammed or not. The CH can detect the presence of jamming in the cluster at member level. The base station can detect the presence of jamming in the wireless sensor network at CH level. The simulation result shows that the proposed technique performs extremely well and achieves jamming detection rate as high as 99.85%.

• KIPS Transactions on Computer and Communication Systems
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• v.2 no.6
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• pp.253-256
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• 2013

For the reliable delivery of safety-related information to vehicles in the VANET, a reliable VANET routing protocol is required. In this paper, we propose a routing protocol that works based on the road vehicle density information for fast and reliable communications among vehicles within the city environment VANET. In the proposed mechanism, each vehicle computes the road vehicle density by using beacon messages and the road information. Based on the road vehicle density information, each vehicle establishes a reliable route for packet delivery. Through the NS-2 based simulations, we compare our proposed mechanism with GPSR and show that our mechanism outperforms GPSR in terms of packet delivery success rate.

• Proceedings of the IEEK Conference
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• 2002.07c
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• pp.2075-2078
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• 2002

Internet became the most popular network in spite of its weakness in realtime multimedia service. Many experts believe that the Internet has the potential to become the main multimedia distribution network of the near future. Currently, it does not provide any (BoS guarantees and, even when it does, guaranteed quality delivery of video may turn out to be too expensive. Unavoidable packet losses and delay jitter caused by congestion in a best effort delivery environment require use of intelligent transport techniques for effective video delivery. According to market needs of better quality of service (QoS) fur realtime multimedia services over Internet, they have been standardizing RSVP, IntServ, and DiffServ This paper combines the benefits of QoS mechanisms such as RSVP/IntServ with scalable video encoding. We propose that more important bit stream is given more priority such that limited network resources are guaranteed far the stream. Various prioritizing approaches are proposed and compared to normal approach by using Network Simulator. The calculated QoS parameters such as packet loss rate are used to calculate degree of degradation in video quality. In this Paper, proposed methods can be implemented adaptively to Von protocol, such as H.323, SIP.

• Journal of Advanced Navigation Technology
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• v.18 no.1
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• pp.67-73
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• 2014

Due to the development of the communication technology and the high-performance mobile device, the number of video streaming service user has been increasing. Video streaming services in the restricted network environments need error control to support users for the Ultra High Definition(UHD) video delivery. Packet-level FEC method was used to control the error for delay-sensitive video service, but it has a defect in huge occupation of network resources by redundant packets. In this paper, we proposed a packet-level FEC algorithm which maintains the video quality by previous FEC methods and also reduces the occupation of network resource in the restricted network environments. The proposed algorithm controls the redundant rate with using the delivery characteristic of the video coding technology, and improves the efficiency of network resources by reducing the traffic by around 33% in comparison with the previous FEC methods.

• Journal of Internet Computing and Services
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• v.15 no.2
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• pp.9-18
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• 2014

The sensors attached on/in a person are moved since human body frequency changes their activity, therefore in wireless body area networks, nodal mobility and non-line-of-sight condition will impact on performance of networks such as energy efficiency and reliable communication. We then proposed schemes which study on forwarding decisions against frequent change of topology and channel conditions to increase reliable connections and improve energy efficiency. In this work, we control the size of packets, forwarding rate based on ratio of input links and output links at each node. We also robust the network topology by extending the peer to peer IEEE 802.15.4-based. The adaptive topology from chain-based to grid-based can optimal our schemes. The simulation shows that these approaches are not only extending network lifetime to 48.2 percent but also increase around 6.08 percent the packet delivery ratio. The "hot spots" problem is also resolved with this approach.

• Journal of Electrical Engineering and Technology
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• v.13 no.6
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• pp.2456-2467
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• 2018

Wireless multimedia sensor networks (WMSNs) require real-time quality-of-service (QoS) guarantees to be provided by the network. The end-to-end delay is very critical metric for QoS guarantees in WMSNs. In WMSNs, due to the transmission errors incurred over wireless channels, it is difficult to obtain reliable delivery of data in conjunction with low end-to-end delay. In order to improve the end-to-end delay performance, the system has to drop few packets during network congestion. In this article, our proposal is based on optimization of end-to end delay for WMSNs. We optimize end-to-end delay constraint by assuming that each packet is allowed fixed number of retransmissions. To optimize the end-to-end delay, first, we compute the performance measures of the system, such as end-to-end delay and reliability for different network topologies (e.g., linear topology, tree topology) and against different choices of system parameters (e.g., data rate, number of nodes, number of retransmissions). Second, we study the impact of the end-to-end delay and packet delivery ratio on indoor and outdoor environments in WMSNs. All scenarios are simulated with multiple run-times by using network simulator-2 (NS-2) and results are evaluated and discussed.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.12 no.3
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• pp.1330-1347
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• 2018

Mobile Ad hoc Network (MANET) is a collection of nodes or communication devices that wish to communicate without any fixed infrastructure and predetermined organization of available links. The effort has been made by proposing a scheme to overcome the critical security issue in MANET. The insufficiency of security considerations in the design of Ad hoc On-Demand Distance Vector protocol makes it vulnerable to the threats of collaborative black hole attacks, where hacker nodes attack the data packets and drop them instead of forwarding. To secure mobile ad hoc networks from collaborative black hole attacks, we implement our scheme and considered sensor's energy as a key feature with a better packet delivery ratio, less delay time and high throughput. The proposed scheme has offered an improved solution to diminish collaborative black hole attacks with high performance and benchmark results as compared to the existing schemes EDRIAODV and DRIAODV respectively. This paper has shown that throughput and packet delivery ratio increase while the end to end delay decreases as compared to existing schemes. It also reduces the overall energy consumption and network traffic by maintaining accuracy and high detection rate which is more safe and reliable for future work.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.36 no.8B
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• pp.954-961
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• 2011

To apply Wi-Fi technique to the Broadcast field, we have to study features of IEEE 802.11 Broadcast packet. IEEE 802.11 Broadcast technique cannot guarantee successful packet delivery than IEEE 802.11 Unicast. A promising solution to this problem is the use of FEC(Forward Error Correction) mechanisms. However, the adjustment of the FEC redundancy rate is not a trivial issue due to the dynamic wireless environment. In order to explore the above issues we conducted an experimental study of the packet loss behavior of the IEEE 802.11g protocol. In order to study, we implemented a broadcast test bed. Based on the experimental results, we provide guidelines on wireless lan parameters(packet size, transmission rate(11g), background traffic). From this experimental study, we provide FEC redundancy rate.