• Journal of the Institute of Electronics Engineers of Korea TC
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• v.41 no.10
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• pp.1-9
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• 2004

User-Perceived application-level performance is a key to the adoption and success of CDMA 2000. To predict this performance in advance, a detailed end-to-end simulation model of a CDMA network was built to include application traffic characteristics, network architecture, network element details, and protocol features. We assess the user application performance when a Radio Access Network (RAN) and a Core Network (CN) adopt different transport architectures such as ATM and If. For voice Performance, we found that the vocoder bypass scenario shows 8% performance improvement over the others. For data packet performance, we found that HTTP v.1.1 shows better performance than that of HTTP v.1.0 due to the pipelining and TCP persistent connection. We also found that If transport technology is better solution for higher FER environment since the IP packet overhead is smaller than that of ATM for web browsing data traffic, while it shows opposite effect to small size voice packet in RAN architecture. Though simulation results we showed that the 3G-lX EV system gives much better packet delay performance than 3G-lX RTT, the main conclusion is that end-to-end application-level performance is affected by various elements and layers of the network and thus it must be considered in all phases of the technology evolution process.

• The Journal of The Korea Institute of Intelligent Transport Systems
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• v.4 no.1 s.6
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• pp.107-115
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• 2005

For effective IP based service implementation on the wireless network environments, wireless network including the ITS network have to support QoS guaranteed protocol such as a RSVP. RSVP is a resource reservation protocol for Internet environment, and its scalability makes easy to implement RSVP over the various IP transport technologies. But for the If based ITS wireless network environment, RSVP is not suitable, since by its path setup procedure characteristic. In the wireless access network for ITS service, when a mobile node moves to other domain it must perform registration procedure. But the registration procedure is time consuming steps, so if a RSVP session was already established in the previous domain, the RSYP session may be disconnected and the time to re establish a new RSVP session is long enough to cause serious packet loss. In this paper, we propose a pre-path reservation mechanism for applying the RSVP in wireless access networks for ITS. In the pre-path reservation mechanism, the resource reservation procedure occurred during a mobile node's handoff time. An access point in wireless access network performs this procedure when the mobile node attempts handoff The access point executes pre-path reservation procedure as a proxy, since the mobile node does not have IP address until the address allocation procedure is finished in a new domain.

• International Journal of Internet, Broadcasting and Communication
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• v.11 no.3
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• pp.59-63
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• 2019

VoIP technology is a technology for exchanging voice or video data through IP network. Various protocols are used for this technique, in particular, RTP(Real-time Transport Protocol) protocol is used to exchange voice data. In recent years, with the development of communication technology, there has been an increasing tendency of services such as "Kakao Voice Talk" to exchange voice and video data through IP network. Most of these services provide a service with security guarantee by a user authentication process and an encryption process. However, RTP protocol does not require encryption when transmitting data. Therefore, there is an exposition risk in the voice data using RTP protocol. We will present the risk of the situation where packets are sniffed in VoIP(Voice over IP) communication using RTP protocol. To this end, we configured a VoIP telephone network, applied our own sniffing tool, and analyzed the sniffed packets to show the risk that users' data could be exposed unprotected.

• Journal of Communications and Networks
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• v.1 no.3
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• pp.145-152
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• 1999

A new switch architecture is presented to enhance out-put queuing in photonic packet switches. Its appkication is for a packet switching enviroment based on the optical transport of fixed length packets. This architecture consists of a couple of cas-cading switching elements with output queuing, whose buffer ca-pacity is limited by photonic technology. The introduction of a suitable buffer management allows a very good and balanced ex-ploitation of the available optical memories, realized with fiber de-lay lines. In particular, packet loss performance is here evaluated showing the improvement with respect to the single switch and a way to design large optical switches is shown in order to meet broadband network requirements.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2009.01a
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• pp.49-54
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• 2009

Synchronization between media is an important aspect in the design of multimedia streaming system. This paper proposes a precise media synchronization mechanism for digital video and audio transport over IP networks. To support synchronization between video and audio bitstreams transported over IP networks, RTP/RTCP protocol suite is usually employed. To provide a precise mechanism for media synchronization between video and audio, we suggest an efficient media synchronization algorithm based on NPT (Normal Play Time) which can be derivable from the timestamp information in the header part of RTP packet generated for the transport of video and audio streams. With the proposed method, we do not need to send and process any RTCP SR (sender report) packet which is required for conventional media synchronization scheme, and accordingly could reduce the number of required UDP ports and the amount of control traffic injected into the network.

• Proceedings of the IEEK Conference
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• 2003.11c
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• pp.145-148
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• 2003

A lot of institutes have proposed RMT (Reliable Multicast Transport) techniques over IP multicast environment which provides only best-effort services. The ECTP (Enhanced Communications Transport Protocol) is a kind of RMT technique and an international standard of ITU-T and JTC1/SC6. It is a protocol that provides reliability fer IP multicast environment. In this paper, we tested ECTP applications which are MFTP (Multicast file transport protocol) and WB (white Board) on the APAN (Asia-Pacific Advanced Network) with the Kyushu University in Japan. We measured ECTP traffic using a packet capture tool that programmed by Pcap library that is developed in Lawrence Berkeley institute. In according to the result, we think that ECTP is an efficient protocol in transporting files to many receivers simultaneously.

• Journal of Korea Society of Digital Industry and Information Management
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• v.18 no.2
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• pp.1-8
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• 2022

The message propagation between vehicles must be efficiently performed to quickly transmit information between vehicles in vehicle ad hoc networks (VANETs). Broadcasting can be the most effective solution for propagating these messages. However, broadcasting can cause broadcast storm problems, which can reduce the performance of the overall network. Therefore, rapid information delivery in VANET requires a method that can propagate messages quickly without causing the broadcast storm problems. This paper proposes a lightweight and opportunistic broadcast (LOB) protocol that leverages the features of opportunistic routing to propagate messages quickly while minimizing the load on the network in VANETs where the network topology changes frequently. LOB does not require any routing information like greedy forwarding scheme, and neighboring nodes at the farthest distance within the range of transmission nodes are likely to be selected as forwarding nodes, and multiple forwarding candidate nodes like opportunistic routing scheme can increase packet transmission rates. Simulation results using ns-2 showed that LOB outperformed existing broadcast protocols in terms of packet rate and packet delay.

• Journal of Advanced Navigation Technology
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• v.14 no.2
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• pp.253-259
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• 2010

Wireless mesh network is flexible network like Ad hoc network or bluetooth together based on base station. But, wireless mesh network shows high packet loss and when TCP was created, however as it was design based on wired link, wireless link made more transmission error than wired link. It is existent problem of TCP congestion control algorithm that TCP unfairness and congestion collapse over wireless mesh network. When TCP operation occurs with the packet loss where is not the congestion loss, it brings the performance degradation which is serious. In this paper, in order to improve efficient TCP congestion control algorithm in wireless mesh network, we proposed that TCP can adaptively regulate the congestion window in wireless link.

• The Journal of The Korea Institute of Intelligent Transport Systems
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• v.9 no.2
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• pp.104-109
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• 2010

In this paper, we propose a user authentication technology to protect wiretapping and attacking from others in the telematics environment, which users in vehicle can use internet service in local area network via mobile device. In the proposed user authentication technology, the packet speech data is encrypted by speech-based biometric key, which is generated from the user's speech signal. Thereafter, the encrypted data packet is submitted to the information communication server(ICS). At the ICS, the speech feature of the user is reconstructed from the encrypted data packet and is compared with the preregistered speech-based biometric key for user authentication. Based on implementation of our proposed communication method, we confirm that our proposed method is secure from various attack methods.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2009.01a
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• pp.385-389
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• 2009

Multi-hop wireless mesh networks (WMNs) suffer from significant packet losses due to insufficient available bandwidth and high channel error probability. To conquer packet losses, end-to-end (E2E) error control schemes have been proposed. However, in WMNs, E2E error control schemes are not effective in adapting to the time-varying network condition due to large delay. Thus, in this paper, we propose a network-adaptive error control for video streaming over WMNs that flexibly operates E2E and hop-by-hop (HbH) error control according to network condition. Moreover, to provide lightweight support at intermediate nodes for HbH error control, we use path-partition-based adaptation. To verify the proposed scheme, we implement it and evaluate its transport performance through MPEG-2 video streaming over a real IEEE 802.11a-based WMN testbed.