• ETRI Journal
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• v.29 no.6
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• pp.745-754
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• 2007

In this paper, we present the Enhanced Communication Transport Protocol-Part 5 (ECTP-5), which provides scalable and reliable multicast communication service for many-to-many applications by constructing high quality recovery trees from two-layer logical trees and repairing the losses via unicast automatic repeat request-based error control. In order to realize the protocol, we developed feasible protocol architectures and building blocks including additional functions which deal with engineering details, such as membership dynamics and sender coordination. Experimental results show that ECTP-5 scales well with various session sizes and packet loss rates in terms of control overhead and recovery latency.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.21 no.9
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• pp.2415-2421
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• 1996

This paper describes implementation and performance evaluation of XTP(Xpress Transport Protocol) onthe Windows NT for multicasting in high-speed communication networks. We designed the protocol byan event-driven method and implemented it in form of network driver in a kernel for performace enhancement. Various applications program are used for its functional test and comparison with the TCP protocol.

• Proceedings of the IEEK Conference
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• 2000.06c
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• pp.141-144
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• 2000

Current Internet and Intranet developments are focusing on network multimedia services involving the transport of real-time multimedia streams. In this paper, we present a real-time MPEG2 Encoding/Decoding systems and a MPEG2 stream player running on client systems based on real-time transport protocol(RTP). The MPEG2 player has been implemented by using JMF and java applet. We have tested our prototype systems with one of possible multimedia sources, camcoder, to verify its functional correctness and measure its performance.

• Journal of the Korea Society for Simulation
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• v.17 no.4
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• pp.153-158
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• 2008

The technology of Software Defined Radio (SDR) is a possible solution to interwork flexibly between various wireless transport protocols. Ubiquitous network, like u-health service network, includes sensor devices or nodes which do not facilitate all the same transport protocols to access network. As such this may be in such unreachable situations as poverty of all required AP (Access Point)’s, faults or contention in a path of particular protocol communication, etc. This paper presents research results of modeling and simulation to analyze transport performance of multi-protocol ubiquitous network which includes SDR-based interwork nodes and congestion-controlled AP’s. Focusing mainly on dynamics of overall transport performance rather than protocol execution procedures, this paper employs the Zeigler’s DEVS (Discrete Event Systems Specification) methodology and DEVSim++simulation environment to experiment.

• 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.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.4B
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• pp.227-233
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• 2007

According to the recent development of network technology, the demands of users are diversified and the needs of multimedia traffic are increasing. In general, UDP(User Datagram Protocol) traffic is used to transport multimedia data, which satisfied the real-time and isochronous characteristics. UDP traffic competes with TCP traffic and incur the network congestion. However, TCP traffic performs network congestion control but does not consider the receiver's status. Thus, it is not appropriate in case of streaming services. In this paper, we solve a fairness problems and proposed a network algorithm based on RTP/RTCP(Real-time Transport Protocol/Realtime Transport Control Protocol) in view of receiver status. The POBA(Priority Ordered Buffer Algorithm), which applies priorities in the receiver's buffer and networks, shows that it provides the appropriate environment for streaming services in view of packet loss ratio and buffer utilization of receiver's buffer compared with the previous method.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.9 no.8
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• pp.1696-1702
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• 2005

Real Time Transport Protocol (RTP) is the Internet standard protocol for transport of real time data audio/video IP Telephony, Multimedia Seivece. In case of 8kbps voice codec, the size of packet per data is 20bytes and become more large to minimal 40bytes with adding each layer's header in RTP/UDP/IP. To solve this problem, various header compression skill were suggested on point-to-point networks. But it compress even IP header and cannot be suitable to apply to end-to-end network Thus, We will renew header compression protocol to apply wired router-based network.

• Journal of IKEEE
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• v.11 no.4
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• pp.331-339
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• 2007

The problem of architecting a reliable transport system across an overlay network using split TCP connections as the transport primitive is mainly considered. The considered overlay network uses the application-level switch in each intermediate host. We first argue that natural designs based on store-and-forward principles that are maintained by split TCP connections of hop-by-hop approaches. These approaches in overlay networks do not concern end-to-end TCP semantics. Then, a new transport protocol-Overlay Transport Protocol (OTP)-that manages the end-to-end connection and is responsible for the congestion/flow control between source host and destination host is proposed. The proposed network model for the congestion and flow control mechanisms uses a new window size-Ownd-and a new timer in the source host and destination host. We validate our analytical findings and evaluate the performance of our OTP using a prototype implementation via simulation.

• Journal of KIISE:Computing Practices and Letters
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• v.7 no.1
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• pp.24-37
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• 2001

This paper deals with our experience for the design and implementation of Transport and Session managers in the Integrated Streaming Service Architecture (IS SA) that was developed by the authors. The ISSA is a streaming framework that allows to develop integrated multimedia streaming applications such as VOD system in unicast/multicast and real-time broadcastOive-cast). It also facilities standard real-time transport protocols such as RTp(Real-Time Transport Protocol)/RTCP(Real-time Control Protocol) and RTSP(Real-Time Streaming ProtocoD that allows to user openness. The ISSA supports diverse media fonnats and is independent from underlying networks and operating systems, and compatible with the global real-time multimedia database system (BeeHive) so that streaming media are efficiently retrieved, stored, and serviced. The role of the Transport Manager is to do packetization and depacketization for the different types of multimedia data and delivers the packetized media data in real-time. The Transport Manager uses RTP protocol. The role of the Session Manager is to establish and control the media channel by using RTSP protocol and to deliver the database transactions for the multimedia database by using RTTP(Real-Time Transaction Protocol) protocol. Both the Transport and Session Manager are doing their functions through the network interface in the ISSA that allows developers to various network programming interfaces and provides flexibility to the system.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.11B
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• pp.989-997
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• 2008

The reverse acknowledgment (RA) scheme supports a fast loss recovery for negative acknowledgment (NAK)-based selective repeat automatic repeat request (SR-ARQ) by detecting a retransmission failure quickly before a retransmission timer expires. In this paper, we evaluate the performance of a NAK-based SR-ARQ protocol with the RA scheme and compare it with the conventional NAK-based SR-ARQ protocol. Particularly, we propose a simple analysis model for the transport delay of the NAK-based SR-ARQ protocol considering the traffic condition, the retransmission persistence, the timer-based retransmissions and the RA scheme's behavior. Both NAK-based SR-ARQ protocols with and without the RA scheme are implemented by using the OPNET simulator. We verified the analysis model's accuracy through the simulation results. Also, we evaluate the performance of the NAK-based SR-ARQ protocol with the RA scheme based on analysis and simulation results.