• Journal of KIISE:Computing Practices and Letters
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• v.11 no.4
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• pp.312-333
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• 2005

SIP(Session Initiation Protocol) has been proposed for session control protocol of Internet multimedia communication system like VoIP(Voice over IP). SIP has complicated session control steps to support various kinds of audio and video formats and to assure service quality of real time data communication. Up until now, existing protocol analyzers can not provide such detailed information of SIP based communication system. In this paper, therefore, we propose a new protocol analyzer as a tool that can analyze and diagnose SIP based multimedia communication system throughout the session initiation, data exchange and session change steps. The propose traffic analyzer, which is called STAT(SIP based Traffic Analysis Tool), Is implemented on Winder's environment so that it is generally usable and extensible. Since STAT analyze low level packets captured via Ethernet broadcasting property, it is able to provide session status and real time traffic monitoring information without any affection to the communication system. The STAT which is implemented in this paper. therefore, is expected to be a useful tool for developing and managing of a SIP based multimedia communication system.

• The KIPS Transactions:PartC
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• v.9C no.3
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• pp.375-384
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• 2002

In the current Internet, the QoS of interactive applications is hardly guaranteed because of variable bandwidth, packet loss and delay. Moreover, VoIP which is becoming an important part of the information infra-structure in these days, is susceptible to network packet loss and end-to-end delay. Therefore, it needs error control mechanisms in network level or application level. The FEC-based error control mechanisms are used for interactive audio application such as VoIP. The FEC sends a main information along with redundant information to recover the lost packets and adjusts redundant information depending on network conditions to reduce the bandwidth overhead. However, because most of the error control mechanisms do not consider end-to-end delay but packet loss rate, their performances are poor. In this paper, we propose a new error control algorithm, SCCRP, considering packet loss rate as well as end-to-end delay. Through experiments, we confirm that the SCCRP has a lower packet loss rate and a lower end-to-end delay after reconstruction.

• Proceedings of the IEEK Conference
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• 2000.07b
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• pp.695-698
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• 2000

The services such as real-time audio and video applications have become increasingly popular, especially over the Internet. Furthermore, as being commercialized those contents on the Internet require quality of service (QoS) support to ensure their performance. PPP is the best solution to of for those kinds of services. The reason why we want to employ PPP is this satisfies most of the requirements associated with remote connectivity to an NSP, such as IP address assignment, security, and AAA (authentication, authorization and accounting). In addition, since ISPs and corporations are familiar with PPP based connectivity, easy migration from existing ISP infrastructure is expected, if QoS is guaranteed. But so for PPP has had no field to ensure the quality of service. This article presents the solution by using some tunneling protocols and the draft [1] that proposed additional LCP option fields to negotiate QoS. To communicate each other, after negotiating those option fields, over various protocols such as ATM, Ethernet, and etc. tunneling protocol is used. Following sections will mention those briefly. And the service provision to offer the end-to-end communication with negotiated QoS will also be proposed.

• Proceedings of the IEEK Conference
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• 2002.07b
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• pp.896-899
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• 2002

In this paper, we present the experimentation and demonstration results of Korea-Japan high-speed satellite ATM network using Ka-band Koreasat-3. This experimentation consists of two items - TCP/IP and MPEG-2 video/audio transmission over 155Mbps ATM based satellite network. The goals of this experimentation are to measure TCP performance when the only standard mechanisms approved by IETF in order to improve TCP performance in LFN(long fat network) are used and to derive the effects of quality for the high definition video stream when MPEG-2 TS is transmitted through 155Mbps satellite ATM link. With on the results of the experiments, we demonstrated the applications suitable to the high-speed satellite ATM network. The first TCP/IP and MPEG-2 transmission experiments were done at the rate of 155Mbps using Ka-band KOREASAT-3 between Korea and Japan, and its results will be demonstrated with the ATM-based 3D-HDV(3 dimensional High definition video) and HDTV during 2002 Korea-Japan World Cup Soccer Game.

• The KIPS Transactions:PartC
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• v.12C no.2 s.98
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• pp.217-222
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• 2005

The IEEE 802.11 based wireless local area networks are candidates to lead the broadband connectivity in the home and office scenarios. Recently IEEE proposed the 802.11e as a new standard to provide appropriate Quality of Services to a plethora of emerging real-time multimedia and high demanding applications such as high definition movie and audio distribution, video-conference and voice over IP. This paper studies the IEEE 802.11e/IEEE 802.11b interactions focusing on potential unfairness problems that might appear in networks with heterogeneous wireless LAN technologies as well as in the IEEE 802.11e deployment phase.

• Proceedings of the KIEE Conference
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• 2004.11c
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• pp.86-88
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• 2004

As the rapid incensement of the number of internet users has occurred recently, many multimedia application services have been emerging. To improve quality of service, traffic can be suggested to be classified with priority in EPON(Ethernet Passive Optical Network), which is supporting the multimedia application services. In this paper, multimedia application services treat bandwidth classifying device in serving both delay sensitive traffic for real-time audio, video and voice data such as VoIP(Voice over Internet Protocol), and nonreal-time traffic such as BE(Best Effort). With looking through existing mechanisms, new mechanism to improve the quality will be suggested. The delay performances and packet losses of traffic achieved by supporting bandwidth allocation of upstream traffic in suggested mechanism will be analyzed with simulation.

• The Transactions of the Korea Information Processing Society
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• v.5 no.12
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• pp.3047-3062
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• 1998

This paper deseribes the desihn and the implementation of the Realtime Transport Protocol(RTP)/ Rdaltime Control Protocol(RTCP) (RFC 1889,1890) that is used to transmit the audio/video data to any destination and to feedback the Quality of Service (QoS) information of the received media data to the sender, in the teleconferencing systems proposed by ITU-T. These protocols are implemented with multi thead technique and run on top of UDP/IP-Multicast through the socket interface as the underlying protocol. The upper layer is impelmented such that in can be accessed by the H245 comference control protocol. The RTP packetizes the digitized audio/video data from the encoder info a fixed format, and multieast to the participants. The RTCP monitors RTP packets and extracts the QoS values from it such as round-trip delay, jiter and packet loss to form RTCP packets and non periokically sends them to the sender site. In this Paper, we also descritx the study of measurement and analysis for QoS factors that observed on performing teleconferencing system over Internet. The results from this experiment is indicate that RTT and Jitter value are acceptable even entwork load is high. However, it appears that packet loss rate is high in daytime and most losses periods have length one or two.

• The KIPS Transactions:PartC
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• v.13C no.4 s.107
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• pp.471-480
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• 2006

Recently, multimedia communication services, such as video conferencing and voice over IP, have been rapidly spread. H.323 is an international standard that specifies the components, protocols and procedures that provide multimedia communication services of real-time audio, video, and data communications over packet networks, including IP based networks. H.323 is applied to many commercial services because it supports various network environments and has a good performance. But communication services based on H.323 may have some problem because of current network trouble or mis-implementation of H.323. The understanding of this problem is a critical issue because it improves the quality of service and is easy to service maintenance. In this paper, we implement the analysis system for H.323 protocol wihch includes H.245, H.225.0, RTP, RTCP, and so on. Tills system is able to capture, parse, and present the H.323 protocol in real-time. Through the operation test and performance evaluation, we prove that our system is a useful to analyze and understand the problems for communication services based on H.323.

• Journal of the Institute of Electronics Engineers of Korea SD
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• v.41 no.5
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• pp.91-97
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• 2004

A digital over-sampling noise-shaping coder to achieve the processing accuracy for the audio signal bandwidth is designed. In order to implement an optimized design of the noise-shaping coder as a form of U (intellectual property), circuit design techniques that optimize the multiplication and the ROM architectures are proposed with emphasis on the low-voltage operation under 2.0 V and the minimization of the hardware resources. In the design and verification methodology, the overall architectures and the internal bit width have been determined through behavioral simulations. The overall performances including timing margin have been estimated through transistor-level simulations. Furthermore, the test results of the implemented chip using a 0.35-${\mu}{\textrm}{m}$ standard CMOS process proposed the validity of the proposed circuits and the design methodology.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.8B
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• pp.507-513
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• 2005

As the rapid increasement of the number of internet users has occured recently, many multimedia application services have been emerging. To improve quality of service, traffic can be suggested to be classified with priority in EPON(Ethernet Passive Optical Network), which is supporting the multimedia application services. In this paper, multimedia application services treat bandwidth classifying device in serving both delay sensitive traffic for real-time audio, video and voice data such as Von(Voice over Internet Protocol), and for real-time traffic such as BE(Best Effort). With looking through existing mechanisms, new mechanism to improve the quality will be suggested. The delay performances and packet losses of traffic achieved by supporting bandwidth allocation of upstream traffic in suggested mechanism will be analized with simulations.