• 한국정보통신설비학회:학술대회논문집
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• 2007.08a
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• pp.136-141
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• 2007

The frequency accuracy and phase alignment is necessary for ensuring the quality of service (QoS) for applications such as voice, real-time video, wireless hand-off, and data over a converged access medium at the telecom network. As telecom networks evolve from circuit to packet switching, proper synchronization algorithm should be meditated for IP networks to achieve performance quality comparable to that of legacy circuit-switched networks. The Time of Packet (ToP) specified in IEEE 1588 is able to synchronize distributed clocks with an accuracy of less than one microsecond in packet networks. But, The ToP can be affected by impairments of a network such as packet delay variation. This paper proposes the efficient method to minimize the expectable delay variation when ToP synchronizes the distributed clocks. The simulation results are presented to demonstrate the improved performance case when the efficient ToP transmit algorithm is applied.

• 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 Journal of the Korea Contents Association
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• v.1 no.1
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• pp.130-136
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• 2001

Real-time applications like VoIP in mobile networks need smooth handovers in order to minimize or eliminate packet loss as a mobile host(MH) transitions between network links. In this paper, we design a new variable buffering mechanism for IPv6 by which an MH can request that the router on its current subnet buffers pad(eta on its behalf while the MH completes registration procedures with the router of a new subnet. Performance results show that our proposed queueing scheme with a variable space allocation is quite appropriate for mobile internet environment in terms of the packet loss rate.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2003.10a
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• pp.176-179
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• 2003

The use of Multiparty Conference service based on SIP for VoIP provides is gradually magnified, and the work for continuous development and standardization on SIP is in the process of advancing. But, currently it is impossible for SIP to support identity discovery and distribution of each participant for multiparty conference. In this paper, we propose a SIP extension for guaranteeing security on the multiparty conference using SIP by adding new method and reconstructing header informations. With this, it is also possible to identify discovery and to distribute each participant using SIP extension when a call is established for closed multiparty conference.

• International journal of advanced smart convergence
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• v.12 no.3
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• pp.1-12
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• 2023

A public address (PA) system installed in a building is a system that delivers alerts, announcements, instructions, etc. in an emergency or disaster situation. As for the products used in PA systems, with the development of information and communication technology, PA products with various functions have been introduced to the market. PA systems recently launched in the market may be connected through a single network to enable efficient management and operation, or use voice recognition technology to deliver quick information in case of an emergency. In addition, a system capable of locating a user inside a building using a location-based service and guiding or responding to a safe area in the event of an emergency is being launched on the market. However, the new PA systems currently on the market add some functions to the existing PA system configuration to make system operation more convenient, but they do not change the complex PA system configuration to reduce facility costs, maintenance, and management costs. In this paper, we propose a novel PA system configuration for buildings using audio networks and control hierarchy over peer-to-peer (Anchor) technology based on audio over IP (AoIP), which simplifies the complex PA system configuration and enables convenient operation and management. As a result of the study, through the emergency signal processing algorithm, fire broadcasting was made possible according to the detection of the existence of a fire signal in the Anchor system. In addition, the control device of the PA system was replaced with software to reduce the equipment installation cost, and the PA system configuration was simplified. In the future, it is expected that the PA system using Anchor technology will become the standard for PA facilities.

• The KIPS Transactions:PartC
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• v.13C no.4 s.107
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• pp.427-434
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• 2006

Broadband Convergence Network(BcN) is a critical infrastructure to provide wired-and-wireless high-quality multimedia services by converging communication and broadcasting systems, However, there exist possible danger to spread the damage of an intrusion incident within an individual network to the whole network due to the convergence and newly generated threats according to the advent of various services roaming vertically and horizontally. In order to cope with these new threats, we need to analyze the vulnerabilities of BcN in a system architecture aspect and classify them in a systematic way and to make the results to be utilized in preparing proper countermeasures, In this paper, we propose a new classification of vulnerabilities which has been extended from the ITU-T recommendation X.805, which defines the security related architectural elements. This new classification includes system elements to be protected for each service, possible attack strategies, resulting damage and its criticalness, and effective countermeasures. The new classification method is compared with the existing methods of CVE(Common Vulnerabilities and Exposures) and CERT/CC(Computer Emergency Response Team/Coordination Center), and the result of an application to one of typical services, VoIP(Voice over IP) and the development of vulnerability database and its management software tool are presented in the paper. The consequence of the research presented in the paper is expected to contribute to the integration of security knowledge and to the identification of newly required security techniques.

• Journal of the Institute of Electronics and Information Engineers
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• v.52 no.1
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• pp.32-39
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• 2015

The voice communications with wireless backbone based networks are evolving into a packet switching VoIP systems. In those networks, a call processing scheme is required for management of subscribers and connection between them. A VoIP service scheme for those systems requires reliable subscriber management and connection establishment schemes, but the conventional call processing schemes based on the centralized server has lack of reliability. Thus, the mutual-backup architecture of SIP-servers is required to ensure efficient subscriber management and reliable VoIP call processing capability, and the synchronization and call processing schemes should be changed as the architecture is changed. In this paper, a mutual-backup architecture of SIP-servers is proposed for wireless backbone based networks. A message format for synchronization and information exchange between SIP servers is also proposed in the paper. This paper also proposes a FSM scheme for the fast call processing in unreliable networks to detect multiple servers at a time. The performance analysis results show that the mutual backup server architecture increases the call processing success rates than conventional centralized server architecture. Also, the FSM scheme provides the smaller call processing times than conventional SIP, and the time is not increased although the number of SIP servers in the networks is increased.

• Journal of IKEEE
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• v.5 no.1 s.8
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• pp.8-15
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• 2001

This paper remarks IETF based requirements for IP/UDP/RTP header compression issued in 3GPP2 All IP Ad Hoc Meeting and protocol stacks of the next generation mobile station. All IP Network, for real time application such as Voice over IP (VoIP) multimedia services based on 3GPP2 3G cdma2000. Frames for various protocols expected in the All IP network Mobile Station (MS) are explained with several figures including the bit-for-bit notation of header format based on IETF draft of Robust Header Compression Working Group (ROHC). Especially, this paper includes problems of IS-707 Radio Link Protocol (RLP) for header compression which will be expected to modify in All IP network MS's medium access layer to accommodate real time packet data service[1]. And also, since PPP has also many problems in header compression and mobility aspects in MS protocol stacks for 3G cdma2000 packet data network based on Mobile IP (PN-4286)[2], we introduce the problem of solution for header compression of PPP. Finally. we suggest the guidelines for All IP network MS header compression about expected protocol stacks, radio resource efficiency and performance.

• 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.

• The Journal of the Korea Contents Association
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• v.4 no.2
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• pp.43-49
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• 2004

SIP Caller Preference is an useful function that allows a caller to express preferences about request handling in servers. It can also feat appropriate call processing according to the callee capabilities. However, only the category of the media is considered as a criteria for target selection in the caller preference. In this case, if the callee's media information such as codec is different from the caller, an additional re­negotiation occurs for SIP call setup. Therefore, in this paper, we have suggested an extended caller preference to solve this problem. In our SIP based VoIP system, a network sewer uses detailed media informations for media stream in the session to select the target for SIP call setup. The sewer gives higher priority to the candidate which do not require re-negotiation for call setup, so that an effective call setup can be achieved in our system.