• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.1B
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• pp.101-108
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• 2004

SIP(Session Initiation Protocol) is a suitable protocol for supporting Internet telephony services and currently market requirements fur developing supplementary telephony services such as unconditional call forwarding, call forwarding on busy or no answer, call filtering services have recently grown. CPL(Call Processing Language) is a standard technology that can be used to describe and control internet telephony services. In this paper, we describe the CPL system for supplementary Internet telephony services using SIP as an application level call signaling protocol. Those supplementary services are composed of CPL client which is a SIP UA, SIP Proxy server, Registrar and CPL server In this paper, we describe the design and implementation of the CPL server system in detail which is developed in Linux 7.2 using C and C++ programming languages.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.19 no.5
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• pp.1144-1150
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• 2015

Presence information provides various informations about users such as on-line status, current location, network connection method and connection address, and there are two kinds of presence information, SIP(Session Initiation Protocol) based presence information and XMPP(Extensible Massaging and Presence Protocol) based presence information. In this paper, a multiple server architecture that can handle these two kinds of presence information has been proposed. In this architecture, severs are added dynamically according to number of users to provide system scalability, and load of each server can be effectively controlled. In this system, a new XMPP stanza architecture and presence information data format are designed for load control. Furthermore message exchanging procedures between servers and users for dynamic server control has been also suggested. The performance of the proposed system has been analysed by simulation.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.20 no.12
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• pp.2252-2258
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• 2016

CCMP(centralized conferencing manipulation protocol) enables adding and removing conference participants, changing their roles, adding and removing media streams in conference system. In this paper, by using extended CCMP, a new distributed conference system architecture which can be used to multiple servers distributed conference system in SIP(session initiation protocol) environment has been presented. In this study, according to increasing number of participants, a new extended CCMP architecture which can distribute conference system loads to multiple servers dynamically to decrease loads of servers has been designed. This extended CCMP architecture also can add dynamically new servers from the prepared servers pool. Furthermore, new conference information data format which can represent extended CCMP has been designed, and exchange procedures of extended CCMP control messages which can distribute loads between servers have also been presented. The performance of the proposed system has been analysed by simulation.

• The Journal of the Korea Contents Association
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• v.8 no.5
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• pp.1-8
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• 2008

The centralized conference architecture, one of the conference architectures in SIP(Session Initiation Protocol) environment, is widely used as it has the advantage of conference management and control. However it has been limited in scalability. Therefore we have proposed an efficient distributed conference architecture to improve scalability of centralized conference model. In our architecture, if the number of conference participants exceeds the predefined maximum number, a new conference server is added to the conference dynamically. In this case, the focus of existing server acts as primary focus and the focus of added server acts as secondary focus, and dynamic reallocation of participants between servers is done to equally divide the loads. This process is repeated as the number of conference participants increases. For this behavior, we have proposed procedure of adding the conference server, SIP call signal exchange, signaling procedure for RTP(Real Time Transport Protocol) sessions between conference servers, and procedure of conference event package between conference servers. The performance of our proposed model is evaluated by experiments.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.12 no.12
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• pp.2301-2306
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• 2008

SIP(Session Initiation Protocol) has become an universal standard for multimedia communications for both wired and wireless networks since it has been adopted as a standard protocol for IMS platform in 3GPP standardization organization at November 2000. In this paper, we design and implement a SIP-based softphone client program which provides telephony service between internet users and a call center equipped with VoIP gateway. A softphone client based on PC-to-phone connection should guarantee to provide interoperability with various VoIP gateways and higher portability to be able to operate on different PC environments. The softphone client program in this paper has been developed with SIP 2.0 standard protocol to support interoperability and with JAIN SIP and JMF package to achieve higher portability.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.21 no.7
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• pp.1335-1341
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• 2017

In this paper, a new architecture of multiple conference servers which use extended control channel framework in SIP(session initiation protocol) session has been presented. For this purpose, in this study, a new extended control channel framework architecture which can distribute total conference system loads to multiple servers effectively has been presented. In the implementation, extended control channels have been connected by using SIP sessions that was established between each conference servers, and extended control channel messages which can be transferred through control channels have been designed in this study. These extended control channel messages can distribute system load effectively between multiple conference servers, and conference information data format that can represent extended control channel framework has also been designed. Furthermore, exchange procedures of extended control channel messages have also been presented. The performance of the proposed system has been analysed by simulation. The analysis results show that average SIP messages delay time and average media stream delay time have improved.

• The Journal of the Korea Contents Association
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• v.6 no.2
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• pp.17-26
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• 2006

SIP(Session Initiation Protocol)-based centralized conference service model has advantage of easiness in conference management and service as compared to other models. However when media server which is one of the components of conference server is included in the conference server, this model shows disadvantage of high server work load with increasing numbers of conferences and participants. In this paper, to improve this problem, we have suggested and implemented a new conference service model in which an UA(User Agent) who first make the conference acts as a media server instead of conventional conference server and, the conference server with conference event package takes only part of management of the conference and its participants. Therefore, many more conferences can be held and managed compared to the conventional centralized conference service model because load of the conference server decreases in our suggested model md, furthermore the participants needs only make SIP call connections to the UA who first make the conference for establishing media session.

• Proceedings of the Korea Contents Association Conference
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• 2006.11a
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• pp.26-30
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• 2006

Internet phone service is a new service technology that provides voice call services through Internet not through the pre-existing PSTN. It enables a cheap voice call service regardless of distance. We may expect that the Internet phone service may substitute for the voice call service through the PSTN, but not in a short period. There are several problems to be solved for this transition, such as, voice call quality, numbering scheme, billing, standardization, and support of several functions. In this paper, we provided and designed a UA (User Agent) that can support functions regarding voice call, such as call forwarding, auto-connection, call rejection and restriction of individual call, using SIP (Session Initiation Protocol) which is proposed by SIP-Working Group as the standard Internet phone service management protocol.

• Journal of KIISE:Information Networking
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• v.34 no.1
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• pp.73-80
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• 2007

This paper presents an approach to support dynamic QoS(Quality of Service) requirements of largely emerging SIP(Session Initiation Protocol) multimedia applications in broadband access networks. The topology of QoS-enabled broadband access networks and its operational model to support SIP QoS are firstly suggested. And then the procedures to bind QoS-enabled SIP signaling into an IP QoS mechanism are presented. In this paper, DiffServ-based IP QoS architecture is deployed due to the complexity problem of the other IntServ architecture, and COPS(Common Open Policy Service) and COPS-PR protocol based signaling mechanisms are used to support dynamic DiffServ QoS, correspondent with dynamic SIP QoS. The broadband access network is assumed to support rapidly expanding Metro Ethernet 802.1 D/Q QoS, and how to translate SIP QoS parameters into IP DiffServ classes and DiffServ classes into 802.1 D/Q QoS parameters is also presented in this paper.

• The KIPS Transactions:PartC
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• v.13C no.4 s.107
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• pp.461-470
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• 2006

Recently mobile Stream Control Transmission Protocol (mSCTP) has been proposed as a transport layer approach for supporting mobility. When a mobile terminal (MT) is not located in the home network. a terminal that wishes to communicate with the MT is not able to establish mSCTP association to the MT, since mSCTP does not include the location management mechanism. In order to solve this problem. an interworking approach using the Session Initiation Protocol (SIP) INVITE method has been proposed. However, this approach has shown subsequent delay in acquiring the current location information of the MT when initiating mSCTP association establishment. In this paper, we propose new SIP methods and an approach that minimizes the address acquisition delay (AAD) by utilizing those SIP methods. Mathematical analysis and simulation results show that the proposed approach is more efficient than the previous approach in terms of AAD in all kinds of SIP environments.