• Journal of Korea Multimedia Society
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• v.15 no.8
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• pp.1038-1047
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• 2012

The SIP(Session Initation Protocol) is an application layer protocol that is used to establish, release, and change the call session of the IP telephony. In the SIP signalling networks, when the number of the UA(User Agent) requested the call session increase, the number of messages to be processed by SIP proxy server increase. It often will be caused the overload of the SIP proxy server. In this paper, we proposed the overload control method with a normal queue and a delay queue in the SIP proxy server. When it is estimated the overload of the server by the excess of the high threshold in the normal queue, new INVITE messages will be put into the delay queue to reduce the load of the server. It results in some delay of the call session from the INVITE message. Subsequently when the number of messages in the normal queue is reduced below the low threshold, the INVITE messages in the delay queue is processed. The simulation results showed that the number of the retransmission messages by our proposed method was 45% less than the one by the method with single queue. The results also showed that the average call success rate by the proposed method was 2% higher than the one by the method with single queue.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.10 no.12
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• pp.2321-2328
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• 2006

Today, SIP standard (The signalling protocol for the Internet phone service) raises to be the standard technique because the expandability is high and complexity is low. It is widely investigated and actively advocated to use Si81a1 ring protocol for SIP in VoIP service. SIP service can be applied even outside the Internet phone service; instance messaging and various multimedia technology are just an example. This paper proposed an embodiment proxy server for rambling support to use JAIN SIP API. It provides standard interface for testing the Proxy server for SIP and embodiment of user agent that transfer instant massaging and voice communication.

• 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.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.11A
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• pp.1051-1058
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• 2010

Mobile IP (MIP) and SIP are considered as important technologies to provide the macro mobility in the next generation mobile convergence networks which have heterogeneous access networks. Typically, MIP and SIP are more suitable for the non-real-time TCP connections and the real-time RTP/UDP sessions respectively, hence a handset which uses both of these sessions should simultaneously apply MIP and SIP to perform the efficient mobility management. Existing multi-layered mobility management schemes focus on the signalling order of each protocol. However, simple combining of two protocols cannot provide the performance enhancement of the mobility management. In this paper, a novel multi-layered mobility management algorithm using the combination of SIP and fast MIPv6 (FMIPv6) is proposed. FMIPv6 and SIP mobility is simultaneously performed to reduce the service interrupt time and to guarantee QoS requirement. The delay model is defined to analysis the performance of the algorithm and the simulation results show the performance of the proposed algorithm.