{"title":"Overload Control in a SIP Signaling Network","authors":"Masataka Ohta","volume":12,"journal":"International Journal of Electronics and Communication Engineering","pagesStart":4089,"pagesEnd":4095,"ISSN":"1307-6892","URL":"https:\/\/publications.waset.org\/pdf\/10162","abstract":"
The Internet telephony employs a new type of Internet communication on which a mutual communication is realized by establishing sessions. Session Initiation Protocol (SIP) is used to establish sessions between end-users. For unreliable transmission (UDP), SIP message should be retransmitted when it is lost. The retransmissions increase a load of the SIP signaling network, and sometimes lead to performance degradation when a network is overloaded. The paper proposes an overload control for a SIP signaling network to protect from a performance degradation. Introducing two thresholds in a queue of a SIP proxy server, the SIP proxy server detects a congestion. Once congestion is detected, a SIP signaling network restricts to make new calls. The proposed overload control is evaluated using the network simulator (ns-2). With simulation results, the paper shows the proposed overload control works well.<\/p>\r\n","references":"[1] H. Schulzrinne and J. Rosenberg: \"The Session Initiation Protocol:\r\nInternet-Centric Signaling\", IEEE Communication Magazine, vol.38, 10,\r\npp-134-141, Oct.(2000)\r\n[2] J.Rosenberg, et.el.: \"SIP: Session Initiation Protocol\", RFC3261,\r\nhttp:\/\/www.ietf.org\/rfc\/rfc3261.txt, June(2002)\r\n[3] M.Ohta: \"Simulation study of SIP Signaling in an Overload Condition\",\r\n3rd Int-l Conf. on communications, Internet, and Information Technology,\r\npp.321-326, Nov.(2004)\r\n[4] M. Govind, S. Sundaragopalan, Binu K S, and Subir Saha: \"Retransmission\r\nin SIP over UDP - Traffic Engineering Issues\", Proc. of International\r\nConference on Communication and Broadband Networking, Bangalore,\r\nMay(2003)\r\n[5] J.H.James, B. Chen and L. Garrison: \"Implementing VoIP: A Voice Transmission\r\nPerformance Progress Report\", IEEE Communication Magazine,\r\nvol.42, 6, pp.36-41, June(2004)\r\n[6] Y. Xu, M. Westhead and F. Baker: \"An Investigation of Multilevel\r\nService Provision for Voice over IP Under Catastrophic Congestion\",\r\nIEEE Communication Magazine, vol.42, 6, pp.94-100, June(2004)\r\n[7] G. Camarillo, R. Kantola and H. Schulzrinne: \"Evaluation of Transport\r\nProtocols for the Session Initiation Protocol\", IEEE Network, Vol.17,\r\n5,pp.40-46, Sep.(2003)\r\n[8] T.Eyers and H. Schulzrinne: \"Predicting Internet Telephony Call Setup\r\nDelay\" Proc. 1st IP-Telephony Wksp.,Jan.(2000)\r\n[9] M. Cortes, J. R. Ensor and J. O. Esteban:\"On SIP Performance\", Bell\r\nLabs Tech. J, 9, pp.155-172(2004)\r\n[10] R.P.Ejzak, C.K.Florkey and R.W.Hemmeter: \"Network Overload and\r\nCongestion: A Cpmparison of ISUP and SIP\" Bell Labs Technical\r\nJournal, 9, pp.173-182(2004)\r\n[11] VINT Project: \"Network simulator ns-2\", http:\/\/www.isi.edu\/nsnam\/ns\/","publisher":"World Academy of Science, Engineering and Technology","index":"Open Science Index 12, 2007"}