1 NTP VoIP Platform: A SIP VoIP Platform and Its Services 1 Whai-En Chen, Chai-Hien Gan and Yi-Bing Lin Department of Computer Science National Chiao Tung University 1001 Ta Hsueh Road, Hsinchu, Taiwan, R.O.C., 300 Abstract: - This paper introduces the major components of a SIP-based VoIP platform, referred to as NTP VoIP platform. Based on the NTP VoIP platform, the researchers can develop and deploy their applications and services. For lawful interception, we propose a monitoring system that provides call detail records and interception function. We also propose a conference system for audio and video conferences. In this paper, we provide detailed message flows to show how the monitoring system and the conference system work. Key-Words: - CDR (Call Detail Record), LI (Lawful Interception), VoIP, SIP, RTP 1 This work was sponsored in part by NICI IPv6 Project, NTP VoIP Project 95-2219-E-009-010, 94-2219-E-009-001, and ITRI/NCTU Joint Research Center. 1 Introduction Many applications have been developed for Internet Protocol (IP) networks. Among them, Voice over IP (VoIP) is one of the most important applications. Session Initiation Protocol (SIP), which supports functions to integrate instant messaging, user presence, and multimedia communications [1], is the most popular signaling protocol for VoIP call control. Under the National Telecommunication Development Program (NTP), we have established a VoIP platform referred to as NTP VoIP platform that allows researchers and students to develop and deploy SIP-based VoIP applications. Figure 1. NTP VoIP Platform Figure 1 illustrates the NTP VoIP platform architecture. The components are described as follows. SIP server (Figure 1(1)) provides primary capabilities for call-session control in NTP VoIP platform. A SIP server processes SIP request and response messages as a SIP proxy. It also plays a role as a SIP registrar to store the contact information (e.g., the IP address) of each VoIP user. Two types of SIP servers are deployed in NTP VoIP platform, including the call server developed by Industrial Technology Research Institute (ITRI) and the SIP Express Router (SER) developed by iptel. The ITRI call server is a commercial product and provides a convenient Operation, Administration and Maintenance (OAM) interface. The SER is an open-source server deployed on a Linux or a BSD system. In this paper, we utilize SER to develop a monitoring system and a conference system. PSTN gateway (Figure 1(5)) supports interworking between NTP VoIP platform (i.e., an IP network) and the Public Switched Telephone Network (PSTN), which allows VoIP phone users to reach other PSTN users. Three PSTN gateways have been deployed in NTP VoIP platform, including the Vontel gateway developed by ITRI, Cisco 2621 and Cisco 3700. The Vontel gateway and Cisco 3700 provide E1 interfaces and each E1 interface can support 30 concurrent users. The Cisco 2621 provides four Foreign eXchange Office (FXO) interfaces that support 4 concurrent users. The SIP server dispatches the calls to these gateways based on a load balancing mechanism. SIP User Agent (UA; Figure 1(4)(6)(7)(8)) is a hardware-based or a software-based SIP phone that provides several call functions such as dial, answer, reject, hold/unhold, and call transfer. In Proceedings of the 5th WSEAS International Conference on Applied Computer Science, Hangzhou, China, April 16-18, 2006 (pp756-761)