ABSTRACT
Session Initiation Protocol (SIP) has been widely used in Internet telephony and has been chosen as the protocol for Internet Protocol (IP)-based multimedia call control for third generation (3G) wireless networks. As defined in RFC 3261 [1], SIP is a client server-based control protocol above the transport layer for creating, modifying, and terminating sessions between two or more participants. However, SIP-based telephony can be viewed as an application of peer-to-peer (P2P) architecture where the user agents (UAs) form a self-organizing P2P overlay network to locate and communicate with each other. In consideration of the characteristics of P2P, it is expected to be a perfect method to solve the scalability, robustness, and fault tolerance problems in traditional SIP networks. Skype is a free P2P application based on the Kazaa architecture for Internet telephony and instant messaging [2,3]. The protocol is proprietary and the system has centralized elements for login authentication [2]. Researchers have proposed several pure P2P architectures for SIP-based IP telephony systems [4,5]. The P2P SIP working group has been formed within the Internet Engineering Task Force (IETF) for adapting P2P with features suitable for SIP.
