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Thanks to IP (Internet Protocol) telephony, voice, data, and video can be exchanged using the same communication infrastructure. Most Internet users have become familiar with IP telephony thanks to the popularity of applications like Skype. VoIP relies on a variety of protocols, including signaling protocols such as SIP, H323 and media stream transport protocols such as RTP. Our work is mainly based on SIP (Session Initiation Protocol). In order to circumvent all drawbacks of centralized systems on which SIP relies, a Peer-to-Peer (P2P) SIP-based communication system was proposed. Many research activities are being carried out on the integration of P2P principles into SIP to capitalize on the benefits of decentralization. We contribute to research activities on P2P SIP by proposing a novel architecture that achieves interoperability and ensures portability. Moreover, our proposal contains efficient proxy topology building and routing mechanisms. Furthermore, we define a NAT traversal solution to allow effective transmission of SIP signaling messages and media streams in presence of NAT devices. Finally, we propose a secure signaling messages routing scheme in our P2P SIP framework.