P2P-SIP
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Our P2P-SIP
 . Overview
 . Architecture
 . Implementation

P2P-VoIP links
 . p2psip.org
 . P2P-SIP blog
 . mailing list
 . IETF meeting

P2P VoIP products
 . Skype
 . Nimcat
 . Teleo
 . Peerio

Summary

P2P systems inherently have high scalability, robustness and fault tolerance because there is no centralized server and the network self-organizes itself. This is achieved at the cost of higher latency for locating the resources of interest in the P2P overlay network. Internet telephony can be viewed as an application of P2P architecture where the participants form a self-organizing P2P overlay network to locate and communicate with other participants. We propose a pure P2P architecture for the Session Initiation Protocol (SIP)-based IP telephony systems. Our P2P-SIP architecture supports basic user registration and call setup as well as advanced services such as offline message delivery, voice/video mails and multi-party conferencing.

My publications and talks

Implementation

I have implemented a P2P-SIP adaptor, SIPpeer, that allows existing or new SIP user agents to connect to the P2P-SIP network without modifying the user agent. SIPpeer can also act as a SIP user agent, proxy or registration server with command line user interface. Currently, I am modifying SIPpeer for authentication and NAT/firewall traversal.

The current design document is available as an Implementation Report titled "SIPpeer: A SIP-based peer-to-peer Internet telephony client adaptor".

I have also implemented use of an external DHT in SIPpeer as well as in our SIP user agent, SIPc. The design document is available as Technical Report titled "Using an External DHT as a SIP Location Service".

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Related work

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