VoIP Websites

Peer to peer voice service. Users may call landlines and cellphones for a fee; users may call each other for free. Source code is not available.

Open Source telephony switching and private branch exchange (PBX) daemon. Supported signalling protocols: H.323, SIP, MGCP, SCCP (Cisco Skinny).

Videoconferencing application to make audio and video calls to remote users. Supports both SIP and H.323.

Free H.323 gatekeeper (GPL) based on OpenH323.

SIP proxy server. It can operate as registrar server, redirect server and stateful proxy server. SIP is an open standard (IETF) replacement for H323.

Implementation of Session Initiation Protocol (SIP). This library provides an interface to initiate and control SIP based sessions.

A free, scalable telecommunications application server by the GNU Project.

Proxy/masquerading daemon for the SIP protocol. It handles registrations of SIP clients on a private IP network and performs rewriting of the SIP message bodies to make SIP connections work via a masquerading firewall (NAT).

SIP based web phone. Supported audio codecs are G711, LPC10-15, GSM, and SPEEX.

SIP based VoIP user agent. It supports Presence and Instant Messaging.