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

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

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

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.

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

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

SIP based phone. Supports video, communication encryption and push-to-talk.

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

Mini-HOWTO for NAT and H.323 - configuring Linux to support H.323 while using network address translation.

Program that allows two parties to hold a secure voice conversation using Linux and TCP/IP.