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See also:
- Asterisk - Open Source telephony switching and private branch exchange (PBX) daemon. Supported signalling protocols: H.323, SIP, MGCP, SCCP (Cisco Skinny).
- Ekiga - Videoconferencing application to make audio and video calls to remote users. Supports both SIP and H.323.
- Gnome-o-Phone - Graphical VoIP software for the Linux Gnome desktop. Includes RtpTunnel utility for tunneling RTP UDP-based traffic through a firewall.
- GNU Bayonne - A free, scalable telecommunications application server by the GNU Project.
- GNU oSIP - Implementation of Session Initiation Protocol (SIP). This library provides an interface to initiate and control SIP based sessions.
- isdn2h323 - Project homepage for isdn2h323, a Linux-based ISDN to H.323 gateway.
- KPhone - SIP based VoIP user agent. It supports Presence and Instant Messaging.
- Linphone - SIP based web phone. Supported audio codecs are G711, LPC10-15, GSM, and SPEEX.
- Minisip - SIP based phone. Supports video, communication encryption and push-to-talk.
- NAT H323 under Linux - Mini-HOWTO for NAT and H.323 - configuring Linux to support H.323 while using network address translation.
- Nautilus Secure Phone - Program that allows two parties to hold a secure voice conversation using Linux and TCP/IP.
- OpenH323 Gatekeeper - Free H.323 gatekeeper (GPL) based on OpenH323.
- Partysip - SIP proxy server. It can operate as registrar server, redirect server and stateful proxy server. SIP is an open standard (IETF) replacement for H323.
- Siproxd 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).
- Skype - 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.
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