Gnome-o-Phone 
Graphical VoIP software for the Linux Gnome desktop. Includes RtpTunnel utility for tunneling RTP UDP-based traffic through a firewall.
http://gphone.sourceforge.net/OpenH323 Gatekeeper
Free H.323 gatekeeper (GPL) based on OpenH323.
http://www.gnugk.org/
Nautilus Secure Phone
Program that allows two parties to hold a secure voice conversation using Linux and TCP/IP.
http://nautilus.berlios.de/
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.
http://www.nongnu.org/partysip/partysip.html
Minisip
SIP based phone. Supports video, communication encryption and push-to-talk.
http://www.minisip.org/
GNU oSIP
Implementation of Session Initiation Protocol (SIP). This library provides an interface to initiate and control SIP based sessions.
http://www.gnu.org/software/osip/osip.html
isdn2h323
Project homepage for isdn2h323, a Linux-based ISDN to H.323 gateway.
http://www.telos.de/linux/H323/
Ekiga
Videoconferencing application to make audio and video calls to remote users. Supports both SIP and H.323.
http://www.gnomemeeting.org/
GNU Bayonne
A free, scalable telecommunications application server by the GNU Project.
http://www.gnu.org/software/bayonne/bayonne.html
Asterisk
Open Source telephony switching and private branch exchange (PBX) daemon. Supported signalling protocols: H.323, SIP, MGCP, SCCP (Cisco Skinny).
http://www.asterisk.org/
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).
http://siproxd.sourceforge.net/
NAT H323 under Linux
Mini-HOWTO for NAT and H.323 - configuring Linux to support H.323 while using network address translation.
http://pierre.clerissi.free.fr/uHOWTO/nath323.html
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.
http://www.skype.com/
Linphone
SIP based web phone. Supported audio codecs are G711, LPC10-15, GSM, and SPEEX.
http://www.linphone.org/
KPhone
SIP based VoIP user agent. It supports Presence and Instant Messaging.
http://sourceforge.net/projects/kphone
