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Computers > Software > Operating Systems > Linux > Networking > VoIP

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

GNU Bayonne - http://www.gnu.org/software/bayonne/bayonne.html
A free, scalable telecommunications application server by the GNU Project.

GNU oSIP - http://www.gnu.org/software/osip/osip.html
Implementation of Session Initiation Protocol (SIP). This library provides an interface to initiate and control SIP based sessions.

http://skype.com - http://skype.com
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://sourceforge.net/projects/kphone - http://sourceforge.net/projects/kphone
SIP based VoIP user agent. It supports Presence and Instant Messaging.

http://www.ekiga.org/ - http://www.ekiga.org/
Videoconferencing application to make audio and video calls to remote users. Supports both SIP and H.323.

http://www.minisip.org/ - http://www.minisip.org/
SIP based phone. Supports video, communication encryption and push-to-talk.

Linphone - http://www.linphone.org/
SIP based web phone. Supported audio codecs are G711, LPC10-15, GSM, and SPEEX.

NAT H323 under Linux - http://pierre.clerissi.free.fr/uHOWTO/nath323.html
Mini-HOWTO for NAT and H.323 - configuring Linux to support H.323 while using network address translation.

Nautilus Secure Phone - http://nautilus.berlios.de/
Program that allows two parties to hold a secure voice conversation using Linux and TCP/IP.

OpenH323 Gatekeeper - http://www.gnugk.org/
Free H.323 gatekeeper (GPL) based on OpenH323.

Partysip - http://www.nongnu.org/partysip/partysip.html
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 - http://siproxd.sourceforge.net/
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).

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