GNU SIP Witch : A secure peer-to-peer VoIP server

GNU SIP Witch is a secure peer-to-peer VoIP server. Calls can be made even behind NAT firewalls, and without requiring service providers. SIP Witch can be used on the desktop to create bottom-up secure calling networks and as a free software alternative to Skype. It can also be used as a stand-alone SIP-based office telephone server, or to create secure VoIP networks for an existing IP-PBX such as Asterisk, FreeSWITCH, or Yate.

• Licenses : GPLv3
• Operating Systems : OS Independent
• Implementation : C++

Linphone : An audio and video Internet phone with GTK+ and console interfaces

Linphone is an audio and video Internet phone with GTK+ and console interfaces. It uses the SIP protocol, and is compatible with most SIP clients and gateways. It can use various audio and video codecs such as Speex, GSM, G711, ilbc, amr, Theora, H263-1998, MPEG4, H264, and snow.

• Licenses : GPL
• Operating Systems : POSIX, Linux, BSD, FreeBSD, Windows, Mac OS X, iPhoneOS, Android
• Implementation : C

The GNU Gatekeeper : A free H.323 gatekeeper based on the OpenH323 project

The GNU Gatekeeper is a free H.323 gatekeeper based on the OpenH323 project. You can use it to manage a Voice-over-IP network and let endpoints (e.g., Netmeeting) communicate through symbolic names. It also has an external interface for billing and other applications. It runs on a number of Unix versions (including Linux and Solaris) and Windows.

• Licenses : GPLv2
• Operating Systems : Mac OS X, POSIX, BSD, FreeBSD, Windows, Linux, Solaris
• Implementation : C++

Yet Another Telephony Engine (Yate) : A next-generation telephony engine

Yate is a next-generation telephony engine. While currently focused on Voice over Internet Protocol (VoIP) and PSTN, its power lies in its ability to be easily extended. Voice, video, data, and instant messaging can all be unified under Yate's flexible routing engine, maximizing communications efficiency and minimizing infrastructure costs for businesses. It supports SIP, H.323, IAX, MGCP, Jingle, Jabber, E1, T1, analogic, robbed bit, ISDN PRI, BRI, and SS7.

• Licenses : GPL
• Operating Systems : POSIX, BSD, FreeBSD, Linux, Windows
• Implementation : C++

GNU SIP Witch : A pure SIP-based office telephone call server

GNU SIP Witch is a pure SIP-based office telephone call server that supports generic phone system features like call forwarding, hunt groups and call distribution, call coverage and ring groups, holding, and call transfer, as well as offering SIP specific capabilities such as presence and messaging. It supports secure telephone extensions for making calls over the Internet, and intercept/decrypt-free peer-to-peer audio and video extensions. It is not a SIP proxy, a multi-protocol telephone server, or an IP-PBX, and does not try to emulate Asterisk, FreeSWITCH, or Yate.

• Licenses GPLv3
• Operating Systems OS Independent
• Implementation C++

SFLphone : An SIP/IAX2 compatible softphone

SFLphone is an SIP/IAX2 compatible softphone. The goal is to create a robust enterprise-class desktop phone. While it can serve home users very well, it is designed for intensive corporate use.

• Licenses : GPLv3
• Operating Systems : Linux
• Implementation : C++, C, dbus, Glib, GTK 2.0, Qt, XML

GNU Gatekeeper : Free H.323 gatekeeper based on the OpenH323 project

The GNU Gatekeeper is a free H.323 gatekeeper based on the OpenH323 project. You can use it to manage a Voice-over-IP network and let endpoints (e.g., Netmeeting) communicate through symbolic names. It also has an external interface for billing and other applications. It runs on a number of Unix versions (including Linux and Solaris) and Windows.

• Licenses : GPLv2
• Operating Systems : Mac OS X, POSIX, BSD, FreeBSD, Windows, Linux, Solaris
• Implementation : C++

Linphone : An audio and video Internet phone with GTK+ and console interfaces

Linphone is an audio and video Internet phone with GTK+ and console interfaces. It uses the SIP protocol, and is compatible with most SIP clients and gateways. It can use various audio and video codecs such as Speex, GSM, G711, ilbc, Theora, H263-1998, MPEG4, H264, and snow.

• Licenses : GPL
• Operating Systems : POSIX, Linux, BSD, FreeBSD, Windows
• Implementation : C

Flash Operator Panel displays information about your Asterisk PBX activity in real time via a standard Web browser with the Flash plugin

Flash Operator Panel displays information about your Asterisk PBX activity in real time via a standard Web browser with the Flash plugin. The display and button layout is configurable, so you can have more than 100 buttons on the screen at once. It also supports contexts: you can have one server running and many different client displays (for hosted PBX, different departments, etc). It can monitor several asterisk servers at once. It can integrate with CRM software, by popping up a Web page (and passing the CLID) when a specified button is ringing. It also can be used to enable click-to-dial for Web-based applications.

• Licenses GPL
• Implementation Perl