The SIP Switch is a Web application which allows you to use multiple SIP accounts on the same phone. This lets you use the call plans of different VoIP companies (using SIP) with one IP phone device. It does the third party SIP registration so you can receive all your calls on the sipswitch account. On dial out, you can set prefixes to make your outgoing calls with such or such provider.
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, G722, ilbc, amr, Theora, H263-1998, MPEG4, H264, VP8, and snow.
Skype uses P2P (peer-to-peer) technology to provide voice- and video-based communication with other Internet users. The technology is extremely advanced, but easy to use. It features excellent sound quality, end-to-end encryption, and automatic negotiation of firewalls or routers. Among major features are SkypeOut and SkypeIn, adding the possibility to make low-cost calls to land line phones and having a fixed number to be able to receive calls from land line phones.
Bayonne is the telephony server of the GNU project. It offers a script-driven threaded multi-line state event telephony service on GNU/Linux, xBSD, and Microsoft Windows for building voice response systems, and uses telephony plugins for runtime driver configuration. It also features "TGI" for making Perl applications "telephony aware". It may be used to build telephony-based system administration, home automation, automated attendant, v-commerce, and voice messaging systems.
Speex is a patent-free compression format designed especially for speech. It is specialized for voice communications at low bit-rates in the 2-45 kbps range. Possible applications include Voice over IP (VoIP), Internet audio streaming, audio books, and archiving of speech data (e.g. voice mail).
Mediastreamer is a portable C library that allows you to create and run audio and video streams. It is designed for any kind of voice over IP applications. It features RTP connectivity, audio codecs (Speex, iLBC, G711, GSM, and AMR), video codecs (MPEG4, H263, H264, and Theora), sound card I/O, wav file streaming, webcam video capture, echo-cancellation, conferencing, parametric equalization, and various other utilities. It has a modular design that makes it extensible through plugins. This is the media-streaming component of linphone, a GPL SIP video phone.
RAT is an RTP audio conferencing and streaming application that allows users to particpate in point-to-point and multi-point audio conferences over the Internet. Developing features include multiple sampling rates, mono/stereo, loss concealment, and IPv4/IPv6 support, and both versions feature encryption and run on a range of platforms.
GYach Enhanced is a feature-rich, improved version of the original Gyach. It is the first Yahoo! client for Linux with voice chat capabilities. It offers almost all of the features you would expect to find in the official Windows Yahoo! client. The program offers support for chat, conferences, buddy lists, and My Yahoo content. In addition, Gyach Enhanced offers many features not available in the official Yahoo! clients for Windows, Mac, and Linux. Webcam support is under development and planned for the future. Unlike the original Gyach, GYach Enhanced is designed for Linux only.
ADM (Asterisk Desktop Manager) aims to integrate your desktop with the Asterisk PBX and hardware IP phone by providing some useful features such as automatic on-call volume reduction, one click dialing (from the clipboard), CRM integration via a browser popup, BlueTooth presence detection and automatic call redirection when you walk out of the office, and transfer of the current call from the desktop.