Oreka is an enterprise telephony recording and retrieval system with a Web-based user interface. It supports recording voice from VoIP SIP, Cisco Skinny (aka SCCP), raw RTP, and audio sound devices, and runs on multiple operating systems and database systems. It can record audio from most PBX and telephony systems, such as BroadWorks, Metaswitch, Asterisk, FreeSwitch, OpenSIPS, Avaya, Nortel, Mitel, Siemens, Cisco Call Manager, Cosmocom, NEC, etc. It is being used in call centers and contact centers for quality monitoring (QM) purposes.
Ekiga (formely known as GnomeMeeting) is a soft phone, video conferencing, and instant messenger application for use over the Internet. It supports HD sound quality and video up to DVD size and quality. It is interoperable with many other standards compliant software, hardware, and service providers as it uses both of the major telephony standards, SIP and H.323.
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.
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.
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.
Baudline is a time-frequency browser designed for scientific visualization of the spectral domain. Signal analysis is performed by Fourier, correlation, and raster transforms that create colorful spectrograms with vibrant detail. Conduct test and measurement experiments with the built in function generator, or play back audio files with a multitude of effects and filters. The baudline signal analyzer combines fast digital signal processing, versatile high speed displays, and continuous capture tools for hunting down and studying elusive signal characteristics.
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.
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).