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. It supports SIP, H.323, IAX, MGCP, Jingle, Jabber, E1, T1, analogic, robbed bit, ISDN PRI, BRI, and SS7. YateClient is an Instant Messenger and Voice application for Jabber/Jingle IM, SIP, H.323, and IAX VoIP protocols.
Kamailio (formerly OpenSER) is a high-performance SIP (RFC3261) server with a flexible architecture and many extensions. The server implements proxy, registrar, redirect, and location SIP/VoIP services. It has support for UDP, TCP, TLS, and SCTP transport layers, DNSsec, ENUM, AAA via database, RADIUS, DIAMETER, gateways to SMS and XMPP, least cost routing, load balancing, NAT traversal, and call processing language. Kamailio implements SIMPLE presence and instant messaging extensions, and includes an embedded XCAP server and MSRP relay. It can be also used as a routing SIP sever for WebRTC via WebSocket.
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.
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.
OTRS is a platform independent Web-based help desk system that supports service organization of any kind (e.g. IT service, customer and technical product service, complaint management, public services, etc.) to increase their efficiency. It increases transparency as well as service quality and lowers your total cost of ownership. It has been certified ITIL V3 compatible by PinkVERIFY for incident, problem, change, service asset and configuration, request fulfillment, and knowledge management. Other ITIL processes like service catalog and service level management are supported as well.
Speech synthesis for asterisk is an Asterisk AGI script that uses Google Translate to convert text to speech and play it back to the user. It supports a variety of different languages, local caching of voice data, and a choice between 8 kHz or 16 kHz sample rates to provide the best possible sound quality along with the use of wideband codecs.
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.