SipUnit provides a class library that allows software developers to create automated unit tests for SIP applications. Session Initiation Protocol (SIP) is widely used for providing telephone services over the Internet. SipUnit extends the JUnit framework to incorporate SIP-specific assertions, and it provides a high-level API for performing the SIP operations needed to interact with or invoke a test target. A test program using the SipUnit API is written in Java and acts as a network element that sends/receives SIP requests and responses.
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.
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.
gnokii is a multisystem tool suite for mobile phones. It provides a library to communicate with a phone hiding the communication protocol. The library handles SMS, phonebook, calendar, phone calls, and other mobile phone capabilities. It supports Nokia-FBUS mobiles, AT-capable phones (most of the mobiles), as well as Symbian-based phones.
Flite For Asterisk provides the "Flite" dialplan application, which allows you to use the Flite TTS Engine with Asterisk. This module invokes the Flite TTS engine locally, and uses it to render text to speech. It supports voice selection and 8kHz or 16kHz sample rates to provide the best possible sound quality along with the use of wideband codecs. It works with Asterisk 1.6.x, 1.8, and 10.