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.
baresip is a bare-bones SIP user agent. It supports SIP, SDP, RTP/RTCP, and STUN/TURN/ICE, and IPv4 and IPv6, and is RFC-compliant and has portable C89 and C99 source code. A modular plugin architecture provides stdio, cons, and evdev user interfaces, celt, g711, g722, gsm, ilbc, l16, and speex audio codecs, alsa, coreaudio, gst, portaudio, oss, winwav, and mda audio drivers, speex_pp, speex_aec, speex_resamp, and sndfile audio filters, the avcodec video codec, avformat, quicktime, qtcapture, v4l, and v4l2 video sources, sdl, opengl, and x11 video display drivers, and srtp media encoding.
GROUP-E is collaboration software which integrates groupware, project management, and business server on one platform. The solution is based on a LAMP architecture (Linux, Apache, MySQL, PHP). GROUP-E offers project management, transparent Samba (file server) integration, integration of Cyrus IMAP server with administration and personal SIEVE filters, support for SyncML 1.0, LDAP-based user management with single sign-on authentication, and LDAP contact databases. GROUP-E also provides ActiveSync for synchronization.
Newfies-Dialer is a voice broadcast application designed and built to automate the delivery of interactive phone calls to contacts, clients, and the general public. It is scalable from a single server to supporting distributed call processing across multiple cloud-based telephony servers to make millions of outbound calls per day. The multi-user Web interface allows every aspect of the campaign to be controlled with detailed reporting. A Newfies-Dialer Voice Broadcasting Platform is assembled entirely from free and open source components including Freeswitch, Django, Plivo, Celery, and RabbitMQ.
The Ozeki VoIP Service Checker is a tool you can use for checking the availability of VoIP services. It can be used to verify whether a given service exists, see whether it can be reached from a particular network, and check whether the user details (such as the username and password) to access the service are correct. It was developed because many mobile Internet providers block VoIP traffic.
Speech synthesis for Asterisk using MS Translator text-to-speech service to synthesize speech and play it back to the user. It supports a variety of languages, local caching of voice data, and 8kHz or 16kHz sample rates to provide the best possible sound quality along with the use of wideband codecs.
EMIPLIB is a library to facilitate the development of programs that need to stream several kinds of media over IP. The library consists of several kinds of components that can be linked together in various ways, thereby providing a flexible framework. It also provides some ready-to-use classes for the transmission of audio and video over IP. Streams originating from the same participant can be synchronized.