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.
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.
WikiPBX is a PBX Web interface for FreeSWITCH. Multiple "accounts" are supported per server instance: each account is effectively a completely independent PBX. Configuration is layered so that XML files go on top of what is stored in the database. This allows you to use a database, but stays out of your way if you choose to use flat files. Extensions, SIP endpoints, and gateways can be configured via a Web interface. Live calls can be viewed, hanged up, and transferred. Call history (CDR records) can be viewed over the Web interface. There is a Web interface for managing IVRs. "Sound clips" can be easily recorded for use in dialplan or IVRs. Audio or text-to-speech can be injected into live calls.
libre is a generic library for real-time communications with asynchronous I/O support. It is written in portable POSIX source code that conforms to the ANSI C89 and ISO C99 standards. It is robust and fast, with a low memory footprint. It also features RFC compliance and support for IPv4 and IPv6. Protocol implementations include SIP, SDP, RTP/RTCP, BFCP, DNS, STUN/TURN/ICE, HTTP, and WebSockets.
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.
Billion PBX is IP phone system that uses the SIP protocol. In addition to voice calls it supports call forwarding, voicemail, callback, and much more, and features simple installation and configuration, reliability, low hardware requirements, and a flexible pricing policy. Its main features are subscriber registration by SIP Registrar and RAS protocols, call routing by SIP and H.323, call authorization, subscriber self-service, Call Data Records (CDR), embedded rating and billing, integration with external billing systems, NAT traversal, IVR, report generation, and a convenient administrative Web interface allowing remote management.