GNU TelephonyOpen-source telephony, preserved and being revived
Open-source telephony, preserved and being revived
Historical archive and independent revival
GNU Telephony: Preserving Open Communications, Building What Comes Next
GNU Telephony belongs to an important chapter in free software history. This independent website reconnects historical technical references and creates a careful foundation for future open-source work.
GNU Telephony, also called GnuComm, was an ecosystem of free-software components for telephone services, signalling protocols, real-time media, scripting, and communications application development. GNU Bayonne focused on telephone and voice-application services. GNU SIP Witch addressed SIP-based infrastructure. GNU ccRTP supported real-time media transport, while ZRTP-related libraries addressed media-session key agreement. Common C++, uCommon, ccScript, and ccAudio supplied supporting application facilities.
It was not a single consumer calling application. Understanding the distinct roles of servers, transport libraries, security protocols, and application foundations is essential when reading the older material.
Explore the historical projects
These are independently written historical explanations. Recovered originals are identified separately with their authorship, date, version, and licence.
The overarching initiative created to provide a complete, free software telephony and voice communications infrastructure, evolving from GnuComm to address VoIP, encrypted media, and decentralized routing.
A pure SIP-based office telephone server and peer-to-peer call routing daemon designed to enable secure, direct audio/video communication while bypassing corporate and government surveillance intermediaries.
The standard GNU implementation of Phil Zimmermann’s ZRTP protocol (RFC 6189), enabling Diffie-Hellman key exchange over RTP to establish authenticated SRTP sessions with Short Authentication String (SAS) verification.
Announced in March 2011 as a community response to centralized telecommunications and proprietary networks like Skype, GNU Free Call aimed to create a resilient, serverless, peer-to-peer encrypted telephone network.
The foundational GNU telephone application server supporting Interactive Voice Response (IVR), voice messaging, conference bridging, and automated call processing over PSTN, ISDN, Dialogic hardware, and SIP.
A versatile, high-performance C++ class framework implementing the Real-time Transport Protocol (RTP, RFC 3550) and RTCP control packets, featuring adaptive jitter buffers, QoS monitoring, and multicast support.
A deeply portable, comprehensive C++ foundation library offering cross-platform multi-threading, socket abstractions, memory-mapped I/O, IPC synchronization, configuration parsers, and XML serialization.
A redesigned, highly lightweight C++ runtime library optimized specifically for memory-constrained embedded VoIP hardware, low-overhead daemon processes, and SIP Witch server components.
A portable C++ class framework for manipulating telephony audio files, sample formats, codecs, and stream-oriented audio data in GNU Telephony applications.
An open XML dialect and interpreter interface developed for GNU Bayonne to describe interactive voice call flows, audio playback, touch-tone recognition, and external database lookups in a standardized markup.
TwoSquares is supporting reconstruction of this archive and early planning for a community-oriented open-source initiative. This does not transfer ownership of historical work or imply appointment or endorsement by the GNU Project or Free Software Foundation. Scope, repositories, maintainers, and release dates will be published only when confirmed.
Developers, technical writers, researchers, designers, archivists, and open-source maintainers are invited to express an interest in contributing.