GNU Telephony Projects Directory
The GNU Telephony suite was structured into modular servers, real-time media libraries, and system abstraction frameworks. Every entry below links directly to the restored canonical historical documentation.
GNU Telephony Ecosystem
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.
- Comprehensive open-source VoIP stack and server software
- End-to-end media security without centralized key authorities
- Adherence to open standards (SIP, RTP, ZRTP)
- Cross-platform C++ frameworks designed for multi-threaded telephony engines
GNU_TelephonyGNU SIP Witch
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.
- Direct peer-to-peer media routing without transcoding bottlenecks
- Integrated NAT traversal and secure SIP registration
- Automatic media encryption negotiation via GNU ZRTP
- Decentralized mesh routing capabilities for GNU Free Call
GNU_SIP_WitchGNU ZRTP (libzrtpcpp)
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.
- Diffie-Hellman Key Exchange (DH-2048, DH-3072, Elliptic Curves)
- Short Authentication String (SAS) vocal verification against MitM attacks
- Key continuity via retained shared secrets between trusted peers
- Compatible with ccRTP, Linphone, SIP Witch, and Twinkle
GNU_ZRTPGNU Free Call
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.
- Peer-to-peer routing operating over SIP Witch daemons
- Integration with FreedomBox and decentralized distributed hash tables (DHT)
- Resistance to centralized wiretapping and platform censorship
- Cryptographic identity management tied to personal devices
GNU_Free_Call_AnnouncementGNU Bayonne
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.
- Modular driver model for Dialogic, Voicetronix, CAPI, and SIP
- Scriptable multi-line voice response execution engine
- BayonneXML integration for web-driven voice applications
- Historical cornerstone of the early GnuComm project
GNU_BayonneGNU ccRTP
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.
- Full RFC 3550 RTP and RTCP compliance with jitter calculation
- Multicast, unicast, and secure payload transport
- Extensible packet queueing and synchronization mechanisms
- Foundation for GNU ZRTP audio/video streaming
GNU_ccRTPGNU Common C++
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.
- Portable POSIX and Win32 thread and mutex wrappers
- TCP/UDP socket streaming and network client/server templates
- Design pattern classes for thread pools and event loops
- Core dependency for Bayonne 1.x and early VoIP servers
GNU_Common_C%2B%2BGNU uCommon C++
A redesigned, highly lightweight C++ runtime library optimized specifically for memory-constrained embedded VoIP hardware, low-overhead daemon processes, and SIP Witch server components.
- Minimal footprint and zero-overhead memory design
- Fast memory pools and thread-safe data structures
- Clean modern C++ idioms replacing legacy Common C++ patterns
- Direct runtime platform for GNU SIP Witch
GNU_uCommon_C%2B%2BGNU ccScript3
An event-driven scripting engine used by GNU Bayonne to describe call flows, application states, transitions, and telephony service behaviour.
- Event-driven state and transition execution
- Telephony-oriented scripting for GNU Bayonne
- Extensible commands and application integration
- Historical source releases preserved by GNU mirrors
GNU_ccScript3GNU ccAudio2
A portable C++ class framework for manipulating telephony audio files, sample formats, codecs, and stream-oriented audio data in GNU Telephony applications.
- Telephony-oriented audio file and stream handling
- Portable C++ interfaces used by GNU Telephony software
- Support for common voice sample formats and conversion workflows
- Packaging history across free software operating systems
GNU_ccAudio2BayonneXML
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.
- Declarative voice interaction documents
- Dynamic backend script integration via XMLRPC and CGI
- Precursor to standardized VoiceXML adoption in free software IVR
- Seamless multi-channel call state transitions
BayonneXML