Free Code Chat Software

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A range of software related to voice and video chat, originally aggregated by Danyl Strype as part of research for Core Us, a proposed Disintermedia project for empowering global scale organizing using free code voice and video chat tools, including increasing the ability of conferences to support remote participation.

Note: This list is probably more useful to developers who want to know what free code is available for implementing voice and video. If you're looking for a list of apps, to get an idea which ones might suit your needs, try Free Code Chat Apps.

Audio only

  • Adhearsion Foundation: a whole stack of web telephony software projects, although as of late 2019, only Adhearsion itself and Blather have recent commits, and most of them seem to have ceased development in late 2015/ 2016 (including Candy). There's been no updates on the Foundation website since then either.
  • GNUComm and GNU Telephony (GPL 2 or later): A suite of telephony software supported by the GNU Project, which includes 7 GNU projects: Bayonne, SIP Witch, and their supporting libraries (Common C++ / uCommon, ccAudio, ccScript, ccRTP, and ZRTP / ZRTP4J). The GNU Telephony website appears to have gone down and the page on gnu.org has no updates on development progress since 2013. GNUComm is listed in the Free Software Directory, as are some of the subsidiary projects (see links above).
  • SIPML5 (BSD, formerly GPLv3) and WebRTC2SIP: A SIP client written in Javascript, and a WebRTC back-end allowing the browser to be a phone.
  • Voicechat (GNU GPLv3+, Java SE 7): a voice chat app that "supports uPnP, conversations with multiple users and basic compression with comfort noise". Last commits 3 years ago, probably discontinued.
  • YATE (GPL v2): Internet telephony engine designed to be easily extensible.

Video only

  • HasciiCam (C, GPLv2+): Uses the AA-Lib library to convert video from a camera (eg web cam) into ASCII text, which can be transferred across a network and displayed without much lower use of resources than a full-data video stream. Listed in the Free Software Directory.

Audio-visual

  • Jami (GPLv3+): A GNU project formerly known as SFLphone and then GNU Ring. A P2P text, voice, and video chat client, cross-platform (desktop and mobile). Listed in the Free Software Directory.
  • Jitsi Meet (Apache 2.0, formerly "MIT"): WebRTC client for text, voice and video conferencing, encrypted by default, with desktop and presentation sharing.
  • Linphone (core library: GNU GPLv3, desktop, Windows 10, iOS clients: GPLv2, Android client: GNU GPLv3, flexisip server: GNU AGPL): Clients and server for voice/ video calling, using the SIP protocol.
  • RocketChat ("MIT): a web-based chat server billed as a "Slack-like online chat, built with Meteor". It supports WebRTC for voice/ video conferencing, with a plug-in for Jitsi Videobridge to increase the number of users it can support in a conference.
  • Spreed (AGPLv3): A WebRTC server for voice and video calls and conferencing, using Go and Node.JS.
  • Sylk Suite (Server: GPLv3+, client: AGPLv3+, API: "MIT"): WebRTC client/server for text, voice, and video conferencing, screensharing, and file transfers.
  • Talky.io: A WebRTC platform which has released a number of elements of their stack as free code including SimpleWebRTC and OTalk (although there are still some proprietary bits in the stack).
  • Tox: encrypted, cross-platform, peer-to-peer, text, voice, and video chat. Client software for various platforms, in various languages, mostly still in alpha/ beta, built around Toxcore (C, GPLv3). Listed in the Free Software Directory, which also has pages for some of its client apps.
  • UnHangout (GPLv3, Javascript): Developed at MIT, specifically as a replacement for the Goggle product with a similar name. Seems to be a JS-based web interface built around a Jitsi Meet core. As of late 2019, no new commits to the code repo since 2018.

Live mixers

Standards/ Protocols

SIP (Session Initiation Protocol): a server/client protocol mainly used for voice chat:

Tox: peer-to-peer, encrypted, text, voice, and video chat.

XMPP (eXtensible Messaging and Presence Protocol): client/server protocol for Instant Messaging (short emails in realtime) based on Jabber

WebRTC: An open standard developed by Goggle, Mozilla, and Opera for realtime voice and video conversation, which is being standardized by the W3C.

Codecs

If we are going to stream, store and replay audio and video from conferences, we don't want to use patent-encumbered, proprietary formats like MP3, and H.264. Instead, we want to use patent-free/ royalty-free multimedia file formats, developed by groups like Xiph.org.

Some existing options are:

Audio

  • FLAC - developed by Xiph, lossless codec
  • Opus - developed by Xiph (replaces Speex)
  • Vorbis - developed by Xiph, lossy codec. Equivalent to MP3, although it compresses to smaller files with better sound quality. Technically obsoleted by Opus as well, but there will be an argument for using Vorbis until such time as Opus is more widely supported by hardware and software media players.

Video

  • Dirac - created by BBC Research, "low-resolution web content to broadcasting HD and beyond, to near-lossless studio editing". Original homepage, linked from BBC site, now redirects to spam site. No releases added to Sourceforge files page since 2011. Discontinued?
  • Theora - developed by Xiph as the video layer for Ogg video files
  • VP8/ VP9 - bought by Goggle and released as a royalty-free open codec, for use with WebM

Containers

Formats which allow video, audio, subtitles and other components in different formats to be one included in one file

  • Ogg - developed by Xiph
  • Matroska - developed by the Matroska organisation in France
  • WebM - developed by Goggle, uses a fork of the Matroska container, VP8 (or 9) video, Vorbis or Opus audio, and WebVTT for text

Peer-to-Peer Network

Whatever end-user applications and codecs are used, it seems reasonable that any conferencing system will be more scalable if it uses some kind of P2P network, where everyone who joins the conference contributes some of their own computing resources to the networking effort. This is how Skype originally worked, before Microsoft acquired it and moved it to a purely client/server model. Some existing open P2P network protocols that could be used:

  • BitTorrent Live (Can't yet find source code or license information, may not be free code): Originally pitched as a system for streaming live events like sports games and concerts, using the same P2P swarming principles underlying BitTorrent downloads. Seems to have pivoted towards a kind of blockchain-powered, P2P, alternative to YT.
  • GNUNet (GNU GPLv3): "GNU's Framework for Secure Peer-to-Peer Networking", subsidiary projects include a pre-alpha voice chat app called Conversation.
  • Retroshare (current licensing situation is unclear): "using a web-of-trust to authenticate peers and OpenSSL to encrypt all communication... provides filesharing, chat, messages, forums and channels ...". An experimental voice and video client, under the same name, is being built on top of this network stack.

Discontinued

  • Speak Freely (discontinued, license unclear, FSD says "public domain", homepage says GNU GPL) - one of the earliest voice chat apps, "written in 1991 by John Walker, founder of Autodesk, and Brian C. Wiles".
  • Subrosa (GPLv3, discontinued): WebRTC client/server for text, voice and video conferencing, encrypted by default. Listed in the Free Software Directory.

More Information

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