Free Code Chat Software

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.

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.

• Mumble/ Murmur (New BSD): an 'IRC with voice' style system, developed for in-game chat

• SIPML5 (BSD, formerly GPLv3) and WebRTC2SIP: A SIP client written in Javascript, and a WebRTC back-end allowing the browser to be a phone.

• 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.

• Jitsi VideoBridge (Apache 2.0, formerly LGPL): video relaying server for multiparty conferences.

Audio-visual

• BigBlueButton (LGPL): voice/video web conferencing server developed for distance education. Currently depends on Flash, but has a rough plan for full transition to HTML5 and WebRTC.

• Eventstreamr (GNU AGPLv3, now discontinued): "single and multi room audio visual stream management". No new commits on this repo since 2015.

• GNU Jami (GPLv3+): P2P text, voice, and video chat client, cross-platform (desktop and mobile)

• Jitsi Meet (Apache 2.0, formerly "MIT"): WebRTC client for text, voice and video conferencing, encrypted by default, with desktop and presentation sharing.

• NextCloud Talk (AGPL): a WebRTC plug-in for the NextCloud app suite, based on Spreed.

• OpenCast (Educational Community License 2.0, a variant of Apache 2.0, Java/ HTML): a system developed for use in recording and streaming university lectures by the Apero Foundation.

• Palava.tv: Like Jitsi Meet, a WebRTC audio and video conferencing stack that includes the Palava-portal ("MIT", Ruby, Javascript), Palava-client (LGPLv3, Ruby, Javascript), and Palava-machine (AGPLv3, Ruby). Listed on the FSF Free Software Directory.

• RetroShare (current licensing situation is unclear): A P2P communications client which can support both voice and video chat

• 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.

• Signal (client apps: GPLv3, server: AGPLv3): Developed by Open Whisper Systems, Signal consists of apps for Android and iOS and a routing server, end-to-end encrypted by default.

• Spreed (AGPLv3): A WebRTC server for voice and video calls and conferencing, using Go and Node.JS.

• Subrosa (GPLv3, discontinued): WebRTC client/server for text, voice and video conferencing, encrypted by default.

• 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)

• 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.

• Wire (client apps: GPLv3, server: AGPLv3):: encrypted one-to-one and group chat, with encryption. Has both web app and desktop/ mobile apps.

Live mixers

dvswitch (GNU GPL): an audio-visual mixing system that depends on LibAV

gst-switch (GNU GPL): written in C and uses GStreamer

snowmix (GNU GPLv3)

Voctomix (Expat or "MIT") a live video mixer wrapped around GStreamer (GNU LGPL), developed by the C3VOC in response to their frustrations with dvswitch, snowmix and gst-switch

Standards/ Protocols

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

• the standard;
• Wikipedia: Session Initiation Protocol

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

• Wikipedia: Tox (protocol)

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

• Wikipedia: XMPP
• Jingle (open protocol extending XMPP): designed for one-on-one voice/ video chat
  • Wikipedia: Jingle (protocol)
  • Muji (open protocol extending Jingle): allow for multi-user voice chat

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

• Some documentation on how to implement can be found on WebRTC Quick Start.
• Wikipedia: WebRTC

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

• Opus - developed by Xiph (replaces Speex)
• Vorbis - developed by Xiph, lossy codec (equivalent to MP3, also obsoleted by Opus?)
• FLAC - developed by Xiph, lossless codec

Video

• Theora - developed by Xiph
• Dirac - created by BBC Research, "low-resolution web content to broadcasting HD and beyond, to near-lossless studio editing"
• VP8/ VP9 - bought by Goggle and released as a royalty-free open codec

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:

FreeNet (GPL): uses standard IP networking in an attempt to create an uncensorable alternative to the world wide web,

GNUNet (GPL): "GNU's Framework for Secure Peer-to-Peer Networking",

BitTorrent Live: A system for streaming live events using the same P2P swarming principles underlying BitTorrent downloads

Retroshare (GPL): "using a web-of-trust to authenticate peers and OpenSSL to encrypt all communication... provides filesharing, chat, messages, forums and channels

See also:

• Chatting, Audio and Video Calls: A document by RiseUp Labs, the folks behind We.Riseup.net
• 74 free code telephony projects from 2007
• The state of FOSS real-time communications - a discussion on Reddit