DEVELOP.md



#Designing a realtime audio/video conferencing service
Features
Functionality to consider when considering tool/platform.

topology

mesh (P2P)
all processing at end-points, participants limited by bandwidth
routing (SFU)
light server processing, participants limited by bandwidth
mixing (MCU)
heavy server processing


stream efficiency - recipient

suspending select streams at receiving end
i.e. pause reception of stream (not just omit further processing it)


stream efficiency - forwarding

simulcast
i.e. encode multiple streams that an SFU can "hop" between

issue: enabling simulcast is done in 3 incompatible ways

[Plan B]: SDP munging

test: SDP contains a=ssrc-group:SIM entry


test: SDP contains a=rid and a=simulcast entry mentioning "rid="
support:

Firefox >= 46 << 72


[rfc8853][Simulcast in SDP]: rid

test: SDP contains a=rid and a=simulcast entry not mentioning "rid="
support:

Firefox >= 68
Chromium >= M74


issue: using simulcast need sendEncodings in RTCRtpTransceiverInit

workaround: adapter.js shim

janus demos


spatial/temporal/quality SVC
i.e. encode a stream that an SFU can efficiently "slice" without recoding


stream efficiency - source

suspending simulcast streams
https://webrtchacks.com/suspending-simulcast-streams/
SCReAM
i.e. adapt encoding based on SCReAM feedback from stream remote end


stream efficiency - conference admin or shared room setting

Skip video streams beyond a threshold of participants
Skip video streams tied to quiet audio streams
Skip streams of explicitly tagged non-speaker participants
Mix audio streams (not stream each individually)


security

PERC
ZRTP
i.e. end-to-end encryption (not only client-to-server)


meeting management

Personalized Meeting rooms
Scheduled/Meet-me Meetings
Instant/Direct Meetings
Presence
Recording
Text chat
Screen sharing
feedback on own audio level
feedback on encoding and streaming qualities


conference management

Conference Recording
force-mute participants
"Raise a hand" for muted participants


meeting room

Dual stream for dual screen


Dial in from telephone
Dial in from SIP audio-only
Dial in from SIP with video
Dial in from SIP with SIMPLE text chat

See also
SFU
Janus Gateway -
WebRTC SFU/bridge/broker
written in C,
shared under the GPL-3 license
on Github
Mediasoup -
WebRTC SFU
written in C,
shared under the ISC license
on Github.
Medooze SFU -
WebRTC SFU
written in Node.js,
shared under the Expat license
on Github.
Spreed WebRTC -
WebRTC SFU
written in NodeJS and Go,
shared under the AGPL-3 license
on Github.
Ion SFU -
WebRTC SFU
written in Go,
shared under the Expat license
on Github.
Jitsi Videobridge -
XMPP SFU
written in Java,
shared under the Apache-2.0 license
on Github.
Seems to require "plan B" SDP.
Open WebRTC Toolkit -
WebRTC SFU and MCU
written in C++,
shared under the Apache-2.0 license
on Github.
MCU
Kurento Media Server -
WebRTC MCU
written in C++,
shared under the Apache-2.0 license
on Github.
Licode -
WebRTC MCU
written in C++,
shared under the Expat license
on Github.
Medooze WebRTC Media Server -
WebRTC/SIP MCU
written in C++,
shared under the GPL-2 license
on Github.
Red5 Server -
WebRTC SFU
written in Java,
shared under the Apache-2.0 license
on Github.
Ant Media Server -
WebRTC SFU
written in Java,
shared under the Apache-2.0 license
on Github.
Fork of Red5 Server.
Bridge/management
SylkServer -
SIP/XMPP Application Server
with bridge to WebRTC
using Janus,
written in Python2,
shared under the GPL-3 license
on Github.
Spreed standalone signaling server -
Nextcloud Talk bridge to WebRTC
using Janus and NATS,
written in Go,
shared under the AGPL-3 license
on Github.
Jigasi -
WebRTC bridge to Jitsi Videobridge
written in Java,
shared under the Apache-2.0 license
on Github.
Matrix Synapse -
Federated text chat
with WebRTC direct chat,
and with optional WebRTC conferencing
using Jigasi and Jitsi Videobridge,
written in Python3,
shared under the Apache-2.0 license
on Github.
BigBlueButton -
using Kurento Media Server
(only for one-way streaming
also using internal SFU),
written in Java,
shared under the LGPL-3 license
on Github.
OpenMeetings -
using Kurento Media Server and Red5 Server,
written in Java and JavaScript,
shared under the Apache-2.0 license
on Github.
WebRTC features "might be not production ready".
in release 5.0.0-M3 (newest as of 2020-04-10).
openVidu -
using Kurento Media Server,
written in Java and TypeScript,
shared under the Apache-2.0 license
on Github.
Wire Audio Video Signaling -
Wire Application Server
using for MLS,
and using internal FIXME for WebRTC SFU,
written in Haskell, Rust, C,
FIXME.
Web frontend
multiparty-meeting
using Mediasoup
(and optionally drachtio and Kurento Media Server),
written in JavaScript
hosted at https://letsmeet.no/
BitLink
using Mediasoup,
written in JavaScript
hosted at https://app.bitlink.live/
Jangouts
using Janus,
written in CoffeeScript
Roomler
using Janus,
written in VueJS
hosted at https://roomler.live/
tawk.space
using Janus
written in CoffeeScript
hosted at https://tawk.space/
PULT
using Janus, Mumble, and VNC,
written in JavaScript.
open-museum
using Janus
written in Svelte
hosted at https://open-museum.ch/
SIP2SIP
using SylkServer and Janus
hosted at https://sip2sip.info/
and https://webrtc.sipthor.net/
Roll Call
audio-only
hosted at https://roll.call
Spreed.ME
using Spreed WebRTC
Nextcloud Talk
using Spreed standalone signaling server
Galène
using Ion SFU
written in Go.
Jitsi Meet
using Jigasi and Jitsi Videobridge
hosted at https://meet.jit.si/
and https://beta.meet.jit.si/.
WorkAdventure
using Jigasi and Jitsi Videobridge
written in TypeScript
hosted at https://workadventu.re/
mConf
using Kurento Media Server
written in Java and Ruby
Veeting
cloud SFU service
using Janus
hosted at https://rooms.veeting.com/
(free trial at https://rooms.veeting.com/home-office)
SkyWay Conference
cloud SFU service
written in TypeScript
hosted at https://conf.webrtc.ecl.ntt.com/
Talky
cloud SFU service
hosted at https://talky.io/
Me
cloud SFU service
GoToMeeting
cloud SFU service
Zoom Meetings
cloud SFU service
supporting "up to 50 participants at once"
(but client bandwidth and resource demands and stability of such session is unknown)
Hangouts Meet
cloud SFU service
Webex Meetings
cloud SFU service
Wowza Streaming Cloud
cloud streaming service
Skype
cloud SFU service
suporting "up to 25 participants at once"
(but client bandwidth and resource demands and stability of such session is unknown)
MoxieMeet
cloud SFU service
requiring Google account
supporting "up to 32 users all on video together"
(but client bandwidth and resource demands and stability of such session is unknown)
TeamViewer
cloud SFU service
Plan B: https://datatracker.ietf.org/doc/html/draft-uberti-rtcweb-plan-00
"Plan B: draft spec expired 2013 on signaling multiple media sources in WebRTC by munging SDP"
Unified Plan: https://tools.ietf.org/html/draft-roach-mmusic-unified-plan-00
"Unified Plan: draft spec expired 2014 on signaling multiple media sources in WebRTC by setting Restriction Identifier (RID)"
Simulcast in SDP: https://datatracker.ietf.org/doc/html/rfc8853
"Simulcast in SDP: spec on signaling multiple media sources in WebRTC by setting Restriction Identifier (RID)"