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 source efficiency

Support simulcast
i.e. encode multiple streams that an SFU can "hop" between
Support spatial/temporal/quality SVC
i.e. encode a stream that an SFU can efficiently "slice" without recoding
Support suspending simulcast streams
https://webrtchacks.com/suspending-simulcast-streams/
Support SCReAM


security

Support PERC
Support ZRTP


stream forwarding efficiency
meeting management

Personalized Meeting rooms
Scheduled/Meet-me Meetings
Instant/Direct Meetings
Presence Support
Recording
Text chat
Screen sharing


conference stream efficiency

Skip video streams beyond a threshold of participants
Skip video streams tied to quiet audio streams
Skip streams of explicitly tagged non-speaker participants


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
Tools
Janus Gateway
WebRTC SFU/bridge/broker
written in C
Mediasoup
WebRTC SFU
written in C
Kurento
WebRTC MCU
written in C++
drachtio
SIP "SFU"
written in C++
Licode
WebRTC MCU
written in C++
Medooze WebRTC Media Server
WebRTC/SIP MCU
written in C++
SylkServer
SIP "SFU"
written in Python
Spreed WebRTC
WebRTC SFU
written in NodeJS and Go
Jitsi Videobridge
XMPP SFU
written in Java
Jigasi
WebRTC bridge to Jitsi Videobridge
written in Java
Platforms
multiparty-meeting
using Mediasoup
(and optionally drachtio and Kurento)
written in JavaScript
hosted at https://letsmeet.no/
Jangouts
using Janus
written in CoffeeScript
tawk.space
using Janus
written in CoffeeScript
hosted at https://tawk.space/
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 WebRTC
Jitsi Meet
using Jigasi and Jitsi Videobridge
hosted at https://meet.jit.si/
Matrix
using Jigasi and Jitsi Videobridge
BigBlueButton
using Kurento
written in Java
mConf
using Kurento
written in Java and Ruby
OpenMeeting
written in Java
Wire
proprietary-protocol Free Software stack
written in Haskell, Rust, C
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
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