# Hosted realtime audio/video conferencing

> Use <https://sip2sip.info/> for now, and try limit max. 6 video participants


## Recommendations

Ideal would be to self-host,
but no such solutions are readyly available in Debian yet.
Current recommendation is therefore to use a cloud service
built with Free software, light on resources,
and realistic to fully self-host later if needed.

Generally most reliable is <https://sip2sip.info/>,
so use that unless you need a specific feature unavailable there.


### sip2sip.info

Login is intuitive and step-wise,
using ad-hoc identity.
Connect to <https://sip2sip.info/> and select room name,
and you are redirected to a room URL
(room URL is however long to type and at another commercial site).

Media selection is done reliably
before entering the room
(except if you cancel browser dialog, then login page is trapped).

Streaming and rendering is efficient.
Video focus can optionally be locked onto 1-2 "speakers".
Unknown how larger meetings are handled.

Includes text chat
(crucial in case of audio trouble).


### letsmeet.no

Login is intuitive and step-wise,
using either ad-hoc or Feide (EduGAIN, Twitter, Facebook, LinkedIn) identity.
Connect to <https://letsmeet.no/> and select room name,
and you are redirected to an intuitive room URL.

Media selection is done unreliably
(not reliably detecting available devices)
when entering the room
(if browser dialog is cancelled you enter room in a broken unrepariable state)

Streaming and rendering is inefficient
(heavy CPU load).
Video focus can optionally be limited to an arbitrary amount of "speakers"
(limiting only by number not locked onto specific participants).
Unknown how larger meetings are handled.

Includes text chat
(crucial in case of audio trouble).


### tawk.space

No login.
Connect to <https://tawk.space/>
which is one big "room".

Media selection is done only using browser dialog.
(if browser dialog is cancelled you enter room in a broken unrepariable state)

Streaming is unreliable
(may fail to initiate video, and video freezes after ~10 seconds);
rendering is efficient.
Unknown how larger meetings are handled.

Includes text-based scratch-space.

Includes per-participant pointer.


## 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](https://webrtcglossary.com/simulcast/)
      i.e. encode multiple streams that an SFU can "hop" between
    + Support spatial/temporal/quality [SVC](https://webrtcglossary.com/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](https://github.com/EricssonResearch/scream)
  * security
    + Support [PERC](https://webrtcglossary.com/perc/)
    + Support [ZRTP](https://en.wikipedia.org/wiki/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](https://janus.conf.meetecho.com/)
WebRTC SFU/bridge/broker
written in C

[Mediasoup](https://mediasoup.org/)
WebRTC SFU
written in C

[Kurento](https://www.kurento.org/)
WebRTC MCU
written in C++

[drachtio](https://drachtio.org/)
SIP "SFU"
written in C++

[Licode](https://lynckia.com/licode/)
WebRTC MCU
written in C++

[Medooze WebRTC Media Server](https://github.com/medooze/media-server)
WebRTC/SIP MCU
written in C++

[SylkServer](https://sylkserver.com/)
SIP "SFU"
written in Python

[Spreed WebRTC](https://github.com/strukturag/spreed-webrtc)
WebRTC SFU
written in NodeJS and Go

[Jitsi Videobridge](https://jitsi.org/jitsi-videobridge/)
XMPP SFU
written in Java

[Jigasi](https://jitsi.org/jitsi-meet/)
WebRTC bridge to Jitsi Videobridge
written in Java

### Platforms

[multiparty-meeting](https://github.com/havfo/multiparty-meeting)
using Mediasoup
(and optionally drachtio and Kurento)
written in JavaScript
hosted at <https://letsmeet.no/>

[Jangouts](https://github.com/jangouts/jangouts)
using Janus
written in CoffeeScript

[tawk.space](https://github.com/invisible-college/tawk.space)
using Janus
written in CoffeeScript
hosted at <https://tawk.space/>

[SIP2SIP](https://ag-projects.com/sip2sip/)
using SylkServer and Janus
hosted at <https://sip2sip.info/>
and <https://webrtc.sipthor.net/>

[Roll Call](https://github.com/mikeal/roll-call)
audio-only
hosted at <https://roll.call>

[Spreed.ME](https://www.spreed.me/)
using Spreed WebRTC

[Nextcloud Talk](https://nextcloud.com/talk/)
using Spreed WebRTC

[Jitsi Meet](https://jitsi.org/jitsi-meet/)
using Jigasi and Jitsi Videobridge
hosted at <https://meet.jit.si/>

[Matrix](https://matrix.org/)
using Jigasi and Jitsi Videobridge

[BigBlueButton](https://bigbluebutton.org/)
using Kurento
written in Java

[mConf](http://mconf.org/)
using Kurento
written in Java and Ruby

[OpenMeeting](https://openmeetings.apache.org/)
written in Java

[Wire](https://wire.com/)
proprietary-protocol [Free Software](https://github.com/wireapp/wire) stack
written in Haskell, Rust, C

[Talky](https://about.talky.io/)
cloud SFU service
hosted at <https://talky.io/>

[Me](https://join.me/)
cloud SFU service

GoToMeeting
cloud SFU service

[Zoom Meetings](https://zoom.us/)
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