jitsi-videobridge

I don't think there is comprehensive information about that. We all do pretty much the same, implement the Selective Forwarding Unit (SFU) model. The idea is the SFU will receive a number of streams for each user (usually up to 3, in low, mid and high resolutions) and then it will forward one of them to each other participant depending on things like available bandwidth, requested video size, etc. There are a number of Open Source implementations available, in case you want to study it deeper: - Jitsi videobridge: https://github.com/jitsi/jitsi-videobridge - Janus: https://github.com/meetecho/janus-gateway - mediasoup: https://github.com/versatica/mediasoup - (p)ion: https://github.com/pion/ion-sfu - meddoze: https://github.com/medooze/media-server-node And probably more!

SVC spatial and quality layers [1] sound like a really good solution to the bandwidth issue. From my (extremely limited) understanding, basically if you skip certain packets you get a lower resolution/quality stream. A client sends a single stream at the best quality it can tolerate, and then the SFU can forward whichever layers to each client depending on what resolution that client wants. What's the state of this in jitsi? I can only find limited info about SVC [2] and that is only on temporal layer. How much bandwidth does it even save in practice - maybe it's not worth the complexity trade-off? [1] http://webrtchacks.staging.wpengine.com/chrome-vp9-svc/ [2] https://github.com/jitsi/jitsi-videobridge/blob/master/doc/s...

It is possible to have conference calls with e2e encryption. Many conference calls are implemented using what's called a Selective Forwarding Unit (SFU) and the sending clients send multiple resolutions (either independent, called "Simulcast" or dependent, called "SVC"). In that case, the adaptation is done by the server in selecting which resolution to forward at any given time. This is fairly common practice in the industry. For example: https://github.com/jitsi/jitsi-videobridge and https://tools.ietf.org/html/draft-aboba-avtcore-sfu-rtp-00 and https://www.w3.org/TR/webrtc-svc/. For those types of conference calls, the server only needs to know the sizes of the various streams and which packet is for what stream. It does not need to see the decrypted media, so one can implement e2e encryption for such types of group calls. This is less

First of all, there are 2 different kinds of video calls: 1:1 and group calls. For 1:1 calls, e2e encryption doesn't cause any problem at all. For group calls, it depends on how it's implemented, but many group calls are implemented using what's called a Selective Forwarding Unit (SFU) and the sending clients send multiple resolutions (either independent, called "Simulcast" or dependent, called "SVC"). In that case, the adaptation is done by the server in selecting which resolution to forward at any given time. This is fairly common practice in the industry. For example: https://github.com/jitsi/jitsi-videobridge and https://tools.ietf.org/html/draft-aboba-avtcore-sfu-rtp-00 and https://www.w3.org/TR/webrtc-svc/. For those types of group calls, the server only needs to know the sizes of the various streams and which packet is for what

The default installs of Jitsi Meet use Jitsi videobridge to avoid p2p for calls of >2 participants. For <=2 people, p2p WebRTC is used (though a STUN server might be needed to traverse NATs for each person [0]). For >2 people, the Jitsi Videobridge is used instead of p2p [1]: it takes in all media streams from all clients (possibly even in different resolutions) and selectively forwards them to clients based on bandwidth [2] [0]: https://en.wikipedia.org/wiki/STUN [1]: https://jitsi.org/jitsi-videobridge/ [2]: https://github.com/jitsi/jitsi-videobridge/blob/master/doc/s...

Awesome client-side resource! WebRTC is poised to hit the big time now that it is supported by iOS 11. For both of my questions below I will link options from previous discussions[1]; any experience you can share would be appreciated. 1) What is the best option to implement a server that can tie into WebRTC for audio/video/screensharing? https://github.com/mappum/electron-webrtc - simple but bulky Node.js WebRTC via a hidden Electron process https://chromium.googlesource.com/external/webrtc - C++ Chromium WebRTC https://github.com/jitsi/jitsi-videobridge - Java video conferencing https://github.com/Kurento/kurento-media-server | https://github.com/OpenVidu/openvidu - Node.js media server transitioning after Twilio hired key devs https://www.wowza.com/products/capabilities/webrtc-streaming... - commercial; in preview 2) What is the best mi

Jitsi also has a Web RTC driven video chat when combined with an XMPP server: http://www.youtube.com/watch?v=x-YGw3bgB_s https://jitsi.org/Projects/JitsiVideobridge https://github.com/jitsi/jitsi-videobridge