Scalable WebRTC peer-to-peer broadcasting demo.
This module simply initializes socket.io and configures it in a way that single broadcast can be relayed over unlimited users without any bandwidth/CPU usage issues. Everything happens peer-to-peer!
RTCMultiConnection v3 and Scalable Broadcast
RTCMultiConnection v3 now naively supports scalable-broadcast:
|Scalable Audio/Video Broadcast||Demo||Source|
|Scalable Screen Broadcast||Demo||Source|
|Scalable Video Broadcast||Demo||Source|
|Scalable File Sharing||Demo||Source|
Note: These (below) are old demos. Above (RTCMultiConnection-v3) demos are preferred (and up-to-dated).
index.html- share video or screen or audio over unlimited users using p2p methods.
share-files.html- share files with unlimited users using p2p methods!
|Firefox||Stable / Aurora / Nightly|
|Google Chrome||Stable / Canary / Beta / Dev|
host means the browser that is used to forward remote-stream.
|Chrome||Audio+Video||Chrome,Firefox||Remote audio tracks are skipped.|
|Chrome||Audio||None||Chrome can NOT forward remote-audio|
- First column shows browser name
- Second column shows type of remote-stream forwarded
- Third column shows browsers that can receive the remote forwarded stream
- Fourth column shows sender's i.e. host's issues
Chrome-to-Firefox interoperability also works!
Android devices are NOT tested yet. Opera is also NOT tested yet (though Opera uses same chromium code-base).
Currently you can't share audio in Chrome out of this big. In case of audio+video stream, chrome will skip remote-audio tracks forwarding. However chrome will keep receiving remote-audio from Firefox!
Firefox additionally allows remote-stream-forwarding for:
- Streams captured from
- Streams captured from
- Streams captured or generated by
AudioContexti.e. WebAudio API
Is stream keeps quality?
Obviously "nope". It will have minor side-effects (e.g. latency in milliseconds/etc.).
If you'll be testing across tabs on the same system, then you'll obviously notice quality lost; however it will NOT happen if you test across different systems.
In the image, you can see that each NEW-peer is getting stream from most-recent peer instead of getting stream directly from the moderator.
npm install webrtc-scalable-broadcast
cd node_modules cd webrtc-scalable-broadcast # and run the server.js file node server.js
cd ./node_modules/webrtc-scalable-broadcast/ node ./server.js
Or install using WGet:
mkdir webrtc-scalable-broadcast && cd webrtc-scalable-broadcast wget http://dl.webrtc-experiment.com/webrtc-scalable-broadcast.tar.gz tar -zxvf webrtc-scalable-broadcast.tar.gz ls -a node server.js
Or directly download the TAR/archive on windows:
And now open:
server.js fails to run:
# if fails, lsof -n -i4TCP:8888 | grep LISTEN kill process-ID # and try again node server.js
How it works?
Above image showing terminal logs explains it better.
For more details, to understand how this broadcasting technique works:
Assuming peers 1-to-10:
Peer1 is the only peer that invokes
getUserMedia. Rest of the peers will simply forward/relay remote stream.
peer1 captures user-media peer1 starts the room
peer2 joins the room peer2 gets remote stream from peer1 peer2 opens a "parallel" broadcasting peer named as "peer2-broadcaster"
peer3 joins the room peer3 gets remote stream from peer2 peer3 opens a "parallel" broadcasting peer named as "peer3-broadcaster"
peer4 joins the room peer4 gets remote stream from peer3 peer4 opens a "parallel" broadcasting peer named as "peer4-broadcaster"
peer5 joins the room peer5 gets remote stream from peer4 peer5 opens a "parallel" broadcasting peer named as "peer5-broadcaster"
and 10th peer:
peer10 joins the room peer10 gets remote stream from peer9 peer10 opens a "parallel" broadcasting peer named as "peer10-broadcaster"
- Peer9 gets remote stream from peer8
- Peer15 gets remote stream from peer14
- Peer50 gets remote stream from peer49
and so on.