WebRTC One-to-Many audio sharing/broadcasting Demo
Usecase
Improved audio quality! Used for live monitoring and studio talkback
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- This WebRTC experiment is aimed to transmit audio stream in one-to-many style.
- It setups multiple peer connections to support multi-user connectivity feature. Rememebr, WebRTC doesn't supports 3-way handshake!
- Out of multi-peers establishment; many RTP-ports are opened according to number of media streamas referenced to each peer connection.
- Multi-ports establishment will cause huge CPU and bandwidth usage!
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If 10 users join your broadcasted room, 20 RTP ports will be opened on your browser:
- 10 RTP ports for outgoing audio streams
- 10 RTP ports for incoming audio streams
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Difference between one-way broadcasting and one-to-many broadcasting
For 10 users session, maximum 10 RTP ports for outgoing audio stream will be opened.
On each participant's side; only one incoming RTP port will be opened.
Unlike one-way broadcasting; one-to-many broadcasting experiment opens both outgoing as well as incoming RTP ports for each participant.
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For signaling; please check following page:
https://github.com/muaz-khan/WebRTC-Experiment/blob/master/Signaling.md
Remember, you can use any signaling implementation exists out there without modifying any single line! Just skip below code and open above link!
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Browser Support
This WebRTC Audio Broadcasting Experiment works fine on following web-browsers:
| Browser | Support |
|---|---|
| Firefox | Stable / Aurora / Nightly |
| Google Chrome | Stable / Canary / Beta / Dev |
| Android | Chrome Beta |
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License
WebRTC Audio Broadcasting Experiment is released under MIT licence . Copyright (c) 2013 Muaz Khan.