- 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!
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
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.
For signaling; please check following page:
Remember, you can use any signaling implementation exists out there without modifying any single line! Just skip below code and open above link!
This WebRTC Audio Broadcasting Experiment works fine on following web-browsers:
|Firefox||Stable / Aurora / Nightly|
|Google Chrome||Stable / Canary / Beta / Dev|