- This WebRTC experiment is for one-to-many transmission of audio/video streams.
- It sets up multiple peer connections to support the multi-user connectivity feature. Rememebr, WebRTC doesn't supports 3-way handshake!
- Multi-peers establishment opens many RTP-ports according to the number of media streamas referenced to each peer connection.
- Multi-ports establishment causes huge CPU and bandwidth usage!
If 10 users join your broadcasted room, 40 RTP ports will be opened on your browser:
- 10 RTP ports for outgoing audio streams
- 10 RTP ports for outgoing video streams
- 10 RTP ports for incoming audio streams
- 10 RTP ports for incoming video streams
Difference between one-way broadcasting and one-to-many broadcasting
For 10 users session, in one-way broadcasting:
- 10 RTP ports for outgoing audio stream
- 10 RTP ports for outgoing video stream
i.e. total 20 outgoing RTP ports will be opened on your browser.
On each participant's side; only 2 incoming RTP ports 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 without modifying a single line! Just skip below code and open above link!
This WebRTC Video Broadcasting Experiment works fine on following web-browsers:
|Firefox||Stable / Aurora / Nightly|
|Google Chrome||Stable / Canary / Beta / Dev|