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1. This WebRTC experiment is for one-to-many transmission of audio/video streams.
2. It sets up multiple peer connections to support the multi-user connectivity feature. Rememebr, WebRTC doesn't supports 3-way handshake!
3. Multi-peers establishment opens many RTP-ports according to the number of media streamas referenced to each peer connection.
4. Multi-ports establishment causes huge CPU and bandwidth usage!

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If 10 users join your broadcasted room, 40 RTP ports will be opened on your browser:

1. 10 RTP ports for outgoing audio streams
2. 10 RTP ports for outgoing video streams
3. 10 RTP ports for incoming audio streams
4. 10 RTP ports for incoming video streams

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For 10 users session, in one-way broadcasting:

1. 10 RTP ports for outgoing audio stream
2. 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.

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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 without modifying a single line! Just skip below code and open above link!

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This WebRTC Video Broadcasting Experiment works fine on following web-browsers:

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WebRTC Video Broadcasting Experiment is released under MIT licence . Copyright (c) 2013 Muaz Khan.