6 comments on “Your Browser as a Audio Conference Server with WebRTC & Web Audio (Alexey Aylarov)

  1. Nice!
    Is that possible to mix multiple audio input streams using this technique in a SFU model and play them via a single AUDIO control?
    This way we would be able to control (e.g. set volume) all the audio streams just by one audio control.

    • Yes, it should work w/o problem – just attach media streams from PCs to the destination media stream and attach it to audioContext

  2. Hi there, very good your post.

    My question is, can I test the demo with just one ip public ?

    I tried to test the demo from two operations system, my current system windows 10 and another virtual system, but I get this notification: Online Users Nobody is online at the moment .

    Thanks for your support.

  3. Regarding the sample code provided, is that supposed to be in the onstream function of the host(the host’s browser-server)? Also, my current webRTC setup is using the updated ‘tracks’ approach, do I need to use the deprecated ‘streams’ approach to get this to work?

    Is the idea that as soon as the host receives a stream from a newly connected peer, it is added to the mixer, which all peers that were already in the mixer will automatically receive, or do I need to recreate everyone’s stream(include audio from all mics except for the receiving peer) and send it back out, causing another negotiation event?

    Sorry, I am new to, both, webRTC and Web Audio APIs

  4. Is it possible to do a two-way mp3 audio streaming using this approach? I assume that mp3 to opus conversion should be made somehow without a noticeable delay…

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