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> Even if you do go peer-to-peer with WebRTC the quality would still be improved by going through a central server (less upstream bandwidth required) Why is th
by wwwtyro 13y ago
> Even if you do go peer-to-peer with WebRTC the quality would still be improved by going through a central server (less upstream bandwidth required)
Why is that?
- tommoor 13y agoI should have prefaced that with when there are more than 2 participants... as you would only need to upload your audio/video stream once.
- saykou 13y agoyeah that makes no sense, unless there's is more then on participant, in that case a full peer to peer would have to upstream (n-1) streams
- viraptor 13y ago1. Possible saving on bandwidth for the clients if they know good codecs A and B respectively, so they can't use them between eachother, but a transcoding third party can help. 2. This doesn't help with the amount of bandwidth used, but sometimes due to peering (that VoIP companies really care about) you can get lower delay going via them rather than directly. This is not very common, but it happens (especially if your broadband provider is your VoIP provider for example). Maybe the poster above had another reason for it too. There's loads of edge cases in the ITSP world.