I read Jitsi’s post here a while ago about applying E2E encryption
on a SFU infastructure, using WebRTC Insertable Streams.
I’m a bit confused whether this technique could work on Asterisk (ConfBridge SFU or regular calls)
when the RTP is just passed through without transcoding. Doesn’t Asterisk SFU mix the audio channels to one channel, and wouldn’t that cause issues with this technique?
Anyone who can shine a light on this? Thanks!
The video payload is relayed to each unaltered. The audio portion is, as you say, mixed so it would need to be in the clear for Asterisk.
Hey @jcolp. Thanks for clarifying this. I suppose it wouldn’t be possible to have the audio channels relayed unaltered as well?
There is no feature or functionality to do it. The architecture of streams could certainly do it.
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