Re:JackTrip WebRTC: high quality, uncompressed, low-delay au...
by Blueskytech on Friday January 29, 2021 @01:17PM
(#25960461)
Completely agree, while proprietary what Audinate has done with Dante is nothing short or a miracle. With their PCI-E FPA I have less then a 2ms round trip latency with my Hilo interface at 96Khz resolution. Additionally the virtual sound card can do 4ms latency in one direction with most modern CPUs.
Re:JackTrip WebRTC: high quality, uncompressed, low-delay au...
by jdc on Thursday January 28, 2021 @11:30PM
(#25953928)
Dante is proprietary for one thing.
Re:JackTrip WebRTC: high quality, uncompressed, low-delay au...
by fxtentacle on Friday January 29, 2021 @04:45AM
(#25955668)
Here's a GPL implementation
https://github.com/jsharkey/wycliffe
Re:JackTrip WebRTC: high quality, uncompressed, low-delay au...
by jdc on Saturday January 30, 2021 @01:11AM
(#25966693)
Thanks! Have you had any luck running it? It doesn't seem to have any documentation.
Re:JackTrip WebRTC: high quality, uncompressed, low-delay au...
by musicale on Thursday January 28, 2021 @08:03PM
(#25952132)
Good question; JackTrip normally uses UDP and I think most browsers do not support sending UDP packets directly without an extension.
If you could send UDP directly then WebRTC could presumably be avoided.
Re:JackTrip WebRTC: high quality, uncompressed, low-delay au...
by mdonahoe on Thursday January 28, 2021 @10:01PM
(#25953225)
Doesn’t webrtc use UDP?
Re:JackTrip WebRTC: high quality, uncompressed, low-delay au...
by fxtentacle on Friday January 29, 2021 @04:47AM
(#25955686)
In some cases on some web browsers WebRTC can be configured to use UDP.
But that's exactly the problem. Usually, nobody knows what's actually happening in that 20000+ lines of code WebRTC stack.
I really don't get why everything needs to be in a web browser nowadays. WebRTC is simply the wrong choice for low-delay audio streaming. Audinate's DANTE protocol has been around for 14 years, is affordable, hardware-accelerated, and has practically no avoidable latency when sending audio over the network.
Why add 20+ms of latency just to use WebRTC?