-
Notifications
You must be signed in to change notification settings - Fork 13
New issue
Have a question about this project? Sign up for a free GitHub account to open an issue and contact its maintainers and the community.
By clicking “Sign up for GitHub”, you agree to our terms of service and privacy statement. We’ll occasionally send you account related emails.
Already on GitHub? Sign in to your account
N15 latency control should be formulated in a technology-agnostic way #91
Comments
There seems to be an implicit assumption that the browser is receiving media from the cloud and sending back game console input. But it should also be possible for cloud gaming participants to send media to each other. My question is what API work is required to "ensure a low and consistent latency for audio, video and data". One might argue WebRTC already attempts to achieve low latency for audio/video via congestion control algorithms such as gcc. Since SCTP typically doesn't implement low-latency congestion control, latency issues seem more likely there. |
This issue was mentioned in WEBRTCWG-2023-09-12 (Page 27) |
This issue was mentioned in WEBRTCWG-2023-12-05 (Page 19) |
The current text of N15 is:
This makes the assumption that 1) data transport is the only thing that needs controlling, and 2) that SCTP heartbeat and RTO are appropriate controls.
It should be restated in terms of what one wants to achieve, for instance:
The text was updated successfully, but these errors were encountered: