[webrtc-nv-use-cases] Use case: Transmit stored pre-encoded content over RTP (#82)

alvestrand has just created a new issue for https://github.com/w3c/webrtc-nv-use-cases:

== Use case: Transmit stored pre-encoded content over RTP ==
This has come up in a couple of contexts, including the provision of "wait signals" and the insertion of pre-recorded segments into an otherwise live conference application.

The important points are:
- The media is pre-recorded (but the media may be available in multiple formats/qualities)
- The desired transmission mechanism is RTP

I think this can be achieved by:
- Providing a means to create frames based on existing encoded video + metadata
- Providing a means to enqueue those frames on an existing RTCRtpSender
- Providing a means to take signals from the RTCRtpSender about available bandwidth and requests for new keyframes and have them processed in an application-specific manner

Responses to congestion signals may involve switching the source of frames to a lower quality source (much like DASH does), or it may involve switching the source to a video showing "wait a bit", or it may involve frame decimation of some kind (assuming the signal is encoded in a decimation-compatible format such as an SVC encoding). These decisions don't need to be part of the WebRTC component.


Please view or discuss this issue at https://github.com/w3c/webrtc-nv-use-cases/issues/82 using your GitHub account


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Received on Monday, 5 December 2022 14:32:44 UTC