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I'm trying to create a low latency stream (sub second) using GStreamer and Python's aiortc library for creating a WebRTC peer for the stream data. I've modified the server example from aiortc and can send an audio file and hook into the video response but what classes/process do I need to use to leverage a GStreamer RTSP video stream?

Do I need to decode the samples with something like an appsink and send each frame individually or is there an aiortc class that can take the RTSP uri and stream the result for me to the peer?

I'm currently running with GStreamer 1.10.4.

ahanson
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  • Did you ever solve this question? I'm interested in the same problem. We are stuck with an old version of Gstreamer (1.12) that doesn't have webrtcbin – Jeremiah Rose May 18 '22 at 05:07

1 Answers1

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This seems like a promising start, but you will need to do some NAL unit parsing. Also I believe this implementation decodes and re-encodes each frame, but if the encoded video formats are compatible, you ought to be able to send it without these extra steps.

Ani
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