Does gstreamer webrtc support Adaptive Bitrate (ABR)?

[Justin]

Hello,

I just started playing around with the gstreamer webrtc examples using the
1.18.5 release on Ubuntu 20.04.  Does the latest webrtcbin have built-in
adaptive bitrate algorithms for video/audio streams?  I remember reading
somewhere in a 2017/2018 article that the ABR functionality for
webrtcbin was still being developed at the time.  What is its current
status?

If it's not currently supported inside of webrtcbin, does anyone have
suggested methods of adding ABR manually in a webrtc pipeline?

Best,
Justin
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