Webrtc 1.16.0 Blocked in the GstOutputSelector (Regression from 1.14.4)

Hey Team, I recently upgraded my gstreamer version to 1.16.0. I used to have a working pipeline where I transcode an RTMP stream to vp8/opus to send out to a website. Since upgrading to 1.16.0, the pipeline appears to do everything correctly except actually send the media. I see a blocked pad in the outputselector which I'm assuming is the culprit but might be a red herring.

Here's the pipeline for reference:

In chrome://webrtc-internals I have both a video and audio candidate pair that send and receive requests.

Any Ideas?