Occasionally no audio between webrtcbin and chrome

[Trey Hutcheson]

Gstreamer 1.16.2; arch linux.

Ok, this one is really weird. I've got a media server (written in rust, if
that matters)
that juggles multiple gstreamer pipelines, with webrtc being the central
use case. Occasionally when a browser based peer (google chrome) joins, it
receives no audio.

The pipeline is pretty basic; an audiotestsrc emits a test tone, encoding
as opus, to webrtcbin. The browser side should simply pipe the audio to the
local speaker. About 50% of the time, after signalling and ice connectivity
has been established, there is no audio. In the cases when there is no
audio, I've looked at chrome://webrtc-internals, and the audio track for
the browser peer is not receiving any samples; no audio level, audio
energy, etc.

So I've tested a variety of scenarios. If I restart the browser between
each test, there is no change. If I restart my media server between each
iteration, then I receive audio 100% of the time. I added an identity
element after `rtpopuspay` to dump the data to the console, and the data is
definitely there. I monitored udp traffic with wireshark; the number and
size of packets to the browser is roughly the same when there is audio and
when there is not.

So then I turned up gstreamer logging to see if it gave me any insight, and
viola! I now have audio 100% of the time. When I set GST_DEBUG to 4 or
higher, I get audio on every test iteration. Anything below 3, and the
browser emits audio only occasionally.

Does anyone have insight on what might be going on here?
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