How to prevent huge RTP h264 livestream quality drops?

[christina xxxxx]

Hi all,

First of all, I'm a beginner to gstreamer, but I've been enjoying using it
so far!

I have a Pi Zero with a Pi camera and a mic. I'm livestreaming audio/video
from the Pi to a remote server with a NGinx frontend running a Janus webRTC
gateway. I'm using gstreamer from the Pi to send RTP to Janus. I am looking
for < 1s latency (but does not have to be <500ms) and relatively consistent
A/V quality.

My full Pi command:

gst-launch-1.0 -v rpicamsrc bitrate=500000 inline-headers=TRUE
keyframe-interval=4 preview=0 !
video/x-h264,width=1280,height=720,framerate=20/1,profile=baseline ! \

h264parse ! rtph264pay config-interval=1 pt=96 ! udpsink host=XXXX
port=8004 \

alsasrc device=mic_mono buffer-time=1000000 !
audio/x-raw,format=S32LE,rate=22050,channels=1 ! \

audioconvert ! audioresample ! opusenc complexity=3 cbr=TRUE ! rtpopuspay !
udpsink host=XXXX port=8005


My video quality is great until there's a lot of movement in front of the
camera. When this happens, the video quality drops by a lot, with every
frame looking like this: http://imgur.com/a/ArPi4 Normally it looks like
this: http://imgur.com/a/WAaoo

I understand UDP is unreliable and packet drops may occur. But can anyone
suggest how I could optimize my settings to prevent such huge drops in
video quality? Perhaps if I added a little more latency, or some other
settings I can tweak?

Thanks so much!

Christina
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