Hi everyone,
I'm running Janus webrtc server on a Raspberry Pi and forwarding RTP data to UDP sockets where it is picked up by GStreamer and played via Alsa. I had this solution running on a Raspberry Pi Zero W with no issues, but it lacked computing resources so I upgraded to Raspberry Pi 3A+. For some reason, I can't get audio from the browser to play locally on RPi 3A+with the same solution as the one that worked on RPi Zero. Here's my GStreamer pipeline:
gst-launch-1.0 -vvv \
rtpbin name=rtpbin latency=100 \
udpsrc port=50000 caps="application/x-rtp, media=audio, encoding-name=OPUS, clock-rate=48000" ! rtpbin.recv_rtp_sink_0 \
udpsrc port=50001 caps="application/x-rtcp" ! rtpbin.recv_rtcp_sink_0 \
rtpbin. ! rtpopusdepay ! opusdec ! audioconvert ! audiorate ! audioresample ! alsasink
I tried putting a filesink instead of rtpopusdepay and I'm definitely receiving bytes. I'm not sure how to parse them for any meaningful info, but here's a link:
Any suggestions are welcome! I've been banging my head against this problem for a while. It's especially frustrating since I had it working on a comparable piece of hardware before.
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