I have an RTP audio source from WebRTC that I'd like to decode and push to a bluetooth device using bluealsa in GStreamer on a Raspberry Pi Zero W. I've tested this approach with a USB sound card and it worked great. However, when I specified device="bluealsa" in alsasink, the pipeline still starts and I can hear the sound for the first half second but then it abruptly stops. Is my pipeline missing anything? Thanks!
gst-launch-1.0 \ rtpbin name=rtpbin latency=100 \ udpsrc port=60000 caps="application/x-rtp, media=audio, encoding-name=OPUS, clock-rate=48000" ! rtpbin.recv_rtp_sink_0 \ udpsrc port=60001 caps="application/x-rtcp" ! rtpbin.recv_rtcp_sink_0 \ rtpbin. ! rtpopusdepay ! queue ! opusdec ! alsasink device="bluealsa"