Can't get Audio from USB Microphone together with Raspberry Pi Camera to work

[nineff]

Question

Hi, I've been stuck on this for hours now and I'm not sure of anything anymore...

I have a Raspberry Pi 5 with the latest 64bit raspi-os and a Camera Module 3. They are working perfectly and streaming them over webRTC with

paths:
  cam:
    source: rpiCamera

works very well.

The issues start when I try to combine them with audio. The Microphone works and I can record sound with arecord, but I can't get them to work together.
The configuration suggested in the documentation as well as here in the discussions both lead to the same behavior.

Config:

paths:
  cam:
    source: rpiCamera
    sourceOnDemand: true

  cam_with_audio:
    runOnDemand: >
      gst-launch-1.0
      rtspclientsink name=s location=rtsp://localhost:$RTSP_PORT/cam_with_audio
      rtspsrc location=rtsp://127.0.0.1:$RTSP_PORT/cam latency=0 ! rtph264depay ! s.
      alsasrc device=hw:3,0 ! opusenc bitrate=16000 ! s.
    runOnDemandRestart: yes

Logs:

025/09/21 14:43:41 INF MediaMTX v1.15.0
2025/09/21 14:43:41 INF configuration loaded from /home/niko/mediamtx/mediamtx.yml
2025/09/21 14:43:41 INF [WebRTC] listener opened on :80 (HTTP), :8189 (ICE/UDP)
2025/09/21 14:43:41 INF [SRT] listener opened on :8890 (UDP)
2025/09/21 14:43:51 INF [WebRTC] [session 72beca4d] created by [fd4c:6056:f38a:0:f6a4:75ff:fea0:b39c]:51488
2025/09/21 14:43:51 INF [path cam_with_audio] runOnDemand command started
Setting pipeline to PAUSED ...
Pipeline is live and does not need PREROLL ...
Progress: (open) Opening Stream
Progress: (open) Opening Stream
Progress: (connect) Connecting to rtsp://localhost:554/cam_with_audio
Pipeline is PREROLLED ...
Prerolled, waiting for progress to finish...
Progress: (connect) Connecting to rtsp://127.0.0.1:554/cam
Redistribute latency...
ERROR: from element /GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0: Could not open resource for reading and writing.
Additional debug info:
../gst/rtsp/gstrtspsrc.c(8331): gst_rtspsrc_retrieve_sdp (): /GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0:
Failed to connect. (Generic error)
ERROR: pipeline doesn't want to preroll.
Setting pipeline to NULL ...
ERROR: from element /GstPipeline:pipeline0/GstRTSPClientSink:s: Could not open resource for reading and writing.
Additional debug info:
../gst/rtsp-sink/gstrtspclientsink.c(3303): gst_rtsp_client_sink_connect_to_server (): /GstPipeline:pipeline0/GstRTSPClientSink:s:
Failed to connect. (Generic error)
ERROR: pipeline doesn't want to preroll.
Freeing pipeline ...
2025/09/21 14:43:51 INF [path cam_with_audio] runOnDemand command exited: command exited with code 0

The error is the same if I don't use on runOnDemand. Sadly, I don't know weather this is a gSteramer or a Mediamtx issue.
Can somebody please point me in the right direction?
Why is reading and writing even needed while opening the ressource? can't i use read only?