Help with simple path to automatically transcode AAC audio from RTMP to opus for WebRTC

[sreplans]

Question

Hi all,
I am trying to do a very simple task that I know is possible (I've seen other post on this very discussion forum).
I am publishing a stream to the server using RTPM, and as you know the audio is encoded as AAC (no I can't change it).
here is my path setting. The issue I am having is that the command starts fine, runs for a second or two then exists, and it restarts over and over. this works ok on my local mac, but not on this remote linux (Debian 10) server

2024/10/10 16:15:00 INF MediaMTX v1.9.1
2024/10/10 16:15:00 INF configuration loaded from /root/mediamtx.yml
2024/10/10 16:15:00 INF [RTSP] listener opened on :8554 (TCP), :8000 (UDP/RTP), :8001 (UDP/RTCP)
2024/10/10 16:15:00 INF [RTMP] listener opened on :1935
2024/10/10 16:15:00 INF [HLS] listener opened on :8888
2024/10/10 16:15:00 INF [WebRTC] listener opened on :8889 (HTTP), :8189 (ICE/UDP)
2024/10/10 16:15:00 INF [SRT] listener opened on :8890 (UDP)
2024/10/10 16:15:07 INF [RTMP] [conn 73.204.186.143:62353] opened
2024/10/10 16:15:07 INF [path v/mystream] runOnReady command started
2024/10/10 16:15:07 INF [RTMP] [conn 73.204.186.143:62353] is publishing to path 'v/mystream', 2 tracks (H264, MPEG-4 Audio)
2024/10/10 16:15:07 INF [RTMP] [conn 127.0.0.1:37054] opened
2024/10/10 16:15:07 INF [RTMP] [conn 127.0.0.1:37054] is reading from path 'v/mystream', 2 tracks (H264, MPEG-4 Audio)
Input #0, flv, from 'rtmp://localhost:1935/v/mystream':
Duration: N/A, start: 0.044000, bitrate: N/A
Stream #0:0: Video: h264 (High 4:4:4 Predictive), yuv444p(progressive), 1280x720 [SAR 1:1 DAR 16:9], 30.30 fps, 30 tbr, 1k tbn
Stream #0:1: Audio: aac (LC), 44100 Hz, mono, fltp
Stream mapping:
Stream #0:0 -> #0:0 (copy)
Stream #0:1 -> #0:1 (aac (native) -> opus (libopus))
Press [q] to stop, [?] for help
2024/10/10 16:15:10 INF [path v/mystream] runOnReady command exited: command exited with code 0
2024/10/10 16:15:10 INF [RTMP] [conn 127.0.0.1:37054] closed: write tcp 127.0.0.1:1935->127.0.0.1:37054: write: broken pipe
2024/10/10 16:15:15 INF [RTMP] [conn 127.0.0.1:14106] opened
2024/10/10 16:15:15 INF [RTMP] [conn 127.0.0.1:14106] is reading from path 'v/mystream', 2 tracks (H264, MPEG-4 Audio)
Input #0, flv, from 'rtmp://localhost:1935/v/mystream':
Duration: N/A, start: 6.476000, bitrate: N/A
Stream #0:0: Video: h264 (High 4:4:4 Predictive), yuv444p(progressive), 1280x720 [SAR 1:1 DAR 16:9], 30.30 fps, 30 tbr, 1k tbn
Stream #0:1: Audio: aac (LC), 44100 Hz, mono, fltp
Stream mapping:
Stream #0:0 -> #0:0 (copy)
Stream #0:1 -> #0:1 (aac (native) -> opus (libopus))
Press [q] to stop, [?] for help
2024/10/10 16:15:18 INF [path v/mystream] runOnReady command exited: command exited with code 0
2024/10/10 16:15:18 INF [RTMP] [conn 127.0.0.1:14106] closed: write tcp 127.0.0.1:1935->127.0.0.1:14106: write: connection reset by peer
2024/10/10 16:15:18 INF [path v/mystream] runOnReady command stopped
2024/10/10 16:15:18 INF [RTMP] [conn 73.204.186.143:62353] closed: EOF

Here is my simple path settings:

paths:
v/webrtc:
v/mystream:
runOnReady: >
ffmpeg -i rtmp://localhost:1935/$MTX_PATH
-loglevel info -hide_banner -vcodec copy -ac 2 -c:a libopus -b:a 128k -async 50
-max_delay 0 -max_muxing_queue_size 1024 -f rtsp -rtsp_transport tcp rtsp://localhost:$RTSP_PORT/v/webrtc
runOnReadyRestart: yes

[sreplans]

duhhhhh
runOnReadyRestart: false :)