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• FFMPEG send RTP audio at 8k bytes/sec [closed]

  10 mai, par Muzza

  I'm trying to use FFMPEG to mimick a device that transmits G711U audio over UDP/RTP at 8k bytes per second.
The device im mimicking sends rtp packets every 20ms with 160byte payload.

  


  I've had limited success using the following command

  


  ffmpeg -f dshow -i audio="Microphone (Realtek(R) Audio)" -ac 1 -ar 8000 -ab 8 -acodec pcm_mulaw -f rtp rtp://127.0.0.1:12345?pkt_size=160


  


  This sends G711U encoded audio, in 160byte chunks, but streams at 64kB/s, not the 8kB/s that my device is expected, so the device errors out ?

  


  Any idea's would be massively appreciated !

  


  Thank you

  


  Log from FFMPEG

  


  >ffmpeg -f dshow -i audio="Microphone (Realtek(R) Audio)" -ac 1 -ar 8000 -ab 8 -acodec pcm_mulaw -f rtp rtp://127.0.0.1:12345?pkt_size=160
ffmpeg version 2025-04-23-git-25b0a8e295-essentials_build-www.gyan.dev Copyright (c) 2000-2025 the FFmpeg developers
  built with gcc 14.2.0 (Rev3, Built by MSYS2 project)
  configuration: --enable-gpl --enable-version3 --enable-static --disable-w32threads --disable-autodetect --enable-fontconfig --enable-iconv --enable-gnutls --enable-libxml2 --enable-gmp --enable-bzlib --enable-lzma --enable-zlib --enable-libsrt --enable-libssh --enable-libzmq --enable-avisynth --enable-sdl2 --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxvid --enable-libaom --enable-libopenjpeg --enable-libvpx --enable-mediafoundation --enable-libass --enable-libfreetype --enable-libfribidi --enable-libharfbuzz --enable-libvidstab --enable-libvmaf --enable-libzimg --enable-amf --enable-cuda-llvm --enable-cuvid --enable-dxva2 --enable-d3d11va --enable-d3d12va --enable-ffnvcodec --enable-libvpl --enable-nvdec --enable-nvenc --enable-vaapi --enable-libgme --enable-libopenmpt --enable-libopencore-amrwb --enable-libmp3lame --enable-libtheora --enable-libvo-amrwbenc --enable-libgsm --enable-libopencore-amrnb --enable-libopus --enable-libspeex --enable-libvorbis --enable-librubberband
  libavutil      60.  2.100 / 60.  2.100
  libavcodec     62.  0.101 / 62.  0.101
  libavformat    62.  0.100 / 62.  0.100
  libavdevice    62.  0.100 / 62.  0.100
  libavfilter    11.  0.100 / 11.  0.100
  libswscale      9.  0.100 /  9.  0.100
  libswresample   6.  0.100 /  6.  0.100
  libpostproc    59.  1.100 / 59.  1.100
[aist#0:0/pcm_s16le @ 00000198256b73c0] Guessed Channel Layout: stereo
Input #0, dshow, from 'audio=Microphone (Realtek(R) Audio)':
  Duration: N/A, start: 135470.702000, bitrate: 1411 kb/s
  Stream #0:0: Audio: pcm_s16le, 44100 Hz, stereo, s16, 1411 kb/s, Start-Time 135470.702s
Stream mapping:
  Stream #0:0 -> #0:0 (pcm_s16le (native) -> pcm_mulaw (native))
Press [q] to stop, [?] for help
[pcm_mulaw @ 00000198256cf240] Bitrate 8 is extremely low, maybe you mean 8k
Output #0, rtp, to 'rtp://127.0.0.1:12345?pkt_size=160':
  Metadata:
    encoder         : Lavf62.0.100
  Stream #0:0: Audio: pcm_mulaw, 8000 Hz, mono, s16 (8 bit), 64 kb/s
    Metadata:
      encoder         : Lavc62.0.101 pcm_mulaw
SDP:
v=0
o=- 0 0 IN IP4 127.0.0.1
s=No Name
c=IN IP4 127.0.0.1
t=0 0
a=tool:libavformat 62.0.100
m=audio 12345 RTP/AVP 0
b=AS:64

[out#0/rtp @ 00000198256cdd00] video:0KiB audio:973KiB subtitle:0KiB other streams:0KiB global headers:0KiB muxing overhead: 8.467470%
size=    1055KiB time=00:02:04.51 bitrate=  69.4kbits/s speed=   1x
Exiting normally, received signal 2.


  


  Wireshark :
Wireshark Log

  


  Shows packets being sent every 0.20ms

  


• v4l2loopback+ffmpeg input for uvc gadget

  13 mai, par Mosi

  I'm trying to use an MP4 video file as the input for a UVC Gadget setup on my Raspberry Pi 4 Model B, but I'm running into an issue when streaming through V4L2.

  


  Goal :

  


  To emulate a webcam that streams a looping MP4 video file to a Windows 11 host.

  


  My setup :

  


  

  • Hardware : Raspberry Pi 4 Model B

  


  • OS : Raspberry Pi OS Lite 64-bit (2025-05-06-raspios-bookworm-arm64-lite)

  


  • Kernel : 6.12.25+rpt-rpi-v8

  


  • Host System : Windows 11

  


  • UVC Gadget version : v0.3.0

  


  


  Workflow :

  


  [MP4 Video] → [FFmpeg] → [V4L2 Loopback] → [UVC Gadget] → Windows sees virtual webcam


  


  What works :

  


  The UVC Gadget works perfectly when I use a real webcam as the source (e.g., /dev/video0). Windows detects the virtual webcam and displays a smooth video stream.

  


  The problem :

  


  When I try to use an MP4 video file through FFmpeg and send it to the loopback device (/dev/video3), the UVC Gadget fails with the following error :

  


  ---

  


  Command I'm using :

  


  ffmpeg -re -stream_loop -1 -i input.mp4 -vf scale=640:480 \
  -c:v rawvideo -pix_fmt yuyv422 -r 30 -f v4l2 /dev/video3


  


  Then I run :

  


  sudo uvc-gadget -d /dev/video3 uvc.0


  


  Output :

  


  bRequestType 21 bRequest 01 wValue 0200 wIndex 0001 wLength 0022
streaming request (req SET_CUR cs 02)
setting commit control, length = 34
Setting format to 0x56595559 640x480
=== Setting frame rate to 30 fps
Starting video stream.
--> [At this point I open the camera on the Windows host]
/dev/video3: 2 buffers requested.
Failed to export buffer 0.
Failed to export buffers on source: Inappropriate ioctl for device (25)


  


  ---

  


  Things I've tried :

  


  

  • Multiple FFmpeg formats, resolutions, and pixel formats

  


  • Various ffmpeg buffer and framerate tweaks

  


  • Different UVC Gadget versions

  


  • GitHub related projects (showcamera, etc.)

  


  • Older Raspberry Pi OS versions

  


  


  Most guides and GitHub projects I found are outdated (5+ years old), and newer methods seem undocumented or incompatible with current kernel/UVC gadget tools.

  


  ---

  


  My question :

  


  How can I stream an MP4 file as a virtual webcam using UVC Gadget without getting ioctl errors ?
  
Is there a proper way to set up FFmpeg and loopback devices so that UVC Gadget can read the stream correctly ?

  


  Any modern working example or tips would be very appreciated. Thanks in advance !

  


• FFmpeg RTSP stream to remote MediaMTX server disconnects after a few seconds [closed]

  13 juin, par Rorschy

  I'm new to RTSP and MediaMTX, and I'm trying to live stream my screen using FFmpeg and MediaMTX for a specific use case.

  


  Everything works perfectly when both FFmpeg and MediaMTX run on the same machine.
However, when I move MediaMTX to a remote server, the stream becomes unstable — I can't maintain a connection or view the stream reliably.

  


  Here is the FFmpeg command I'm using from the client machine :

  


  ffmpeg -f gdigrab -framerate 10 -offset_x 0 -offset_y 0 -video_size 1920x1080 -i desktop -f lavfi -i anullsrc -vcodec libx264 -tune zerolatency -g 30 -sc_threshold 0 -preset ultrafast -tune zerolatency -f rtsp rtsp:///live/stream


  


  And here’s the relevant MediaMTX log output on the remote server :

  


  2025/06/12 14:28:44 INF [RTSP] [conn :35798] opened
2025/06/12 14:28:44 INF [RTSP] [session 2e487869] created by :35798
2025/06/12 14:28:44 INF [RTSP] [session 2e487869] is publishing to path 'live/stream', 2 tracks (H264, MPEG-4 Audio)
2025/06/12 14:28:45 INF [WebRTC] [session 8a909818] created by :47296
2025/06/12 14:28:45 WAR [WebRTC] [session 8a909818] skipping track 2 (MPEG-4 Audio)
2025/06/12 14:28:47 INF [WebRTC] [session dd0d3af7] created by :46306
2025/06/12 14:28:47 WAR [WebRTC] [session dd0d3af7] skipping track 2 (MPEG-4 Audio)
2025/06/12 14:28:49 INF [WebRTC] [session 5f853024] created by :46320
2025/06/12 14:28:49 WAR [WebRTC] [session 5f853024] skipping track 2 (MPEG-4 Audio)
2025/06/12 14:28:51 INF [WebRTC] [session 3edba9a8] created by :46342
2025/06/12 14:28:51 WAR [WebRTC] [session 3edba9a8] skipping track 2 (MPEG-4 Audio)
2025/06/12 14:28:53 INF [WebRTC] [session 4be5bd9b] created by :46352
2025/06/12 14:28:53 WAR [WebRTC] [session 4be5bd9b] skipping track 2 (MPEG-4 Audio)
2025/06/12 14:28:54 INF [RTSP] [conn :35798] closed: terminated
2025/06/12 14:28:54 INF [RTSP] [session 2e487869] destroyed: session timed out
2025/06/12 14:28:54 INF [WebRTC] [session 8a909818] closed: terminated
2025/06/12 14:28:54 INF [WebRTC] [session 3edba9a8] closed: terminated
2025/06/12 14:28:54 INF [WebRTC] [session 5f853024] closed: terminated


  


  My questions :

  


  

  1. What could be causing the RTSP stream to disconnect when streaming to a remote MediaMTX server ?

  


  2. Are there any recommended network settings or MediaMTX configuration tweaks to ensure a stable stream over the internet ?

  


  


  Any help or guidance would be greatly appreciated. Thanks !