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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

  


• Could not find com.arthenica:ffmpeg-kit-full:6.0-2

  6 mai, par gabocalero

  I started receiving this message when I build my project

  


  > Could not find com.arthenica:ffmpeg-kit-full:6.0-2.
  Searched in the following locations:
    - https://dl.google.com/dl/android/maven2/com/arthenica/ffmpeg-kit-full/6.0-2/ffmpeg-kit-full-6.0-2.pom
    - https://repo.maven.apache.org/maven2/com/arthenica/ffmpeg-kit-full/6.0-2/ffmpeg-kit-full-6.0-2.pom
    - https://jcenter.bintray.com/com/arthenica/ffmpeg-kit-full/6.0-2/ffmpeg-kit-full-6.0-2.pom
    - https://jitpack.io/com/arthenica/ffmpeg-kit-full/6.0-2/ffmpeg-kit-full-6.0-2.pom
    - https://pkgs.dev.azure.com/MicrosoftDeviceSDK/DuoSDK-Public/_packaging/Duo-SDK-Feed/maven/v1/com/arthenica/ffmpeg-kit-full/6.0-2/ffmpeg-kit-full-6.0-2.pom
    - https://oss.sonatype.org/content/repositories/snapshots/com/arthenica/ffmpeg-kit-full/6.0-2/ffmpeg-kit-full-6.0-2.pom
  Required by:
      project :presentation > project :domain


  


  These are the project's repositories

  


  allprojects {     
   repositories {
      google()
      mavenCentral()
      jcenter()
      maven { url 'https://jitpack.io' }
      maven { url "https://oss.sonatype.org/content/repositories/snapshots" }     
   } 
}


  


  And this is the dependency I'm adding

  


      implementation(libs.arthenica.ffmpeg.full)


  


  This is my libs.version.toml

  


  ffmpeg = "6.0-2"
arthenica-ffmpeg-full = {  group = "com.arthenica", name = "ffmpeg-kit-full", version.ref = "ffmpeg" }


  


  AFAICS in the project's github page, it will not be maintained anymore


  


  In the short term, do you know any other repository that still servers this dependency ?

  


  On the other hand, do you know any other project to replace Arthenica ffmpeg-kit-full dependency ?

  


  Thank you very much

  


• RTMP Disconnects quickly when the stream is turned on and No index.m3u8 files are being generated in the assigned directory [closed]

  23 avril, par Kartik Salve

  23/4/2025 09:52:58 9408 [INFO] [rtmp connect] id=7K1RWSO1 ip=::1 app=live args={"app":"live","flashVer":"LNX 9,0,124,2","tcUrl":"rtmp://localhost:1935/live","fpad":false,"capabilities":15,"audioCodecs":4071,"videoCodecs":252,"videoFunction":1}
23/4/2025 09:52:58 9408 [INFO] [rtmp play] Join stream. id=7K1RWSO1 streamPath=/live/test  streamId=1 
23/4/2025 09:52:59 9408 [INFO] [rtmp play] Close stream. id=7K1RWSO1 streamPath=/live/test streamId=1
23/4/2025 09:52:59 9408 [INFO] [rtmp disconnect] id=7K1RWSO1


  


  These are the logs, ffmpeg pushes the stream in a different folder when tried manually.

  


  

  • I tried changing the location of my file outside of oneDrive to avoid any permission conflicts.

  


  • Manually checked if FFmpeg is correct with this command : C:\ffmpeg\bin\ffmpeg.exe -i rtmp://localhost/live/test -c:v copy -c:a aac -f hls -hls_time 2 -hls_list_size 5 -hls_flags delete_segments output/index.m3u8

  


  • Downgraded NMS to a stable version.