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• Want to send a video from desktop to Wowza using ffmpeg

  3 mars 2018, par ST94

  I am trying to stream a local .mp4 video file from my laptop to Wowza Streaming Engine using ffmpeg. Both systems are able to ping each other. I give the following command on my laptop

  ffmpeg -re -i bunny_1080p_60fps_normal.mp4 -vcodec libx264 -acodec aac -ar 48000 -strict experimental -f flv "rtmp://192.168.1.22:1935/live/myStream"

  192.168.1.22 is the IP address of Wowza server residing on another system running Ubuntu 17.04.

  I see the following on the command prompt of my laptop when I give the above command,

  ffmpeg version N-89894-g18e2ac032e Copyright (c) 2000-2018 the FFmpeg developers
  
    built with gcc 7.2.0 (GCC)
  
    configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-bzlib --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libmfx --enable-amf --enable-cuda --enable-cuvid --enable-d3d11va --enable-nvenc --enable-dxva2 --enable-avisynth
  
    libavutil      56.  7.100 / 56.  7.100
  
    libavcodec     58.  9.100 / 58.  9.100
  
    libavformat    58.  5.101 / 58.  5.101
  
    libavdevice    58.  0.101 / 58.  0.101
  
    libavfilter     7. 11.101 /  7. 11.101
  
    libswscale      5.  0.101 /  5.  0.101
  
    libswresample   3.  0.101 /  3.  0.101
  
    libpostproc    55.  0.100 / 55.  0.100
  
  Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'bunny_1080p_60fps_normal.mp4':
  
    Metadata:
  
      major_brand     : isom
  
      minor_version   : 1
  
      compatible_brands: isomavc1
  
      creation_time   : 2013-12-16T17:59:32.000000Z
  
      title           : Big Buck Bunny, Sunflower version
  
      artist          : Blender Foundation 2008, Janus Bager Kristensen 2013
  
      comment         : Creative Commons Attribution 3.0 - http://bbb3d.renderfarming.net
  
      genre           : Animation
  
      composer        : Sacha Goedegebure
  
    Duration: 00:10:34.53, start: 0.000000, bitrate: 4486 kb/s
  
      Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 1920x1080 [SAR 1:1 DAR 16:9], 4001 kb/s, 60 fps, 60 tbr, 60k tbn, 120 tbc (default)
  
      Metadata:
  
        creation_time   : 2013-12-16T17:59:32.000000Z
  
        handler_name    : GPAC ISO Video Handler
  
      Stream #0:1(und): Audio: mp3 (mp4a / 0x6134706D), 48000 Hz, stereo, s16p, 160 kb/s (default)
  
      Metadata:
  
        creation_time   : 2013-12-16T17:59:37.000000Z
  
        handler_name    : GPAC ISO Audio Handler
  
      Stream #0:2(und): Audio: ac3 (ac-3 / 0x332D6361), 48000 Hz, 5.1(side), fltp, 320 kb/s (default)
  
      Metadata:
  
        creation_time   : 2013-12-16T17:59:37.000000Z
  
        handler_name    : GPAC ISO Audio Handler
  
      Side data:
  
        audio service type: main
  
  [rtmp @ 000001ed7c4108c0] No credentials set
  
  [rtmp @ 000001ed7c4108c0] Server error: [ AccessManager.Reject ] : [ code=403 need auth; authmod=adobe ] :
  
  rtmp://192.168.1.22:1935/live/myStream: Unknown error occurred

  Can anyone please tell me what the exact procedure is to stream a video file to Wowza .

  How will I be able to view the stream at Wowza ?

• 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

  


• Node.JS Live Streaming Audio with FFMPEG

  20 avril 2021, par nicnacnic

  I'm trying to create an Express server to live stream audio captured from another application (Discord in this case). I'm able to get a server up and running, but there are a couple issues that need to be solved. Here's my server code so far.

  


  const app = express();
app.get("/", function(req, res) {
    res.sendFile(__dirname + "/index.html");
});
app.get("/audio", function(req, res) {
    const stream = ffmpeg(audio).inputOptions(["-f", "s16le", "-ar", "48k", "-ac", "2"]).format('wav');
    res.writeHead(200, { "Content-Type": "audio/wav" });
    stream.pipe(res);
});
app.listen(8080)


  


  

  1. Silent sections of audio need to be added. When there's no activity on the input, there's no data written to the audio variable. This causes weird behavior, for example I can speak and the audio comes through a second later. Then, if I wait 10 seconds then speak again, the audio comes through 4-5 seconds later. I believe this is a problem with the way I'm using ffmpeg to transcode, but I have no idea how to fix it.

  


  2. Refreshing the client crashes the program. Every time I refresh the client I get an ffmpeg error. Error: Output stream closed. This error doesn't happen if I close it, only on reload.

  


  3. The audio is not synced between clients. Every time I open a new connection, the audio starts playing from the beginning instead of being synced with each other and playing the audio live.

  


  


  This is how it's supposed to work : it captures audio from my app in PCM, converts the audio to WAV with ffmpeg, and then streams the audio live to the clients. The audio needs to be synced with all the clients as best as possible to reduce delay. And I'm using fluent-ffmpeg instead of just regular ffmpeg for the transcoding.
Thanks !