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• Unable to record mediasoup producer using FFmpeg on real server

  30 novembre 2020, par Sarvesh Patil

  I have built a nice app in react native for audio calling, many thanks to MediaSoup !!

  


  To take it to next level, I need to record some of my calls.
I used this tutorial for reference :
mediasoup recording demo

  


  I followed the FFmpeg way and have reached a point where I have created a plainTransport with

  


      router.createPlainTransport({
        // No RTP will be received from the remote side
        comedia: false,
        // FFmpeg and GStreamer don't support RTP/RTCP multiplexing ("a=rtcp-mux" in SDP)
        rtcpMux: false,
        listenIp: {ip:"0.0.0.0", announcedIp:"MY_PUBLIC_IP"},
    });



  


  Then I connect to this transport :

  


      rtpPlainTransport.connect({
        ip: 127.0.0.1,
        port: "port1",
        rtcpPort: "port2",
    });



  


  My first doubt : is the ip address in .connect({}) parameters supplied above correct ?

  


  Second, the FFMPEG command requires an SDP header. This is mine :

  


      v=0
    o=- 0 0 IN IP4 127.0.0.1
    s=-
    c=IN IP4 127.0.0.1
    t=0 0
    m=audio port1 RTP/AVPF 111
    a=rtcp:port2
    a=rtpmap:111 opus/48000/2
    a=fmtp:111 minptime=10;useinbandfec=1


  


  When I start recording, the FFMPEG process does not receive any data.
Moreover, on stopping, I get the following message

  


  


  Output file is empty, nothing was encoded (check -ss / -t / -frames
parameters if used) Exiting normally, received signal 2. Recording
process exit, code : 255, signal : null

  


  


  I was able to make the recording save on localhost with 127.0.0.1 when the server was itself running on localhost.

  


  However, with my actual server hosted with Nginx, I'm not able to figure out what is going wrong.

  


  I can see data being sent on my audio port :

  


  1 0.000000000    127.0.0.1 → 127.0.0.1    UDP 117 10183 → 5004 Len=75
2 0.020787740    127.0.0.1 → 127.0.0.1    UDP 108 10183 → 5004 Len=66
3 0.043201757    127.0.0.1 → 127.0.0.1    UDP 118 10183 → 5004 Len=76


  


  What do I do with FFmpeg so that it starts the recording !?

  


  Can someone please help ?

  


• Revision 1c0769107b : vp10 cleanup : remove nonrd and real-time code Code cleanup. Change-Id : I668bd5

  26 août 2015, par Yunqing Wang

  Changed Paths :
   Modify /vp10/encoder/encodeframe.c

  
   Modify /vp10/encoder/encoder.c

  
   Modify /vp10/encoder/ethread.c

  
   Delete /vp10/encoder/pickmode.c

  
   Delete /vp10/encoder/pickmode.h

  
   Modify /vp10/encoder/ratectrl.c

  
   Modify /vp10/encoder/rd.c

  
   Modify /vp10/encoder/rdopt.c

  
   Modify /vp10/encoder/speed_features.c

  
   Modify /vp10/encoder/speed_features.h

  
   Modify /vp10/vp10cx.mk

  
  
  vp10 cleanup : remove nonrd and real-time code

  Code cleanup.

  Change-Id : I668bd5a4e5fbe96969d51d9ee777fc5f1d8abfe6

• Real time HLS editing with FFmpeg filter_complex without re-encoding full stream

  28 mai, par Devdeep Ghosh

  I'm new to FFmpeg and working on a lightweight video editor for HLS streams that allows cutting and trimming without re-encoding the entire video. I'm using FFmpeg to only process the segments that intersect with "exclusion" ranges (timestamps to remove), while copying unmodified segments directly. The goal is speed and accuracy.

  


  The logic identifies which .ts segments overlap with exclusion windows, applies filter_complex to trim just those segments, and then programatically reassemble everything into a new .m3u8 playlist. However, the final output sometimes freezes or desynchronizes—audio gets out of sync or playback freezes where segments were excluded.

  


  Here's the FFmpeg logic I'm using to trim segments :

  


  const videoFilter = `[0:v]select='not(between(t,${start},${end}))'[vout]`;
const audioFilter = `[0:a]aselect='not(between(t,${start},${end}))'[aout]`;


const ffmpegArgs = [
  "-i", inputPath,
  "-filter_complex", `${videoFilter};${audioFilter};[vout]setpts='PTS-STARTPTS'[v];[aout]asetpts='PTS-STARTPTS'[a]`,
  "-map", "[v]",
  "-map", "[a]",
  "-c:v", "libx264",
  "-c:a", "aac",
  "-preset", "ultrafast",
  "-hls_flags", "independent_segments",
  "-avoid_negative_ts", "make_zero",
  "-y", outputPath,
];


  


  Is this approach actually viable for cutting .ts segments from an HLS stream without full re-encoding ?

  


  Am I misunderstanding how FFmpeg handles timestamps when trimming ?

  


  Is there a better or more reliable way to do this (e.g. concat, -ss/-to, GOP alignment, etc.) ?