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• Streaming RTMP to JANUS-Gateway only showing bitrate but no video

  14 avril 2021, par Ilyas OUMNIH

  I'm currently using the streaming plugin as follows
Fancy artchitecture here

  



  OBS--------RTMP--------->NGINX-Server------FFMPEG(input RTMP output RTP)--------->JANUS---------webrtc-------->Client

  



  When using the ffmpeg command (bellow), on the Janus streaming interface, we only see the bitrate that corresponds to that of the ffmpeg output in the console but we don't see any video.

  



  ffmpeg -i rtmp://localhost/live/test -an -c:v copy -flags global_header -bsf dump_extra -f rtp rtp://localhost:8004 


  



  


  (using "-c:v copy" so that no encoding is used and hence reducing the
 latency)

  


  



  The video shows fine if I use "-c:v libx264", the only issue is that it is CPU intensive and adds latency.

  



  Previously I had tried using RTSP as input for FFMPEG and in this case the video show fine with almost no latency even though I use "-c:v copy".

  



  So I don't realy get why for RTSP the copy works fine, but for RTMP I have to use the libx264 codec. If anyone has an idea about this I am all ears :)

  


• Any advice on streaming av1 with gstreamer to mediamtx and webrtcbin ?

  8 février 2024, par Israel Robotnick

  I have gstreamer 1.22.7 with the RS plugin for av1 support.
I'm trying to stream AV1 rtp to mediamtx using gstreamer, but the bigger goal is that my rtspsrc->webrtcbin pipeline will work with av1 as it works with h264\vp8\vp9.

  


  I have gstreamer 1.22.7 with the RS plugin for av1 support.
I've created a few av1 files with ffmpeg using svtav1 and rav1e encoders :

  


  ffmpeg -i h264.mp4 -an -c:v libsvtav1 -preset 5 -crf 30 -g 60 -svtav1-params tune=0:fast-decode=1 -pix_fmt yuv420p test1.mp4

ffmpeg -i h264.mp4 -an -c:v librav1e -preset 5 -crf 30 -g 60 -rav1e-params speed=5:low_latency=true -pix_fmt yuv420p test2.mp4


  


  ffmpeg does not currently support AV1 streaming to rtp\rtsp, so im using gstreamer to do so :

  


  gst-launch-1.0 filesrc location=test1.mp4 ! qtdemux ! av1parse ! rtspclietsink location=rtsp://127.0.0.1:8554/test1


  


  From what I've read, mediaMTX\chrome\VLC in their latest versions support av1 streaming in webrtc\rtsp,
but there are no examples whatsoever on how to do so.

  


  Gstreamer preroll, playing and recording when publishing. Everything seems to be fine. Same in mediamtx logs.

  


  When I try to connect a client to the rtsp path via VLC\FFplay\gstreamer rtspsrc->webrtcbin pipeline I don't
et any image. (though webrtc internals show packets arrive fine, but VLC\ffmpeg cant connect)

  


  Any ideas what can be wrong ? Anyone have experience with encoding+streaming AV1 with gstreamer rtspclientsink ?
If you have any tips on redirecting it to webrtcbin (what I do is rtspsrc...parsebin ! queue ! rtpav1pay ! webrtcbin, which seems to connect to chrome and create the av1 sdp, but there in no image) I would appreciate them as well ( :

  


• Trouble setting up rtmp audio streaming with ffmpeg and nginx

  20 septembre 2021, par igoryonya

  Can't figure out how to set up rtmp streaming with ffmpeg.
I've been trying to figure it out for about 2 weeks now.
I need to stream the default sound from my sound card over rtmp stream.
I don't need a video, only an audio.

  


  So, i've installed nginx with rtmp module and added the following code at the end of my "/etc/nginx/nginx.conf" file :

  


  rtmp {
        server {
                listen 1935;
                application cam1 {
                        live on;
                }
        }
}


  


  restarted my nginx installation :

  


  sudo systemctl status nginx.service


  


  Checked that the 1935 port is listening :

  


  ~$ sudo netstat -nlp|egrep -i 'nginx'
tcp        0      0 0.0.0.0:1935            0.0.0.0:*               LISTEN      1885931/nginx: mast 
tcp        0      0 0.0.0.0:80              0.0.0.0:*               LISTEN      1885931/nginx: mast 
tcp6       0      0 :::80                   :::*                    LISTEN      1885931/nginx: mast


  


  Launched an ffmpeg to capture my sound card, encode it into ogg format and stream it to the rtmp server (on the same computer with nginx installed) :

  


  ffmpeg -re -f pulse -i default -c:a libvorbis -content_type 'audio/ogg' -vn -f ogg rtmp://127.0.0.1/cam1/test


  


  And it looks like, it's streaming, by showing statistics.

  


  Then, I try to open it with the vlc player to test the stream and the yellow bar on the progress bar just going back and forth and never starts playing.
I tried to use vlc on the same computer and on the computer on the same network.

  


  Assuming, the computer, that is streaming rtmp has an IP : 192.168.0.1, I use the following connection URL in VLC :

  


  rtmp://192.168.0.1/cam1/test