Gstreamer webrtcbin working sample pipeline

Asked Nov 12 '18 at 17:15

Active Nov 15 '18 at 10:01

Viewed 4,949 times

Can someone show up to date webrtcbin pipeline? At the moment i use these pipelines and they do not work.

Send:

gst-launch-1.0 webrtcbin bundle-policy=max-bundle name=sendrecv  stun-server=stun://stun.l.google.com:19302 audiotestsrc is-live=true wave=red-noise ! audioconvert ! audioresample ! queue ! opusenc ! rtpopuspay ! application/x-rtp,media=audio,encoding-name=OPUS,payload=97 ! sendrecv.

Receive:

gst-launch-1.0 webrtcbin bundle-policy=max-bundle name=sendrecv  stun-server=stun://stun.l.google.com:19302 ! rtpopusdepay ! opusdec ! audioconvert ! autoaudiosink async=false

Thanks!!!

asked Nov 12 '18 at 17:15

Monalisa

Did you try the link https://github.com/centricular/gstwebrtc-demos ? And may be you could mention what is the issue you are facing if you have already tried the centricular demo. – AshSivan Nov 15 '18 at 10:01
Did you ever this working? – Ani Jul 11 '19 at 20:21
make sure you have GStreamer 1.18+ – Ryan Barrett Nov 20 '20 at 02:52

Gstreamer webrtcbin working sample pipeline

0 Answers0