Gstreamer webrtcbin working sample pipeline

Question

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!!!

Incoordination

Did you try the link 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. – Maddening

Did you ever this working? – Froebel

make sure you have GStreamer 1.18+ – Yetah

Answer 1

The short answer is, you will not be able to run a WebRTC pipeline from the commandline. Another user already posted some example code that you will need to adapt for your use case: https://github.com/centricular/gstwebrtc-demos

The commandline example is missing a critical piece of WebRTC: the signalling server. The sending and receiving ends of the pipeline need to be able to swap two bits of information before a connection is established via WebRTC: The SDP, and the ICE candidates. This will allow them to negotiate the format and the parameters of the stream (SDP) and a way to connect to each other via a peer to peer connection (ICE). Without this, a connection cannot be established.

Note that the WebRTC specification does not specify how the signalling server needs to be implemented. It is perfectly valid to exchange the SDP and ICE candidates via email for example, however it makes more sense for the signalling server to be an actual server.

Cromorne