Really struggling with gstreamer + webrtc

I am running this example from here

gst-launch-1.0 videotestsrc ! videoconvert ! openh264enc ! rtph264pay ! \
'application/x-rtp,media=video,encoding-name=H264,payload=97,clock-rate=90000' ! \
whip.sink_0 audiotestsrc ! audioconvert ! opusenc ! rtpopuspay ! \
'application/x-rtp,media=audio,encoding-name=OPUS,payload=96,clock-rate=48000,encoding-params=(string)2' ! \
whip.sink_1 whipsink name=whip auth-token=$WHIP_TOKEN whip-endpoint=$WHIP_ENDPOINT

it gives me this error:

WARNING: erroneous pipeline: could not link rtph264pay0 to whip, whip can't handle caps application/x-rtp, media=(string)video, encoding-name=(string)H264, payload=(int)97, clock-rate=(int)90000

I am very new to gstreamer, I have no idea how to fix it and haven’t slept all night… would really appreciate any help!!

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