(no subject)

[Ottavio Campana-2]

Hello,

I am trying to bridge H.264 over RSTP from a camera to webRTC for seeing the video in the browser.

I started by setting up the example https://gitlab.freedesktop.org/gstreamer/gst-examples/-/blob/master/webrtc/sendrecv/gst/webrtc-sendrecv.c and it works.

My successive step was moving from VP8 to H.264, thus in the function start_pipeline I changed

"videotestsrc is-live=true pattern=ball ! videoconvert ! queue ! vp8enc deadline=1 ! rtpvp8pay ! "

"queue ! " RTP_CAPS_VP8 "96 ! sendrecv. "

to

videotestsrc is-live=true pattern=ball ! videoconvert ! queue ! x264enc ! rtph264pay config-interval=-1 ! "

"queue ! " RTP_CAPS_H264 "96 ! sendrecv. "

and this also works. At this point I tried to change the pipeline, by fetching the nal units from a RTSP source and by sending them to the webrtcbin

"rtspsrc location=rtsp://192.168.69.159/live2.sdp latency=0 ! queue ! rtph264depay ! h264parse ! rtph264pay config-interval=-1 ! "

"queue ! " RTP_CAPS_H264 "96 ! sendrecv. "

With firefox 85.0.1 the example hangs with "Sending SDP answer". Neither video nor audio flaw between webrtc-sednrecv and firefox. On the other hand, if I try it with chrome 88.0.4323.150 it works as expected.

I suspect that there must be something wrong in my pipe, but I cannot understand what. Can you please give me a hint?

Thank you,

Ottavio

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