Question about bridging H.264 from RTSP to webRTC

[Ottavio Campana]

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


-- 
Non c'è più forza nella normalità, c'è solo monotonia
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://lists.freedesktop.org/archives/gstreamer-devel/attachments/20210209/54d06cee/attachment.htm>