Hi, all:
I am trying to send rtp data to a webrtc media
server(
https://github.com/medooze/media-server-node)
if i generate a a rtmp source with ffmpeg and mux the rtmp to rtp with
gstreamer it works.
the command is:
ffmpeg:
ffmpeg -f lavfi -re -i color=black:s=640x480:r=15 -filter:v
"drawtext=text='%{localtime\:%T}':fontcolor=white:fontsize=80:x=20:y=20" \
-vcodec libx264 -tune zerolatency -preset ultrafast -bsf:v h264_mp4toannexb
\
-g 15 -keyint_min 15 -profile:v baseline -pix_fmt yuv420p -r 15 -f flv
rtmp://localhost/live/live
gstreamer command:
gst-launch-1.0 -v rtmpsrc location=rtmp://localhost/live/live ! flvdemux !
h264parse ! rtph264pay config-interval=-1 pt=100 ! udpsink host=127.0.0.1
port={port}
if i use gstreamer's videotestsrc, it does not work(brower can not play,
send many pli rtcp):
the gstreamer command is:
gst-launch-1.0 -v videotestsrc is-live=true !
video/x-raw,format=I420,framerate=15/1 ! x264enc bframes=0
speed-preset=veryfast key-int-max=15 !
video/x-h264,stream-format=byte-stream,profile=baseline ! h264parse !
rtph264pay config-interval=-1 pt=100 ! udpsink host=127.0.0.1 port={port}
so what't the diffrence between this, does i miss something?
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