[gst-rpwc]
type = rtp
id = 1
description = Test Stream
audio = no
video = yes
videoport = 8004
videopt = 96
videortpmap = VP8/90000
videofmtp = profile-level-id=42e028\;packetization-mode=1
ffmpeg -f qtkit -video_device_index 0 -i "" -video_size 640x480 -f mpeg1video -b:v 800k -r 30
Edit: I tried ffmpeg -f qtkit -video_device_index 0 -i "" -video_size 640x480 -f mpeg1video -b:v 800k -r 30 http://x.x.x.x:8004 but I got connection refused.
So just videofmtp = profile-level-id=42e028
Excuse my ignorance I'm pretty new to ffmpeg and the world of encoding. Is the video format for VP8 as follows?ffmpeg -f qtkit -video_device_index 0 -i "" -video_size 640x480 -vcodec libvpx_vp8 -f alsa -b:v 800k -r 30 rtp://x.x.x.x:8004I get invalid output format for -f alsa, also for -f video4linux2. What other output formats would work with OSX for VP8 encoding?
[gst-rpwc]
type = rtp
id = 1
description = RPWC H264 test streaming
audio = no
video = yes
videoport = 8004
videopt = 100
videortpmap = VP8/90000
And my publisher looks like
ffmpeg -f avfoundation -video_size 640x480 -framerate 30 -i "0" -b:v 800k -c:v libvpx rtp://178.62.29.190:8004
Input #0, avfoundation, from '0':
Duration: N/A, start: 84921.465000, bitrate: N/A
Stream #0:0: Video: rawvideo (UYVY / 0x59565955), uyvy422, 640x480, 30 fps, 30 tbr, 1000k tbn, 1000k tbc
[NULL @ 0x7fb50a012000] Unable to find a suitable output format for 'rtp://178.62.29.190:8004'
rtp://178.62.29.190:8004: Invalid argument
ffmpeg -f avfoundation -video_size 640x480 -framerate 30 -i "0" -an -c:v libvpx -f rtp rtp://x.x.x.x:8004
HTTP/Janus sessions watchdog started
[test] New video stream! (ssrc=1621663919)
Creating new session: 67846169
Creating new handle in session 67846169: 3003307565
[3003307565] Creating ICE agent (ICE Full mode, controlling)
[WARN] [3003307565] Still waiting for the DTLS stack for component 1 in stream 1...
[3003307565] The DTLS handshake has been completed
WebRTC media is now available
ffmpeg -f avfoundation -video_size 640x480 -framerate 25 -i "0" -an -c:v libvpx -quality realtime -error-resilient default -f rtp rtp://x.x.x.x:8004
Any other hidden params of ffmpeg I could try to make it more efficient?