FFmpeg

我们有 whipinto 和 whepfrom 来把 rtp <-> whip/whep 进行转换

For Example:

ffmpeg -> whipinto -> live777 -> whepfrom -> ffplay

Video: VP8

Video Test Src

# send RTP and Create SDP file
ffmpeg -re -f lavfi -i testsrc=size=640x480:rate=30 \
-vcodec libvpx -f rtp 'rtp://127.0.0.1:5002' -sdp_file input.sdp

For ffplay. You Need a sdp file

ffplay -protocol_whitelist rtp,file,udp -i input.sdp

You can use whipinto tools receiver RTP and convert to WebRTC

# Use SDP file as input
whipinto -i input.sdp -w http://localhost:7777/whip/777

whepfrom -o output.sdp -w http://localhost:7777/whep/777

For ffplay. You Need a sdp file

ffplay -protocol_whitelist rtp,file,udp -i output.sdp

Video: VP9

WARNING

Packetizing VP9 is experimental and its specification is still in draft state. Please set -strict experimental in order to enable it.

ffmpeg -re -f lavfi -i testsrc=size=640x480:rate=30 \
-strict experimental -vcodec libvpx-vp9 \
-f rtp 'rtp://127.0.0.1:5002' -sdp_file input.sdp

whipinto -i input.sdp -w http://localhost:7777/whip/777

Video: H264

ffmpeg -re -f lavfi -i testsrc=size=640x480:rate=30 -vcodec libx264 \
-x264-params "level-asymmetry-allowed=1:packetization-mode=1:profile-level-id=42001f" \
-f rtp 'rtp://127.0.0.1:5002' -sdp_file input.sdp

whipinto -i input.sdp -w http://localhost:7777/whip/777

Video: AV1

DANGER

RTP Unsupported codec av1

Audio: Opus

ffmpeg -re -f lavfi -i sine=frequency=1000 \
-acodec libopus -f rtp 'rtp://127.0.0.1:5004'

Audio: G722

ffmpeg -re -f lavfi -i sine=frequency=1000 \
-acodec g722 -f rtp 'rtp://127.0.0.1:5004?pkt_size=1200'

Both: Audio and Video

ffmpeg -re \
-f lavfi -i sine=frequency=1000 \
-f lavfi -i testsrc=size=640x480:rate=30 \
-acodec libopus -vn -f rtp rtp://127.0.0.1:5002 \
-vcodec libvpx -an -f rtp rtp://127.0.0.1:5004 -sdp_file input.sdp