WhipInto

RTP/RTSP to WHIP tool

This tool has three working mode:

• rtp
• rtsp as client
• rtsp as server

RTP

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

TIP

You need to generate an sdp file first

For example: Use ffmpeg -sdp_file flag

RTP Only video

ffmpeg -re -f lavfi -i testsrc=size=640x480:rate=30 \
-vcodec libvpx -f rtp 'rtp://127.0.0.1:5003' -sdp_file input.sdp

RTP Only audio

ffmpeg -re -f lavfi -i sine=frequency=1000 \
-acodec libopus -f rtp 'rtp://127.0.0.1:5005' -sdp_file input.sdp

RTP 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://127.0.0.1:11111 \
-vcodec libvpx -an -f rtp rtp://127.0.0.1:11113 -sdp_file input.sdp

RTSP Server

It's default mode

This example is whipinto as RTSP Server, use ffmpeg as client use RTSP push stream

whipinto -w http://localhost:7777/whip/777

Only video

ffmpeg -re -f lavfi -i testsrc=size=640x480:rate=30 \
-vcodec libvpx -f rtsp 'rtsp://127.0.0.1:8554'

NOTE:

For H264 FFmpeg RTSP client

Your must use -x264-params repeat_headers=1

ffmpeg -re -f lavfi -i testsrc=size=640x480:rate=30 \
-vcodec libx264 \
-profile:v baseline -level 3.1 -pix_fmt yuv420p \
-g 15 -keyint_min 15 -b:v 1000k \
-minrate 1000k -maxrate 1000k -bufsize 1000k \
-preset ultrafast -tune zerolatency \
-x264-params repeat_headers=1 \
-f rtsp rtsp://127.0.0.1:8554

Only audio

ffmpeg -re -f lavfi -i sine=frequency=1000 \
-acodec libopus -f rtsp 'rtsp://127.0.0.1:8554'

Audio and Video

ffmpeg -re \
-f lavfi -i sine=frequency=1000 \
-f lavfi -i testsrc=size=640x480:rate=30 \
-acodec libopus -vcodec libvpx \
-f rtsp 'rtsp://127.0.0.1:8554'

Use transport tcp

ffmpeg -re \
-f lavfi -i sine=frequency=1000 \
-f lavfi -i testsrc=size=640x480:rate=30 \
-acodec libopus -vcodec libvpx \
-rtsp_transport tcp \
-f rtsp 'rtsp://127.0.0.1:8554'

RTSP Client

whipinto as a client, pull stream from RTSP Server

whipinto -i rtsp://127.0.0.1:8554 -w http://localhost:7777/whip/777

Use transport tcp

whipinto -i rtsp://localhost:8554/test-rtsp?transport=tcp -w http://localhost:7777/whip/test-rtsp

About pkt_size=1200

WARNING

WebRTC must need pkt_size<=1200

If pkt_size > 1200 (most tool default > 1200, for example: ffmpeg default 1472), we need to de-payload after re-payload

But now, We support re-size pkt_size in VP8 and VP9, You can use any pkt_size value in VP8 and VP9

Codec	AV1	VP9	VP8	H264	OPUS	G722
pkt_size > 1200	💩	⭐	⭐	⭐	⭐	💩

• ⭐ It's working
• 💩 Don't support