WhipInto

RTP/RTSP to WHIP tool

This tool has three working mode:

• rtp
• rtsp as client
• rtsp as server

Options

Option	Default	Description
-i, --input	sdp://0.0.0.0:8554	Input source: sdp:// (RTP/SDP file or RTSP server mode), rtsp:// (RTSP client mode), synth:// (generated test frames)
-w, --whip	required	WHIP endpoint URL
-t, --token	none	Bearer token for WHIP authentication
--command	none	Run a command as child process
--ice-server	stun:stun.l.google.com:19302	ICE server for gathering, repeatable; format <url>[,<username>[,<credential>]] (empty string disables ICE servers)
-v	warn	Increase verbosity (-v info, -vv debug, -vvv trace)

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=1280x720:rate=30 \
-vcodec libx264 -pix_fmt yuv420p \
-g 60 -keyint_min 60 -crf 23 \
-preset ultrafast -tune zerolatency \
-profile:v main -level 4.1 \
-f rtp 'rtp://127.0.0.1:5002' -sdp_file input.sdp

RTP Only audio

ffmpeg -re -f lavfi -i sine=frequency=1000 \
-acodec libopus \
-ar 48000 -ac 2 \
-b:a 48k -application voip \
-frame_duration 10 -vbr constrained \
-f rtp 'rtp://127.0.0.1:5004' -sdp_file input.sdp

RTP Audio and Video

ffmpeg -re \
-f lavfi -i sine=frequency=1000 \
-f lavfi -i testsrc=size=1280x720: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=1280x720:rate=30 \
-vcodec libvpx -f rtsp 'rtsp://127.0.0.1:8554'

ffmpeg -re -f lavfi -i testsrc=size=1280x720:rate=30 \
-vcodec -pix_fmt yuv420p \
-g 60 -keyint_min 60 -crf 23 \
-preset ultrafast -tune zerolatency \
-profile:v main -level 4.1 \
-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=1280x720: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=1280x720: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

Synthetic input

With the rsmpeg feature enabled, -i also accepts a synth:// URL that generates test frames in-process (no external encoder needed):

whipinto -i 'synth://h264?audio=opus&width=1280&height=720&fps=30' \
  -w http://localhost:7777/whip/777

Format: synth://<vcodec>?<parameters>; <vcodec> is one of vp8, vp9, h264, h265, av1 and every parameter is optional:

Parameter	Default	Description
audio	none	Audio codec: opus, g722 (omit for no audio)
width	640	Video width in pixels
height	480	Video height in pixels
fps	30	Video frame rate
duration	none	Stop publishing after this many seconds
ice	--ice-server value	ICE server spec <url>[,<username>[,<credential>]], repeatable; replaces the CLI list for this input (an empty value disables ICE servers)

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 AV1, VP8 and VP9, You can use any pkt_size value in AV1, VP8 and VP9

Codec	AV1	VP9	VP8	H264	H265	OPUS	G722
pkt_size > 1200	⭐	⭐	⭐	⭐	⭐	⭐	💩

• ⭐ It's working
• 💩 Don't support