WhipInto

RTP/RTSP to WHIP tool

Streaming a camera from a Raspberry Pi / ARM board?

Do not pair whipinto with a software-encoding ffmpeg on these boards — the CPU cannot keep up. The native-rpi / native-rkmpp live777 builds capture and encode in hardware, in-process; see the Raspberry Pi deployment guide. whipinto is for sources that are already encoded (IP cameras, RTP/RTSP feeds, files, synthetic test sources).

This tool has two working modes:

• rtp
• rtsp as client

Use whepfrom for the reverse WHEP → RTP/RTSP direction, or follow the three-terminal loop.

Get it and prerequisites

Download a package or prebuilt binary using the installation guide, or build from this repository with cargo build --release --bin whipinto. You need a reachable Live777 WHIP endpoint and an already-encoded RTP/RTSP source. FFmpeg is required only for the examples below; synth:// input requires a build with --features=rsmpeg.

Bearer token

If live777.toml contains tokens = ["live777"] under [auth], pass the same token with -t:

whipinto -i input.sdp -t live777 -w http://127.0.0.1:7777/whip/777

Do not put tokens in URLs or commit them to shell history and configuration files intended for publication.

Options

Option	Default	Description
-i, --input	input.sdp	Input source: SDP file path (RTP), 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
--cable-url	none	Cable UDP bridge URL for DataChannel <-> UDP forwarding, e.g. udp://0.0.0.0:9001?host=127.0.0.1&port=9000
--cable-label	-	DataChannel label; joins the matching cable on the server
--ice-server	none	ICE server for gathering, repeatable; format <url>[,<username>[,<credential>]]. Defaults to none — host candidates only (WHIP/WHEP endpoints advertise their own ICE servers via Link headers)
-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 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

Troubleshooting

Run with -vv to include WebRTC and bridge diagnostics. Authentication, HTTP status codes and ICE failures are covered once in Troubleshooting; the rows below are specific to this tool.

Symptom	What to check
404 Not Found	Check the /whip/<stream> URL, server port and reverse-proxy path. Confirm you are publishing to Live777 rather than a WHEP endpoint.
Connected, but no media	Confirm the sender is emitting to the SDP ports, the SDP codecs match the encoded input, and RTSP/SDP inputs are reachable. Check packet size and codec support above.