WhipInto

RTP/RTSP to WHIP tool

这个工具应该有三种模式：

• rtp
• rtsp as client
• rtsp as server

RTP

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

TIP

你需要先创建一个 SDP 文件

可以用 ffmpeg -sdp_file flag 来创建 SDP 文件

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

默认是这种模式

这个例子是用 whipinto 作为 RTSP Server，用 ffmpeg 作为 client 用 RTSP 推流

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'

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'

RTSP Client

whipinto 作为一个客户端，从其他的 RTSP Server 来拉流

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

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