Getting Started

Introduction

Current support encode

protocol	video codecs	audio codecs
WHIP	AV1, VP9, VP8, H264	Opus, G722
WHEP	AV1, VP9, VP8, H264	Opus, G722

Current client support

Application	AV1	VP9	VP8	H264	OPUS	G722
Browser whip	⭐	⭐	⭐	⭐	⭐	⭐
Browser whep	⭐	⭐	⭐	⭐	⭐	⭐
Gstreamer whip	📺 1	⭐	⭐	⭐	⭐	⭐
Gstreamer whep	📺 2	⭐	⭐	⭐	⭐	⭐
Gstreamer rtp-into	📺 1	⭐	⭐	⭐	⭐	⭐
Gstreamer rtp-from	📺 2	⭐	⭐	⭐	⭐	⭐
FFmpeg rtp-into	💩	⭐	⭐	⭐	⭐	⭐
FFmpeg rtp-from	💩	⭐	⭐	⭐	⭐	⭐
VLC rtp-into	💩	💩	⭐	⭐	⭐	💩
VLC rtp-from	💩	💩	⭐	⭐	⭐	💩
OBS Studio whip	📺 3	💩	💩	⭐	⭐	💩

• ⭐ It's working
• 💩 Don't support
• 💡 I don't know, No testing
• 📺 Note have some problem
  1. Working, But Browser can't player this video, Gstreamer to Gstreamer is working
  2. I don't know why av1 and whep error
  3. OBS av1 codec can't play

Components

Live777 Core (liveion)

A Pure Single SFU Server for WebRTC.

Only WHIP / WHEP protocol supported.

a core SFU server, If you need a single server, use this

Live777 Manager (liveman)

Live777 Cluster manager.

If I have so many servers (live777 core cluster), I need this manage them all.

Cluster mode must liveman

whipinto and whepfrom

Stream protocol convert tool.

• RTP => WHIP
• WHEP => RTP
• TODO: RTSP => WHIP
• TODO: WHEP => RTSP

whipinto

NOTE: About pkt_size=1200

WebRTC must need pkt_size=1200

If pkt_size > 1200 (most tool > 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

---

This tool is rtp2whip

whipinto -c vp8 -u http://localhost:7777/whip/777 --port 5003

whepfrom

This tool is whep2rtp

Build

cargo build --package=whepfrom --release

Use WHEP protocol pull stream convert rtp sender

whepfrom -c vp8 -u http://localhost:7777/whep/777 -t localhost:5004

Web WHIP/WHEP client

Open your browser, enter the URL: http://localhost:7777/

Debugger

example: http://localhost:7777/tools/debugger.html

You can use this test WebRTC-SVC

Single Page Player

example: http://localhost:7777/tools/player.html?id=web-0&autoplay&controls&muted&reconnect=3000

URL params:

• id: string, live777 Stream ID
• autoplay: boolean
• controls: boolean
• muted: boolean, whether to mute by default
• reconnect: number, reconnect timeout in millisecond

Minimize Running

Run Live777 using docker:

DANGER

You must use network host mode

docker run --name live777-server --rm --network host ghcr.io/binbat/live777-server:latest live777

Open your browser, enter the URL: http://localhost:7777/