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LiveForge

High-performance multi-protocol live streaming server written in Go.

LiveForge is an open-source, modular media streaming server that ingests, transmuxes, and delivers audio/video in real time. It supports RTMP, RTSP, SRT, WebRTC (WHIP/WHEP), HLS, LL-HLS, DASH, HTTP-FLV, HTTP-TS, FMP4, GB28181, and WebSocket streaming -- all from a single binary with zero external dependencies.

Supported Protocols

Direction	Protocols
Ingest (Publish)	RTMP, RTSP (TCP + UDP), SRT, WebRTC WHIP, GB28181
Playback (Subscribe)	RTMP, RTSP, SRT, WebRTC WHEP, HLS, LL-HLS, DASH, HTTP-FLV, HTTP-TS, FMP4, WebSocket

Any ingest protocol can be played back via any delivery protocol. For example: RTMP push to WebRTC pull, or GB28181 camera to HLS playback.

Supported Codecs

• Video: H.264, H.265/HEVC, VP8, VP9, AV1
• Audio: AAC, Opus, G.711 (PCMU/PCMA), MP3, G.722, G.729, Speex

Key Features

• Audio Transcoding — On-demand codec bridging (AAC ↔ Opus ↔ G.711 ↔ MP3) between protocols
• GB28181 — Full video surveillance support: SIP signaling, live invite, playback, PTZ, alarm, device simulator
• Cluster Deployment — Multi-protocol forwarding and origin pull (RTMP/SRT/RTSP/RTP/GB28181) with HTTP scheduler
• LL-HLS — Low-Latency HLS with partial segments and blocking playlist reload
• Prometheus Metrics — Server-level and per-stream monitoring with Grafana-ready queries
• Testing Tools — Built-in lf-test CLI and GB28181 device simulator

Quick Start

Build and Run

git clone https://github.com/im-pingo/liveforge.git
cd liveforge
go build -o liveforge ./cmd/liveforge
./liveforge -c configs/liveforge.yaml

Publish a Stream

RTMP (OBS / FFmpeg):

ffmpeg -re -i input.mp4 -c copy -f flv rtmp://localhost:1935/live/stream1

RTSP:

ffmpeg -re -i input.mp4 -c copy -f rtsp rtsp://localhost:8554/live/stream1

SRT:

ffmpeg -re -i input.mp4 -c copy -f mpegts "srt://localhost:6000?streamid=publish:/live/stream1"

WebRTC (Browser):

Open http://localhost:8090/console, click "+ WebRTC Publish", select camera/mic, and start streaming.

GB28181:

Configure your IP camera to point at localhost:5060, or use the built-in simulator:

go run ./tools/gb28181-sim -server 127.0.0.1:5060

Play a Stream

Protocol	URL
RTMP	rtmp://localhost:1935/live/stream1
RTSP	rtsp://localhost:8554/live/stream1
SRT	srt://localhost:6000?streamid=subscribe:/live/stream1
HLS / LL-HLS	http://localhost:8080/live/stream1.m3u8
DASH	http://localhost:8080/live/stream1.mpd
HTTP-FLV	http://localhost:8080/live/stream1.flv
HTTP-TS	http://localhost:8080/live/stream1.ts
FMP4	http://localhost:8080/live/stream1.mp4
WebSocket	ws://localhost:8080/ws/live/stream1.flv
WebRTC	Open console, select stream, preview via WebRTC tab

Navigation

See the sidebar for the full list of documentation pages covering configuration, protocols, GB28181, cluster deployment, audio transcoding, LL-HLS, authentication, recording, webhooks, REST API, metrics, testing tools, architecture, usage scenarios, and troubleshooting.