Protocol Overview

Monibuca V6 adopts a plugin-based architecture where each protocol is provided as an independent plugin. Pre-compiled binaries and Docker images include support for all protocols, which can be enabled on demand through configuration files.

Protocol Comparison

Protocol	Publish	Subscribe	Latency	Typical Use Cases
RTMP	✅	✅	1-3s	OBS/FFmpeg publishing, live streaming distribution
RTSP	✅	✅	0.5-2s	Camera integration, security surveillance
HTTP-FLV	❌	✅	1-3s	Web playback, low-latency live streaming
HLS	❌	✅	5-30s	Large-scale distribution, mobile playback, CDN compatible
WebRTC	✅	✅	< 500ms	Ultra-low latency interaction, video conferencing
SRT	✅	✅	120ms+	Remote contribution, lossy network transport, broadcast-grade ingest
GB28181	✅	✅	1-3s	National standard video surveillance, platform cascading
WebTransport	✅	✅	< 500ms	Next-generation low-latency web transport

Protocol Configuration

Each protocol plugin can be enabled or disabled on demand in the configuration file:

# config.yaml — Enable protocols as needed
rtmp:
  enable: true
  tcp:
    listenaddr: ":1935"

rtsp:
  enable: true
  tcp:
    listenaddr: ":8554"

hls:
  enable: true

webrtc:
  enable: true
  portrange: "udp:9000-9100"

# Set unwanted protocols to false or omit them
srt:
  enable: false

Note

Pre-compiled binaries and Docker images include all protocol plugins by default. Set enable: false to disable any protocol you don’t need.

Cross-Protocol Transcoding

Monibuca V6 supports automatic audio transcoding between different protocols to ensure cross-protocol playback compatibility.

Opus ↔ AAC Transcoding

WebRTC and WebTransport use Opus audio encoding, while traditional protocols such as RTMP, HTTP-FLV, and HLS use AAC audio encoding. When streams are forwarded between these protocols, the engine automatically performs audio transcoding:

Scenario	Transcoding Direction	Description
WebRTC publish → RTMP subscribe	Opus → AAC	Automatic transcoding
WebRTC publish → HLS subscribe	Opus → AAC	Automatic transcoding
RTMP publish → WebRTC subscribe	AAC → Opus	Automatic transcoding
RTMP publish → RTMP subscribe	No transcoding needed	Direct forwarding
WebRTC publish → WebRTC subscribe	No transcoding needed	Direct forwarding

To enable audio transcoding, activate the transcode feature:

[features]
transcode = ["codec/transcode"]

Video Codec Compatibility

Video codecs (H.264/H.265) are universal across all protocols and do not require transcoding. Video codec support by protocol:

Codec	RTMP	RTSP	HTTP-FLV	HLS	WebRTC	SRT	GB28181	WebTransport
H.264	✅	✅	✅	✅	✅	✅	✅	✅
H.265	✅	✅	✅	✅	✅	✅	✅	✅

Protocol Selection Guide

• Publishing: RTMP is recommended (best compatibility) or SRT (more stable on lossy networks)
• Web playback: HTTP-FLV (low latency) or HLS (high compatibility) is recommended
• Ultra-low latency: WebRTC or WebTransport is recommended
• Security surveillance: RTSP or GB28181 is recommended
• Large-scale distribution: HLS is recommended (CDN-friendly)