go2rtc

go2rtc 用于统一接入 RTSP、WebRTC / MSE、转流与双向音频。

基础示例

go2rtc:
  streams:
    front_main:
      - rtsp://user:password@192.168.1.10:554/main
    front_sub:
      - rtsp://user:password@192.168.1.10:554/sub

WebRTC 示例

go2rtc:
  webrtc:
    candidates:
      - 192.168.1.100:8555
      - stun:stun.l.google.com:19302
    port_min: 8555
    port_max: 8559

验证

先在 go2rtc 管理页确认视频可播放，再把 rtsp://127.0.0.1:8554/STREAM_NAME 交给摄像头输入。这样可减少摄像头并发连接数量。

常见问题

• RTSP 账号密码包含特殊字符时注意 URL 编码。
• WebRTC 不通时检查 8555/UDP、NAT 与候选地址。
• 双向音频依赖摄像头音频编码、协议与 go2rtc 支持情况。

go2rtc

go2rtc keeps a small number of stable upstream camera sessions and reuses them for FFmpeg, WebRTC/MSE, Home Assistant, or other consumers. This is valuable for cameras with connection limits or fragile RTSP implementations.

Recommended topology

Camera → go2rtc → Securex FFmpeg/browser. Detection, recording, and live playback can then reuse the same upstream source.

Example

go2rtc:
  streams:
    front:
      - rtsp://user:password@192.168.1.50:554/stream1
    front_sub:
      - rtsp://user:password@192.168.1.50:554/stream2

cameras:
  front:
    ffmpeg:
      inputs:
        - path: rtsp://127.0.0.1:8554/front_sub
          input_args: preset-rtsp-restream
          roles: [detect]
        - path: rtsp://127.0.0.1:8554/front
          input_args: preset-rtsp-restream
          roles: [record]

Naming

• Use clear main/sub names such as front and front_sub.
• Keep camera and go2rtc names easy to relate.
• URL-encode reserved characters in credentials.

Verification order

1. Verify continuous playback in go2rtc first.
2. Point FFmpeg to the local restream next.
3. Test WebRTC/MSE last; do not change all layers simultaneously.

Common issues

• go2rtc plays but no recording: inspect FFmpeg inputs and roles.
• Low-latency playback fails: inspect WebRTC ports, candidates, firewall/NAT.
• Repeated disconnects: prove source stability at the go2rtc layer first.

go2rtc

go2rtc mantiene pocas conexiones hacia las cámaras y reutiliza los streams para FFmpeg, WebRTC/MSE, Home Assistant u otros clientes.

Topología recomendada

Cámara → go2rtc → FFmpeg/navegador. Así detección, grabación y Live comparten la fuente.

Ejemplo

go2rtc:
  streams:
    front:
      - rtsp://user:password@192.168.1.50:554/stream1
    front_sub:
      - rtsp://user:password@192.168.1.50:554/stream2

cameras:
  front:
    ffmpeg:
      inputs:
        - path: rtsp://127.0.0.1:8554/front_sub
          input_args: preset-rtsp-restream
          roles: [detect]
        - path: rtsp://127.0.0.1:8554/front
          input_args: preset-rtsp-restream
          roles: [record]

Nombres

• Use nombres claros main/sub.
• Relacione nombres de cámara y go2rtc.
• Codifique caracteres reservados en credenciales.

Orden de prueba

1. Compruebe go2rtc.
2. Después configure FFmpeg con el restream local.
3. Finalmente pruebe WebRTC/MSE.

Problemas

• go2rtc funciona pero no graba: revise inputs/roles.
• Falla baja latencia: revise puertos/candidates/firewall.
• Desconexiones: compruebe primero la fuente.