go2rtc

go2rtc - ultimate camera streaming application with support RTSP, WebRTC, FFmpeg, RTMP, etc.

• zero-dependency and zero-config small app for all OS (Windows, macOS, Linux, ARM)
• zero-delay for all supported protocols (lowest possible streaming latency)
• zero-load on CPU for supported codecs
• on the fly transcoding for unsupported codecs via FFmpeg
• multi-source 2-way codecs negotiation
• streaming from private networks via Ngrok or SSH-tunnels

Codecs negotiation

For example, you want to watch RTSP-stream from Dahua IPC-K42 camera in your Chrome browser.

• this camera support 2-way audio standard ONVIF Profile T
• this camera support codecs H264, H265 for send video, and you select H264 in camera settings
• this camera support codecs AAC, PCMU, PCMA for send audio (from mic), and you select AAC/16000 in camera settings
• this camera support codecs AAC, PCMU, PCMA for receive audio (to speaker), you don't need to select them
• your browser support codecs H264, VP8, VP9, AV1 for receive video, you don't need to select them
• your browser support codecs OPUS, PCMU, PCMA for send and receive audio, you don't need to select them
• you can't get camera audio directly, because its audio codecs doesn't match with your browser codecs
  • so you decide to use transcoding via FFmpeg and add this setting to config YAML file
  • you have chosen OPUS/48000/2 codec, because it is higher quality than the PCMU/8000 or PCMA/8000
• now you have stream with two sources - RTSP and FFmpeg

go2rtc automatically match codecs for you browser and all your stream sources. This called multi-source 2-way codecs negotiation. And this is one of the main features of this app.

PS. You can select PCMU or PCMA codec in camera setting and don't use transcoding at all. Or you can select AAC codec for main stream and PCMU codec for second stream and add both RTSP to YAML config, this also will work fine.

streams:
  dahua:
    - rtsp://admin:password@192.168.1.123/cam/realmonitor?channel=1&subtype=0&unicast=true&proto=Onvif
    - ffmpeg:rtsp://admin:password@192.168.1.123/cam/realmonitor?channel=1&subtype=0&unicast=true&proto=Onvif#audio=opus

Installation

Download binary for your OS from latest release:

• go2rtc_win64.exe - Windows 64-bit
• go2rtc_win32.exe - Windows 32-bit
• go2rtc_linux_amd64 - Linux 64-bit
• go2rtc_linux_i386 - Linux 32-bit
• go2rtc_linux_arm64 - Linux ARM 64-bit (ex. Raspberry 64-bit OS)
• go2rtc_linux_arm - Linux ARM 32-bit (ex. Raspberry 32-bit OS)
• go2rtc_linux_mipsel - Linux on MIPS (ex. Xiaomi Gateway 3)
• go2rtc_mac_amd64 - Mac with Intel
• go2rtc_mac_arm64 - Mac with M1

Don't forget to fix the rights chmod +x go2rtc_linux_xxx on Linux and Mac.

Configuration

Create file go2rtc.yaml next to the app.

• by default, you need to config only your streams links
• api server will start on default 3000 port
• rtsp server will start on default 554 port
• webrtc will use random UDP port for each connection
• ffmpeg will use default transcoding options (you need to install it manually)

Available modules:

• streams
• api - HTTP API (important for WebRTC support)
• rtsp - RTSP Server (important for FFmpeg support)
• webrtc - WebRTC Server (important for external access)
• ngrok - Ngrok integration (external access for private network)
• ffmpeg - FFmpeg integration
• hass - Home Assistant integration
• log - logs config

Module: Streams

go2rtc support different stream source types. You can config only one link as stream source or multiple.

Available source types:

• rtsp - most cameras on market
• rtmp
• ffmpeg - FFmpeg integration
• exec - advanced FFmpeg and GStreamer integration
• hass - Home Assistant integration

Source: RTSP

• Support RTSP and RTSPS links with multiple video and audio tracks
• Support 2-way audio ONLY for ONVIF Profile T cameras (back channel connection)

Attention: proprietary 2-way audio standards are not supported!

streams:
  sonoff_camera: rtsp://rtsp:12345678@192.168.1.123:554/av_stream/ch0

If your camera has two RTSP links - you can add both of them as sources. This is useful when streams has different codecs, as example AAC audio with main stream and PCMU/PCMA audio with second stream.

Attention: Dahua cameras has different capabilities for different RTSP links. For example, it has support multiple codecs for 2-way audio with &proto=Onvif in link and only one codec without it.

streams:
  dahua_camera:
    - rtsp://admin:password@192.168.1.123/cam/realmonitor?channel=1&subtype=0&unicast=true&proto=Onvif
    - rtsp://admin:password@192.168.1.123/cam/realmonitor?channel=1&subtype=1

Source: RTMP

You can get stream from RTMP server, for example Frigate. Support ONLY H264 video codec without audio.

streams:
  rtmp_stream: rtmp://192.168.1.123/live/camera1

Source: FFmpeg

You can get any stream or file or device via FFmpeg and push it to go2rtc. The app will automatically start FFmpeg with the proper arguments when someone starts watching the stream.

Format: ffmpeg:{input}#{params}. Examples:

streams:
  # [FILE] all tracks will be copied without transcoding codecs
  file1: ffmpeg:~/media/BigBuckBunny.mp4

  # [FILE] video will be transcoded to H264, audio will be skipped
  file2: ffmpeg:~/media/BigBuckBunny.mp4#video=h264

  # [FILE] video will be copied, audio will be transcoded to pcmu
  file3: ffmpeg:~/media/BigBuckBunny.mp4#video=copy&audio=pcmu

  # [HLS] video will be copied, audio will be skipped
  hls: ffmpeg:https://devstreaming-cdn.apple.com/videos/streaming/examples/bipbop_16x9/gear5/prog_index.m3u8#video=copy

  # [MJPEG] video will be transcoded to H264
  mjpeg: ffmpeg:http://185.97.122.128/cgi-bin/faststream.jpg?stream=half&fps=15#video=h264

  # [RTSP] video and audio will be copied
  rtsp: ffmpeg:rtsp://rtsp:12345678@192.168.1.123:554/av_stream/ch0#video=copy&audio=copy

All trascoding formats has built-in templates. But you can override them via YAML config. You can also add your own formats to config and use them with source params.

ffmpeg:
  bin: ffmpeg                                        # path to ffmpeg binary
  link: -hide_banner -i {input}                      # if input is link
  file: -hide_banner -re -stream_loop -1 -i {input}  # if input not link
  rtsp: -hide_banner -fflags nobuffer -flags low_delay -rtsp_transport tcp -i {input}  # if input is RTSP link
  output: -rtsp_transport tcp -f rtsp {output}  # output

  h264:       "-codec:v libx264 -g 30 -preset superfast -tune zerolatency -profile main -level 4.1"
  h264/ultra: "-codec:v libx264 -g 30 -preset ultrafast -tune zerolatency"
  h264/high:  "-codec:v libx264 -g 30 -preset superfast -tune zerolatency"
  h265:       "-codec:v libx265 -g 30 -preset ultrafast -tune zerolatency"
  opus:       "-codec:a libopus -ar 48000 -ac 2"
  pcmu:       "-codec:a pcm_mulaw -ar 8000 -ac 1"
  pcmu/16000: "-codec:a pcm_mulaw -ar 16000 -ac 1"
  pcmu/48000: "-codec:a pcm_mulaw -ar 48000 -ac 1"
  pcma:       "-codec:a pcm_alaw -ar 8000 -ac 1"
  pcma/16000: "-codec:a pcm_alaw -ar 16000 -ac 1"
  pcma/48000: "-codec:a pcm_alaw -ar 48000 -ac 1"
  aac/16000:  "-codec:a aac -ar 16000 -ac 1"

Source: Exec

FFmpeg source just a shortcut to exec source. You can get any stream or file or device via FFmpeg or GStreamer and push it to go2rtc via RTSP protocol:

streams:
  stream1: exec:ffmpeg -hide_banner -re -stream_loop -1 -i ~/media/BigBuckBunny.mp4 -c copy -rtsp_transport tcp -f rtsp {output}

Source: Hass

Support import camera links from Home Assistant config files:

• support ONLY Generic Camera, setup via GUI

hass:
  config: "~/.homeassistant"

streams:
  generic_camera: hass:Camera1  # Settings > Integrations > Integration Name

Module: API

The HTTP API is the main part for interacting with the application.

• you can use WebRTC only when HTTP API enabled
• you can disable HTTP API with listen: "" and use, for example, only RTSP client/server protocol
• you can enable HTTP API only on localhost with listen: "localhost:3000" setting
• you can change API base_path and host go2rtc on your main app webserver suburl
• all files from static_dir hosted on root path: /

api:
  listen: ":3000"    # HTTP API port ("" - disabled)
  base_path: ""      # API prefix for serve on suburl
  static_dir: "www"  # folder for static files ("" - disabled)

Module: RTSP

You can get any stream as RTSP-stream with codecs filter:

rtsp://192.168.1.123/{stream_name}?video={codec}&audio={codec1}&audio={codec2}

• you can omit the codecs, so one first video and one first audio will be selected
• you can set ?video=copy or just ?video, so only one first video without audio will be selected
• you can set multiple video or audio, so all of them will be selected

rtsp:
  listen: ":554"

Module: WebRTC

WebRTC usually works without problems in the local network. But external access may require additional settings. It depends on what type of internet do you have.

• by default, WebRTC use two random UDP ports for each connection (for video and audio)
• you can enable one additional TCP port for all connections and use it for external access

Static public IP

• add some TCP port to YAML config (ex. 8555)
• forward this port on your router (you can use same 8555 port or any other)
• add your external IP-address and external port to YAML config

webrtc:
  listen: ":8555"  # address of your local server (TCP)
  candidates:
    - 216.58.210.174:8555  # if you have static public IP-address

Dynamic public IP

• add some TCP port to YAML config (ex. 8555)
• forward this port on your router (you can use same 8555 port or any other)
• add stun word and external port to YAML config
  • go2rtc automatically detects your external address with STUN-server

webrtc:
  listen: ":8555"  # address of your local server (TCP)
  candidates:
    - stun:8555  # if you have dynamic public IP-address

Private IP

• add some TCP port to YAML config (ex. 8555)
• setup integration with Ngrok service

webrtc:
  listen: ":8555"  # address of your local server (TCP)

ngrok:
  command: ...

Own TCP-tunnel

If you have personal VPS, you can create TCP-tunnel and setup in the same way as "Static public IP". But use your VPS IP-address in YAML config.

Using TURN-server

TODO...

webrtc:
  ice_servers:
    - urls: [stun:stun.l.google.com:19302]
    - urls: [turn:123.123.123.123:3478]
      username: your_user
      credential: your_pass

Module: Ngrok

With Ngrok integration you can get external access to your streams in situation when you have Internet with private IP-address.

• you may need external access for two different things:
  • WebRTC stream, so you need tunnel WebRTC TCP port (ex. 8555)
  • go2rtc web interface, so you need tunnel API HTTP port (ex. 3000)
• Ngrok support authorization for your web interface
• Ngrok automatically adds HTTPS to your web interface

Ngrok free subscription limitations:

• you will always get random external address (not a problem for webrtc stream)
• you can forward multiple ports but use only one Ngrok app

go2rtc will automatically get your external TCP address (if you enable it in ngrok config) and use it with WebRTC connection (if you enable it in webrtc config).

You need manually download Ngrok agent app for your OS and register in Ngrok service.

Tunnel for only WebRTC Stream

You need to add your Ngrok token and WebRTC TCP port to YAML:

ngrok:
  command: ngrok tcp 8555 --authtoken eW91IHNoYWxsIG5vdCBwYXNzCnlvdSBzaGFsbCBub3QgcGFzcw

Tunnel for WebRTC and Web interface

You need to create ngrok.yaml config file and add it to go2rtc config:

ngrok:
  command: ngrok start --all --config ngrok.yaml

Ngrok config example:

version: "2"
authtoken: eW91IHNoYWxsIG5vdCBwYXNzCnlvdSBzaGFsbCBub3QgcGFzcw
tunnels:
  api:
    addr: 3000  # use the same port as in go2rtc config
    proto: http
    basic_auth:
      - admin:password  # you can set login/pass for your web interface
  webrtc:
    addr: 8555  # use the same port as in go2rtc config
    proto: tcp

Module: Hass

go2rtc compatible with Home Assistant RTSPtoWebRTC integration API.

• add integration with link to go2rtc HTTP API:
  • Hass > Settings > Integrations > Add Integration > RTSPtoWebRTC > http://192.168.1.123:3000/
• add generic camera with RTSP link:
  • Hass > Settings > Integrations > Add Integration > Generic Camera > rtsp://...
• use Picture Entity or Picture Glance lovelace card
• open full screen card - this is should be WebRTC stream

Module: Log

You can set different log levels for different modules.

log:
  level: info  # default level
  api: trace
  exec: debug
  ngrok: info
  rtsp: warn
  streams: error
  webrtc: fatal