search

AlexxIT / go2rtc

Ultimate camera streaming application with support RTSP, RTMP, HTTP-FLV, WebRTC, MSE, HLS, MP4, MJPEG, HomeKit, FFmpeg, etc.
https://github.com/AlexxIT/Blog
MIT License
3.7k stars 302 forks source link
ffmpeg h264 h265 hassio hls home-assistant homekit http-flv media-server mjpeg mp4 ngrok rtmp rtp rtsp rtsp-server streaming webcam-streaming webrtc

readme

![go2rtc](assets/logo.gif)
[![stars](https://img.shields.io/github/stars/AlexxIT/go2rtc?style=flat-square&logo=github)](https://github.com/AlexxIT/go2rtc/stargazers) [![docker pulls](https://img.shields.io/docker/pulls/alexxit/go2rtc?style=flat-square&logo=docker&logoColor=white&label=pulls)](https://hub.docker.com/r/alexxit/go2rtc) [![releases](https://img.shields.io/github/downloads/AlexxIT/go2rtc/total?color=blue&style=flat-square&logo=github)](https://github.com/AlexxIT/go2rtc/releases) [![goreport](https://goreportcard.com/badge/github.com/AlexxIT/go2rtc)](https://goreportcard.com/report/github.com/AlexxIT/go2rtc)

Ultimate camera streaming application with support RTSP, WebRTC, HomeKit, FFmpeg, RTMP, etc.

Inspired by:

Fast start

  1. Download binary or use Docker or Home Assistant Add-on or Integration
  2. Open web interface: http://localhost:1984/

Optionally:

Developers:

go2rtc: Binary

Download binary for your OS from latest release:

Don't forget to fix the rights chmod +x go2rtc_xxx_xxx on Linux and Mac.

go2rtc: Docker

The Docker container alexxit/go2rtc supports multiple architectures including amd64, 386, arm64, and arm. This container offers the same functionality as the Home Assistant Add-on but is designed to operate independently of Home Assistant. It comes preinstalled with FFmpeg, ngrok, and Python.

go2rtc: Home Assistant Add-on

  1. Install Add-On:
    • Settings > Add-ons > Plus > Repositories > Add https://github.com/AlexxIT/hassio-addons
    • go2rtc > Install > Start
  2. Setup Integration

go2rtc: Home Assistant Integration

WebRTC Camera custom component can be used on any Home Assistant installation, including HassWP on Windows. It can automatically download and use the latest version of go2rtc. Or it can connect to an existing version of go2rtc. Addon installation in this case is optional.

go2rtc: Dev version

Latest, but maybe unstable version:

Configuration

Configuration options and a complete list of settings can be found in the wiki.

Available modules:

Module: Streams

go2rtc support different stream source types. You can config one or multiple links of any type as stream source.

Available source types:

Read more about incoming sources

Two way audio

Supported for sources:

Two way audio can be used in browser with WebRTC technology. The browser will give access to the microphone only for HTTPS sites (read more).

go2rtc also support play audio files and live streams on this cameras.

Source: RTSP

streams:
  sonoff_camera: rtsp://rtsp:12345678@192.168.1.123/av_stream/ch0
  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
  amcrest_doorbell:
    - rtsp://username:password@192.168.1.123:554/cam/realmonitor?channel=1&subtype=0#backchannel=0
  unifi_camera: rtspx://192.168.1.123:7441/fD6ouM72bWoFijxK
  glichy_camera: ffmpeg:rtsp://username:password@192.168.1.123/live/ch00_1

Recommendations

Other options

Format: rtsp...#{param1}#{param2}#{param3}

RTSP over WebSocket

streams:
  # WebSocket with authorization, RTSP - without
  axis-rtsp-ws:  rtsp://192.168.1.123:4567/axis-media/media.amp?overview=0&camera=1&resolution=1280x720&videoframeskipmode=empty&Axis-Orig-Sw=true#transport=ws://user:pass@192.168.1.123:4567/rtsp-over-websocket
  # WebSocket without authorization, RTSP - with
  dahua-rtsp-ws: rtsp://user:pass@192.168.1.123/cam/realmonitor?channel=1&subtype=1&proto=Private3#transport=ws://192.168.1.123/rtspoverwebsocket

Source: RTMP

You can get stream from RTMP server, for example Nginx with nginx-rtmp-module.

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

Source: HTTP

Support Content-Type:

Source also support HTTP and TCP streams with autodetection for different formats: MJPEG, H.264/H.265 bitstream, MPEG-TS.

streams:
  # [HTTP-FLV] stream in video/x-flv format
  http_flv: http://192.168.1.123:20880/api/camera/stream/780900131155/657617

  # [JPEG] snapshots from Dahua camera, will be converted to MJPEG stream
  dahua_snap: http://admin:password@192.168.1.123/cgi-bin/snapshot.cgi?channel=1

  # [MJPEG] stream will be proxied without modification
  http_mjpeg: https://mjpeg.sanford.io/count.mjpeg

  # [MJPEG or H.264/H.265 bitstream or MPEG-TS]
  tcp_magic: tcp://192.168.1.123:12345

  # Add custom header
  custom_header: "https://mjpeg.sanford.io/count.mjpeg#header=Authorization: Bearer XXX"

PS. Dahua camera has bug: if you select MJPEG codec for RTSP second stream - snapshot won't work.

Source: ONVIF

New in v1.5.0

The source is not very useful if you already know RTSP and snapshot links for your camera. But it can be useful if you don't.

WebUI > Add webpage support ONVIF autodiscovery. Your server must be on the same subnet as the camera. If you use docker, you must use "network host".

streams:
  dahua1: onvif://admin:password@192.168.1.123
  reolink1: onvif://admin:password@192.168.1.123:8000
  tapo1: onvif://admin:password@192.168.1.123:2020

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}#{param1}#{param2}#{param3}. 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#video=h264

  # [RTSP] video with rotation, should be transcoded, so select H264
  rotate: ffmpeg:rtsp://rtsp:12345678@192.168.1.123/av_stream/ch0#video=h264#rotate=90

All trascoding formats has built-in templates: h264, h265, opus, pcmu, pcmu/16000, pcmu/48000, pcma, pcma/16000, pcma/48000, aac, aac/16000.

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
  h264: "-codec:v libx264 -g:v 30 -preset:v superfast -tune:v zerolatency -profile:v main -level:v 4.1"
  mycodec: "-any args that supported by ffmpeg..."
  myinput: "-fflags nobuffer -flags low_delay -timeout 5000000 -i {input}"
  myraw: "-ss 00:00:20"

Read more about hardware acceleration.

PS. It is recommended to check the available hardware in the WebUI add page.

Source: FFmpeg Device

You can get video from any USB-camera or Webcam as RTSP or WebRTC stream. This is part of FFmpeg integration.

Format: ffmpeg:device?{input-params}#{param1}#{param2}#{param3}

streams:
  linux_usbcam:   ffmpeg:device?video=0&video_size=1280x720#video=h264
  windows_webcam: ffmpeg:device?video=0#video=h264
  macos_facetime: ffmpeg:device?video=0&audio=1&video_size=1280x720&framerate=30#video=h264#audio=pcma

PS. It is recommended to check the available devices in the WebUI add page.

Source: Exec

Exec source can run any external application and expect data from it. Two transports are supported - pipe (from v1.5.0) and RTSP.

If you want to use RTSP transport - the command must contain the {output} argument in any place. On launch, it will be replaced by the local address of the RTSP server.

pipe reads data from app stdout in different formats: MJPEG, H.264/H.265 bitstream, MPEG-TS. Also pipe can write data to app stdin in two formats: PCMA and PCM/48000.

The source can be used with:

Pipe commands support parameters (format: exec:{command}#{param1}#{param2}):

streams:
  stream: exec:ffmpeg -re -i /media/BigBuckBunny.mp4 -c copy -rtsp_transport tcp -f rtsp {output}
  picam_h264: exec:libcamera-vid -t 0 --inline -o -
  picam_mjpeg: exec:libcamera-vid -t 0 --codec mjpeg -o -
  pi5cam_h264: exec:libcamera-vid -t 0 --libav-format h264 -o -
  canon: exec:gphoto2 --capture-movie --stdout#killsignal=2#killtimeout=5
  play_pcma: exec:ffplay -fflags nobuffer -f alaw -ar 8000 -i -#backchannel=1
  play_pcm48k: exec:ffplay -fflags nobuffer -f s16be -ar 48000 -i -#backchannel=1

Source: Echo

Some sources may have a dynamic link. And you will need to get it using a bash or python script. Your script should echo a link to the source. RTSP, FFmpeg or any of the supported sources.

Docker and Hass Add-on users has preinstalled python3, curl, jq.

Check examples in wiki.

streams:
  apple_hls: echo:python3 hls.py https://developer.apple.com/streaming/examples/basic-stream-osx-ios5.html

Source: Expr

New in v1.8.2

Like echo source, but uses the built-in expr expression language (read more).

Source: HomeKit

Important:

go2rtc support import paired HomeKit devices from Home Assistant. So you can use HomeKit camera with Hass and go2rtc simultaneously. If you using Hass, I recommend pairing devices with it, it will give you more options.

You can pair device with go2rtc on the HomeKit page. If you can't see your devices - reload the page. Also try reboot your HomeKit device (power off). If you still can't see it - you have a problems with mDNS.

If you see a device but it does not have a pair button - it is paired to some ecosystem (Apple Home, Home Assistant, HomeBridge etc). You need to delete device from that ecosystem, and it will be available for pairing. If you cannot unpair device, you will have to reset it.

Important:

Recommended settings for using HomeKit Camera with WebRTC, MSE, MP4, RTSP:

streams:
  aqara_g3:
    - hass:Camera-Hub-G3-AB12
    - ffmpeg:aqara_g3#audio=aac#audio=opus

RTSP link with "normal" audio for any player: rtsp://192.168.1.123:8554/aqara_g3?video&audio=aac

This source is in active development! Tested only with Aqara Camera Hub G3 (both EU and CN versions).

Source: Bubble

New in v1.6.1

Other names: ESeeCloud, dvr163.

streams:
  camera1: bubble://username:password@192.168.1.123:34567/bubble/live?ch=0&stream=0

Source: DVRIP

New in v1.2.0

Other names: DVR-IP, NetSurveillance, Sofia protocol (NETsurveillance ActiveX plugin XMeye SDK).

streams:
  only_stream: dvrip://username:password@192.168.1.123:34567?channel=0&subtype=0
  only_tts: dvrip://username:password@192.168.1.123:34567?backchannel=1
  two_way_audio:
    - dvrip://username:password@192.168.1.123:34567?channel=0&subtype=0
    - dvrip://username:password@192.168.1.123:34567?backchannel=1

Source: Tapo

New in v1.2.0

TP-Link Tapo proprietary camera protocol with two way audio support.

streams:
  # cloud password without username
  camera1: tapo://cloud-password@192.168.1.123
  # admin username and UPPERCASE MD5 cloud-password hash
  camera2: tapo://admin:UPPERCASE-MD5@192.168.1.123
  # admin username and UPPERCASE SHA256 cloud-password hash
  camera3: tapo://admin:UPPERCASE-SHA256@192.168.1.123
echo -n "cloud password" | md5 | awk '{print toupper($0)}'
echo -n "cloud password" | shasum -a 256 | awk '{print toupper($0)}'

Source: Kasa

New in v1.7.0

TP-Link Kasa non-standard protocol more info.

streams:
  kc401: kasa://username:password@192.168.1.123:19443/https/stream/mixed

Tested: KD110, KC200, KC401, KC420WS, EC71.

Source: GoPro

New in v1.8.3

Support streaming from GoPro cameras, connected via USB or Wi-Fi to Linux, Mac, Windows. Read more.

Source: Ivideon

Support public cameras from service Ivideon.

streams:
  quailcam: ivideon:100-tu5dkUPct39cTp9oNEN2B6/0

Source: Hass

Support import camera links from Home Assistant config files:

hass:
  config: "/config"  # skip this setting if you Hass Add-on user

streams:
  generic_camera: hass:Camera1  # Settings > Integrations > Integration Name
  aqara_g3: hass:Camera-Hub-G3-AB12

WebRTC Cameras (from v1.6.0)

Any cameras in WebRTC format are supported. But at the moment Home Assistant only supports some Nest cameras in this fomat.

Important. The Nest API only allows you to get a link to a stream for 5 minutes. Do not use this with Frigate! If the stream expires, Frigate will consume all available ram on your machine within seconds. It's recommended to use Nest source - it supports extending the stream.

streams:
  # link to Home Assistant Supervised
  hass-webrtc1: hass://supervisor?entity_id=camera.nest_doorbell
  # link to external Hass with Long-Lived Access Tokens
  hass-webrtc2: hass://192.168.1.123:8123?entity_id=camera.nest_doorbell&token=eyXYZ...

RTSP Cameras

By default, the Home Assistant API does not allow you to get dynamic RTSP link to a camera stream. So more cameras, like Tuya, and possibly others can also be imported by using this method.

Source: ISAPI

New in v1.3.0

This source type support only backchannel audio for Hikvision ISAPI protocol. So it should be used as second source in addition to the RTSP protocol.

streams:
  hikvision1:
    - rtsp://admin:password@192.168.1.123:554/Streaming/Channels/101
    - isapi://admin:password@192.168.1.123:80/

Source: Nest

New in v1.6.0

Currently only WebRTC cameras are supported.

For simplicity, it is recommended to connect the Nest/WebRTC camera to the Home Assistant. But if you can somehow get the below parameters - Nest/WebRTC source will work without Hass.

streams:
  nest-doorbell: nest:?client_id=***&client_secret=***&refresh_token=***&project_id=***&device_id=***

Source: Roborock

New in v1.3.0

This source type support Roborock vacuums with cameras. Known working models:

Source support load Roborock credentials from Home Assistant custom integration. Otherwise, you need to log in to your Roborock account (MiHome account is not supported). Go to: go2rtc WebUI > Add webpage. Copy roborock://... source for your vacuum and paste it to go2rtc.yaml config.

If you have graphic pin for your vacuum - add it as numeric pin (lines: 123, 456, 678) to the end of the roborock-link.

Source: WebRTC

New in v1.3.0

This source type support four connection formats.

whep

WebRTC/WHEP - is replaced by WebRTC/WISH standard for WebRTC video/audio viewers. But it may already be supported in some third-party software. It is supported in go2rtc.

go2rtc

This format is only supported in go2rtc. Unlike WHEP it supports asynchronous WebRTC connection and two way audio.

openipc (from v1.7.0)

Support connection to OpenIPC cameras.

wyze (from v1.6.1)

Supports connection to Wyze cameras, using WebRTC protocol. You can use docker-wyze-bridge project to get connection credentials.

kinesis (from v1.6.1)

Supports Amazon Kinesis Video Streams, using WebRTC protocol. You need to specify signalling WebSocket URL with all credentials in query params, client_id and ice_servers list in JSON format.

streams:
  webrtc-whep:    webrtc:http://192.168.1.123:1984/api/webrtc?src=camera1
  webrtc-go2rtc:  webrtc:ws://192.168.1.123:1984/api/ws?src=camera1
  webrtc-openipc: webrtc:ws://192.168.1.123/webrtc_ws#format=openipc#ice_servers=[{"urls":"stun:stun.kinesisvideo.eu-north-1.amazonaws.com:443"}]
  webrtc-wyze:    webrtc:http://192.168.1.123:5000/signaling/camera1?kvs#format=wyze
  webrtc-kinesis: webrtc:wss://...amazonaws.com/?...#format=kinesis#client_id=...#ice_servers=[{...},{...}]

PS. For kinesis sources you can use echo to get connection params using bash/python or any other script language.

Source: WebTorrent

New in v1.3.0

This source can get a stream from another go2rtc via WebTorrent protocol.

streams:
  webtorrent1: webtorrent:?share=huofssuxaty00izc&pwd=k3l2j9djeg8v8r7e

Incoming sources

By default, go2rtc establishes a connection to the source when any client requests it. Go2rtc drops the connection to the source when it has no clients left.

Examples

Incoming: Browser

New in v1.3.0

You can turn the browser of any PC or mobile into an IP-camera with support video and two way audio. Or even broadcast your PC screen:

  1. Create empty stream in the go2rtc.yaml
  2. Go to go2rtc WebUI
  3. Open links page for you stream
  4. Select camera+microphone or display+speaker option
  5. Open webrtc local page (your go2rtc should work over HTTPS!) or share link via WebTorrent technology (work over HTTPS by default)

Incoming: WebRTC/WHIP

New in v1.3.0

You can use OBS Studio or any other broadcast software with WHIP protocol support. This standard has not yet been approved. But you can download OBS Studio dev version:

Stream to camera

New in v1.3.0

go2rtc support play audio files (ex. music or TTS) and live streams (ex. radio) on cameras with two way audio support (RTSP/ONVIF cameras, TP-Link Tapo, Hikvision ISAPI, Roborock vacuums, any Browser).

API example:

POST http://localhost:1984/api/streams?dst=camera1&src=ffmpeg:http://example.com/song.mp3#audio=pcma#input=file

Publish stream

New in v1.8.0

You can publish any stream to streaming services (YouTube, Telegram, etc.) via RTMP/RTMPS. Important:

You can use API:

POST http://localhost:1984/api/streams?src=camera1&dst=rtmps://...

Or config file:

publish:
  # publish stream "video_audio_transcode" to Telegram
  video_audio_transcode:
    - rtmps://xxx-x.rtmp.t.me/s/xxxxxxxxxx:xxxxxxxxxxxxxxxxxxxxxx
  # publish stream "audio_transcode" to Telegram and YouTube
  audio_transcode:
    - rtmps://xxx-x.rtmp.t.me/s/xxxxxxxxxx:xxxxxxxxxxxxxxxxxxxxxx
    - rtmp://xxx.rtmp.youtube.com/live2/xxxx-xxxx-xxxx-xxxx-xxxx

streams:
  video_audio_transcode:
    - ffmpeg:rtsp://user:pass@192.168.1.123/stream1#video=h264#hardware#audio=aac
  audio_transcode:
    - ffmpeg:rtsp://user:pass@192.168.1.123/stream1#video=copy#audio=aac

Module: API

The HTTP API is the main part for interacting with the application. Default address: http://localhost:1984/.

Important! go2rtc passes requests from localhost and from unix socket without HTTP authorisation, even if you have it configured! It is your responsibility to set up secure external access to API. If not properly configured, an attacker can gain access to your cameras and even your server.

API description.

Module config

api:
  listen: ":1984"    # default ":1984", HTTP API port ("" - disabled)
  username: "admin"  # default "", Basic auth for WebUI
  password: "pass"   # default "", Basic auth for WebUI
  base_path: "/rtc"  # default "", API prefix for serve on suburl (/api => /rtc/api)
  static_dir: "www"  # default "", folder for static files (custom web interface)
  origin: "*"        # default "", allow CORS requests (only * supported)
  tls_listen: ":443" # default "", enable HTTPS server
  tls_cert: |        # default "", PEM-encoded fullchain certificate for HTTPS
    -----BEGIN CERTIFICATE-----
    ...
    -----END CERTIFICATE-----
  tls_key: |         # default "", PEM-encoded private key for HTTPS
    -----BEGIN PRIVATE KEY-----
    ...
    -----END PRIVATE KEY-----
  unix_listen: "/tmp/go2rtc.sock"  # default "", unix socket listener for API

PS:

Module: RTSP

You can get any stream as RTSP-stream: rtsp://192.168.1.123:8554/{stream_name}

You can enable external password protection for your RTSP streams. Password protection always disabled for localhost calls (ex. FFmpeg or Hass on same server).

rtsp:
  listen: ":8554"    # RTSP Server TCP port, default - 8554
  username: "admin"  # optional, default - disabled
  password: "pass"   # optional, default - disabled
  default_query: "video&audio"  # optional, default codecs filters

By default go2rtc provide RTSP-stream with only one first video and only one first audio. You can change it with the default_query setting:

Read more about codecs filters.

Module: RTMP

New in v1.8.0

You can get any stream as RTMP-stream: rtmp://192.168.1.123/{stream_name}. Only H264/AAC codecs supported right now.

Incoming stream in RTMP-format tested only with OBS Studio and Dahua camera. Different FFmpeg versions has differnt problems with this format. 

rtmp:
  listen: ":1935"  # by default - disabled!

Module: WebRTC

In most cases WebRTC uses direct peer-to-peer connection from your browser to go2rtc and sends media data via UDP. It can't pass media data through your Nginx or Cloudflare or Nabu Casa HTTP TCP connection! It can automatically detects your external IP via public STUN server. It can establish a external direct connection via UDP hole punching technology even if you not open your server to the World.

But about 10-20% of users may need to configure additional settings for external access if mobile phone or go2rtc server behing Symmetric NAT.

webrtc:
  listen: ":8555"  # address of your local server and port (TCP/UDP)

Static public IP

webrtc:
  candidates:
    - 216.58.210.174:8555  # if you have static public IP-address

Dynamic public IP

webrtc:
  candidates:
    - stun:8555  # if you have dynamic public IP-address

Private IP

ngrok:
  command: ...

Hard tech way 1. 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.

Hard tech way 2. Using TURN-server

If you have personal VPS, you can install TURN server (e.g. coturn, config example).

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

Module: HomeKit

New in v1.7.0

HomeKit module can work in two modes:

Important

Minimal config

streams:
  dahua1: rtsp://admin:password@192.168.1.123/cam/realmonitor?channel=1&subtype=0
homekit:
  dahua1:  # same stream ID from streams list, default PIN - 19550224

Full config

streams:
  dahua1:
    - rtsp://admin:password@192.168.1.123/cam/realmonitor?channel=1&subtype=0
    - ffmpeg:dahua1#video=h264#hardware  # if your camera doesn't support H264, important for HomeKit
    - ffmpeg:dahua1#audio=opus           # only OPUS audio supported by HomeKit

homekit:
  dahua1:                   # same stream ID from streams list
    pin: 12345678           # custom PIN, default: 19550224
    name: Dahua camera      # custom camera name, default: generated from stream ID
    device_id: dahua1       # custom ID, default: generated from stream ID
    device_private: dahua1  # custom key, default: generated from stream ID

Proxy HomeKit camera

streams:
  aqara1:
    - homekit://...
    - ffmpeg:aqara1#audio=aac#audio=opus  # optional audio transcoding

homekit:
  aqara1:  # same stream ID from streams list

Module: WebTorrent

New in v1.3.0

This module support:

Securely and free. You do not need to open a public access to the go2rtc server. But in some cases (Symmetric NAT) you may need to set up external access to WebRTC module.

To generate sharing link or incoming link - goto go2rtc WebUI (stream links page). This link is temporary and will stop working after go2rtc is restarted!

You can create permanent external links in go2rtc config:

webtorrent:
  shares:
    super-secret-share:  # share name, should be unique among all go2rtc users!
      pwd: super-secret-password
      src: rtsp-dahua1   # stream name from streams section

Link example: https://alexxit.github.io/go2rtc/#share=02SNtgjKXY&pwd=wznEQqznxW&media=video+audio

TODO: article how it works...

Module: ngrok

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

The ngrok free subscription has the following limitations:

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 to manually download the ngrok agent app for your OS and register with the ngrok service.

Tunnel for only WebRTC Stream

You need to add your ngrok authtoken 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: 1984  # 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

See the ngrok agent documentation for more details on the ngrok configuration file.

Module: Hass

The best and easiest way to use go2rtc inside the Home Assistant is to install the custom integration WebRTC Camera and custom lovelace card.

But go2rtc is also compatible and can be used with RTSPtoWebRTC built-in integration.

You have several options on how to add a camera to Home Assistant:

  1. Camera RTSP source => Generic Camera
  2. Camera any source => go2rtc config => Generic Camera
    • Install any go2rtc
    • Add your stream to go2rtc config
    • Hass > Settings > Integrations > Add Integration > ONVIF > Host: 127.0.0.1, Port: 1984
    • Hass > Settings > Integrations > Add Integration > Generic Camera > Stream Source URL: rtsp://127.0.0.1:8554/camera1 (change to your stream name, leave everything else as is)

You have several options on how to watch the stream from the cameras in Home Assistant:

  1. Camera Entity => Picture Entity Card => Technology HLS, codecs: H264/H265/AAC, poor latency.
  2. Camera Entity => RTSPtoWebRTC => Picture Entity Card => Technology WebRTC, codecs: H264/PCMU/PCMA/OPUS, best latency.
    • Install any go2rtc
    • Hass > Settings > Integrations > Add Integration > RTSPtoWebRTC > http://127.0.0.1:1984/
    • RTSPtoWebRTC > Configure > STUN server: stun.l.google.com:19302
    • Use Picture Entity or Picture Glance lovelace card
  3. Camera Entity or Camera URL => WebRTC Camera => Technology: WebRTC/MSE/MP4/MJPEG, codecs: H264/H265/AAC/PCMU/PCMA/OPUS, best latency, best compatibility.
    • Install and add WebRTC Camera custom integration
    • Use WebRTC Camera custom lovelace card

You can add camera entity_id to go2rtc config if you need transcoding:

streams:
  "camera.hall": ffmpeg:{input}#video=copy#audio=opus

PS. Default Home Assistant lovelace cards don't support 2-way audio. You can use 2-way audio from Add-on Web UI. But you need use HTTPS to access the microphone. This is a browser restriction and cannot be avoided.

PS. There is also another nice card with go2rtc support - Frigate Lovelace Card.

Module: MP4

Provides several features:

  1. MSE stream (fMP4 over WebSocket)
  2. Camera snapshots in MP4 format (single frame), can be sent to Telegram
  3. HTTP progressive streaming (MP4 file stream) - bad format for streaming because of high start delay. This format doesn't work in all Safari browsers, but go2rtc will automatically redirect it to HLS/fMP4 it this case.

API examples:

Read more about codecs filters.

PS. Rotate and scale params don't use transcoding and change video using metadata.

Module: HLS

New in v1.1.0

HLS is the worst technology for real-time streaming. It can only be useful on devices that do not support more modern technology, like WebRTC, MSE/MP4.

The go2rtc implementation differs from the standards and may not work with all players.

API examples:

Read more about codecs filters.

Module: MJPEG

Important. For stream as MJPEG format, your source MUST contain the MJPEG codec. If your stream has a MJPEG codec - you can receive MJPEG stream or JPEG snapshots via API.

You can receive an MJPEG stream in several ways:

With this example, your stream will have both H264 and MJPEG codecs:

streams:
  camera1:
    - rtsp://rtsp:12345678@192.168.1.123/av_stream/ch0
    - ffmpeg:camera1#video=mjpeg

API examples:

PS. This module also supports streaming to the server console (terminal) in the animated ASCII art format (read more):

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

Security

By default go2rtc starts the Web interface on port 1984 and RTSP on port 8554, as well as use port 8555 for WebRTC connections. The three ports are accessible from your local network. So anyone on your local network can watch video from your cameras without authorization. The same rule applies to the Home Assistant Add-on.

This is not a problem if you trust your local network as much as I do. But you can change this behaviour with a go2rtc.yaml config:

api:
  listen: "127.0.0.1:1984" # localhost

rtsp:
  listen: "127.0.0.1:8554" # localhost

webrtc:
  listen: ":8555" # external TCP/UDP port

If you need Web interface protection without Home Assistant Add-on - you need to use reverse proxy, like Nginx, Caddy, ngrok, etc.

PS. Additionally WebRTC will try to use the 8555 UDP port for transmit encrypted media. It works without problems on the local network. And sometimes also works for external access, even if you haven't opened this port on your router (read more). But for stable external WebRTC access, you need to open the 8555 port on your router for both TCP and UDP.

Codecs filters

go2rtc can automatically detect which codecs your device supports for WebRTC and MSE technologies.

But it cannot be done for RTSP, HTTP progressive streaming, HLS technologies. You can manually add a codec filter when you create a link to a stream. The filters work the same for all three technologies. Filters do not create a new codec. They only select the suitable codec from existing sources. You can add new codecs to the stream using the FFmpeg transcoding.

Without filters:

Some examples:

Codecs madness

AVC/H.264 video can be played almost anywhere. But HEVC/H.265 has a lot of limitations in supporting with different devices and browsers. It's all about patents and money, you can't do anything about it.

Device WebRTC MSE HTTP* HLS
latency best medium bad bad
- Desktop Chrome 107+
- Desktop Edge
- Android Chrome 107+		 H264
PCMU, PCMA
OPUS		 H264, H265
AAC, FLAC
OPUS		 H264, H265
AAC, FLAC
OPUS, MP3		 no
Desktop Firefox H264
PCMU, PCMA
OPUS		 H264
AAC, FLAC*
OPUS		 H264
AAC, FLAC*
OPUS		 no
- Desktop Safari 14+
- iPad Safari 14+
- iPhone Safari 17.1+		 H264, H265*
PCMU, PCMA
OPUS		 H264, H265
AAC, FLAC*		 no! H264, H265
AAC, FLAC*
iPhone Safari 14+ H264, H265*
PCMU, PCMA
OPUS		 no! no! H264, H265
AAC, FLAC*
macOS Hass App no no no H264, H265
AAC, FLAC*

HTTP* - HTTP Progressive Streaming, not related with Progressive download, because the file has no size and no end

Audio

Apple devices

Codec names

Built-in transcoding

There are no plans to embed complex transcoding algorithms inside go2rtc. FFmpeg source does a great job with this. Including hardware acceleration support.

But go2rtc has some simple algorithms. They are turned on automatically, you do not need to set them up additionally.

PCM for MSE/MP4/HLS

Go2rtc can pack PCMA, PCMU and PCM codecs into an MP4 container so that they work in all browsers and all built-in players on modern devices. Including Apple QuickTime:

PCMA/PCMU => PCM => FLAC => MSE/MP4/HLS

Resample PCMA/PCMU for WebRTC

By default WebRTC support only PCMA/8000 and PCMU/8000. But go2rtc can automatically resample PCMA and PCMU codec with with a different sample rate. Also go2rtc can transcode PCM codec to PCMA/8000, so WebRTC can play it:

PCM/xxx => PCMA/8000 => WebRTC
PCMA/xxx => PCMA/8000 => WebRTC
PCMU/xxx => PCMU/8000 => WebRTC

Important

Codecs negotiation

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

Now you have stream with two sources - RTSP and FFmpeg:

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#audio=opus

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.

Projects using go2rtc

Distributions

Cameras experience

TIPS

Using apps for low RTSP delay

Snapshots to Telegram

read more

FAQ

Q. What's the difference between go2rtc, WebRTC Camera and RTSPtoWebRTC?

go2rtc is a new version of the server-side WebRTC Camera integration, completely rewritten from scratch, with a number of fixes and a huge number of new features. It is compatible with native Home Assistant RTSPtoWebRTC integration. So you can use default lovelace Picture Entity or Picture Glance.

Q. Should I use go2rtc addon or WebRTC Camera integration?

go2rtc is more than just viewing your stream online with WebRTC/MSE/HLS/etc. You can use it all the time for your various tasks. But every time the Hass is rebooted - all integrations are also rebooted. So your streams may be interrupted if you use them in additional tasks.

Basic users can use WebRTC Camera integration. Advanced users can use go2rtc addon or Frigate 12+ addon.

Q. Which RTSP link should I use inside Hass?

You can use direct link to your cameras there (as you always do). go2rtc support zero-config feature. You may leave streams config section empty. And your streams will be created on the fly on first start from Hass. And your cameras will have multiple connections. Some from Hass directly and one from go2rtc.

Also you can specify your streams in go2rtc config file and use RTSP links to this addon. With additional features: multi-source codecs negotiation or FFmpeg transcoding for unsupported codecs. Or use them as source for Frigate. And your cameras will have one connection from go2rtc. And go2rtc will have multiple connection - some from Hass via RTSP protocol, some from your browser via WebRTC/MSE/HLS protocols.

Use any config what you like.

Q. What about lovelace card with support 2-way audio?

At this moment I am focused on improving stability and adding new features to go2rtc. Maybe someone could write such a card themselves. It's not difficult, I have some sketches.